diff --git a/tools/build-system/README.launchpad b/tools/build-system/README.launchpad
deleted file mode 100644
index aa0861b03356e631e6208007a2eeee00616818dd..0000000000000000000000000000000000000000
--- a/tools/build-system/README.launchpad
+++ /dev/null
@@ -1,10 +0,0 @@
-To push packages for a new Ubuntu distribution on launchpad, you'll need to
-modify the following files:
-
-tools/build-system/launchpad/dput.conf
-tools/build-system/setenv.sh
-
-See commit 1b19e3869aa5e4632b8b043f47024297218ed2c5 for an example.
-
-In addition, you'll have to modify the sflphone-package-manager job on Jenkins, specifically
-Configure->Build->Execute Shell (add new distro).
diff --git a/tools/build-system/build_tarball.sh b/tools/build-system/build_tarball.sh
deleted file mode 100755
index d34e2b6b6cbfb6519002cb863a1394d79d65c05c..0000000000000000000000000000000000000000
--- a/tools/build-system/build_tarball.sh
+++ /dev/null
@@ -1,76 +0,0 @@
-#!/bin/bash
-#
-# Script to build the source tarball for distribution on sflphone.org
-# Inclusion of KDE is a requirement. Run get-kde.sh to have it.
-#
-# Author: Francois Marier <francois@debian.org>
-
-
-# Exit on error
-set -o errexit
-
-# This is an environment variable provided by Jenkins. It points to the repository's root
-cd ${WORKSPACE}
-
-if [ ! -e daemon/configure.ac ] ; then
- echo "This script must be run in the root directory of the sflphone repository" >&2
- exit 1
-fi
-
-if [ $# -ne 1 ] ; then
- echo "Usage: $(basename $0) SOFTWARE_VERSION_NUMBER" >&2
- exit 2
-fi
-
-if [ ! -d kde ] ; then
- echo 'No "kde" directory. Make sure get-kde.sh ran at some point.' >&2
- exit 1
-fi
-
-# Use the version fed by launch-build-machine-jenkins.sh
-SOFTWARE_VERSION=$1
-BUILDDIR=sflphone-$SOFTWARE_VERSION
-
-if [ -e $BUILDDIR ] ; then
- echo "The build directory ($BUILDDIR) already exists. Delete it first." >&2
- exit 3
-fi
-
-# Populate the tarball directory
-mkdir $BUILDDIR
-SRCITEMS=$(echo *)
-# Exclude existing tarballs from the created tarball
-SRCITEMS=${SRCITEMS//*.tar.gz}
-# ${SRCITEMS//$BUILDDIR} is used to remove $BUILDDIR from $SRCITEMS
-# See bash parameter expansion
-cp -r ${SRCITEMS//$BUILDDIR} $BUILDDIR/
-
-pushd $BUILDDIR
-# No dash in Version:
-sed /^Version/s/[0-9].*/${SOFTWARE_VERSION%%-*}/ tools/build-system/rpm/sflphone.spec > sflphone.spec
-
-# Remove unwanted files
-rm -rf lang/
-rm -rf tools/
-rm -rf .git/
-find -name .gitignore -delete
-find -name .project -type f -delete
-find -name .cproject -type f -delete
-find -name .settings -type d -exec rm -rf {} +
-
-# Generate the configure files
-pushd daemon
-./autogen.sh
-find -name \*.spec -delete
-popd
-
-pushd gnome
-NOCONFIGURE=1 ./autogen.sh
-popd
-
-find -name autom4te.cache -type d -exec rm -rf {} +
-find -name *.in~ -type f -delete
-popd # builddir
-
-tar zcf sflphone-${SOFTWARE_VERSION}.tar.gz sflphone-${SOFTWARE_VERSION}
-rm -rf $BUILDDIR
diff --git a/tools/build-system/get-kde.sh b/tools/build-system/get-kde.sh
deleted file mode 100755
index 25630c27215de58ef11d64d5c225bf1b6de5af77..0000000000000000000000000000000000000000
--- a/tools/build-system/get-kde.sh
+++ /dev/null
@@ -1,38 +0,0 @@
-#!/bin/bash
-# Get the KDE client
-# To get all files, you have to create the tarball from scratch,
-# then extract files from it. The directory is renamed "kde".
-# $WORKSPACE is declared in setenv.sh
-set -o errexit
-source $(dirname $0)/setenv.sh
-cd "$WORKSPACE"
-baseurl='https://projects.kde.org/projects'
-config_uri='/playground/network/sflphone-kde/repository/revisions/master/raw/data/config.ini'
-createtarball_uri='/kde/kdesdk/kde-dev-scripts/repository/revisions/master/raw/createtarball/create_tarball.rb'
-
-set -x
-
-# timeout in seconds
-let -i timeout=300
-let -i timestamp=$(date +%s)
-while ! curl --fail --remote-name ${baseurl}${config_uri}
-do
- if [ $(date +%s) -gt $(( $timestamp + $timeout)) ]; then
- break
- fi
- sleep 15
-done
-let -i timestamp=$(date +%s)
-while ! curl --fail --remote-name ${baseurl}${createtarball_uri}
-do
- if [ $(date +%s) -gt $(( $timestamp + $timeout)) ]; then
- break
- fi
- sleep 15
-done
-
-ruby create_tarball.rb --noaccount --application sflphone-kde
-rm -rf kde
-rm -rf sflphone-kde-*.tar.*
-rm create_tarball.rb config.ini
-mv sflphone-kde-* kde
diff --git a/tools/build-system/hudson-sflphone-script.sh b/tools/build-system/hudson-sflphone-script.sh
deleted file mode 100755
index 0a795b41c019d8bd3eb1595cf59e9b205927ee15..0000000000000000000000000000000000000000
--- a/tools/build-system/hudson-sflphone-script.sh
+++ /dev/null
@@ -1,249 +0,0 @@
-#!/bin/bash -e
-#
-# Copyright (C) 2004-2016 Savoir-faire Linux Inc.
-#
-# Author: Emmanuel Milou <emmanuel.milou@savoirfairelinux.com>
-#
-# This program is free software; you can redistribute it and/or modify
-# it under the terms of the GNU General Public License as published by
-# the Free Software Foundation; either version 3 of the License, or
-# (at your option) any later version.
-#
-# This program is distributed in the hope that it will be useful,
-# but WITHOUT ANY WARRANTY; without even the implied warranty of
-# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
-# GNU General Public License for more details.
-#
-# You should have received a copy of the GNU General Public License
-# along with this program; if not, write to the Free Software
-# Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
-#
-
-# Script used by Hudson continious integration server to build SFLphone
-
-XML_RESULTS="cppunitresults.xml"
-TEST=0
-BUILD=
-CODE_ANALYSIS=0
-DOXYGEN=0
-#daemon opts
-DOPTS="--prefix=/usr"
-#gnome opts
-GOPTS="--prefix=/usr --enable-video"
-
-#compiler defaults
-export CC=gcc
-export CXX=g++
-
-CONFIGDIR=~/.config
-SFLCONFDIR=${CONFIGDIR}/sflphone
-
-function exit_clean {
- popd
- exit $1
-}
-
-function run_code_analysis {
- # Check if cppcheck is installed on the system
- if [ `which cppcheck &>/dev/null ; echo $?` -ne 1 ] ; then
- pushd src
- cppcheck . --enable=all --xml --inline-suppr 2> cppcheck-report.xml
- popd
- fi
-}
-
-
-function gen_doxygen {
- # Check if doxygen is installed on the system
- if [ `which doxygen &>/dev/null ; echo $?` -ne 1 ] ; then
- pushd doc/doxygen
- doxygen core-doc.cfg.in
- popd
- fi
-}
-
-function launch_functional_test_daemon {
- # Run the python functional tests for the daemon
-
- # make sure no other instance are currently running
- killall sflphoned
- killall sipp
-
- # make sure the configuration directory created
- CONFDIR=~/.config
- SFLCONFDIR=${CONFDIR}/sflphone
-
- eval `dbus-launch --auto-syntax`
-
- if [ ! -d ${CONFDIR} ]; then
- mkdir ${CONFDIR}
- fi
-
- if [ ! -d ${SFLCONFDIR} ]; then
- mkdir ${SFLCONFDIR}
- fi
-
- # make sure the most recent version of the configuration
- # is installed
- pushd tools/pysflphone
- cp -f sflphoned.functest.yml ${SFLCONFDIR}
- popd
-
- # launch sflphone daemon, wait some time for
- # dbus registration to complete
- pushd daemon
- ./src/sflphoned &
- sleep 3
- popd
-
- # launch the test script
- pushd tools/pysflphone
- nosetests --with-xunit test_sflphone_dbus_interface.py
- popd
-}
-
-function build_contrib {
- if [ -d contrib ] ; then
- pushd contrib
- mkdir -p native
- pushd native
- ../bootstrap
- # list dependencies that aren't detected by contrib
- make list
-
- # FIXME: this is very slow but it's the best we can do until we migrate
- # to a builder with more up to date packages
- if [ "$DEBUG_CONTRIB" != "" ]; then
- make
- else
- make -j
- fi
- popd
- else
- # We're on 1.4.x
- pushd libs
- ./compile_pjsip.sh
- fi
- popd
-}
-
-function build_daemon {
- pushd daemon
-
- # Build dependencies first
- build_contrib
-
- # Run static analysis code tool
- if [ $CODE_ANALYSIS == 1 ]; then
- run_code_analysis
- fi
-
- # Compile the daemon
- ./autogen.sh || exit_clean 1
- #FIXME: this is a temporary hack around linking failure on jenkins
- ./configure $DOPTS
- make clean
- make -j
- # Remove the previous XML test file
- rm -rf $XML_RESULTS
- # Compile unit tests
- make check
- popd
-}
-
-function build_gnome {
- # Compile the plugins
- pushd plugins
- ./autogen.sh || exit_clean 1
- ./configure $GOPTS
- make -j
- popd
-
- # Compile the client
- pushd gnome
- ./autogen.sh || exit_clean 1
- ./configure $GOPTS
- make clean
- make -j 1
- make check
- popd
-}
-
-function build_kde {
- # Compile the KDE client
- pushd kde
- mkdir -p build
- cd build
- cmake ../
- make -j
- popd
-}
-
-
-if [ "$#" -eq 0 ]; then # Script needs at least one command-line argument.
- echo "$0 accepts the following options:
- -b select 'daemon' or 'gnome' component
- -v enable video support
- -c use clang compiler
- -a run static code analysis after build
- -t run unit tests after build
- -m disable most optional options"
- exit $E_OPTERR
-fi
-
-pushd "$(git rev-parse --show-toplevel)"
-git clean -f -d -x
-
-while getopts ":b: t a v c" opt; do
- case $opt in
- b)
- echo "-b is set with option $OPTARG" >&2
- if [ ! -d $OPTARG ]
- then
- echo "$OPTARG directory is missing, exiting"
- exit_clean $E_OPTERR
- fi
- BUILD=$OPTARG
- ;;
- t)
- echo "-t is set, unit tests will be run after build" >&2
- TEST=1
- ;;
- a)
- echo "-a is set, static code analysis will be run after build" >&2
- CODE_ANALYSIS=1
- ;;
- v)
- echo "-v is set, video support is disabled" >&2
- DOPTS="--disable-video $DOPTS"
- ;;
- m)
- echo "-m is set, disabling dbus, video, nm and pulse" >&2
- DOPTS="--disable-video --without-dbus --without-pulse --without-networkmanager $DOPTS"
- ;;
- c)
- echo "-c is set, clang compiler is used" >&2
- export CC=clang
- export CXX=clang++
- DOPTS="--without-dbus $DOPTS"
- ;;
- \?)
- echo "Invalid option: -$OPTARG" >&2
- exit_clean 1
- ;;
- :)
- echo "Option -$OPTARG requires an argument." >&2
- exit_clean 1
- ;;
- esac
-done
-
-# Call appropriate build function, with parameters if needed
-build_$BUILD
-
-if [ $TEST == 1 ]; then
- launch_functional_test_daemon
-fi
-
-# SUCCESS
-exit_clean 0
diff --git a/tools/build-system/launch-build-machine-jenkins.sh b/tools/build-system/launch-build-machine-jenkins.sh
deleted file mode 100755
index 017b164a57320b031e282c9212c4288dbf184a0e..0000000000000000000000000000000000000000
--- a/tools/build-system/launch-build-machine-jenkins.sh
+++ /dev/null
@@ -1,315 +0,0 @@
-#!/bin/bash
-# run with --help for documentation
-
-set -x
-set -e
-
-#Check dependencies
-
-for cmd in curl ruby git svn
-do
- if ! command -v $cmd; then
- echo "$cmd is missing" >&2
- exit 1
- fi
-done
-
-source $(dirname $0)/setenv.sh
-
-./$(dirname $0)/get-kde.sh
-
-IS_RELEASE=
-VERSION_INDEX="1"
-IS_KDE_CLIENT=
-DO_PUSH=1
-DO_LOGGING=1
-DO_UPLOAD=1
-SNAPSHOT_TAG=`date +%Y%m%d`
-TAG_NAME_PREFIX=
-VERSION_NUMBER="1.4.2"
-
-LAUNCHPAD_PACKAGES=("sflphone-daemon" "sflphone-kde" "sflphone-gnome" "sflphone-plugins" "sflphone-daemon-video" "sflphone-gnome-video")
-
-cat << EOF
-_________________________
-| SFLPhone build system |
--------------------------
-EOF
-
-
-for PARAMETER in $*
-do
- case ${PARAMETER} in
- --help)
- echo
- echo "Options :"
- echo " --skip-push"
- echo " --skip-upload"
- echo " --kde-client"
- echo " --no-logging"
- echo " --release"
- echo " --version-index=[1,2,...]"
- echo
- exit 0;;
- --skip-push)
- unset DO_PUSH;;
- --skip-upload)
- unset DO_UPLOAD;;
- --kde-client)
- IS_KDE_CLIENT=1;;
- --no-logging)
- unset DO_LOGGING;;
- --release)
- IS_RELEASE=1;;
- --tag=*)
- TAG=(${PARAMETER##*=});;
- --version-index=*)
- VERSION_INDEX=(${PARAMETER##*=});;
- *)
- echo "Unknown parameter : ${PARAMETER}"
- exit -1;;
- esac
-done
-
-#########################
-# LAUNCHPAD
-#########################
-
-# change to working directory
-cd ${LAUNCHPAD_DIR}
-
-if [ "$?" -ne "0" ]; then
- echo " !! Cannot cd to launchpad directory"
- exit -1
-fi
-
-# logging
-if [ ${DO_LOGGING} ]; then
-
- rm -f ${ROOT_DIR}/packaging.log >/dev/null 2>&1
-
- # open file descriptor
- exec 3<> ${ROOT_DIR}/packaging.log
-
- # redirect outputs (stdout & stderr)
- exec 1>&3
- exec 2>&3
-fi
-
-if [ ${RELEASE_MODE} ]; then
- echo "Release mode"
-else
- echo "Snapshot mode"
-fi
-
-if [ ${IS_KDE_CLIENT} ]; then
- TAG_NAME_PREFIX="kde."
-fi
-
-#########################
-# COMMON PART
-#########################
-
-cd ${REFERENCE_REPOSITORY}
-
-echo "Update reference sources"
-git checkout . && git checkout -f master && git pull
-
-# Get the version
-if [ -n "$TAG" ]; then
- CURRENT_RELEASE_TAG_NAME="$TAG"
-else
- CURRENT_RELEASE_TAG_NAME=`git describe --tags --abbrev=0`
-fi
-
-PREVIOUS_RELEASE_TAG_NAME=`git describe --tags --abbrev=0 ${CURRENT_RELEASE_TAG_NAME}^`
-CURRENT_RELEASE_COMMIT_HASH=`git show --pretty=format:"%H" -s ${CURRENT_RELEASE_TAG_NAME} | tail -n 1`
-PREVIOUS_RELEASE_COMMIT_HASH=`git show --pretty=format:"%H" -s ${PREVIOUS_RELEASE_TAG_NAME} | tail -n 1`
-CURRENT_COMMIT=`git show --pretty=format:"%H" -s | tail -n 1`
-CURRENT_RELEASE_TYPE=${CURRENT_RELEASE_TAG_NAME##*.}
-PREVIOUS_RELEASE_TYPE=${PREVIOUS_RELEASE_TAG_NAME##*.}
-
-if [ ${IS_KDE_CLIENT} ]; then
- CURRENT_RELEASE_VERSION=${CURRENT_RELEASE_TAG_NAME%.*}
- CURRENT_RELEASE_VERSION=${CURRENT_RELEASE_VERSION#*.}
- PREVIOUS_VERSION=${PREVIOUS_RELEASE_TAG_NAME%.*}
- PREVIOUS_VERSION=${PREVIOUS_VERSION#*.}
-else
- CURRENT_RELEASE_VERSION=${CURRENT_RELEASE_TAG_NAME}
- PREVIOUS_VERSION=${PREVIOUS_RELEASE_TAG_NAME}
-fi
-
-
-echo "Retrieve build info"
-# retrieve info we may need
-if [ ${IS_KDE_CLIENT} ]; then
- TAG_NAME_PREFIX="kde."
- LAUNCHPAD_PACKAGES=( "sflphone-kde" )
-fi
-
-
-cd ${LAUNCHPAD_DIR}
-
-COMMIT_HASH_BEGIN=""
-COMMIT_HASH_END=""
-SOFTWARE_VERSION=""
-LAUNCHPAD_CONF_PREFIX=""
-
-if [ ${IS_RELEASE} ]; then
- SOFTWARE_VERSION="${CURRENT_RELEASE_VERSION}"
- COMMIT_HASH_BEGIN="${PREVIOUS_RELEASE_COMMIT_HASH}"
- LAUNCHPAD_CONF_PREFIX="sflphone"
-else
- SOFTWARE_VERSION="${VERSION_NUMBER}-rc${SNAPSHOT_TAG}"
- COMMIT_HASH_BEGIN="${CURRENT_RELEASE_COMMIT_HASH}"
- LAUNCHPAD_CONF_PREFIX="sflphone-nightly"
-fi
-
-VERSION="${SOFTWARE_VERSION}~ppa${VERSION_INDEX}~SYSTEM"
-
-echo "Clean build directory"
-git clean -f -x ${LAUNCHPAD_DIR}/* >/dev/null
-git checkout ${LAUNCHPAD_DIR}
-
-# If release, checkout the latest tag
-if [ ${IS_RELEASE} ]; then
- git checkout ${CURRENT_RELEASE_TAG_NAME}
-
- # When we need to apply an emergency patch for the release builds
- # This should only be used to temporarily patch packaging tools, not
- # daemon/client code (or anything else that build_tarball would grab).
- if [ -d /tmp/sflphone_release_patch ]; then
- echo "Applying patch(es) to packaging tools..."
- git apply --verbose /tmp/sflphone_release_patch/*
- rm -rf /tmp/sflphone_release_patch
- REQUIRE_RESET=1
- fi
-fi
-
-get_dir_name() {
- case $1 in
- sflphone-daemon)
- echo daemon
- ;;
- sflphone-daemon-video)
- echo daemon
- ;;
- sflphone-plugins)
- echo plugins
- ;;
- sflphone-gnome)
- echo gnome
- ;;
- sflphone-gnome-video)
- echo gnome
- ;;
- sflphone-kde)
- echo kde
- ;;
- *)
- exit 1
- ;;
- esac
-}
-
-# Looping over the packages
-for LAUNCHPAD_PACKAGE in ${LAUNCHPAD_PACKAGES[*]}
-do
- echo " Package: ${LAUNCHPAD_PACKAGE}"
-
- echo " --> Clean old sources"
- git clean -f -x ${LAUNCHPAD_DIR}/${LAUNCHPAD_PACKAGE}/* >/dev/null
-
- DEBIAN_DIR="${LAUNCHPAD_DIR}/${LAUNCHPAD_PACKAGE}/debian"
-
- echo " --> Retrieve new sources"
- DIRNAME=`get_dir_name ${LAUNCHPAD_PACKAGE}`
- cp -r ${REFERENCE_REPOSITORY}/${DIRNAME}/* ${LAUNCHPAD_DIR}/${LAUNCHPAD_PACKAGE}
-
- echo " --> Update software version number (${SOFTWARE_VERSION})"
- echo "${SOFTWARE_VERSION}" > ${LAUNCHPAD_DIR}/${LAUNCHPAD_PACKAGE}/VERSION
-
- echo " --> Update debian changelog"
-
-cat << END > ${WORKING_DIR}/sfl-git-dch.conf
-WORKING_DIR="${REFERENCE_REPOSITORY}"
-SOFTWARE="${LAUNCHPAD_PACKAGE}"
-VERSION="${VERSION}"
-DISTRIBUTION="SYSTEM"
-CHANGELOG_FILE="${DEBIAN_DIR}/changelog"
-COMMIT_HASH_BEGIN="${COMMIT_HASH_BEGIN}"
-COMMIT_HASH_END="${COMMIT_HASH_END}"
-IS_RELEASE=${IS_RELEASE}
-export DEBFULLNAME="Emmanuel Milou"
-export DEBEMAIL="emmanuel.milou@savoirfairelinux.com"
-export EDITOR="echo"
-END
-
- ${WORKING_DIR}/sfl-git-dch-2.sh ${WORKING_DIR}/sfl-git-dch.conf ${REFERENCE_REPOSITORY}/${DIRNAME}/
- if [ "$?" -ne "0" ]; then
- echo "!! Cannot update debian changelogs"
- exit -1
- fi
-
- if [ "${LAUNCHPAD_PACKAGE}" == "sflphone-kde" ]; then
- version_kde=$(echo ${VERSION} | grep -e '[0-9]*\.[0-9.]*' -o | head -n1)
- sed -i -e "s/Standards-Version: [0-9.A-Za-z]*/Standards-Version: ${version_kde}/" ${LAUNCHPAD_DIR}/${LAUNCHPAD_PACKAGE}/debian/control
- tar -C ${LAUNCHPAD_DIR}/ -cjf ${LAUNCHPAD_DIR}/sflphone-kde_${version_kde}.orig.tar.bz2 ${LAUNCHPAD_PACKAGE}
- fi
-
- rm -f ${WORKING_DIR}/sfl-git-dch.conf >/dev/null 2>&1
-
- cd ${LAUNCHPAD_DIR}
-
- cp ${DEBIAN_DIR}/changelog ${DEBIAN_DIR}/changelog.generic
-
- for LAUNCHPAD_DISTRIBUTION in ${LAUNCHPAD_DISTRIBUTIONS[*]}
- do
-
- LOCAL_VERSION="${SOFTWARE_VERSION}~ppa${VERSION_INDEX}~${LAUNCHPAD_DISTRIBUTION}"
-
- cp ${DEBIAN_DIR}/changelog.generic ${DEBIAN_DIR}/changelog
-
- sed -i "s/SYSTEM/${LAUNCHPAD_DISTRIBUTION}/g" ${DEBIAN_DIR}/changelog
-
- cd ${LAUNCHPAD_DIR}/${LAUNCHPAD_PACKAGE}
- if [ "${DIRNAME}" == "daemon" ]; then
- if [ -d contrib ]; then
- mkdir -p contrib/native
- pushd contrib/native
- ../bootstrap
- # only fetch it, don't build it
- make dht
- else
- pushd libs
- #./compile_pjsip.sh #This script should not attempt to compile
- fi
- popd
- fi
- if [ "${LAUNCHPAD_PACKAGE}" != "sflphone-kde" ]; then
- ./autogen.sh
- fi
- debuild -S -sa -kF5362695
- cd ${LAUNCHPAD_DIR}
-
- if [ ${DO_UPLOAD} ] ; then
- dput -f --debug --no-upload-log -c ${LAUNCHPAD_DIR}/dput.conf ${LAUNCHPAD_CONF_PREFIX}-${LAUNCHPAD_DISTRIBUTION} ${LAUNCHPAD_PACKAGE}_${LOCAL_VERSION}_source.changes
- fi
- done
-
- cp ${DEBIAN_DIR}/changelog.generic ${DEBIAN_DIR}/changelog
-done
-
-# Archive source tarball for Debian maintainer
-# and for RPM package building
-${WORKING_DIR}/build_tarball.sh ${SOFTWARE_VERSION}
-
-# Undo any modifications caused by temporary patches
-if [ "$REQUIRE_RESET" == "1" ]; then
- git reset --hard
-fi
-
-# close file descriptor
-exec 3>&-
-
-exit 0
diff --git a/tools/build-system/launchpad/dput.conf b/tools/build-system/launchpad/dput.conf
deleted file mode 100644
index 8d6fe9d5cbcb42e31bab13cc23be7a7f563b3eea..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/dput.conf
+++ /dev/null
@@ -1,27 +0,0 @@
-[sflphone-trusty]
-fqdn = ppa.launchpad.net
-method = ftp
-incoming = ~savoirfairelinux/ppa/ubuntu/trusty
-login = anonymous
-allow_unsigned_uploads = 0
-
-[sflphone-utopic]
-fqdn = ppa.launchpad.net
-method = ftp
-incoming = ~savoirfairelinux/ppa/ubuntu/utopic
-login = anonymous
-allow_unsigned_uploads = 0
-
-[sflphone-nightly-trusty]
-fqdn = ppa.launchpad.net
-method = ftp
-incoming = ~savoirfairelinux/sflphone-nightly/ubuntu/trusty
-login = anonymous
-allow_unsigned_uploads = 0
-
-[sflphone-nightly-utopic]
-fqdn = ppa.launchpad.net
-method = ftp
-incoming = ~savoirfairelinux/sflphone-nightly/ubuntu/utopic
-login = anonymous
-allow_unsigned_uploads = 0
diff --git a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/changelog b/tools/build-system/launchpad/mozilla-telify-sflphone/debian/changelog
deleted file mode 100644
index 17f7f40a397bc9ee8a91173c25e16280cc6a328e..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/changelog
+++ /dev/null
@@ -1,13 +0,0 @@
-mozilla-telify-sflphone (1.0) unstable; urgency=low
-
- [ Julien Bonjean ]
- * Package update
-
- -- Julien Bonjean <julien.bonjean@savoirfairelinux.com> Fri, 21 Apr 2010 19:51:54 +0100
-
-mozilla-telify-sflphone (0.4.7.3) unstable; urgency=low
-
- [ Julien Bonjean ]
- * Package creation
-
- -- Julien Bonjean <julien.bonjean@savoirfairelinux.com> Fri, 20 Nov 2009 19:51:54 +0100
diff --git a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/compat b/tools/build-system/launchpad/mozilla-telify-sflphone/debian/compat
deleted file mode 100644
index 7f8f011eb73d6043d2e6db9d2c101195ae2801f2..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/compat
+++ /dev/null
@@ -1 +0,0 @@
-7
diff --git a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/control b/tools/build-system/launchpad/mozilla-telify-sflphone/debian/control
deleted file mode 100644
index e0bbb912d313d64c9322715f52c3619c7dd0f483..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/control
+++ /dev/null
@@ -1,19 +0,0 @@
-Source: mozilla-telify-sflphone
-Section: web
-Priority: optional
-Maintainer: Julien Bonjean <julien.bonjean@savoirfairelinux.com>
-Uploaders: Julien Bonjean <julien.bonjean@savoirfairelinux.com>
-Build-Depends: debhelper (>= 7), unzip
-Homepage: http://www.sflphone.org
-Standards-Version: 3.8.3
-DM-Upload-Allowed: yes
-
-Package: mozilla-telify-sflphone
-Depends: firefox-gnome-support, sflphone-gnome
-Architecture: all
-Description: This package provides telify firefox plugin and handler for SFLphone.
- Telify recognizes phone numbers on web pages and converts them to clickable links.
- Additionally, any text can be selected and handled as a phone number (including
- vanity number conversion) by selecting the corresponding context menu item.
- http://www.codepad.de/en/software/firefox-add-ons/telify.html
-
diff --git a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/control.debian b/tools/build-system/launchpad/mozilla-telify-sflphone/debian/control.debian
deleted file mode 100644
index 77147717cca137e5bd882e5cf7176efe9889fb5c..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/control.debian
+++ /dev/null
@@ -1,19 +0,0 @@
-Source: mozilla-telify-sflphone
-Section: web
-Priority: optional
-Maintainer: Julien Bonjean <julien.bonjean@savoirfairelinux.com>
-Uploaders: Julien Bonjean <julien.bonjean@savoirfairelinux.com>
-Build-Depends: debhelper (>= 7), unzip
-Homepage: http://www.sflphone.org
-Standards-Version: 3.8.3
-DM-Upload-Allowed: yes
-
-Package: mozilla-telify-sflphone
-Depends: iceweasel-gnome-support, sflphone-gnome
-Architecture: all
-Description: This package provides telify firefox plugin and handler for SFLphone.
- Telify recognizes phone numbers on web pages and converts them to clickable links.
- Additionally, any text can be selected and handled as a phone number (including
- vanity number conversion) by selecting the corresponding context menu item.
- http://www.codepad.de/en/software/firefox-add-ons/telify.html
-
diff --git a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/control.ubuntu b/tools/build-system/launchpad/mozilla-telify-sflphone/debian/control.ubuntu
deleted file mode 100644
index e0bbb912d313d64c9322715f52c3619c7dd0f483..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/control.ubuntu
+++ /dev/null
@@ -1,19 +0,0 @@
-Source: mozilla-telify-sflphone
-Section: web
-Priority: optional
-Maintainer: Julien Bonjean <julien.bonjean@savoirfairelinux.com>
-Uploaders: Julien Bonjean <julien.bonjean@savoirfairelinux.com>
-Build-Depends: debhelper (>= 7), unzip
-Homepage: http://www.sflphone.org
-Standards-Version: 3.8.3
-DM-Upload-Allowed: yes
-
-Package: mozilla-telify-sflphone
-Depends: firefox-gnome-support, sflphone-gnome
-Architecture: all
-Description: This package provides telify firefox plugin and handler for SFLphone.
- Telify recognizes phone numbers on web pages and converts them to clickable links.
- Additionally, any text can be selected and handled as a phone number (including
- vanity number conversion) by selecting the corresponding context menu item.
- http://www.codepad.de/en/software/firefox-add-ons/telify.html
-
diff --git a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/copyright b/tools/build-system/launchpad/mozilla-telify-sflphone/debian/copyright
deleted file mode 100644
index a0990367ef8b03c70c29d285e22ef85907e1d0b7..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/copyright
+++ /dev/null
@@ -1 +0,0 @@
-TBD
diff --git a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/files b/tools/build-system/launchpad/mozilla-telify-sflphone/debian/files
deleted file mode 100644
index 320b727519610179c97214659e4e3be20f8cebc3..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/files
+++ /dev/null
@@ -1 +0,0 @@
-mozilla-telify-sflphone_1.0_all.deb web optional
diff --git a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone.debhelper.log b/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone.debhelper.log
deleted file mode 100644
index 89ec40ebaf21fd29fbf68d5f444d2332c792b359..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone.debhelper.log
+++ /dev/null
@@ -1,11 +0,0 @@
-dh_prep
-dh_installdirs
-dh_install
-dh_installchangelogs
-dh_link
-dh_compress
-dh_fixperms
-dh_installdeb
-dh_gencontrol
-dh_md5sums
-dh_builddeb
diff --git a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone.install b/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone.install
deleted file mode 100644
index 937e53876e144e58b5c23f555deec380159f457b..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone.install
+++ /dev/null
@@ -1,2 +0,0 @@
-tmp/telify usr/share/
-sflphone-handler usr/bin/
diff --git a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone.links b/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone.links
deleted file mode 100644
index f234168dd005f777b71662153559dbe689f1aebf..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone.links
+++ /dev/null
@@ -1 +0,0 @@
-usr/share/telify usr/lib/firefox-addons/extensions/{6c5f349a-ddda-49ad-bdf0-326d3fe1f938}
diff --git a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone.links.debian b/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone.links.debian
deleted file mode 100644
index f8f52cec904e91fe248e874af8edaf64b6ff04cb..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone.links.debian
+++ /dev/null
@@ -1 +0,0 @@
-usr/share/telify usr/lib/iceweasel/extensions/{6c5f349a-ddda-49ad-bdf0-326d3fe1f938}
diff --git a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone.links.ubuntu b/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone.links.ubuntu
deleted file mode 100644
index f234168dd005f777b71662153559dbe689f1aebf..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone.links.ubuntu
+++ /dev/null
@@ -1 +0,0 @@
-usr/share/telify usr/lib/firefox-addons/extensions/{6c5f349a-ddda-49ad-bdf0-326d3fe1f938}
diff --git a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone.substvars b/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone.substvars
deleted file mode 100644
index abd3ebebc30de133ecc51d80b32908bd4a077b9a..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone.substvars
+++ /dev/null
@@ -1 +0,0 @@
-misc:Depends=
diff --git a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/DEBIAN/control b/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/DEBIAN/control
deleted file mode 100644
index b6e813410f360d6b3711eda0c47bd2f9b10a56bf..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/DEBIAN/control
+++ /dev/null
@@ -1,14 +0,0 @@
-Package: mozilla-telify-sflphone
-Version: 0.4.7.3
-Architecture: all
-Maintainer: Julien Bonjean <julien.bonjean@savoirfairelinux.com>
-Installed-Size: 1296
-Depends: firefox-gnome-support, sflphone-gnome
-Section: web
-Priority: optional
-Homepage: http://www.sflphone.org
-Description: This package provides telify firefox plugin and handler for SFLphone.
- Telify recognizes phone numbers on web pages and converts them to clickable links.
- Additionally, any text can be selected and handled as a phone number (including
- vanity number conversion) by selecting the corresponding context menu item.
- http://www.codepad.de/en/software/firefox-add-ons/telify.html
diff --git a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/DEBIAN/md5sums b/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/DEBIAN/md5sums
deleted file mode 100644
index 20effcb346c717f0a35c8187eeff044efe8baf5f..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/DEBIAN/md5sums
+++ /dev/null
@@ -1,282 +0,0 @@
-65ce74599376d092487f618de1e986fb usr/bin/sflphone-handler
-1dfa9e4bdb5667ed2452cb1842598638 usr/share/telify/chrome/content/jshashtable.js
-09146002421216f18e238dc9914df02a usr/share/telify/chrome/content/icon32.png
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-d070239a3728cafad19bb8aef31739aa usr/share/telify/chrome/content/messagebox.js
-5624cf500dd7c023075672d3fad866d6 usr/share/telify/chrome/content/config.xul
-f6ea223fbe6e7f85e00e2164546b9bd4 usr/share/telify/chrome/content/messagebox.xul
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-f4de0420b7dd240b29d42e2af3bbf50c usr/share/telify/chrome/content/util.js
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diff --git a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/DEBIAN/postinst b/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/DEBIAN/postinst
deleted file mode 100755
index 1039df3268eae8fca20a728add6c4939f7494360..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/DEBIAN/postinst
+++ /dev/null
@@ -1,16 +0,0 @@
-#!/bin/bash
-
-gconftool-2 --direct --config-source xml::/etc/gconf/gconf.xml.defaults -t string -s /desktop/gnome/url-handlers/tel/command "/usr/bin/sflphone-handler %s"
-gconftool-2 --direct --config-source xml::/etc/gconf/gconf.xml.defaults -s /desktop/gnome/url-handlers/tel/needs_terminal false -t bool
-gconftool-2 --direct --config-source xml::/etc/gconf/gconf.xml.defaults -t bool -s /desktop/gnome/url-handlers/tel/enabled true
-
-gconftool-2 --direct --config-source xml::/etc/gconf/gconf.xml.defaults -t string -s /desktop/gnome/url-handlers/callto/command "/usr/bin/sflphone-handler %s"
-gconftool-2 --direct --config-source xml::/etc/gconf/gconf.xml.defaults -s /desktop/gnome/url-handlers/callto/needs_terminal false -t bool
-gconftool-2 --direct --config-source xml::/etc/gconf/gconf.xml.defaults -t bool -s /desktop/gnome/url-handlers/callto/enabled true
-
-gconftool-2 --direct --config-source xml::/etc/gconf/gconf.xml.defaults -t string -s /desktop/gnome/url-handlers/sip/command "/usr/bin/sflphone-handler %s"
-gconftool-2 --direct --config-source xml::/etc/gconf/gconf.xml.defaults -s /desktop/gnome/url-handlers/sip/needs_terminal false -t bool
-gconftool-2 --direct --config-source xml::/etc/gconf/gconf.xml.defaults -t bool -s /desktop/gnome/url-handlers/sip/enabled true
-
-exit 0
-
diff --git a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/bin/sflphone-handler b/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/bin/sflphone-handler
deleted file mode 100755
index c3686ff8043d377d441fc3e249ad4addd2db8b56..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/bin/sflphone-handler
+++ /dev/null
@@ -1,52 +0,0 @@
-#!/bin/sh
-#
-# This script can be used as a callto: (or other) protocol handler in
-# Mozilla Firefox-based browser.
-# In Firefox use Preferences > Applications and set the callto handler
-# to this script.
-
-# The sflphone daemon config file
-RESFILE=~/.config/sflphone/sflphonedrc
-
-# Parse sflphonedrc and get default account id string
-if [ -f "$RESFILE" ]; then
-
- # Use first ID
- ACCOUNTID=`grep Accounts.order $RESFILE | sed -e 's/Accounts.order=//' -e 's/\/.*//'`
-
- # Accounts.order is not set
- if [ -z $ACCOUNTID ]; then
-
- # Use first account declared in sflphone config
- ACCOUNTID="`grep -m 1 Account: $RESFILE | sed -e 's/\[//' -e 's/\]//'`"
- fi
-
-else
- echo Fatal: Cant find sflphonedrc config file.
- exit 1
-fi
-
-# Check 1st argument (phone number)
-if [ -z $1 ]; then
- echo "Error: argument 1 (phone number) not provided."
- exit 1
-fi
-
-# Cleanup destination, keeping numbers only
-TO="`echo $1 | sed -e 's/[^0123456789]//g'`"
-
-# Generate call id.
-CALLID=${RANDOM}$$
-
-dbus-send \
- --type="method_call" \
- --dest="org.sflphone.SFLphone" \
- "/org/sflphone/SFLphone/CallManager" \
- "org.sflphone.SFLphone.CallManager.placeCall" \
- string:"$ACCOUNTID" \
- string:"$CALLID" \
- string:"$TO"
-
-exit 0
-
-# EOF
diff --git a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/doc/mozilla-telify-sflphone/changelog.gz b/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/doc/mozilla-telify-sflphone/changelog.gz
deleted file mode 100644
index 186ab5d35fe47e1e0590fbfa544a38e8604cbdfd..0000000000000000000000000000000000000000
Binary files a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/doc/mozilla-telify-sflphone/changelog.gz and /dev/null differ
diff --git a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/telify/chrome.manifest b/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/telify/chrome.manifest
deleted file mode 100644
index bc9507c8816097cc6790ce6b16406987793c0971..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/telify/chrome.manifest
+++ /dev/null
@@ -1,5 +0,0 @@
-content telify chrome/content/
-locale telify en-US chrome/locale/en-US/
-locale telify de-DE chrome/locale/de-DE/
-
-overlay chrome://browser/content/browser.xul chrome://telify/content/browser.xul
diff --git a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/telify/chrome/content/ask32.png b/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/telify/chrome/content/ask32.png
deleted file mode 100644
index d56ba2c2449cb156979af2896e82be831f258d93..0000000000000000000000000000000000000000
Binary files a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/telify/chrome/content/ask32.png and /dev/null differ
diff --git a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/telify/chrome/content/browser.xul b/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/telify/chrome/content/browser.xul
deleted file mode 100644
index 68f72fbff72784512e9051b78473baaf39b76643..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/telify/chrome/content/browser.xul
+++ /dev/null
@@ -1,87 +0,0 @@
-<?xml version="1.0"?>
-<?xml-stylesheet href="chrome://global/skin/global.css" type="text/css"?>
-<?xml-stylesheet href="chrome://telify/content/dialog.css" type="text/css"?>
-
-<!DOCTYPE overlay SYSTEM "chrome://telify/locale/lang.dtd">
-<overlay xmlns='http://www.mozilla.org/keymaster/gatekeeper/there.is.only.xul'>
-
- <stringbundleset id="stringbundleset">
- <stringbundle id="idTelifyStringBundle" src="chrome://telify/locale/lang.properties"/>
- </stringbundleset>
-
- <script type='application/x-javascript' src='chrome://telify/content/jshashtable.js'></script>
- <script type='application/x-javascript' src='chrome://telify/content/util.js'></script>
- <script type='application/x-javascript' src='chrome://telify/content/pref.js'></script>
- <script type='application/x-javascript' src='chrome://telify/locale/country_locale.js'></script>
- <script type='application/x-javascript' src='chrome://telify/content/country_data.js'></script>
- <script type='application/x-javascript' src='chrome://telify/locale/locale.js'></script>
- <script type='application/x-javascript' src='chrome://telify/content/telify.js'></script>
-
- <statusbar id="status-bar">
- <statusbarpanel id="idTelify_status" collapsed="true">
- <popup id="idTelify_status_popup" onpopupshowing="objTelify.modifyPopup(event)">
- <menuitem id="idTelify_status_activity" oncommand="objTelify.toggleActive()" />
- <menuitem id="idTelify_status_blacklist" oncommand="objTelify.toggleBlacklist()" />
- </popup>
- <hbox id="idTelify_statusicon" context="idTelify_status_popup" class="statusbarpanel-menu-iconic" src="chrome://telify/content/icon18_active.png" />
- </statusbarpanel>
- </statusbar>
-
- <menupopup id="contentAreaContextMenu">
- <menu id="idTelify_menu_context" label="&menu.selection;" collapsed="true" insertbefore="context-sep-stop"
- class="menu-iconic" image="chrome://telify/content/icon_menu.png">
- <menupopup id="idTelify_popup_context">
- <menuitem id="idTelify_context" class="menuitem-iconic"/>
- <menuseparator id="idTelify_sep_context"/>
- <menuitem id="idTelify_tld_context" class="menuitem-iconic"/>
- <menuitem id="idTelify_context0" class="menuitem-iconic"/>
- <menuitem id="idTelify_context1" class="menuitem-iconic"/>
- <menuitem id="idTelify_context2" class="menuitem-iconic"/>
- <menuitem id="idTelify_context3" class="menuitem-iconic"/>
- <menuitem id="idTelify_context4" class="menuitem-iconic"/>
- <menuitem id="idTelify_context5" class="menuitem-iconic"/>
- <menuitem id="idTelify_context6" class="menuitem-iconic"/>
- <menuitem id="idTelify_context7" class="menuitem-iconic"/>
- <menuitem id="idTelify_context8" class="menuitem-iconic"/>
- <menuitem id="idTelify_context9" class="menuitem-iconic"/>
- <menuseparator />
- <menuitem id="idTelify_edit_context" class="menuitem-iconic" label="&menu.edit_number;" image="chrome://telify/content/edit22x15.png"/>
- </menupopup>
- </menu>
- </menupopup>
-
- <popupset>
- <menupopup id="idTelify_popup_dial">
- <menuitem id="idTelify_dial" class="menuitem-iconic"/>
- <menuseparator id="idTelify_sep_dial"/>
- <menuitem id="idTelify_tld_dial" class="menuitem-iconic"/>
- <menuitem id="idTelify_dial0" class="menuitem-iconic"/>
- <menuitem id="idTelify_dial1" class="menuitem-iconic"/>
- <menuitem id="idTelify_dial2" class="menuitem-iconic"/>
- <menuitem id="idTelify_dial3" class="menuitem-iconic"/>
- <menuitem id="idTelify_dial4" class="menuitem-iconic"/>
- <menuitem id="idTelify_dial5" class="menuitem-iconic"/>
- <menuitem id="idTelify_dial6" class="menuitem-iconic"/>
- <menuitem id="idTelify_dial7" class="menuitem-iconic"/>
- <menuitem id="idTelify_dial8" class="menuitem-iconic"/>
- <menuitem id="idTelify_dial9" class="menuitem-iconic"/>
- <menuseparator />
- <menuitem id="idTelify_edit_dial" class="menuitem-iconic" label="&menu.edit_number;" image="chrome://telify/content/edit22x15.png"/>
- </menupopup>
- </popupset>
-
- <menupopup id="menu_ToolsPopup">
- <menu id="idTelify_menu" label="Telify" insertafter="devToolsSeparator">
- <menupopup onpopupshowing="objTelify.modifyPopup(event)">
- <menuitem id="idTelify_menu_activity" oncommand="objTelify.toggleActive()"/>
- <menuitem id="idTelify_menu_blacklist" oncommand="objTelify.toggleBlacklist()"/>
- <menuseparator />
- <menuitem id="idTelify_menu_config" label="&menu.onlinehelp;" oncommand="objTelifyLocale.openOnlineHelp()"/>
- <menuseparator />
- <menuitem id="idTelify_menu_config" label="&menu.config;" oncommand="objTelifyPrefs.showConfigDialog()"/>
- </menupopup>
- </menu>
- </menupopup>
-
-</overlay>
-
diff --git a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/telify/chrome/content/config.js b/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/telify/chrome/content/config.js
deleted file mode 100644
index 25e529121ac47f94550cb2413dbf33e93e3f8360..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/telify/chrome/content/config.js
+++ /dev/null
@@ -1,196 +0,0 @@
-/*
-Creative Commons License: Attribution-No Derivative Works 3.0 Unported
-http://creativecommons.org/licenses/by-nd/3.0/
-(c)2009 Michael Koch
-*/
-
-var objTelifyConfig = {
-
-tmplIndex: 0,
-customLabelDefault: "",
-
-setConfigValues: function()
-{
- objTelifyPrefs.telPrefs.setCharPref(objTelifyPrefs.PREF_IDD_PREFIX, document.getElementById("idTelifyPref_idd_prefix").value);
- objTelifyPrefs.telPrefs.setIntPref(objTelifyPrefs.PREF_HREFTYPE, document.getElementById("idTelifyPref_hreftype").value);
- objTelifyPrefs.telPrefs.setIntPref(objTelifyPrefs.PREF_HIGHLIGHT, document.getElementById("idTelifyPref_highlight").value);
- objTelifyPrefs.telPrefs.setIntPref(objTelifyPrefs.PREF_NUMHISTORY, document.getElementById("idTelifyPref_num_history").value);
- objTelifyPrefs.telPrefs.setBoolPref(objTelifyPrefs.PREF_STATUSICON, document.getElementById("idTelifyPref_statusicon").value == 1);
- objTelifyPrefs.telPrefs.setBoolPref(objTelifyPrefs.PREF_DIAL_CC_DIRECT, document.getElementById("idTelifyPref_dialcc").value == 1);
-
- objTelifyPrefs.telPrefs.setCharPref(objTelifyPrefs.PREF_CUSTOM_URL, document.getElementById("idTelifyPref_url_input").value);
- objTelifyPrefs.telPrefs.setIntPref(objTelifyPrefs.PREF_CUSTOM_TMPL, this.tmplIndex);
- for (var i=1; i<objTelifyPrefs.NUM_CUSTOM_PARAMS+1; i++) {
- objTelifyPrefs.telPrefs.setCharPref(objTelifyPrefs.PREF_CUSTOM_PARAM+i, document.getElementById("idTelifyPref_param"+i+"_value").value);
- }
- objTelifyPrefs.telPrefs.setIntPref(objTelifyPrefs.PREF_CUSTOM_OPENTYPE, document.getElementById("idTelifyPref_opentype").value);
-},
-
-onAccept: function()
-{
- this.setConfigValues();
- return true;
-},
-
-onHelp: function()
-{
- objTelifyLocale.openOnlineHelp();
- return true;
-},
-
-initConfig: function()
-{
- objTelifyPrefs.initTelifyPrefs();
- document.getElementById("idTelifyPref_idd_prefix").value = objTelifyPrefs.idd_prefix;
- document.getElementById("idTelifyPref_hreftype").value = objTelifyPrefs.hrefType;
- this.hrefTypeChanged(objTelifyPrefs.hrefType);
- document.getElementById("idTelifyPref_highlight").value = objTelifyPrefs.highlight;
- document.getElementById("idTelifyPref_num_history").value = objTelifyPrefs.numHistory;
- document.getElementById("idTelifyPref_statusicon").value = (objTelifyPrefs.fStatusIcon ? 1 : 0);
- document.getElementById("idTelifyPref_dialcc").value = (objTelifyPrefs.fDialCCDirect ? 1 : 0);
-
- document.getElementById("idTelifyPref_url_input").value = objTelifyPrefs.custom_url;
- this.tmplIndex = objTelifyPrefs.custom_tmpl;
- for (var i=1; i<objTelifyPrefs.NUM_CUSTOM_PARAMS+1; i++) {
- document.getElementById("idTelifyPref_param"+i+"_value").value = objTelifyPrefs.custom_param[i];
- }
- document.getElementById("idTelifyPref_opentype").value = objTelifyPrefs.custom_opentype;
-
- this.customLabelDefault = document.getElementById("idTelifyPref_custom_caption").label
-
- var popup = document.getElementById("idTelifyPref_url_popup");
- for (var i=0; i<telify_custom_preset.length; i++) {
- var item = document.createElement("menuitem");
- item.setAttribute("label", telify_custom_preset[i][0]);
- popup.appendChild(item);
- }
-
- this.setTemplate(this.tmplIndex, true);
-
- document.getElementById("idTelifyPref_version_label").value = "Telify v"+objTelifyUtil.getAddonVersion();
-},
-
-getTemplateParam: function(nr)
-{
- if (nr == 0) return objTelifyPrefs.telStrings.getString("phonenr_tmpl");
- var param = document.getElementById("idTelifyPref_param"+nr+"_value").value;
- var label = document.getElementById("idTelifyPref_param"+nr+"_caption").value;
- if (label.value == "") param = "";
- return param;
-},
-
-createResultDOM: function(node)
-{
- if (node == null) return 0; // safety
- if (node.nodeType == Node.TEXT_NODE) {
- var text = node.data;
- var len = text.length;
- var escape = 0;
- for (var i=0; i<len-1; i++) {
- if (escape == 1) {escape = 0; continue;}
- var c = text.charAt(i);
- if (c == '\\') {escape = 1; continue}
- if (c != '$') continue;
- c = text.charAt(i+1);
- var nr = "0123456789".indexOf(c);
- if (nr < 0 || nr > objTelifyPrefs.NUM_CUSTOM_PARAMS) continue;
- var prev_node = document.createTextNode(text.substr(0, i));
- var next_node = document.createTextNode(text.substr(i+2));
- var hilite_node = document.createElement("span");
- hilite_node.setAttribute("class", (nr == 0 ? "tmpl_number" : "tmpl_param"));
- var param_node = document.createTextNode(this.getTemplateParam(nr));
- hilite_node.appendChild(param_node);
- var parentNode = node.parentNode;
- parentNode.replaceChild(next_node, node);
- parentNode.insertBefore(hilite_node, next_node);
- parentNode.insertBefore(prev_node, hilite_node);
- break;
- }
- } else {
- for (var i=0; i<node.childNodes.length; i++) {
- this.createResultDOM(node.childNodes[i]);
- }
- }
-},
-
-urlChanged: function()
-{
- var url = document.getElementById("idTelifyPref_url_input").value;
- var result = document.getElementById("idTelifyPref_url_result");
- while (result.childNodes[0]) result.removeChild(result.childNodes[0]);
- if (url == "") {
- var item = document.createElement("span");
- var empty_url = objTelifyPrefs.telStrings.getString("empty_url")
- item.appendChild(document.createTextNode(empty_url));
- item.setAttribute("class", "tmpl_empty");
- result.appendChild(item);
- } else {
- var item = document.createTextNode(url);
- result.appendChild(item);
- this.createResultDOM(result);
- }
-},
-
-setTemplate: function(nr, init)
-{
- var caption = document.getElementById("idTelifyPref_custom_caption");
- caption.label = this.customLabelDefault;
- if (telify_custom_preset[nr][0].length) caption.label += " ("+telify_custom_preset[nr][0]+")";
- if (!init) document.getElementById("idTelifyPref_url_input").value = telify_custom_preset[nr][1];
- for (var j=0; j<objTelifyPrefs.NUM_CUSTOM_PARAMS; j++) {
- var label = document.getElementById("idTelifyPref_param"+(j+1)+"_caption");
- var param = document.getElementById("idTelifyPref_param"+(j+1)+"_value");
- var row = document.getElementById("idTelifyPref_param"+(j+1)+"_row");
- label.value = telify_custom_preset[nr][2+j];
- if (label.value != "") label.value += ":";
- if (label.value == "") param.setAttribute("disabled", true); else param.removeAttribute("disabled");
- }
- this.urlChanged();
-},
-
-tmplChanged: function()
-{
- var obj = document.getElementById("idTelifyPref_url_input");
- for (var i=0; i<telify_custom_preset.length; i++) {
- if (obj.value == telify_custom_preset[i][0]) {
- this.tmplIndex = i;
- this.setTemplate(i, false);
- break;
- }
- }
-},
-
-paramChanged: function(nr, value)
-{
- this.urlChanged();
-},
-
-enableDOMTree: function(node, enable)
-{
- if (node == null) return;
- if (enable) {
- if (node.removeAttribute) node.removeAttribute("disabled");
- } else {
- if (node.setAttribute) node.setAttribute("disabled", true);
- }
- for (var i=0; i<node.childNodes.length; i++) {
- this.enableDOMTree(node.childNodes[i], enable);
- }
-},
-
-hrefTypeChanged: function(nr)
-{
- var group = document.getElementById("idTelifyPref_custom_group");
- if (nr == objTelifyPrefs.HREFTYPE_CUSTOM) {
- group.removeAttribute("collapsed");
- window.sizeToContent();
- } else {
- //alert(group.clientHeight);
- group.setAttribute("collapsed", true);
- //window.resizeTo(500, 500);
- window.resizeBy(0, -200);
- window.sizeToContent();
- }
-}
-
-};
diff --git a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/telify/chrome/content/config.xul b/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/telify/chrome/content/config.xul
deleted file mode 100644
index 967d4db7227e48bde4587aa0b59a409aedd6aa33..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/telify/chrome/content/config.xul
+++ /dev/null
@@ -1,180 +0,0 @@
-<?xml version="1.0"?>
-<?xml-stylesheet href="chrome://global/skin/global.css" type="text/css"?>
-<?xml-stylesheet href="chrome://telify/content/dialog.css" type="text/css"?>
-
-<!DOCTYPE dialog SYSTEM "chrome://telify/locale/lang.dtd">
-
-<dialog id="dlgTelifyConfig"
- xmlns="http://www.mozilla.org/keymaster/gatekeeper/there.is.only.xul"
- buttons="accept,cancel,help"
- onload="objTelifyConfig.initConfig()"
- ondialogaccept="objTelifyConfig.onAccept()"
- ondialogcancel=""
- ondialoghelp="objTelifyConfig.onHelp()"
- title="&dialog.config.title;">
-
- <stringbundleset id="stringbundleset">
- <stringbundle id="idTelifyStringBundle" src="chrome://telify/locale/lang.properties"/>
- </stringbundleset>
-
- <script type='application/x-javascript' src='chrome://telify/content/util.js'></script>
- <script type='application/x-javascript' src='chrome://telify/content/pref.js'></script>
- <script type='application/x-javascript' src='chrome://telify/content/config.js'></script>
- <script type='application/x-javascript' src='chrome://telify/locale/locale.js'></script>
- <script type='application/x-javascript' src='chrome://telify/locale/custom_preset.js'></script>
-
- <hbox>
-
- <groupbox style="padding-bottom:8px;">
- <caption label="&dialog.config.general;"/>
- <grid>
- <columns>
- <column flex="1"/>
- <column flex="2"/>
- </columns>
- <rows>
- <row align="center">
- <label value="&dialog.config.hreftype;:"/>
- <menulist id="idTelifyPref_hreftype" onselect="objTelifyConfig.hrefTypeChanged(this.value)">
- <menupopup>
- <menuitem label="&dialog.config.hreftype0;" value="0"/>
- <menuitem label="&dialog.config.hreftype1;" value="1"/>
- <menuitem label="&dialog.config.hreftype2;" value="2"/>
- <menuitem label="&dialog.config.hreftype3;" value="3"/>
- <menuitem label="&dialog.config.hreftype_custom;" value="9"/>
- </menupopup>
- </menulist>
- </row>
-
- <row align="center">
- <label value="&dialog.config.dialcc;:"/>
- <menulist id="idTelifyPref_dialcc">
- <menupopup>
- <menuitem label="&dialog.config.dialcc_menu;" value="0"/>
- <menuitem label="&dialog.config.dialcc_direct;" value="1"/>
- </menupopup>
- </menulist>
- </row>
-
- <row align="center">
- <label value="&dialog.config.highlight;:"/>
- <menulist id="idTelifyPref_highlight">
- <menupopup>
- <menuitem label="&dialog.config.highlight0;" value="0"/>
- <menuitem label="&dialog.config.highlight1;" value="25"/>
- <menuitem label="&dialog.config.highlight2;" value="50"/>
- <menuitem label="&dialog.config.highlight3;" value="100"/>
- </menupopup>
- </menulist>
- </row>
- <row align="center">
- <label value="&dialog.config.num_history;:"/>
- <menulist id="idTelifyPref_num_history">
- <menupopup>
- <menuitem label="3" value="3"/>
- <menuitem label="4" value="4"/>
- <menuitem label="5" value="5"/>
- <menuitem label="6" value="6"/>
- <menuitem label="7" value="7"/>
- <menuitem label="8" value="8"/>
- <menuitem label="9" value="9"/>
- <menuitem label="10" value="10"/>
- </menupopup>
- </menulist>
- </row>
- <row align="center">
- <label value="&dialog.config.statusicon;:"/>
- <menulist id="idTelifyPref_statusicon">
- <menupopup>
- <menuitem label="&dialog.config.statusicon0;" value="0" />
- <menuitem label="&dialog.config.statusicon1;" value="1" />
- </menupopup>
- </menulist>
- </row>
- <row align="center">
- <label value="&dialog.config.idd_prefix;"/>
- <menulist id="idTelifyPref_idd_prefix" editable="true">
- <menupopup>
- <menuitem label="" value=""/>
- <menuitem label="00" value="00"/>
- <menuitem label="001" value="001"/>
- <menuitem label="011" value="011"/>
- <menuitem label="0011" value="0011"/>
- </menupopup>
- </menulist>
- </row>
- </rows>
- </grid>
- </groupbox>
-
- <groupbox style="padding-bottom:8px;">
- <caption label="&dialog.config.about;"/>
- <vbox style="width:96px;">
- <spacer style="height:0px;"/>
- <hbox>
- <image src="chrome://telify/content/icon96.png" style="width:96px;height:96px;margin-left:0px;"/>
- <spacer/>
- </hbox>
- <spacer style="height:4px;"/>
- <label id="idTelifyPref_version_label" value="" style="font-weight:bold;"/>
- <label value="www.codepad.de" href="http://www.codepad.de" class="text-link"/>
- <spacer flex="1"/>
- </vbox>
- </groupbox>
-
- </hbox>
-
- <groupbox id="idTelifyPref_custom_group" style="padding-bottom:8px;">
- <caption id="idTelifyPref_custom_caption" label="&dialog.config.custom;"/>
- <vbox>
- <description id="idTelifyPref_url_result" class="urlresult">
- </description>
-
- <menulist id="idTelifyPref_url_input" editable="true"
- oninput="objTelifyConfig.urlChanged(this.value)" onselect="objTelifyConfig.tmplChanged(this.value)">
- <menupopup id="idTelifyPref_url_popup">
- </menupopup>
- </menulist>
-
- <grid>
- <columns>
- <column flex="0"/>
- <column flex="2"/>
- <column flex="3"/>
- </columns>
- <rows>
- <row align="center" id="idTelifyPref_param1_row" style="margin-top:4px;">
- <label id="idTelifyPref_param1_caption"/>
- <textbox id="idTelifyPref_param1_value" emptytext="&dialog.config.replaces; $1 &dialog.config.in_template;"
- oninput="objTelifyConfig.paramChanged(1, this.value)"/>
- <spacer flex="2"/>
- </row>
- <row align="center" id="idTelifyPref_param2_row">
- <label id="idTelifyPref_param2_caption"/>
- <textbox id="idTelifyPref_param2_value" emptytext="&dialog.config.replaces; $2 &dialog.config.in_template;"
- oninput="objTelifyConfig.paramChanged(2, this.value)"/>
- </row>
- <row align="center" id="idTelifyPref_param3_row">
- <label id="idTelifyPref_param3_caption"/>
- <textbox id="idTelifyPref_param3_value" emptytext="&dialog.config.replaces; $3 &dialog.config.in_template;"
- oninput="objTelifyConfig.paramChanged(3, this.value)"/>
- </row>
- <row align="center" style="margin-top:4px;">
- <label value="&dialog.config.opentype;:"/>
- <menulist id="idTelifyPref_opentype">
- <menupopup>
- <menuitem label="&dialog.config.opentype0;" value="0"/>
- <menuitem label="&dialog.config.opentype1;" value="1"/>
- <menuitem label="&dialog.config.opentype2;" value="2"/>
- <menuitem label="&dialog.config.opentype3;" value="3"/>
- </menupopup>
- </menulist>
- </row>
- </rows>
- </grid>
-
-
- </vbox>
- </groupbox>
-
-</dialog>
diff --git a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/telify/chrome/content/country_data.js b/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/telify/chrome/content/country_data.js
deleted file mode 100644
index 63acf6089a1a45b873005a49f7933cb9b7790743..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/telify/chrome/content/country_data.js
+++ /dev/null
@@ -1,258 +0,0 @@
-/*
-Creative Commons License: Attribution-No Derivative Works 3.0 Unported
-http://creativecommons.org/licenses/by-nd/3.0/
-(c)2009 Michael Koch
-*/
-// must be saved as UTF-8
-var telify_country_data = [
-["", "", "", ""],
-["+1", "USA", "us,mil,gov,edu", "1"],
-["+1340", "U.S. Virgin Islands", "vi", "1"],
-["+1670", "Northern Mariana Islands", "mp", "1"],
-["+1671", "Guam", "gu", "1"],
-["+1684", "American Samoa", "as", "1"],
-["+1787", "Puerto Rico", "pr", "1"],
-["+1939", "Puerto Rico", "", "1"],
-["+1", "Canada", "ca", "1"],
-["+1264", "Anguilla", "ai", "1"],
-["+1268", "Antigua and Barbuda", "ag", "1"],
-["+1242", "Bahamas", "bs", "1"],
-["+1246", "Barbados", "bb", "1"],
-["+1441", "Bermuda", "bm", "1"],
-["+1284", "British Virgin Islands", "vg", "1"],
-["+1345", "Cayman Islands", "ky", "1"],
-["+1767", "Dominica", "dm", "1"],
-["+1808", "Midway Island", "", "1"],
-["+1809", "Dominican Republic", "do", "1"],
-["+1829", "Dominican Republic", "", "1"],
-["+1849", "Dominican Republic", "", "1"],
-["+1473", "Grenada", "gd", "1"],
-["+1876", "Jamaica", "jm", "1"],
-["+1664", "Montserrat", "ms", "1"],
-["+1869", "Saint Kitts and Nevis", "kn", "1"],
-["+1758", "Saint Lucia", "lc", "1"],
-["+1784", "Saint Vincent and the Grenadines", "vc", "1"],
-["+1868", "Trinidad and Tobago", "tt", "1"],
-["+1649", "Turks and Caicos Islands", "tc", "1"],
-["+20", "Egypt", "eg", "0"],
-["+212", "Morocco", "ma", ""],
-["+213", "Algeria", "dz", "7"],
-["+216", "Tunisia", "tn", "0"],
-["+218", "Libya", "ly", "0"],
-["+220", "Gambia", "gm", ""],
-["+221", "Senegal", "sn", "0"],
-["+222", "Mauritania", "mr", "0"],
-["+223", "Mali", "ml", "0"],
-["+224", "Guinea", "gn", "0"],
-["+225", "Ivory Coast", "ci", "0"],
-["+226", "Burkina Faso", "bf", ""],
-["+227", "Niger", "ne", "0"],
-["+228", "Togo", "tg", ""],
-["+229", "Benin", "bj", ""],
-["+230", "Mauritius", "mu", "0"],
-["+231", "Liberia", "lr", "22"],
-["+232", "Sierra Leone", "sl", "0"],
-["+233", "Ghana", "gh", ""],
-["+234", "Nigeria", "ng", "0"],
-["+235", "Chad", "td", ""],
-["+236", "Central African Republic", "cf", ""],
-["+237", "Cameroon", "cm", ""],
-["+238", "Cape Verde", "cv", ""],
-["+239", "São Tomé and Príncipe", "st", "0"],
-["+240", "Equatorial Guinea", "gq", ""],
-["+241", "Gabon", "ga", ""],
-["+242", "Congo (Republic)", "cg", ""],
-["+243", "Congo (Democratic Republic)", "cd", ""],
-["+244", "Angola", "ao", "0"],
-["+245", "Guinea-Bissau", "gw", ""],
-["+246", "Diego Garcia", "", ""],
-["+247", "Ascension Island", "ac", ""],
-["+248", "Seychelles", "sc", "0"],
-["+249", "Sudan", "sd", "0"],
-["+250", "Rwanda", "rw", "0"],
-["+251", "Ethiopia", "et", "0"],
-["+252", "Somalia", "so", ""],
-["+253", "Djibouti", "dj", ""],
-["+254", "Kenya", "ke", "0"],
-["+255", "Tanzania", "tz", "0"],
-["+256", "Uganda", "ug", "0"],
-["+257", "Burundi", "bi", ""],
-["+258", "Mozambique", "mz", "0"],
-["+260", "Zambia", "zm", "0"],
-["+261", "Madagascar", "mg", "0"],
-["+262", "Réunion", "re", "0"],
-["+262", "Mayotte", "yt", "0"],
-["+263", "Zimbabwe", "zw", "0"],
-["+264", "Namibia", "na", "0"],
-["+265", "Malawi", "mw", ""],
-["+266", "Lesotho", "ls", "0"],
-["+267", "Botswana", "bw", ""],
-["+268", "Swaziland", "sz", ""],
-["+269", "Comoros", "km", ""],
-["+27", "South Africa", "za", "0"],
-["+290", "Saint Helena", "sh", ""],
-["+290", "Tristan da Cunha", "", "0"],
-["+291", "Eritrea", "er", "0"],
-["+297", "Aruba", "aw", ""],
-["+298", "Faroe Islands", "fo", ""],
-["+299", "Greenland", "gl", ""],
-["+30", "Greece", "gr", ""],
-["+31", "Netherlands", "nl", "0"],
-["+32", "Belgium", "be", "0"],
-["+33", "France", "fr", "0"],
-["+34", "Spain", "es", "0"],
-["+350", "Gibraltar", "gi", ""],
-["+351", "Portugal", "pt", ""],
-["+352", "Luxembourg", "lu", ""],
-["+353", "Ireland", "ie", "0"],
-["+354", "Iceland", "is", "0"],
-["+355", "Albania", "al", "0"],
-["+356", "Malta", "mt", "0"],
-["+357", "Cyprus (South)", "cy", ""],
-["+358", "Finland", "fi", "0"],
-["+359", "Bulgaria", "bg", "0"],
-["+36", "Hungary", "hu", "06"],
-["+370", "Lithuania", "lt", "8"],
-["+371", "Latvia", "lv", "8"],
-["+372", "Estonia", "ee", ""],
-["+373", "Moldova", "md", "0"],
-["+374", "Armenia", "am", "8"],
-["+37447", "Nagorno-Karabakh", "", "0"],
-["+37497", "Nagorno-Karabakh (Mobile)", "", "0"],
-["+375", "Belarus", "by", "8"],
-["+376", "Andorra", "ad", ""],
-["+377", "Monaco", "mc", "0"],
-["+37744", "Kosovo (Mobile)", "", "0"],
-["+378", "San Marino", "sm", "0"],
-["+379", "Vatican City", "va", ""],
-["+380", "Ukraine", "ua", "8"],
-["+381", "Serbia", "rs", "0"],
-["+381", "Kosovo", "", "0"],
-["+382", "Montenegro", "me", "0"],
-["+385", "Croatia", "hr", "0"],
-["+386", "Slovenia", "si", "0"],
-["+38649", "Kosovo (Mobile)", "", "0"],
-["+387", "Bosnia and Herzegovina", "ba", "0"],
-["+389", "Macedonia", "mk", "0"],
-["+39", "Italy and Vatican City", "it", ""],
-["+40", "Romania", "ro", "0"],
-["+41", "Switzerland", "ch", "0"],
-["+420", "Czech Republic", "cz", ""],
-["+421", "Slovakia", "sk", "0"],
-["+423", "Liechtenstein", "li", ""],
-["+43", "Austria", "at", "0"],
-["+44", "United Kingdom", "uk,gb", "0"],
-["+45", "Denmark", "dk", ""],
-["+46", "Sweden", "se", "0"],
-["+47", "Norway", "no", ""],
-["+48", "Poland", "pl", "0"],
-["+49", "Germany", "de", "0"],
-["+500", "Falkland Islands", "fk", ""],
-["+501", "Belize", "bz", "0"],
-["+502", "Guatemala", "gt", ""],
-["+503", "El Salvador", "sv", ""],
-["+504", "Honduras", "hn", "0"],
-["+505", "Nicaragua", "ni", "0"],
-["+506", "Costa Rica", "cr", ""],
-["+507", "Panama", "pa", "0"],
-["+508", "Saint-Pierre and Miquelon", "pm", "0"],
-["+509", "Haiti", "ht", "0"],
-["+51", "Peru", "pe", "0"],
-["+52", "Mexico", "mx", "01"],
-["+53", "Cuba", "cu", "0"],
-["+5399", "Guantanamo Bay", "", "0"],
-["+54", "Argentina", "ar", "0"],
-["+55", "Brazil", "br", "0"],
-["+56", "Chile", "cl", "0"],
-["+57", "Colombia", "co", "0"],
-["+58", "Venezuela", "ve", "0"],
-["+590", "Guadeloupe", "gp", ""],
-["+591", "Bolivia", "bo", "0"],
-["+592", "Guyana", "gy", "0"],
-["+593", "Ecuador", "ec", "0"],
-["+594", "French Guiana", "gf", ""],
-["+595", "Paraguay", "py", "0"],
-["+596", "Martinique", "mq", ""],
-["+597", "Suriname", "sr", ""],
-["+598", "Uruguay", "uy", "0"],
-["+599", "Netherlands Antilles", "an", ""],
-["+60", "Malaysia", "my", "0"],
-["+61", "Australia", "au", "0"],
-["+62", "Indonesia", "id", "0"],
-["+63", "Philippines", "ph", "0"],
-["+64", "New Zealand", "nz", ""],
-["+65", "Singapore", "sg", ""],
-["+66", "Thailand", "th", "0"],
-["+670", "East Timor", "tp,tl", ""],
-["+672", "Australian external territories", "", ""],
-["+673", "Brunei", "bn", "0"],
-["+674", "Nauru", "nr", "0"],
-["+675", "Papua New Guinea", "pg", ""],
-["+676", "Tonga", "to", ""],
-["+677", "Solomon Islands", "sb", ""],
-["+678", "Vanuatu", "vu", ""],
-["+679", "Fiji", "fj", ""],
-["+680", "Palau", "pw", ""],
-["+681", "Wallis and Futuna", "wf", ""],
-["+682", "Cook Islands", "ck", "00"],
-["+683", "Niue Island", "nu", "0"],
-["+685", "Samoa", "ws", ""],
-["+686", "Kiribati", "ki", "0"],
-["+687", "New Caledonia", "nc", "0"],
-["+688", "Tuvalu", "tv", ""],
-["+689", "French Polynesia", "pf", ""],
-["+690", "Tokelau", "tk", ""],
-["+691", "Micronesia", "fm", "1"],
-["+692", "Marshall Islands", "mh", "1"],
-["+7", "Russia", "ru,su", "8"],
-["+7", "Kazakhstan", "kz", "8"],
-["+81", "Japan", "jp", "0"],
-["+82", "South Korea", "kr", "0"],
-["+84", "Vietnam", "vn", "0"],
-["+850", "North Korea", "", "0"],
-["+852", "Hong Kong", "hk", ""],
-["+853", "Macao", "mo", "0"],
-["+855", "Cambodia", "kh", "0"],
-["+856", "Laos", "la", "0"],
-["+86", "China", "cn", "0"],
-["+870", "Inmarsat SNAC", "", ""],
-["+871", "Inmarsat (Atlantic East)", "", ""],
-["+872", "Inmarsat (Pacific)", "", ""],
-["+873", "Inmarsat (Indian)", "", ""],
-["+874", "Inmarsat (Atlantic West)", "", ""],
-["+880", "Bangladesh", "bd", "0"],
-["+881", "Global Mobile Satellite System", "", ""],
-["+882", "International Networks", "", ""],
-["+883", "International Networks", "", ""],
-["+886", "Taiwan", "tw", ""],
-["+90", "Turkey", "tr", "0"],
-["+90392", "Cyprus (North)", "", "0"],
-["+91", "India", "in", "0"],
-["+92", "Pakistan", "pk", "0"],
-["+93", "Afghanistan", "af", "0"],
-["+94", "Sri Lanka", "lk", "0"],
-["+95", "Burma", "mm", ""],
-["+960", "Maldives", "mv", "0"],
-["+961", "Lebanon", "lb", "0"],
-["+962", "Jordan", "jo", "0"],
-["+963", "Syria", "sy", "0"],
-["+964", "Iraq", "iq", "0"],
-["+965", "Kuwait", "kw", "0"],
-["+966", "Saudi Arabia", "sa", "0"],
-["+967", "Yemen", "ye", "0"],
-["+968", "Oman", "om", "0"],
-["+971", "United Arab Emirates", "ae", ""],
-["+972", "Israel", "il", "0"],
-["+973", "Bahrain", "bh", ""],
-["+974", "Qatar", "qa", "0"],
-["+975", "Bhutan", "bt", ""],
-["+976", "Mongolia", "mn", "0"],
-["+977", "Nepal", "np", "0"],
-["+98", "Iran", "ir", "0"],
-["+992", "Tajikistan", "tj", "8"],
-["+993", "Turkmenistan", "tm", "8"],
-["+994", "Azerbaijan", "az", "8"],
-["+995", "Georgia", "ge", "8"],
-["+996", "Kyrgyzstan", "kg", "0"],
-["+998", "Uzbekistan", "uz", "8"],
-];
diff --git a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/telify/chrome/content/dialog.css b/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/telify/chrome/content/dialog.css
deleted file mode 100644
index 89a400e72fefeb04e680fcb1138f7129feaaeefa..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/telify/chrome/content/dialog.css
+++ /dev/null
@@ -1,33 +0,0 @@
-.telInputCC {
- width: 6em;
-}
-
-#idTelify_popup_dial .menu-iconic-icon {
- width: 22px;
- height: 15px;
-}
-
-#idTelify_popup_context .menu-iconic-icon {
- width: 22px;
- height: 15px;
-}
-
-.tmpl_empty {
- color: #a0a0a0;
-}
-
-.tmpl_number {
- color: #008000;
-}
-
-.tmpl_param {
- color: #000080;
-}
-
-.urlresult {
- height:4.2em;
- padding:2px 3px;
- border:1px solid #e0e0e0;
- background-color:#f0f0f0;
- margin-left:4px;
-}
diff --git a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/telify/chrome/content/edit22x15.png b/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/telify/chrome/content/edit22x15.png
deleted file mode 100644
index cb4c614f0b8250a5424018630cc4663962591633..0000000000000000000000000000000000000000
Binary files a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/telify/chrome/content/edit22x15.png and /dev/null differ
diff --git a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/telify/chrome/content/editNumber.js b/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/telify/chrome/content/editNumber.js
deleted file mode 100644
index 64896ad5c87f04ff0cf1581be739da27fb571215..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/telify/chrome/content/editNumber.js
+++ /dev/null
@@ -1,180 +0,0 @@
-/*
-Creative Commons License: Attribution-No Derivative Works 3.0 Unported
-http://creativecommons.org/licenses/by-nd/3.0/
-(c)2009 Michael Koch
-*/
-
-var objTelifyEditNumber = {
-
-checkKey: function(event, allowed)
-{
- if (event.which < 32) return
- var key = String.fromCharCode(event.which)
- if (allowed.indexOf(key) >= 0) return;
- event.preventDefault();
-},
-
-
-createListItem: function()
-{
- var item = document.createElement('listitem');
- for (var i=0; i<arguments.length; i++) {
- var cell = document.createElement('listcell');
- cell.setAttribute("label", arguments[i]);
- item.appendChild(cell);
- }
- return item;
-},
-
-
-updateCountrySelection: function()
-{
- var list = document.getElementById("idTelifyCountryCodeList");
- var editcc = document.getElementById("idTelifyInputCC");
- if (editcc.value == "" || editcc.value == "+" || editcc.value.charAt(0) != '+') {
- list.scrollToIndex(0);
- list.selectedIndex = 0;
- editcc.style.color = "#ff0000";
- return;
- }
- var index = 0;
- var maxlen = 1;
- for (var i=0; i<telify_country_data.length; i++) {
- if (editcc.value == telify_country_data[i][0]) {
- index = i;
- break;
- }
- for (var j=1; j<editcc.value.length; j++) {
- if (editcc.value.charAt(j) == telify_country_data[i][0].charAt(j)) {
- if (j+1 > maxlen) {
- maxlen = j+1;
- index = i;
- }
- } else {
- break;
- }
- }
- }
- if (index >= 0) {
- list.scrollToIndex(index);
- if (editcc.value == telify_country_data[index][0]) {
- list.selectedIndex = index;
- editcc.style.color = "#000000";
- } else {
- list.clearSelection();
- editcc.style.color = "#ff0000";
- }
- } else {
- list.scrollToIndex(0);
- list.clearSelection();
- editcc.style.color = "#ff0000";
- }
-},
-
-
-ccChanged: function()
-{
- var editcc = document.getElementById("idTelifyInputCC");
- if (editcc.value.length == 1 && editcc.value.charAt(0) != '+') {
- editcc.value = "+" + editcc.value;
- }
- this.updateCountrySelection();
-},
-
-
-updateNumberEdit: function()
-{
- var list = document.getElementById("idTelifyCountryCodeList");
- var fClear = false;
- if (list.getRowCount() != telify_country_data.length) {
- while (list.getRowCount() > 0) list.removeItemAt(0);
- fClear = true;
- }
- for (var i=0; i<telify_country_data.length; i++) {
- var item = this.createListItem(telify_country_data[i][0], telify_country_data[i][1]);
- if (fClear) {
- list.appendChild(item);
- } else {
- list.replaceChild(item, list.getItemAtIndex(i));
- }
- }
- this.updateCountrySelection();
-},
-
-
-updateListSelection: function()
-{
- var list = document.getElementById("idTelifyCountryCodeList");
- var editcc = document.getElementById("idTelifyInputCC");
- if (list.selectedCount > 0) {
- editcc.value = telify_country_data[list.selectedIndex][0];
- editcc.style.color = "#000000";
- }
-},
-
-
-compareCol1: function(a, b)
-{
- var v = a[0].localeCompare(b[0]);
- if (v == 0) return a[1].localeCompare(b[1]);
- return v;
-},
-
-
-compareCol2: function(a, b)
-{
- var v = a[1].localeCompare(b[1]);
- if (v == 0) return a[0].localeCompare(b[0]);
- return v;
-},
-
-
-last_sorted_column: -1,
-
-sortCountryCodeList: function(column)
-{
- var telPrefs = objTelifyPrefs.getPrefObj();
- if (column < 0) {
- column = telPrefs.getIntPref(objTelifyPrefs.PREF_COLSORTCC);
- } else {
- telPrefs.setIntPref(objTelifyPrefs.PREF_COLSORTCC, column);
- }
- if (column == this.last_sorted_column) return;
- if (column == 0) {
- telify_country_data.sort(this.compareCol1);
- document.getElementById("idTelifyColCode").setAttribute("sortDirection", "descending");
- document.getElementById("idTelifyColCountry").setAttribute("sortDirection", "natural");
- }
- if (column == 1) {
- telify_country_data.sort(this.compareCol2);
- document.getElementById("idTelifyColCode").setAttribute("sortDirection", "natural");
- document.getElementById("idTelifyColCountry").setAttribute("sortDirection", "descending");
- }
- this.last_sorted_column = column;
- this.updateNumberEdit();
-},
-
-
-setNumberEditReturnValue: function(fOK)
-{
- window.arguments[0].cc = document.getElementById("idTelifyInputCC").value;
- window.arguments[0].nr = document.getElementById("idTelifyInputNr").value;
- window.arguments[0].fOK = fOK;
-},
-
-
-initNumberEdit: function()
-{
- var cc = window.arguments[0].cc;
- var nr = window.arguments[0].nr;
- var index = -1;
- var maxlen = 0;
-
- objTelifyUtil.localizeCountryData();
- document.getElementById("idTelifyInputCC").value = (cc ? cc : "");
- document.getElementById("idTelifyInputNr").value = nr;
- this.sortCountryCodeList(-1);
-}
-
-};
-
diff --git a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/telify/chrome/content/editNumber.xul b/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/telify/chrome/content/editNumber.xul
deleted file mode 100644
index 251fbc0198542f60424265bec1b80426e10c519f..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/telify/chrome/content/editNumber.xul
+++ /dev/null
@@ -1,45 +0,0 @@
-<?xml version="1.0"?>
-<?xml-stylesheet href="chrome://global/skin/global.css" type="text/css"?>
-<?xml-stylesheet href="chrome://telify/content/dialog.css" type="text/css"?>
-
-<!DOCTYPE dialog SYSTEM "chrome://telify/locale/lang.dtd">
-
-<dialog id="dlgTelifyEditNumber"
- xmlns="http://www.mozilla.org/keymaster/gatekeeper/there.is.only.xul"
- buttons="accept,cancel"
- buttonlabelaccept="&dialog.edit.dial;"
- onload="objTelifyEditNumber.initNumberEdit()"
- ondialogaccept="objTelifyEditNumber.setNumberEditReturnValue(true)"
- ondialogcancel="objTelifyEditNumber.setNumberEditReturnValue(false)"
- title="&dialog.edit.title;">
-
- <stringbundleset id="stringbundleset">
- <stringbundle id="idTelifyStringBundle" src="chrome://telify/locale/lang.properties"/>
- </stringbundleset>
-
- <script type='application/x-javascript' src='chrome://telify/content/jshashtable.js'></script>
- <script type='application/x-javascript' src='chrome://telify/content/util.js'></script>
- <script type='application/x-javascript' src='chrome://telify/content/pref.js'></script>
- <script type='application/x-javascript' src='chrome://telify/content/country_data.js'></script>
- <script type='application/x-javascript' src='chrome://telify/locale/country_locale.js'></script>
- <script type='application/x-javascript' src='chrome://telify/content/editNumber.js'></script>
-
- <vbox>
- <hbox align="center">
- <textbox id="idTelifyInputCC" class="telInputCC" oninput="objTelifyEditNumber.ccChanged()" onkeypress="objTelifyEditNumber.checkKey(event,'+0123456789')"/>
- <label value="–"/>
- <textbox id="idTelifyInputNr" flex="1" onkeypress="objTelifyEditNumber.checkKey(event,'0123456789')"/>
- </hbox>
- <listbox id="idTelifyCountryCodeList" flex="1" width="280" height="250" onselect="objTelifyEditNumber.updateListSelection()">
- <listhead>
- <listheader id="idTelifyColCode" class="telInputCC" label="&dialog.edit.code;" onclick="objTelifyEditNumber.sortCountryCodeList(0)"/>
- <listheader id="idTelifyColCountry" label="&dialog.edit.country;" onclick="objTelifyEditNumber.sortCountryCodeList(1)"/>
- </listhead>
- <listcols>
- <listcol flex="0"/>
- <listcol flex="1"/>
- </listcols>
- </listbox>
- </vbox>
-
-</dialog>
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diff --git a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/telify/chrome/content/jshashtable.js b/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/telify/chrome/content/jshashtable.js
deleted file mode 100644
index 3806f818f98bc13a75438ad1b976ab74b5f08862..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/telify/chrome/content/jshashtable.js
+++ /dev/null
@@ -1,380 +0,0 @@
-/**
- * Copyright 2009 Tim Down.
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-/**
- * jshashtable
- *
- * jshashtable is a JavaScript implementation of a hash table. It creates a
- * single constructor function called Hashtable in the global scope.
- *
- * Author: Tim Down <tim@timdown.co.uk>
- * Version: 1.0
- * Build date: 5 February 2009
- * Website: http://www.timdown.co.uk/jshashtable
- */
-
-var Hashtable = (function() {
- function isUndefined(obj) {
- return (typeof obj === "undefined");
- }
-
- function isFunction(obj) {
- return (typeof obj === "function");
- }
-
- function isString(obj) {
- return (typeof obj === "string");
- }
-
- function hasMethod(obj, methodName) {
- if (obj[methodName]) {
- return isFunction(obj[methodName]);
- } else {
- return false;
- }
- }
-
- function hasEquals(obj) {
- return hasMethod(obj, "equals");
- }
-
- function hasHashCode(obj) {
- return hasMethod(obj, "hashCode");
- }
-
- function keyForObject(obj) {
- if (isString(obj)) {
- return obj;
- } else if (hasHashCode(obj)) {
- // Check the hashCode method really has returned a string
- var hashCode = obj.hashCode();
- if (!isString(hashCode)) {
- return keyForObject(hashCode);
- }
- return hashCode;
- } else if (hasMethod(obj, "toString")) {
- return obj.toString();
- } else {
- return String(obj);
- }
- }
-
- function equals_fixedValueHasEquals(fixedValue, variableValue) {
- return fixedValue.equals(variableValue);
- }
-
- function equals_fixedValueNoEquals(fixedValue, variableValue) {
- if (hasEquals(variableValue)) {
- return variableValue.equals(fixedValue);
- } else {
- return fixedValue === variableValue;
- }
- }
-
- function equals_equivalence(o1, o2) {
- return o1 === o2;
- }
-
- function arraySearch(arr, value, arrayValueFunction, returnFoundItem, equalityFunction) {
- var currentValue;
- for (var i = 0, len = arr.length; i < len; i++) {
- currentValue = arr[i];
- if (equalityFunction(value, arrayValueFunction(currentValue))) {
- return returnFoundItem ? [i, currentValue] : true;
- }
- }
- return false;
- }
-
- function arrayRemoveAt(arr, idx) {
- if (hasMethod(arr, "splice")) {
- arr.splice(idx, 1);
- } else {
- if (idx === arr.length - 1) {
- arr.length = idx;
- } else {
- var itemsAfterDeleted = arr.slice(idx + 1);
- arr.length = idx;
- for (var i = 0, len = itemsAfterDeleted.length; i < len; i++) {
- arr[idx + i] = itemsAfterDeleted[i];
- }
- }
- }
- }
-
- function checkKeyOrValue(kv, kvStr) {
- if (kv === null) {
- throw new Error("null is not a valid " + kvStr);
- } else if (isUndefined(kv)) {
- throw new Error(kvStr + " must not be undefined");
- }
- }
-
- var keyStr = "key", valueStr = "value";
-
- function checkKey(key) {
- checkKeyOrValue(key, keyStr);
- }
-
- function checkValue(value) {
- checkKeyOrValue(value, valueStr);
- }
-
- /*------------------------------------------------------------------------*/
-
- function Bucket(firstKey, firstValue, equalityFunction) {
- this.entries = [];
- this.addEntry(firstKey, firstValue);
-
- if (equalityFunction !== null) {
- this.getEqualityFunction = function() {
- return equalityFunction;
- };
- }
- }
-
- function getBucketEntryKey(entry) {
- return entry[0];
- }
-
- function getBucketEntryValue(entry) {
- return entry[1];
- }
-
- Bucket.prototype = {
- getEqualityFunction: function(searchValue) {
- if (hasEquals(searchValue)) {
- return equals_fixedValueHasEquals;
- } else {
- return equals_fixedValueNoEquals;
- }
- },
-
- searchForEntry: function(key) {
- return arraySearch(this.entries, key, getBucketEntryKey, true, this.getEqualityFunction(key));
- },
-
- getEntryForKey: function(key) {
- return this.searchForEntry(key)[1];
- },
-
- getEntryIndexForKey: function(key) {
- return this.searchForEntry(key)[0];
- },
-
- removeEntryForKey: function(key) {
- var result = this.searchForEntry(key);
- if (result) {
- arrayRemoveAt(this.entries, result[0]);
- return true;
- }
- return false;
- },
-
- addEntry: function(key, value) {
- this.entries[this.entries.length] = [key, value];
- },
-
- size: function() {
- return this.entries.length;
- },
-
- keys: function(keys) {
- var startIndex = keys.length;
- for (var i = 0, len = this.entries.length; i < len; i++) {
- keys[startIndex + i] = this.entries[i][0];
- }
- },
-
- values: function(values) {
- var startIndex = values.length;
- for (var i = 0, len = this.entries.length; i < len; i++) {
- values[startIndex + i] = this.entries[i][1];
- }
- },
-
- containsKey: function(key) {
- return arraySearch(this.entries, key, getBucketEntryKey, false, this.getEqualityFunction(key));
- },
-
- containsValue: function(value) {
- return arraySearch(this.entries, value, getBucketEntryValue, false, equals_equivalence);
- }
- };
-
- /*------------------------------------------------------------------------*/
-
- function BucketItem() {}
- BucketItem.prototype = [];
-
- // Supporting functions for searching hashtable bucket items
-
- function getBucketKeyFromBucketItem(bucketItem) {
- return bucketItem[0];
- }
-
- function searchBucketItems(bucketItems, bucketKey, equalityFunction) {
- return arraySearch(bucketItems, bucketKey, getBucketKeyFromBucketItem, true, equalityFunction);
- }
-
- function getBucketForBucketKey(bucketItemsByBucketKey, bucketKey) {
- var bucketItem = bucketItemsByBucketKey[bucketKey];
-
- // Check that this is a genuine bucket item and not something
- // inherited from prototype
- if (bucketItem && (bucketItem instanceof BucketItem)) {
- return bucketItem[1];
- }
- return null;
- }
-
- /*------------------------------------------------------------------------*/
-
- function Hashtable(hashingFunction, equalityFunction) {
- var bucketItems = [];
- var bucketItemsByBucketKey = {};
-
- hashingFunction = isFunction(hashingFunction) ? hashingFunction : keyForObject;
- equalityFunction = isFunction(equalityFunction) ? equalityFunction : null;
-
- this.put = function(key, value) {
- checkKey(key);
- checkValue(value);
- var bucketKey = hashingFunction(key);
-
- // Check if a bucket exists for the bucket key
- var bucket = getBucketForBucketKey(bucketItemsByBucketKey, bucketKey);
- if (bucket) {
- // Check this bucket to see if it already contains this key
- var bucketEntry = bucket.getEntryForKey(key);
- if (bucketEntry) {
- // This bucket entry is the current mapping of key to value, so replace
- // old value and we're done.
- bucketEntry[1] = value;
- } else {
- // The bucket does not contain an entry for this key, so add one
- bucket.addEntry(key, value);
- }
- } else {
- // No bucket, so create one and put our key/value mapping in
- var bucketItem = new BucketItem();
- bucketItem[0] = bucketKey;
- bucketItem[1] = new Bucket(key, value, equalityFunction);
- bucketItems[bucketItems.length] = bucketItem;
- bucketItemsByBucketKey[bucketKey] = bucketItem;
- }
- };
-
- this.get = function(key) {
- if (key == null) return null;
- checkKey(key);
- var bucketKey = hashingFunction(key);
- // Check if a bucket exists for the bucket key
- var bucket = getBucketForBucketKey(bucketItemsByBucketKey, bucketKey);
- if (bucket) {
- // Check this bucket to see if it contains this key
- var bucketEntry = bucket.getEntryForKey(key);
- if (bucketEntry) {
- // This bucket entry is the current mapping of key to value, so return
- // the value.
- return bucketEntry[1];
- }
- }
- return null;
- };
-
- this.containsKey = function(key) {
- checkKey(key);
-
- var bucketKey = hashingFunction(key);
-
- // Check if a bucket exists for the bucket key
- var bucket = getBucketForBucketKey(bucketItemsByBucketKey, bucketKey);
- if (bucket) {
- return bucket.containsKey(key);
- }
-
- return false;
- };
-
- this.containsValue = function(value) {
- checkValue(value);
- for (var i = 0, len = bucketItems.length; i < len; i++) {
- if (bucketItems[i][1].containsValue(value)) {
- return true;
- }
- }
- return false;
- };
-
- this.clear = function() {
- bucketItems.length = 0;
- bucketItemsByBucketKey = {};
- };
-
- this.isEmpty = function() {
- return bucketItems.length === 0;
- };
-
- this.keys = function() {
- var keys = [];
- for (var i = 0, len = bucketItems.length; i < len; i++) {
- bucketItems[i][1].keys(keys);
- }
- return keys;
- };
-
- this.values = function() {
- var values = [];
- for (var i = 0, len = bucketItems.length; i < len; i++) {
- bucketItems[i][1].values(values);
- }
- return values;
- };
-
- this.remove = function(key) {
- checkKey(key);
-
- var bucketKey = hashingFunction(key);
-
- // Check if a bucket exists for the bucket key
- var bucket = getBucketForBucketKey(bucketItemsByBucketKey, bucketKey);
-
- if (bucket) {
- // Remove entry from this bucket for this key
- if (bucket.removeEntryForKey(key)) {
- // Entry was removed, so check if bucket is empty
- if (bucket.size() === 0) {
- // Bucket is empty, so remove it
- var result = searchBucketItems(bucketItems, bucketKey, bucket.getEqualityFunction(key));
- arrayRemoveAt(bucketItems, result[0]);
- delete bucketItemsByBucketKey[bucketKey];
- }
- }
- }
- };
-
- this.size = function() {
- var total = 0;
- for (var i = 0, len = bucketItems.length; i < len; i++) {
- total += bucketItems[i][1].size();
- }
- return total;
- };
- }
-
- return Hashtable;
-})();
\ No newline at end of file
diff --git a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/telify/chrome/content/messagebox.js b/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/telify/chrome/content/messagebox.js
deleted file mode 100644
index 8be1b9828521bb3aa4b62cd7139dcb5e5e647d9c..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/telify/chrome/content/messagebox.js
+++ /dev/null
@@ -1,43 +0,0 @@
-/*
-Creative Commons License: Attribution-No Derivative Works 3.0 Unported
-http://creativecommons.org/licenses/by-nd/3.0/
-(c)2009 Michael Koch
-*/
-
-var objTelifyMessageBox = {
-
-init: function()
-{
- var title = window.arguments[0].title;
- if (title == null || title == "") title =" Telify";
- document.getElementById("dlgTelifyMessageBox").setAttribute("title", title);
- var msg_node = document.createTextNode(window.arguments[0].msg);
- document.getElementById("idTelify_mb_msg").appendChild(msg_node);
- var flags = window.arguments[0].flags;
- if ((flags & objTelifyUtil.MB_MASK) == 0) flags |= objTelifyUtil.MB_OK; // default button
- if ((flags & objTelifyUtil.MB_OK) == 0) document.documentElement.getButton("accept").collapsed = true;
- if ((flags & objTelifyUtil.MB_CANCEL) == 0) document.documentElement.getButton("cancel").collapsed = true;
- var icon = "info32.png";
- switch (flags & objTelifyUtil.MB_ICON_MASK) {
- case objTelifyUtil.MB_ICON_ERROR: icon = "error32.png"; break;
- case objTelifyUtil.MB_ICON_WARNING: icon = "warn32.png"; break;
- case objTelifyUtil.MB_ICON_ASK: icon = "ask32.png"; break;
- case objTelifyUtil.MB_ICON_INFO: icon = "info32.png"; break;
- }
- document.getElementById("idTelify_mb_icon").setAttribute("src", "chrome://telify/content/"+icon);
-},
-
-onAccept: function()
-{
- window.arguments[0].fResult = true;
- return true;
-},
-
-onCancel: function()
-{
- window.arguments[0].fResult = false;
- return true;
-}
-
-};
-
diff --git a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/telify/chrome/content/messagebox.xul b/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/telify/chrome/content/messagebox.xul
deleted file mode 100644
index 1be8d587f4dc776d2454b566d79f910df780ac51..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/telify/chrome/content/messagebox.xul
+++ /dev/null
@@ -1,34 +0,0 @@
-<?xml version="1.0"?>
-<?xml-stylesheet href="chrome://global/skin/global.css" type="text/css"?>
-<?xml-stylesheet href="chrome://telify/content/dialog.css" type="text/css"?>
-
-<!DOCTYPE dialog SYSTEM "chrome://telify/locale/lang.dtd">
-
-<dialog id="dlgTelifyMessageBox"
- xmlns="http://www.mozilla.org/keymaster/gatekeeper/there.is.only.xul"
- buttons="accept, cancel"
- onload="objTelifyMessageBox.init()"
- ondialogaccept="objTelifyMessageBox.onAccept()"
- ondialogcancel="objTelifyMessageBox.onCancel()"
- title="">
-
- <stringbundleset id="stringbundleset">
- <stringbundle id="idTelifyStringBundle" src="chrome://telify/locale/lang.properties"/>
- </stringbundleset>
-
- <script type='application/x-javascript' src='chrome://telify/content/pref.js'></script>
- <script type='application/x-javascript' src='chrome://telify/content/util.js'></script>
- <script type='application/x-javascript' src='chrome://telify/content/messagebox.js'></script>
-
-
- <groupbox style="background-color:white;padding:8px;">
- <hbox>
- <vbox>
- <image id="idTelify_mb_icon" src="chrome://telify/content/info32.png" style="width:32px;height:32px;margin-right:4px;"/>
- <spacer flex="1"/>
- </vbox>
- <description id="idTelify_mb_msg" style="width:240px;text-align:justify;"/>
- </hbox>
- </groupbox>
-
-</dialog>
diff --git a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/telify/chrome/content/pref.js b/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/telify/chrome/content/pref.js
deleted file mode 100644
index 2d3b635917caa5e8b8d7361020bd7a2e2c819339..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/telify/chrome/content/pref.js
+++ /dev/null
@@ -1,164 +0,0 @@
-/*
-Creative Commons License: Attribution-No Derivative Works 3.0 Unported
-http://creativecommons.org/licenses/by-nd/3.0/
-(c)2009 Michael Koch
-*/
-
-var objTelifyPrefs = {
-
-PREF_BLACKLIST: "blacklist",
-PREF_HIGHLIGHT: "highlight",
-PREF_EXCLUDE: "exclude",
-PREF_DEBUG: "debug",
-PREF_ACTIVE: "active",
-PREF_STATUSICON: "statusicon",
-PREF_HREFTYPE: "linktype",
-PREF_COLSORTCC: "colsortcc",
-PREF_NUMHISTORY: "num_history",
-PREF_IDD_PREFIX: "idd_prefix",
-PREF_DONT_ESCAPE_PLUS: "dont_escape_plus",
-PREF_DIAL_CC_DIRECT: "dial_cc_direct",
-
-NUM_CUSTOM_PARAMS: 3,
-
-PREF_CUSTOM_URL: "custom_url",
-PREF_CUSTOM_TMPL: "custom_tmpl",
-PREF_CUSTOM_PARAM: "custom_param",
-PREF_CUSTOM_OPENTYPE: "custom_opentype",
-
-maxHistory: 10,
-
-telPrefs: null,
-telStrings: null,
-
-blacklist: null,
-excludedHosts: null,
-highlight: null,
-excludedTags: null,
-hrefType: null,
-numHistory: null,
-idd_prefix: null,
-fStatusIcon: null,
-fActive: null,
-fDebug: null,
-fDontEscapePlus: null,
-fDialCCDirect: null,
-
-custom_url: null,
-custom_tmpl: null,
-custom_param: [],
-custom_opentype: null,
-
-HREFTYPE_CUSTOM: 9,
-
-protoList: new Array("tel", "callto", "skype", "sip"),
-
-
-showConfigDialog: function()
-{
- while (true) {
- window.openDialog("chrome://telify/content/config.xul", "dlgTelifyConfig", "centerscreen,chrome,modal").focus;
- if (this.hrefType == this.HREFTYPE_CUSTOM && this.custom_url.indexOf("$0") < 0) {
- var result = objTelifyUtil.showMessageBox("", objTelifyLocale.msgNumberTemplateMissing(),
- objTelifyUtil.MB_OK|objTelifyUtil.MB_CANCEL|objTelifyUtil.MB_ICON_WARNING);
- if (result == false) continue;
- }
- break;
- }
-},
-
-
-getPrefObj: function()
-{
- var obj = Components.classes["@mozilla.org/preferences-service;1"];
- obj = obj.getService(Components.interfaces.nsIPrefService);
- obj = obj.getBranch("telify.settings.");
- obj.QueryInterface(Components.interfaces.nsIPrefBranch2);
- return obj;
-},
-
-
-getCharPref: function(name)
-{
- try {
- return this.telPrefs.getCharPref(name);
- } catch (e) {
- alert(e);
- return "";
- }
-},
-
-
-getIntPref: function(name)
-{
- try {
- return this.telPrefs.getIntPref(name);
- } catch (e) {
- return 0;
- }
-},
-
-
-getBoolPref: function(name)
-{
- try {
- return this.telPrefs.getBoolPref(name);
- } catch (e) {
- return false;
- }
-},
-
-
-getPrefs: function()
-{
- this.blacklist = this.telPrefs.getCharPref(this.PREF_BLACKLIST);
- if (this.blacklist.length > 0) {
- this.excludedHosts = this.blacklist.toLowerCase().split(",");
- } else {
- this.excludedHosts = new Array();
- }
- this.highlight = this.telPrefs.getIntPref(this.PREF_HIGHLIGHT);
- this.highlight = objTelifyUtil.trimInt(this.highlight, 0, 100);
- this.numHistory = this.telPrefs.getIntPref(this.PREF_NUMHISTORY);
- this.numHistory = objTelifyUtil.trimInt(this.numHistory, 1, 10);
- this.idd_prefix = this.telPrefs.getCharPref(this.PREF_IDD_PREFIX);
- var exclude = this.telPrefs.getCharPref(this.PREF_EXCLUDE);
- this.excludedTags = exclude.toLowerCase().split(",");
- this.hrefType = this.telPrefs.getIntPref(this.PREF_HREFTYPE);
- if ((this.hrefType < 0 || this.hrefType >= this.protoList.length) && this.hrefType != this.HREFTYPE_CUSTOM) this.hrefType = 0;
- this.fStatusIcon = this.telPrefs.getBoolPref(this.PREF_STATUSICON);
- var status = document.getElementById("idTelify_status");
- if (status) status.setAttribute("collapsed", !this.fStatusIcon);
- this.fDebug = this.telPrefs.getBoolPref(this.PREF_DEBUG);
- this.fActive = this.telPrefs.getBoolPref(this.PREF_ACTIVE);
- this.fDontEscapePlus = this.telPrefs.getBoolPref(this.PREF_DONT_ESCAPE_PLUS);
- this.fDialCCDirect = this.telPrefs.getBoolPref(this.PREF_DIAL_CC_DIRECT);
- // custom url
- this.custom_url = this.getCharPref(this.PREF_CUSTOM_URL);
- this.custom_tmpl = this.getIntPref(this.PREF_CUSTOM_TMPL);
- for (var i=1; i<this.NUM_CUSTOM_PARAMS+1; i++) {
- this.custom_param[i] = this.getCharPref(this.PREF_CUSTOM_PARAM+i);
- }
- this.custom_opentype = this.getIntPref(this.PREF_CUSTOM_OPENTYPE);
-},
-
-
-prefObserver: {
- observe: function(subject, topic, data) {
- if (topic != "nsPref:changed") return;
- objTelifyPrefs.getPrefs();
- }
-},
-
-
-initTelifyPrefs: function()
-{
- objTelifyPrefs.telPrefs = objTelifyPrefs.getPrefObj();
- objTelifyPrefs.telPrefs.addObserver("", objTelifyPrefs.prefObserver, false);
- objTelifyPrefs.telStrings = document.getElementById("idTelifyStringBundle");
- objTelifyPrefs.getPrefs();
-}
-
-};
-
-
diff --git a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/telify/chrome/content/telify.js b/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/telify/chrome/content/telify.js
deleted file mode 100644
index 58a615ec060bbb4bfeba41cb99ff550fab6b5f9f..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/telify/chrome/content/telify.js
+++ /dev/null
@@ -1,715 +0,0 @@
-/*
-Creative Commons License: Attribution-No Derivative Works 3.0 Unported
-http://creativecommons.org/licenses/by-nd/3.0/
-(c)2009 Michael Koch
-*/
-
-var objTelify = {
-
-digits_min: 7,
-digits_max: 16,
-
-hilite_color: new Array(0,0,255),
-hilite_bgcolor: new Array(255,255,0),
-
-// special chars
-sc_nbsp: String.fromCharCode(0xa0),
-
-// chars which look like dashes
-token_dash:
- String.fromCharCode(0x2013) +
- String.fromCharCode(0x2014) +
- String.fromCharCode(0x2212),
-
-exclPatternList: [
- /^\d{2}\.\d{2} ?(-|–) ?\d{2}\.\d{2}$/, // time range e.g. 08.00 - 17.00
- /^\d{2}\/\d{2}\/\d{2}$/, // date e.g. 09/03/09
- /^\d{1,3}\.\d{1,3}\.\d{1,3}\.\d{1,3}$/, // ip address
- /^[0-3]?[0-9]\.[0-3]?[0-9]\.(19|20)\d{2} - \d{2}\.\d{2}$/, // date and time e.g. 09.03.2009 - 17.59
- /^[0-3]?[0-9][\/\.-][0-3]?[0-9][\/\.-](19|20)\d{2}$/, // date e.g. 09/03/2009, 09.03.2009, 09-03-2009
- /^[0-3]?[0-9][\/\.-][0-3]?[0-9][\/\.-]\d{2} ?(-|–) ?[0-3]?[0-9][\/\.-][0-3]?[0-9][\/\.-]\d{2}$/, // date range short
- /^[0-3]?[0-9][\/\.-][0-3]?[0-9][\/\.-] ?(-|–) ?[0-3]?[0-9][\/\.-][0-3]?[0-9][\/\.-](19|20)\d{2}$/, // date range medium
- /^[0-3]?[0-9][\/\.-][0-3]?[0-9][\/\.-](19|20)\d{2} ?(-|–) ?[0-3]?[0-9][\/\.-][0-3]?[0-9][\/\.-](19|20)\d{2}$/, // date range long
- /^0\.\d+$/, // e.g. 0.12345678
-],
-
-inclLocalList: [
- [/^[1-9]\d{2}[\.-]\d{3}[\.-]\d{4}$/, "+1"], // US
-],
-
-token_trigger: "+(0123456789",
-token_part: " -/()[].\r\n"
- + String.fromCharCode(0xa0) // sc_nbsp
- + String.fromCharCode(0x2013) + String.fromCharCode(0x2014) + String.fromCharCode(0x2212), // token_dash
-token_start: "+(0",
-token_sep: " -/(.",
-token_disallowed_post: ":-",
-token_disallowed_prev: "-,.",
-
-dialHistory: new Array(objTelifyPrefs.maxHistory),
-
-
-getDialHistory: function()
-{
- for (var i=0; i<objTelifyPrefs.maxHistory; i++) {
- try {
- this.dialHistory[i] = objTelifyPrefs.telPrefs.getCharPref("history"+i);
- } catch (e) {
- this.dialHistory[i] = "";
- }
- }
-},
-
-
-saveDialHistory: function()
-{
- for (var i=0; i<objTelifyPrefs.maxHistory; i++) {
- if (this.dialHistory[i] == null) this.dialHistory[i] = "";
- objTelifyPrefs.telPrefs.setCharPref("history"+i, this.dialHistory[i]);
- }
-},
-
-
-updateDialHistory: function(prefix)
-{
- //logmsg("updateDialHistory("+prefix+")");
- var newList = new Array(objTelifyPrefs.maxHistory);
- newList[0] = prefix;
- for (var i=0, j=1; i<objTelifyPrefs.maxHistory && j<objTelifyPrefs.maxHistory; i++) {
- if (this.dialHistory[i] == null || this.dialHistory[i] == "" || this.dialHistory[i] == prefix) continue;
- newList[j++] = this.dialHistory[i];
- }
- this.dialHistory = newList;
- this.saveDialHistory();
-},
-
-
-setStatus: function()
-{
- var statusicon = document.getElementById("idTelify_statusicon");
- if (objTelifyPrefs.fActive) {
- statusicon.setAttribute("src", "chrome://telify/content/icon18_active.png");
- var text = objTelifyPrefs.telStrings.getString("telify_active");
- statusicon.setAttribute("tooltiptext", text);
- } else {
- statusicon.setAttribute("src", "chrome://telify/content/icon18_inactive.png");
- var text = objTelifyPrefs.telStrings.getString("telify_inactive");
- statusicon.setAttribute("tooltiptext", text);
- }
-},
-
-
-toggleBlacklist: function()
-{
- var host = objTelifyUtil.getHost();
- if (host == null) return;
- if (objTelifyPrefs.excludedHosts.indexOf(host) >= 0) {
- objTelifyUtil.arrayRemove(objTelifyPrefs.excludedHosts, host);
- } else {
- objTelifyPrefs.excludedHosts.push(host);
- }
- objTelifyPrefs.blacklist = objTelifyPrefs.excludedHosts.join(",");
- objTelifyPrefs.telPrefs.setCharPref(objTelifyPrefs.PREF_BLACKLIST, objTelifyPrefs.blacklist);
-},
-
-
-toggleActive: function()
-{
- objTelifyPrefs.telPrefs.setBoolPref(objTelifyPrefs.PREF_ACTIVE, !objTelifyPrefs.fActive);
- this.setStatus();
-},
-
-
-getSelectionNumber: function()
-{
- //var sel = content.window.getSelection().toString();
- var sel = document.commandDispatcher.focusedWindow.getSelection().toString();
- sel = this.convertVanityNr(sel);
- sel = objTelifyUtil.stripNumber(sel);
- return sel;
-},
-
-
-dialNumber: function(nr)
-{
- var requ = new XMLHttpRequest();
- var url = objTelifyUtil.createDialURL(nr);
-
- if (objTelifyPrefs.hrefType == objTelifyPrefs.HREFTYPE_CUSTOM) {
- if (objTelifyPrefs.custom_opentype == 1) {
- window.open(url, "_blank");
- return;
- }
- if (objTelifyPrefs.custom_opentype == 2) {
- var browser = top.document.getElementById("content");
- var tab = browser.addTab(url);
- return;
- }
- if (objTelifyPrefs.custom_opentype == 3) {
- var browser = top.document.getElementById("content");
- var tab = browser.addTab(url);
- browser.selectedTab = tab;
- return;
- }
- }
-
- try {
- requ.open("GET", url, true);
- requ.send(null);
- } catch(e) {
- // throws exception because answer is empty (or protocol is unknown)
- if (e.name == "NS_ERROR_UNKNOWN_PROTOCOL") {
- objTelifyUtil.showMessageBox("", objTelifyLocale.msgUnknownProtocol(), objTelifyUtil.MB_ICON_ERROR);
- }
- }
-},
-
-
-modifyPopup: function(event)
-{
- var label, key;
-
- //var selText = content.window.getSelection().toString();
- var selText = document.commandDispatcher.focusedWindow.getSelection().toString();
-
- if (document.popupNode && document.popupNode.getAttribute("class") == "telified") {
- var nr = document.popupNode.getAttribute("nr");
- var nr_parts = objTelifyUtil.splitPhoneNr(nr);
- objTelify.modifyDialPopup(nr_parts[0], nr_parts[1], "context");
- objTelifyUtil.setIdAttr("collapsed", false, "idTelify_menu_context");
- } else if (objTelifyPrefs.fActive && selText.length > 0 && objTelifyUtil.countDigits(selText) > 1) {
- var nr = objTelify.getSelectionNumber();
- var nr_parts = objTelifyUtil.splitPhoneNr(nr);
- objTelify.modifyDialPopup(nr_parts[0], nr_parts[1], "context");
- objTelifyUtil.setIdAttr("collapsed", false, "idTelify_menu_context");
- } else {
- objTelifyUtil.setIdAttr("collapsed", true, "idTelify_menu_context");
- }
-
- if (objTelifyPrefs.fActive) {
- label = objTelifyPrefs.telStrings.getString("telify_deactivate");
- } else {
- label = objTelifyPrefs.telStrings.getString("telify_activate");
- }
- objTelifyUtil.setIdAttr("label", label, "idTelify_menu_activity", "idTelify_status_activity");
-
- var host = objTelifyUtil.getHost();
- if (host) {
- objTelifyUtil.setIdAttr("disabled", !objTelifyPrefs.fActive, "idTelify_menu_blacklist", "idTelify_status_blacklist");
- if (objTelifyPrefs.excludedHosts.indexOf(host) >= 0) key = "host_active_arg"; else key = "host_inactive_arg";
- label = objTelifyUtil.substArgs(objTelifyPrefs.telStrings.getString(key), host);
- objTelifyUtil.setIdAttr("label", label, "idTelify_menu_blacklist", "idTelify_status_blacklist");
- } else {
- objTelifyUtil.setIdAttr("label", "Kein Host aktiv", "idTelify_menu_blacklist", "idTelify_status_blacklist");
- objTelifyUtil.setIdAttr("disabled", true, "idTelify_menu_blacklist", "idTelify_status_blacklist");
- }
-},
-
-
-showEditNumberDialog: function(cc, nr)
-{
- var argObj = {cc: cc, nr: nr, fOK: false};
- window.openDialog("chrome://telify/content/editNumber.xul", "dlgTelifyEditNumber", "centerscreen,chrome,modal", argObj);
- if (argObj.fOK) {
- this.updateDialHistory(argObj.cc);
- var dial = objTelifyUtil.prefixNumber(argObj.cc, argObj.nr, "");
- objTelify.dialNumber(dial);
- }
-},
-
-
-dialMenuSelection: function(cc, nr)
-{
- this.updateDialHistory(cc);
- var dial = objTelifyUtil.prefixNumber(cc, nr, "");
- objTelify.dialNumber(dial);
-},
-
-
-createTargetCountryInfo: function(prefix)
-{
- var cstring = objTelifyUtil.getCountryListString(prefix);
- if (cstring) return "\n" + objTelifyPrefs.telStrings.getString('country_code') + ": " + cstring;
- return "";
-},
-
-
-setDialMenuItem: function(item, code, nr)
-{
- var label = objTelifyUtil.prefixNumber(code, nr, "-");
- item.setAttribute("label", label);
- var cmd = "objTelify.dialMenuSelection('"+code+"','"+nr+"');";
- item.setAttribute("oncommand", cmd);
- label = objTelifyUtil.substArgs(objTelifyPrefs.telStrings.getString('call_arg'), label);
- label += objTelify.createTargetCountryInfo(code);
- item.setAttribute("tooltiptext", label);
- item.setAttribute("image", "chrome://telify/content/flag/"+code.substr(1)+".png");
-},
-
-
-modifyDialPopup: function(cc, nr, id)
-{
- var item = document.getElementById("idTelify_"+id);
- var sep = document.getElementById("idTelify_sep_"+id);
- var numShown = 0;
-
- if (cc) {
- this.setDialMenuItem(item, cc, nr);
- } else {
- item.setAttribute("label", nr);
- var label = objTelifyUtil.substArgs(objTelifyPrefs.telStrings.getString('call_arg'), nr);
- item.setAttribute("tooltiptext", label);
- item.removeAttribute("image");
- item.setAttribute("oncommand", "objTelify.dialNumber('"+nr+"')");
- }
-
- item = document.getElementById("idTelify_edit_"+id);
- if (cc) {
- item.setAttribute("oncommand", "objTelify.showEditNumberDialog('"+cc+"','"+nr+"')");
- } else {
- item.setAttribute("oncommand", "objTelify.showEditNumberDialog(null,'"+nr+"')");
- }
-
- var tldcc = objTelifyUtil.tld2cc(objTelifyUtil.getHostTLD());
- item = document.getElementById("idTelify_tld_"+id);
- if (!cc && tldcc) {
- item.setAttribute("collapsed", false);
- this.setDialMenuItem(item, tldcc, nr);
- numShown = 1;
- } else {
- item.setAttribute("collapsed", true);
- tldcc = null;
- }
-
- this.getDialHistory();
-
- if (!cc && nr.charAt(0) != '+') {
- var numLeft = objTelifyPrefs.numHistory;
- if (tldcc) numLeft--;
- for (var i=0; i<objTelifyPrefs.maxHistory; i++) {
- item = document.getElementById("idTelify_"+id+i);
- if (numLeft == 0 || this.dialHistory[i] == null || this.dialHistory[i].length == 0 || this.dialHistory[i] == cc || this.dialHistory[i] == tldcc) {
- item.setAttribute("collapsed", true);
- } else {
- item.setAttribute("collapsed", false);
- this.setDialMenuItem(item, this.dialHistory[i], nr);
- numLeft--;
- numShown++;
- }
- }
- } else {
- for (var i=0; i<objTelifyPrefs.maxHistory; i++) {
- item = document.getElementById("idTelify_"+id+i);
- item.setAttribute("collapsed", true);
- }
- }
- sep.setAttribute("collapsed", numShown == 0);
-},
-
-
-showDialPopup: function(target, cc, nr)
-{
- var menu = document.getElementById("idTelify_popup_dial");
- var nr_parts = objTelifyUtil.splitPhoneNr(nr);
- this.modifyDialPopup(cc, nr, "dial");
- menu.openPopup(target, "after_start", 0, 0, true, false);
-},
-
-
-onClick: function(event)
-{
- if (event.button != 0) return;
- var class = event.target.getAttribute("class");
- if (class != "telified") return;
- event.preventDefault();
- var nr = event.target.getAttribute("nr");
- var nr_parts = objTelifyUtil.splitPhoneNr(nr);
- if (event.button == 0) {
- if (nr_parts[0] && objTelifyPrefs.fDialCCDirect) {
- objTelify.dialNumber(nr);
- } else {
- objTelify.showDialPopup(event.target, nr_parts[0], nr_parts[1]);
- }
- }
- if (event.button == 2) {
- objTelify.showDialPopup(event.target, nr_parts[0], nr_parts[1]);
- }
-},
-
-
-getNodeBackgroundColor: function(node)
-{
- node = node.parentNode;
- if (node == null) return null;
- if (node.nodeType == Node.ELEMENT_NODE) {
- var style = content.document.defaultView.getComputedStyle(node, "");
- var image = style.getPropertyValue("background-image");
- if (image && image != "none") return null;
- var color = style.getPropertyValue("background-color");
- if (color && color != "transparent") return color;
- }
- return this.getNodeBackgroundColor(node);
-},
-
-
-getNodeColor: function(node)
-{
- node = node.parentNode;
- if (node == null) return null;
- if (node.nodeType == Node.ELEMENT_NODE) {
- var style = content.document.defaultView.getComputedStyle(node, "");
- var color = style.getPropertyValue("color");
- if (color && color != "transparent") return color;
- }
- return this.getNodeColor(node);
-},
-
-
-formatPhoneNr: function(phonenr)
-{
- var substList = [
- [" ", " "], // double spaces to single space
- [this.sc_nbsp, " "], // non-breaking space to plain old space
- ["+ ", "+"], // remove space after +
- ["--", "-"], // double dashes to single dash
- ["(0)", " "], // remove optional area code prefix
- ["[0]", " "], // remove optional area code prefix
- ["-/", "/"],
- ["/-", "/"],
- ["( ", "("],
- [" )", ")"],
- ["\r", " "],
- ["\n", " "],
- ];
-
- // replace dash-like chars with dashes
- for (var i=0; i<phonenr.length; i++) {
- var c = phonenr.charAt(i);
- if (this.token_dash.indexOf(c) >= 0) {
- phonenr = phonenr.substr(0, i) + "-" + phonenr.substr(i+1);
- }
- }
-
- const MAXLOOP = 100; // safety bailout
- var nChanged;
-
- nChanged = 1;
- for (var j=0; nChanged > 0 && j < MAXLOOP; j++) {
- nChanged = 0;
- for (var i=0; i<substList.length; i++) {
- var index;
- while ((index = phonenr.indexOf(substList[i][0])) >= 0) {
- phonenr = phonenr.substr(0, index) + substList[i][1] + phonenr.substr(index+substList[i][0].length);
- nChanged++;
- }
- }
- }
-
- return phonenr;
-},
-
-
-convertVanityNr: function(phonenr)
-{
- const tab_alpha = "ABCDEFGHIJKLMNOPQRSTUVWXYZ";
- const tab_digit = "22233344455566677778889999";
- var newnr = "";
- for (var i=0; i<phonenr.length; i++) {
- var c = phonenr.charAt(i);
- var index = tab_alpha.indexOf(c);
- if (index >= 0) c = tab_digit.substr(index, 1);
- newnr += c;
- }
- return newnr;
-},
-
-
-reject: function(str, reason)
-{
- if (objTelifyPrefs.fDebug == false) return;
- var msg = "Telify: reject '"+str+"' reason: "+reason;
- objTelifyUtil.logmsg(msg);
-},
-
-
-basechar_tab: [
- String.fromCharCode(0xa0) +
- String.fromCharCode(0x2013) +
- String.fromCharCode(0x2014) +
- String.fromCharCode(0x2212),
- " ---"
-],
-
-
-basechar: function(c)
-{
- var index = this.basechar_tab[0].indexOf(c);
- if (index >= 0) c = this.basechar_tab[1].charAt(index);
- return c;
-},
-
-
-telifyTextNode: function(node)
-{
- if (node == null) return 0;
- var text = node.data;
- var len = text.length;
- if (len < this.digits_min) return 0;
- var hlFactor = objTelifyPrefs.highlight/200.0;
-
- for (var i=0; i<len; i++) {
- var c = text.charAt(i);
-
- if (this.token_trigger.indexOf(c) < 0) continue;
-
- c = this.basechar(c);
-
- var str = "" + c;
- var strlen = 1;
- var last_c = c;
- var ndigits = (objTelifyUtil.isdigit(c) ? 1 : 0);
- var index;
- var fStartsWithCountryCode = false;
- var CCfromPattern = null;
-
- // gather allowed chars
- while (strlen < len-i) {
- c = text.charAt(i+strlen);
- c = this.basechar(c);
- if ((c == '+' && ndigits == 0) || (this.token_part.indexOf(c) >= 0)) {
- if (c == last_c && c!=' ') break;
- } else {
- if (!objTelifyUtil.isdigit(c)) break;
- ndigits++;
- }
- str += c;
- strlen++;
- last_c = c;
- }
-
- // check against digit count min value
- if (ndigits < this.digits_min) {
- this.reject(str, "less than "+this.digits_min+" digits");
- i += strlen - 1; continue;
- }
-
- // check allowed prev token
- if (i > 0) {
- var prev_c = text.charAt(i-1);
- if (this.token_disallowed_prev.indexOf(prev_c) >= 0) {
- this.reject(str, "unallowed previous token (reject list)");
- i += strlen - 1; continue;
- }
- if ((prev_c >= 'a' && prev_c <= "z") || (prev_c >= 'A' && prev_c <= "Z")) {
- this.reject(str, "unallowed previous token (letter)");
- i += strlen - 1; continue;
- }
- }
-
- // check if phone number starts with country code
- for (var j=0; j<telify_country_data.length; j++) {
- var cclen = telify_country_data[j][0].length;
- if (cclen < 2 || cclen > 4) continue;
- var pattern = telify_country_data[j][0].substr(1);
- var plen = pattern.length;
- if (str.substr(0, plen) != pattern) continue;
- var c = str.charAt(plen);
- if (this.token_sep.indexOf(c) < 0) continue;
- fStartsWithCountryCode = true;
- break;
- }
-
- // check against special local patterns
- for (var j=0; j<this.inclLocalList.length; j++) {
- var res = this.inclLocalList[j][0].exec(str);
- if (res) {CCfromPattern = this.inclLocalList[j][1]; break;}
- }
-
- // check if phone number starts with allowed token
- if (CCfromPattern == null && fStartsWithCountryCode == false && this.token_start.indexOf(str.charAt(0)) < 0) {
- this.reject(str, "unallowed start token (reject list)");
- i += strlen - 1; continue;
- }
-
- // trim chars at end of string up to an unmatched opening bracket
- index = -1;
- for (var j=strlen-1; j>=0; j--) {
- c = str.charAt(j);
- if (c == ')') break;
- if (c == '(') {index = j; break;}
- }
- if (index == 0) continue;
- if (index > 0) {
- str = str.substr(0, index);
- strlen = str.length;
- }
-
- // check against digit count max value (after we have removed unnecessary digits)
- if (objTelifyUtil.countDigits(str) > this.digits_max) {
- this.reject(str, "more than "+this.digits_max+" digits");
- i += strlen - 1; continue;
- }
-
- // trim non-digit chars at end of string
- while (str.length > 0) {
- c = str.charAt(str.length-1);
- if (!objTelifyUtil.isdigit(c)) {
- str = str.substr(0, str.length-1);
- strlen--;
- } else break;
- }
-
- // check allowed post token
- var post_c = text.charAt(i+strlen);
- if (post_c) {
- if (this.token_disallowed_post.indexOf(post_c) >= 0) {
- this.reject(str, "unallowed post token (reject list)");
- i += strlen - 1; continue;
- }
- if ((post_c >= 'a' && post_c <= "z") || (post_c >= 'A' && post_c <= "Z")) {
- this.reject(str, "unallowed post token (letter)");
- i += strlen - 1; continue;
- }
- }
-
- // check if this is just a number in braces
- // first check for unnecessary opening braces
- if (str.substr(0, 1) == "(" && str.indexOf(")") < 0) {
- str = str.substr(1);
- i++;
- strlen--;
- // now check if it still starts with allowed token
- if (this.token_start.indexOf(str.charAt(0)) < 0) {
- this.reject(str, "unallowed start token (after brace removal)");
- i += strlen - 1;
- continue;
- }
- }
-
- // check against blacklist patterns (date, time ranges etc.)
- index = -1;
- for (var j=0; j<this.exclPatternList.length; j++) {
- var res = this.exclPatternList[j].exec(str);
- if (res) {index = j; break;}
- }
- if (index >= 0) {this.reject(str, "blacklisted pattern #"+index); i += strlen - 1; continue;}
-
-
- // ----------------------------------------------------------------
-
- var display = this.formatPhoneNr(str);
- var href = objTelifyUtil.stripNumber(display);
- if (fStartsWithCountryCode) href = "+"+href;
- //if (CCfromPattern) href = CCfromPattern + href;
-
- // insert link into DOM
-
- var node_prev = content.document.createTextNode(text.substr(0, i));
- var node_after = content.document.createTextNode(text.substr(i+strlen));
-
- //alert("match="+str);
-
- var node_anchor = content.document.createElement("a");
-
- if (hlFactor > 0.0) {
- var color = objTelifyUtil.parseColor(this.getNodeColor(node));
- if (color == null) color = new Array(0,0,0);
- var bgcolor = objTelifyUtil.parseColor(this.getNodeBackgroundColor(node));
- if (bgcolor == null) bgcolor = new Array(255,255,255);
- for (var i=0; i<3; i++) {
- color[i] = color[i] + hlFactor * (this.hilite_color[i] - color[i]);
- bgcolor[i] = bgcolor[i] + hlFactor * (this.hilite_bgcolor[i] - bgcolor[i]);
- }
- var style = "color:#"+objTelifyUtil.color2hex(color)+";background-color:#"+objTelifyUtil.color2hex(bgcolor)+";-moz-border-radius:3px";
- node_anchor.setAttribute("style", style);
- }
-
- node_anchor.setAttribute("title", objTelifyPrefs.telStrings.getString('link_title'));
- node_anchor.setAttribute("class", "telified");
- node_anchor.setAttribute("nr", href);
- node_anchor.setAttribute("href", objTelifyUtil.createDialURL(href));
-
- var node_text = content.document.createTextNode(str);
- node_anchor.appendChild(node_text);
-
- var parentNode = node.parentNode;
- parentNode.replaceChild(node_after, node);
- parentNode.insertBefore(node_anchor, node_after);
- parentNode.insertBefore(node_prev, node_anchor);
-
- return 1;
- }
-
- return 0;
-},
-
-
-recurseNode: function(node)
-{
- if (node == null) return 0; // safety
- if (node.nodeType == Node.TEXT_NODE) {
- return this.telifyTextNode(node);
- } else {
- var nChanged = 0;
- //objTelifyUtil.logmsg("node type="+node.nodeType+" "+node.tagName+" (childs:"+node.childNodes.length+")");
- if (node.nodeType == Node.ELEMENT_NODE) {
- var tagName = node.tagName.toLowerCase();
- if (objTelifyPrefs.excludedTags.indexOf(tagName) >= 0) return 0;
- }
- for (var i=0; i<node.childNodes.length; i++) {
- nChanged += this.recurseNode(node.childNodes[i]);
- }
- if (node.contentDocument) {
- nChanged += this.recurseNode(node.contentDocument.body);
- node.contentDocument.addEventListener("click", objTelify.onClick, false);
- }
- }
- return nChanged;
-},
-
-
-parsePage: function(event)
-{
- if (!objTelifyPrefs.fActive) return;
- //objTelifyUtil.logmsg("eventPhase: "+event.eventPhase+"\n"+content.document.URL);
- if (content.document.body == null) return;
- if (event && event.eventPhase != 1) return;
-
- var host = objTelifyUtil.getHost();
- if (host && objTelifyPrefs.excludedHosts.indexOf(host) >= 0) return;
-
- //if (content.document.body.getAttribute('telified') == 1) return;
- //content.document.body.setAttribute('telified', 1);
-
-/*
- var nChanged = 0;
- var duration = (new Date()).getTime();
- nChanged = objTelify.recurseNode(content.document.body);
- duration = (new Date()).getTime() - duration;
- var label = "Telify\n" + objTelifyPrefs.telStrings.getString('converted') + ": " + nChanged + " (" + duration + " ms)";
- document.getElementById("idTelify_statusicon").setAttribute("tooltiptext", label);
-*/
-
- window.setTimeout("objTelify.recurseNode(content.document.body)", 0);
-
- content.document.addEventListener("click", objTelify.onClick, false);
-},
-
-
-init: function(event)
-{
- window.addEventListener('load', objTelify.init, false);
- objTelifyPrefs.initTelifyPrefs();
- objTelify.setStatus();
- getBrowser().addEventListener("load", objTelify.parsePage, true);
- document.getElementById("contentAreaContextMenu").addEventListener("popupshowing", objTelify.modifyPopup, false);
- objTelifyUtil.addScheme("tel");
- objTelifyUtil.localizeCountryData();
- objTelifyUtil.getAddonVersion();
-}
-
-};
-
-
-window.addEventListener('load', objTelify.init, false);
-
diff --git a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/telify/chrome/content/util.js b/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/telify/chrome/content/util.js
deleted file mode 100644
index 838cd91afba39fcc34f151f1ed42c521603171c9..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/telify/chrome/content/util.js
+++ /dev/null
@@ -1,516 +0,0 @@
-/*
-Creative Commons License: Attribution-No Derivative Works 3.0 Unported
-http://creativecommons.org/licenses/by-nd/3.0/
-(c)2009 Michael Koch
-*/
-
-var objTelifyUtil = {
-
-getBrowser: function()
-{
- var wm = Components.classes["@mozilla.org/appshell/window-mediator;1"].getService(Components.interfaces.nsIWindowMediator);
- var mainWindow = wm.getMostRecentWindow("navigator:browser");
- var browser = mainWindow.getBrowser();
- return browser;
-},
-
-
-getAddonVersion: function()
-{
- var gExtensionManager = Components.classes["@mozilla.org/extensions/manager;1"]
- .getService(Components.interfaces.nsIExtensionManager);
- return gExtensionManager.getItemForID("{6c5f349a-ddda-49ad-bdf0-326d3fe1f938}").version;
-},
-
-
-createDialURL: function(nr)
-{
- var url;
- if (nr.charAt(0) == '+') {
- if (objTelifyPrefs.idd_prefix.length > 0) {
- nr = objTelifyPrefs.idd_prefix + nr.substr(1);
- } else if (objTelifyPrefs.hrefType == objTelifyPrefs.HREFTYPE_CUSTOM && !objTelifyPrefs.fDontEscapePlus) {
- nr = "%2B" + nr.substr(1);
- }
- }
- if (objTelifyPrefs.hrefType == objTelifyPrefs.HREFTYPE_CUSTOM) {
- url = objTelifyPrefs.custom_url;
- url = objTelifyUtil.replaceRefs(url, 0, nr);
- for (var i=1; i<objTelifyPrefs.NUM_CUSTOM_PARAMS+1; i++) {
- url = objTelifyUtil.replaceRefs(url, i, objTelifyPrefs.custom_param[i]);
- }
- } else {
- url = objTelifyPrefs.protoList[objTelifyPrefs.hrefType]+":"+nr;
- }
- return url;
-},
-
-
-token_href: "+0123456789",
-
-stripNumber: function(phonenr)
-{
- var newnr = "";
- for (var i=0; i<phonenr.length; i++) {
- var c = phonenr.charAt(i);
- if (this.token_href.indexOf(c) >= 0) newnr += c;
- }
- return newnr.substr(0, objTelify.digits_max);
-},
-
-
-code2ndd_hashtable: null,
-
-create_code2ndd_hashtable: function()
-{
- this.code2ndd_hashtable = new Hashtable();
- for (var i=0; i<telify_country_data.length; i++) {
- if (telify_country_data[i][0] == "") continue;
- this.code2ndd_hashtable.put(telify_country_data[i][0], telify_country_data[i][3]);
- }
-},
-
-
-prefixNumber: function(prefix, nr, sep)
-{
- if (prefix == null || prefix == "") return this.stripNumber(nr);
- if (this.code2ndd_hashtable == null) this.create_code2ndd_hashtable();
- var ndd = this.code2ndd_hashtable.get(prefix);
- if ((ndd.length > 0) && (nr.substr(0, ndd.length) == ndd)) nr = nr.substr(ndd.length);
- return this.stripNumber(prefix) + sep + this.stripNumber(nr);
-},
-
-
-trim: function(s)
-{
- s = s.replace(/^\s*(.*)/, "$1");
- s = s.replace(/(.*?)\s*$/, "$1");
- return s;
-},
-
-
-localizeCountryData: function()
-{
-/*
- for (var i=0; i < telify_country_data.length; i++) {
- for (var j=0; j<telify_country_locale.length; j++) {
- if (telify_country_data[i][1] == telify_country_locale[j][0]) {
- telify_country_data[i][1] = telify_country_locale[j][1];
- break;
- }
- }
- }
-*/
- var hashtable = new Hashtable();
- for (var i=0; i<telify_country_locale.length; i++) {
- hashtable.put(telify_country_locale[i][0], telify_country_locale[i][1]);
- }
- for (var i=0; i<telify_country_data.length; i++) {
- var value = hashtable.get(telify_country_data[i][1]);
- if (value) telify_country_data[i][1] = value;
- }
-},
-
-
-tld_hashtable: null,
-
-create_tld_hashtable: function()
-{
- this.tld_hashtable = new Hashtable();
- for (var i=0; i<telify_country_data.length; i++) {
- if (telify_country_data[i][2] == "") continue;
- var tld_list = telify_country_data[i][2].toLowerCase().split(",");
- for (var j=0; j<tld_list.length; j++) {
- tld_list[j] = this.trim(tld_list[j]);
- this.tld_hashtable.put(tld_list[j], telify_country_data[i][0]);
- }
- }
-},
-
-
-tld2cc: function(tld)
-{
- if (this.tld_hashtable == null) this.create_tld_hashtable();
- return this.tld_hashtable.get(tld);
-},
-
-
-splitPhoneNr: function(nr)
-{
- var index = -1;
- var maxlen = 0;
- var idd_list = ["00", "011"];
- var oldnr = nr;
-
- if (nr.charAt(0) != '+') {
- for (var i=0; i<idd_list.length; i++) {
- if (nr.substr(0, idd_list[i].length) == idd_list[i]) {
- nr = "+" + nr.substr(idd_list[i].length);
- break;
- }
- }
- }
- if (nr.charAt(0) != '+') return [null, oldnr];
- for (var i=0; i<telify_country_data.length; i++) {
- if (nr.substr(0, telify_country_data[i][0].length) == telify_country_data[i][0]) {
- if (telify_country_data[i][0].length > maxlen) {
- index = i;
- maxlen = telify_country_data[i][0].length;
- }
- }
- }
- if (index >= 0) {
- var cc = telify_country_data[index][0];
- return [cc, nr.substr(cc.length)];
- }
- return [null, oldnr];
-},
-
-
-code2name_hashtable: null,
-
-create_code2name_hashtable: function()
-{
- this.code2name_hashtable = new Hashtable();
- for (var i=0; i<telify_country_data.length; i++) {
- if (telify_country_data[i][0] == "") continue;
- var name = telify_country_data[i][1];
- var prev = this.code2name_hashtable.get(telify_country_data[i][0]);
- if (prev) name = prev + ", " + name;
- this.code2name_hashtable.put(telify_country_data[i][0], name);
- }
-},
-
-
-getCountryListString: function(prefix)
-{
- if (this.code2name_hashtable == null) this.create_code2name_hashtable();
- return this.code2name_hashtable.get(prefix);
-},
-
-
-getHost: function()
-{
- try {
- return content.document.location.host.toLowerCase();
- } catch (e) {
- return null;
- }
-},
-
-
-getHostTLD: function()
-{
- var host = this.getHost();
- if (host) {
- var index = host.lastIndexOf('.');
- if (index >= 0) {
- var tld = host.substr(index+1);
- if (tld.length) return tld;
- }
- }
- return null;
-},
-
-
-MB_MASK: 0xff, MB_OK: 1, MB_CANCEL: 2,
-MB_ICON_MASK: 0xff00, MB_ICON_INFO: 0, MB_ICON_WARNING: 0x0100, MB_ICON_ERROR: 0x0200, MB_ICON_ASK: 0x0300,
-
-showMessageBox: function(title, msg, flags)
-{
- var argObj = {title: title, msg: msg, flags: flags, fResult: true};
- window.openDialog("chrome://telify/content/messagebox.xul", "dlgTelifyMessageBox", "centerscreen,chrome,modal", argObj).focus();
- return argObj.fResult;
-},
-
-
-consoleService: null,
-
-logmsg: function(msg) {
- if (this.consoleService == null) {
- this.consoleService = Components.classes["@mozilla.org/consoleservice;1"];
- this.consoleService = this.consoleService.getService(Components.interfaces.nsIConsoleService);
- }
- this.consoleService.logStringMessage(msg);
-},
-
-
-logerror: function(msg) {
- Components.utils.reportError(msg);
-},
-
-
-arrayRemove: function(a, v)
-{
- for (var i=0; i<a.length; i++) {
- if (a[i] == v) {
- a.splice(i, 1);
- i--;
- }
- }
-},
-
-
-replaceRefs: function(string, nr, param)
-{
- var index;
- while ((index = string.indexOf("$"+nr)) >= 0 && string.charAt(index-1) != '\\') {
- string = string.substr(0, index) + param + string.substr(index+2);
- }
- return string;
-},
-
-
-substArgs: function(text)
-{
- var newText = "";
- for (var i=1; i<arguments.length && i<10; i++) {
- for (var j=0; j<text.length; j++) {
- var c = text.charAt(j);
- if (c == '$') {
- c = text.charAt(j+1);
- if (c >= '1' && c <= '9') {
- var index = c - '0';
- if (index < arguments.length) {
- newText += arguments[index];
- } else {
- this.logerror("substArgs("+text+"): argument for $"+index+" missing");
- }
- j++;
- } else {
- newText += c;
- }
- } else {
- newText += c;
- }
- }
- }
- return newText;
-},
-
-
-setIdAttr: function(name, value)
-{
- for (var i=2; i<arguments.length; i++) {
- var e = document.getElementById(arguments[i]);
- if (e) {
- e.setAttribute(name, value);
- } else {
- this.logerror("unknown element '"+arguments[i]+"'");
- }
- }
-},
-
-
-countDigits: function(text)
-{
- var count = 0;
- for (var i=0; i<text.length; i++) {
- var c = text.charAt(i);
- if (c >= '0' && c <= '9') count++;
- }
- return count;
-},
-
-
-isdigit: function(c)
-{
- return ("0123456789".indexOf(c) >= 0);
-},
-
-
-trimInt: function(value, min, max)
-{
- if (value < min) return min;
- if (value > max) return max;
- return value;
-},
-
-
-parseColor: function(text)
-{
- var exp, res, color;
-
- if (text == null) return null;
-
- exp = /^rgb *\( *(\d{1,3}) *, *(\d{1,3}) *, *(\d{1,3}) *\)$/;
- res = exp.exec(text);
- if (res) {
- color = new Array(parseInt(res[1]), parseInt(res[2]), parseInt(res[3]));
- for (var i=0; i<3; i++) {
- if (color[i] < 0) color[i] = 0;
- if (color[i] > 255) color[i] = 255;
- }
- return color;
- }
-
- exp = /^#?([\da-f]{2})([\da-f]{2})([\da-f]{2})$/i;
- res = exp.exec(text);
- if (res) {
- color = new Array(parseInt(res[1], 16), parseInt(res[2], 16), parseInt(res[3], 16));
- return color;
- }
-
- exp = /^#?([\da-f])([\da-f])([\da-f])$/i;
- res = exp.exec(text);
- if (res) {
- color = new Array(parseInt(res[1], 16), parseInt(res[2], 16), parseInt(res[3], 16));
- for (var i=0; i<3; i++) color[i] = color[i]*16+color[i];
- return color;
- }
-
- return null;
-},
-
-
-color2hex: function(color)
-{
- var hex;
-
- if (color == null || color.length != 3) return "";
- for (var i=0, hex=""; i<3; i++) {
- var d = "0"+Math.floor(color[i]).toString(16);
- hex += d.substr(d.length - 2, 2);
- }
- return hex;
-},
-
-
-esc2xml: function(string)
-{
- var substList = [
- ["&", "&"], // here be dragons: must be first element in list
- ["<", "<"],
- [">", ">"],
- ["\'", "'"],
- ["\"", """],
- ["Ä", "Ä"],
- ["Ö", "Ö"],
- ["Ü", "Ü"],
- ["ä", "ä"],
- ["ö", "ö"],
- ["ü", "ü"],
- ["ß", "ß"],
- ];
-
- for (var i=0; i<substList.length; i++) {
- var index;
- while ((index = string.indexOf(substList[i][0])) >= 0) {
- string = string.substr(0, index) + substList[i][1] + string.substr(index+substList[i][0].length);
- }
- }
-
- return string;
-},
-
-
-iso2utf8: function(s)
-{
- s = s.split("");
- for (var i=0; i<s.length; i++) {
- var c = s[i].charCodeAt(0);
- if (c > 127) s[i] = String.fromCharCode(0xc0 | ((c >> 6) & 3)) + String.fromCharCode(0x80 | (c & 0x3f));
- }
- return s.join("");
-},
-
-addScheme: function(scheme)
-{
- var createNC = function(aProperty) {return "http://home.netscape.com/NC-rdf#" + aProperty;};
-
- var RDF = Components.classes["@mozilla.org/rdf/rdf-service;1"].getService();
- var IRDFService = RDF.QueryInterface(Components.interfaces.nsIRDFService);
-
- var ContainerUtils = Components.classes["@mozilla.org/rdf/container-utils;1"].getService();
- var IRDFContainerUtils = ContainerUtils.QueryInterface(Components.interfaces.nsIRDFContainerUtils);
-
- var Container = Components.classes["@mozilla.org/rdf/container;1"].createInstance();
- var IRDFContainer = Container.QueryInterface(Components.interfaces.nsIRDFContainer);
-
- const mimeTypes = "UMimTyp";
- var fileLocator = Components.classes["@mozilla.org/file/directory_service;1"].getService(Components.interfaces.nsIProperties);
- var file = fileLocator.get(mimeTypes, Components.interfaces.nsIFile);
- var ioService = Components.classes["@mozilla.org/network/io-service;1"].getService(Components.interfaces.nsIIOService);
- var fileHandler = ioService.getProtocolHandler("file").QueryInterface(Components.interfaces.nsIFileProtocolHandler);
- var datasource = IRDFService.GetDataSource(fileHandler.getURLSpecFromFile(file));
- var irds = datasource.QueryInterface(Components.interfaces.nsIRDFRemoteDataSource);
-
- var about, property, value;
-
- about = IRDFService.GetResource("urn:schemes");
- property = IRDFService.GetResource(createNC("Protocol-Schemes"));
- value = IRDFService.GetResource("urn:schemes:root");
- datasource.Assert(about, property, value, true);
-
- about = IRDFService.GetResource("urn:schemes:root");
- if (IRDFContainerUtils.IsSeq(datasource, about) == false) {
- datasource.Assert(about, null, null, true);
- IRDFContainerUtils.MakeSeq(datasource, about);
- }
- IRDFContainer.Init(datasource, about);
- var element = IRDFService.GetResource("urn:scheme:"+scheme);
- if (IRDFContainer.IndexOf(element) < 0) {
- IRDFContainer.AppendElement(element);
- }
-
- about = IRDFService.GetResource("urn:scheme:"+scheme);
- property = IRDFService.GetResource(createNC("value"));
- value = IRDFService.GetLiteral(scheme);
- datasource.Assert(about, property, value, true);
- property = IRDFService.GetResource(createNC("handlerProp"));
- value = IRDFService.GetResource("urn:scheme:handler:"+scheme)
- datasource.Assert(about, property, value, true);
-
- about = IRDFService.GetResource("urn:scheme:handler:"+scheme);
- property = IRDFService.GetResource(createNC("alwaysAsk"));
- value = IRDFService.GetLiteral("true");
- datasource.Assert(about, property, value, true);
- property = IRDFService.GetResource(createNC("useSystemDefault"));
- value = IRDFService.GetLiteral("false");
- datasource.Assert(about, property, value, true);
-/*
- property = IRDFService.GetResource(createNC("possibleApplication"));
- value = IRDFService.GetResource("urn:scheme:possibleApplication:tel");
- datasource.Assert(about, property, value, true);
-
- about = IRDFService.GetResource("urn:scheme:possibleApplication:tel");
- property = IRDFService.GetResource(createNC("prettyName"));
- value = IRDFService.GetLiteral("Nicht konfiguriert");
- datasource.Assert(about, property, value, true);
- property = IRDFService.GetResource(createNC("uriTemplate"));
- value = IRDFService.GetLiteral("urn:handler:web:http://www.mike-koch.de");
- datasource.Assert(about, property, value, true);
-*/
- irds.Flush();
-}
-
-
-/*
- <RDF:Description RDF:about="urn:schemes">
- <NC:Protocol-Schemes RDF:resource="urn:schemes:root"/>
- </RDF:Description>
-
- <RDF:Seq RDF:about="urn:schemes:root">
- <RDF:li RDF:resource="urn:scheme:webcal"/>
- <RDF:li RDF:resource="urn:scheme:mailto"/>
- <RDF:li RDF:resource="urn:scheme:callto"/>
- <RDF:li RDF:resource="urn:scheme:tel"/>
- </RDF:Seq>
-
- <RDF:Description RDF:about="urn:scheme:tel" NC:value="tel">
- <NC:handlerProp RDF:resource="urn:scheme:handler:tel"/>
- </RDF:Description>
-
- <RDF:Description RDF:about="urn:scheme:handler:tel" NC:alwaysAsk="true">
- <NC:externalApplication RDF:resource="urn:scheme:externalApplication:tel"/>
- </RDF:Description>
-
- <RDF:Description RDF:about="urn:scheme:externalApplication:tel"
- NC:prettyName="3GP_Converter.exe"
- NC:path="C:\Programme\3GP_Converter033\3GP_Converter.exe" />
-
-*/
-
-
-};
-
diff --git a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/telify/chrome/content/warn32.png b/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/telify/chrome/content/warn32.png
deleted file mode 100644
index d5f6551d940eb76b48597f3f9bf09e2a3395b090..0000000000000000000000000000000000000000
Binary files a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/telify/chrome/content/warn32.png and /dev/null differ
diff --git a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/telify/chrome/locale/de-DE/country_locale.js b/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/telify/chrome/locale/de-DE/country_locale.js
deleted file mode 100644
index 57e1c7035d4edb12e9b82fdeace05f9f351833e8..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/telify/chrome/locale/de-DE/country_locale.js
+++ /dev/null
@@ -1,158 +0,0 @@
-// caveat: save as UTF-8
-var telify_country_locale = [
-['U.S. Virgin Islands', 'Amerikanische Jungferninseln'],
-['Northern Mariana Islands', 'Nördliche Marianen'],
-['American Samoa', 'Amerikanisch-Samoa'],
-['Canada', 'Kanada'],
-['Antigua and Barbuda', 'Antigua und Barbuda'],
-['British Virgin Islands', 'Britische Junferninseln'],
-['Cayman Islands', 'Kaimaninseln'],
-['Dominican Republic', 'Dominikanische Republik'],
-['Jamaica', 'Jamaika'],
-['Saint Kitts and Nevis', 'Saint Kitts und Nevis'],
-['Saint Lucia', 'St. Lucia'],
-['Saint Vincent and the Grenadines', 'St. Vincent und die Grenadinen'],
-['Trinidad and Tobago', 'Trinidad und Tobago'],
-['Turks and Caicos Islands', 'Turks- und Caicosinseln'],
-['Egypt', 'Ägypten'],
-['Morocco', 'Marokko'],
-['Algeria', 'Algerien'],
-['Tunisia', 'Tunesien'],
-['Libya', 'Libyen'],
-['Mauritania', 'Mauretanien'],
-['Ivory Coast', 'Elfenbeinküste'],
-['Ghana', 'Gana'],
-['Chad', 'Tschad'],
-['Central African Republic', 'Zentralafrikanische Republik'],
-['Cameroon', 'Kamerun'],
-['Cape Verde', 'Kap Verde'],
-['São Tomé and Príncipe', 'São Tomé und Príncipe'],
-['Equatorial Guinea', 'Äquatorialguinea'],
-['Gabon', 'Gabun'],
-['Congo (Republic)', 'Kongo (Republik)'],
-['Congo (Democratic Republic)', 'Kongo (Demokratische Republik)'],
-['Ascension Island', 'Ascension'],
-['Seychelles', 'Seychellen'],
-['Rwanda', 'Ruanda'],
-['Ethiopia', 'Äthiopien'],
-['Djibouti', 'Dschibuti'],
-['Kenya', 'Kenia'],
-['Tanzania', 'Tansania'],
-['Mozambique', 'Mosambik'],
-['Zambia', 'Sambia'],
-['Madagascar', 'Madagaskar'],
-['Zimbabwe', 'Simbabwe'],
-['Botswana', 'Botsuana'],
-['Swaziland', 'Swasiland'],
-['Comoros', 'Komoren'],
-['South Africa', 'Südafrika'],
-['Saint Helena', 'St.Helena'],
-['Faroe Islands', 'Färöer'],
-['Greenland', 'Grönland'],
-['Greece', 'Griechenland'],
-['Netherlands', 'Niederlande'],
-['Belgium', 'Belgien'],
-['France', 'Frankreich'],
-['Spain', 'Spanien'],
-['Luxembourg', 'Luxemburg'],
-['Ireland', 'Irland'],
-['Iceland', 'Island'],
-['Albania', 'Albanien'],
-['Cyprus (South)', 'Zypern (Süden)'],
-['Finland', 'Finnland'],
-['Bulgaria', 'Bulgarien'],
-['Hungary', 'Ungarn'],
-['Lithuania', 'Litauen'],
-['Latvia', 'Lettland'],
-['Estonia', 'Estland'],
-['Moldova', 'Moldawien'],
-['Armenia', 'Armenien'],
-['Nagorno-Karabakh', 'Bergkarabach'],
-['Nagorno-Karabakh (Mobile)', 'Bergkarabach (Handynetz)'],
-['Belarus', 'Weißrussland'],
-['Kosovo (Mobile)', 'Kosovo (Handynetz)'],
-['Serbia', 'Serbien'],
-['Croatia', 'Kroatien'],
-['Slovenia', 'Slowenien'],
-['Kosovo (Mobile)', 'Kosovo (Handynetz)'],
-['Bosnia and Herzegovina', 'Bosnien und Herzegowina'],
-['Macedonia', 'Mazedonien'],
-['Italy and Vatican City', 'Italien und Vatikanstadt'],
-['Romania', 'Rumänien'],
-['Switzerland', 'Schweiz'],
-['Czech Republic', 'Tschechien'],
-['Slovakia', 'Slowakei'],
-['Austria', 'Österreich'],
-['United Kingdom', 'Großbritannien'],
-['Denmark', 'Dänemark'],
-['Sweden', 'Schweden'],
-['Norway', 'Norwegen'],
-['Poland', 'Polen'],
-['Germany', 'Deutschland'],
-['Falkland Islands', 'Falklandinseln'],
-['Saint-Pierre and Miquelon', 'Saint-Pierre und Miquelon'],
-['Mexico', 'Mexiko'],
-['Cuba', 'Kuba'],
-['Argentina', 'Argentinien'],
-['Brazil', 'Brasilien'],
-['Colombia', 'Kolumbien'],
-['Bolivia', 'Bolivien'],
-['French Guiana', 'Französisch-Guayana'],
-['Suriname', 'Surinam'],
-['Netherlands Antilles', 'Niederländische Antillen'],
-['Malaysia', 'Malaisia'],
-['Australia', 'Australien'],
-['Indonesia', 'Indonesien'],
-['Philippines', 'Philippinen'],
-['New Zealand', 'Neuseeland'],
-['Singapore', 'Singapur'],
-['East Timor', 'Ost-Timor'],
-['Australian external territories', 'Australische Außengebiete'],
-['Papua New Guinea', 'Papua-Neuguinea'],
-['Solomon Islands', 'Salomonen'],
-['Fiji', 'Fidschi'],
-['Wallis and Futuna', 'Wallis und Futuna'],
-['Cook Islands', 'Cook-Inseln'],
-['Niue Island', 'Niue'],
-['New Caledonia', 'Neukaledonien'],
-['French Polynesia', 'Französisch-Polynesien'],
-['Micronesia', 'Mikronesien'],
-['Marshall Islands', 'Marshallinseln'],
-['Russia', 'Russland'],
-['Kazakhstan', 'Kasachstan'],
-['South Korea', 'Südkorea'],
-['North Korea', 'Nordkorea'],
-['Hong Kong', 'Hongkong'],
-['Macau', 'Macao'],
-['Cambodia', 'Kambodscha'],
-['Inmarsat (Atlantic East)', 'Inmarsat (Ostatlantik)'],
-['Inmarsat (Pacific)', 'Inmarsat (Pazifik)'],
-['Inmarsat (Indian)', 'Inmarsat (Indien)'],
-['Inmarsat (Atlantic West)', 'Inmarsat (Westatlantik)'],
-['Bangladesh', 'Bangladesch'],
-['Global Mobile Satellite System', 'Globales mobiles Satellitensystem'],
-['International Networks', 'Internationale Netzwerke'],
-['Turkey', 'Türkei'],
-['Cyprus (North)', 'Zypern (Nord)'],
-['India', 'Indien'],
-['Maldives', 'Malediven'],
-['Lebanon', 'Libanon'],
-['Jordan', 'Jordanien'],
-['Syria', 'Syrien'],
-['Iraq', 'Irak'],
-['Saudi Arabia', 'Saudi Arabien'],
-['Yemen', 'Jemen'],
-['United Arab Emirates', 'Vereinigte Arabische Emirate'],
-['Bahrain', 'Barain'],
-['Qatar', 'Katar'],
-['Bhutan', 'Butan'],
-['Mongolia', 'Mongolei'],
-['Tajikistan', 'Tadschikistan'],
-['Azerbaijan', 'Aserbaidschan'],
-['Georgia', 'Georgien'],
-['Kyrgyzstan', 'Kirgisistan'],
-['Uzbekistan', 'Usbekistan'],
-['Guantanamo Bay', 'Guantanamo'],
-['Midway Island', 'Midway Inseln'],
-['Vatican City', 'Vatikanstadt'],
-];
diff --git a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/telify/chrome/locale/de-DE/custom_preset.js b/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/telify/chrome/locale/de-DE/custom_preset.js
deleted file mode 100644
index 7f248be61dd3df535a0d4d0d23404feb3d520058..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/telify/chrome/locale/de-DE/custom_preset.js
+++ /dev/null
@@ -1,8 +0,0 @@
-/* (c)2009 Michael Koch
-*/
-
-/* name, url, parameter #1, parameter #2, parameter #3 */
-var telify_custom_preset = [
- ["", "", "Parameter #1", "Parameter #2", "Parameter #3"],
- ["Vorlage für snom-Telefone", "http://$1/command.htm?number=$0&outgoing_uri=$2", "Telefon-IP", "Ausgehende URI", ""],
-];
diff --git a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/telify/chrome/locale/de-DE/lang.dtd b/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/telify/chrome/locale/de-DE/lang.dtd
deleted file mode 100644
index 797fd04dcf40dce7fb7f609244f43e7e6fe84b61..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/telify/chrome/locale/de-DE/lang.dtd
+++ /dev/null
@@ -1,39 +0,0 @@
-<!ENTITY menu.edit_number "Telefonnummer bearbeiten">
-<!ENTITY menu.selection "Rufnummernauswahl">
-<!ENTITY menu.config "Einstellungen">
-<!ENTITY menu.onlinehelp "Online-Hilfe">
-<!ENTITY dialog.edit.title "Telefonnummer bearbeiten">
-<!ENTITY dialog.edit.code "Vorwahl">
-<!ENTITY dialog.edit.country "Land">
-<!ENTITY dialog.edit.dial "Wählen">
-<!ENTITY dialog.config.title "Telify-Einstellungen">
-<!ENTITY dialog.config.general "Allgemeine Einstellungen">
-<!ENTITY dialog.config.custom "Eigene URL">
-<!ENTITY dialog.config.about "Info">
-<!ENTITY dialog.config.replaces "Ersetzt">
-<!ENTITY dialog.config.in_template "in der Vorlage">
-<!ENTITY dialog.config.empty_url "Vorlage unten eingeben oder aus der Liste auswählen">
-<!ENTITY dialog.config.idd_prefix "Ersetze '+' durch">
-<!ENTITY dialog.config.hreftype "Verwendetes Protokoll">
-<!ENTITY dialog.config.hreftype0 "tel:">
-<!ENTITY dialog.config.hreftype1 "callto:">
-<!ENTITY dialog.config.hreftype2 "skype:">
-<!ENTITY dialog.config.hreftype3 "sip:">
-<!ENTITY dialog.config.hreftype_custom "Eigene URL">
-<!ENTITY dialog.config.dialcc "Bei vorhandener Landesvorwahl">
-<!ENTITY dialog.config.dialcc_menu "Öffne Menu">
-<!ENTITY dialog.config.dialcc_direct "Wähle direkt">
-<!ENTITY dialog.config.highlight "Texthervorhebung">
-<!ENTITY dialog.config.highlight0 "Keine">
-<!ENTITY dialog.config.highlight1 "Leicht">
-<!ENTITY dialog.config.highlight2 "Mittel">
-<!ENTITY dialog.config.highlight3 "Stark">
-<!ENTITY dialog.config.num_history "Anzahl der Nummerneinträge">
-<!ENTITY dialog.config.statusicon "Status-Icon anzeigen">
-<!ENTITY dialog.config.statusicon0 "Nein">
-<!ENTITY dialog.config.statusicon1 "Ja">
-<!ENTITY dialog.config.opentype "Öffne Link">
-<!ENTITY dialog.config.opentype0 "im Hintergrund">
-<!ENTITY dialog.config.opentype1 "in einem neuen Fenster">
-<!ENTITY dialog.config.opentype2 "in einem neuen Tab ohne Fokus">
-<!ENTITY dialog.config.opentype3 "in einem neuen Tab mit Fokus">
diff --git a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/telify/chrome/locale/de-DE/lang.properties b/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/telify/chrome/locale/de-DE/lang.properties
deleted file mode 100644
index 52c37336e6fbfb04829831cb42b8bf9929c824d3..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/telify/chrome/locale/de-DE/lang.properties
+++ /dev/null
@@ -1,13 +0,0 @@
-leave_blank=
-converted=Konvertiert
-telify_active=Telify ist aktiv
-telify_inactive=Telify ist inaktiv
-telify_activate=Telify aktivieren
-telify_deactivate=Telify deaktivieren
-call_arg=$1 anrufen
-host_active_arg=Auf $1 aktivieren
-host_inactive_arg=Auf $1 deaktivieren
-link_title=Wählbare Rufnummer
-country_code=Landesvorwahl
-empty_url=Vorlage unten eingeben oder aus der Liste auswählen
-phonenr_tmpl=[TelNr]
diff --git a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/telify/chrome/locale/de-DE/locale.js b/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/telify/chrome/locale/de-DE/locale.js
deleted file mode 100644
index 539c6126e6c459e0acecc121155d405ea03a6b0a..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/telify/chrome/locale/de-DE/locale.js
+++ /dev/null
@@ -1,25 +0,0 @@
-/* (c)2009 Michael Koch
-*/
-
-var objTelifyLocale = {
-
-openOnlineHelp: function()
-{
- var browser = objTelifyUtil.getBrowser();
- var tab = browser.addTab("http://www.codepad.de/de/download/firefox-add-ons/telify.html");
- browser.selectedTab = tab;
-},
-
-msgNumberTemplateMissing: function()
-{
- return "Ihre Vorlage enthält keinen Platzhalter für die Telefonnummer (d.h. '$0') und wird deshalb keine Telefonnummer übermitteln. "
- + "Wollen Sie das wirklich?";
-},
-
-msgUnknownProtocol: function()
-{
- return "Im diesem System ist keine Anwendung installiert, die sich für das verwendete Protokoll registriert hat. "
- + "Bitte stellen Sie in der Telify-Konfiguration ein geeignetes Protokoll ein oder installieren Sie eine geeignete Anwendung.";
-}
-
-}
diff --git a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/telify/chrome/locale/en-US/country_locale.js b/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/telify/chrome/locale/en-US/country_locale.js
deleted file mode 100644
index 8d0143c5c15e061727e83655aa800824acfe78f5..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/telify/chrome/locale/en-US/country_locale.js
+++ /dev/null
@@ -1,3 +0,0 @@
-var telify_country_locale = [
-/* for en-US this is empty */
-];
diff --git a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/telify/chrome/locale/en-US/custom_preset.js b/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/telify/chrome/locale/en-US/custom_preset.js
deleted file mode 100644
index e8d3b6be973d7adba081867a9ca0b0fadb2265ea..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/telify/chrome/locale/en-US/custom_preset.js
+++ /dev/null
@@ -1,8 +0,0 @@
-/* (c)2009 Michael Koch
-*/
-
-/* name, url, parameter #1, parameter #2, parameter #3 */
-var telify_custom_preset = [
- ["", "", "Parameter #1", "Parameter #2", "Parameter #3"],
- ["snom phones template", "http://$1/command.htm?number=$0&outgoing_uri=$2", "Telefon-IP", "Ausgehende URI", ""],
-];
diff --git a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/telify/chrome/locale/en-US/lang.dtd b/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/telify/chrome/locale/en-US/lang.dtd
deleted file mode 100644
index 0af8755f0e19f2aa45d35f7790b3824f6663b716..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/telify/chrome/locale/en-US/lang.dtd
+++ /dev/null
@@ -1,38 +0,0 @@
-<!ENTITY menu.edit_number "Edit phone number">
-<!ENTITY menu.selection "Phone number selection">
-<!ENTITY menu.config "Preferences">
-<!ENTITY menu.onlinehelp "Online Help">
-<!ENTITY dialog.edit.title "Edit phone number">
-<!ENTITY dialog.edit.code "Code">
-<!ENTITY dialog.edit.country "Country">
-<!ENTITY dialog.edit.dial "Dial">
-<!ENTITY dialog.config.title "Telify Preferences">
-<!ENTITY dialog.config.general "General Settings">
-<!ENTITY dialog.config.custom "Custom URL">
-<!ENTITY dialog.config.about "About">
-<!ENTITY dialog.config.replaces "Replaces">
-<!ENTITY dialog.config.in_template "in template string">
-<!ENTITY dialog.config.idd_prefix "Replace '+' with">
-<!ENTITY dialog.config.hreftype "Used protocol">
-<!ENTITY dialog.config.hreftype0 "tel:">
-<!ENTITY dialog.config.hreftype1 "callto:">
-<!ENTITY dialog.config.hreftype2 "skype:">
-<!ENTITY dialog.config.hreftype3 "sip:">
-<!ENTITY dialog.config.hreftype_custom "Custom URL">
-<!ENTITY dialog.config.dialcc "When number has country code">
-<!ENTITY dialog.config.dialcc_menu "open menu">
-<!ENTITY dialog.config.dialcc_direct "dial directly">
-<!ENTITY dialog.config.highlight "Text highlighting">
-<!ENTITY dialog.config.highlight0 "None">
-<!ENTITY dialog.config.highlight1 "Light">
-<!ENTITY dialog.config.highlight2 "Medium">
-<!ENTITY dialog.config.highlight3 "Strong">
-<!ENTITY dialog.config.num_history "Number of recent country codes">
-<!ENTITY dialog.config.statusicon "Status icon">
-<!ENTITY dialog.config.statusicon0 "Hide">
-<!ENTITY dialog.config.statusicon1 "Show">
-<!ENTITY dialog.config.opentype "Open link">
-<!ENTITY dialog.config.opentype0 "silently in the background">
-<!ENTITY dialog.config.opentype1 "in a new window">
-<!ENTITY dialog.config.opentype2 "in a new tab without focus">
-<!ENTITY dialog.config.opentype3 "in a new tab with focus">
diff --git a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/telify/chrome/locale/en-US/lang.properties b/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/telify/chrome/locale/en-US/lang.properties
deleted file mode 100644
index ce7ed93d4997f6b269a4da7a70a15ffec20e8425..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/telify/chrome/locale/en-US/lang.properties
+++ /dev/null
@@ -1,13 +0,0 @@
-leave_blank=
-converted=Converted
-telify_active=Telify is active
-telify_inactive=Telify is inactive
-telify_activate=Activate Telify
-telify_deactivate=Deactivate Telify
-call_arg=Call $1
-host_active_arg=Activate on $1
-host_inactive_arg=Deactivate on $1
-link_title=phone number
-country_code=Country Code
-empty_url=Enter template below or choose from the list
-phonenr_tmpl=[phonenr]
diff --git a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/telify/chrome/locale/en-US/locale.js b/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/telify/chrome/locale/en-US/locale.js
deleted file mode 100644
index 5240f8585d315d5a39fa0e7bac4a807f99d3d7a5..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/telify/chrome/locale/en-US/locale.js
+++ /dev/null
@@ -1,25 +0,0 @@
-/* (c)2009 Michael Koch
-*/
-
-var objTelifyLocale = {
-
-openOnlineHelp: function()
-{
- var browser = objTelifyUtil.getBrowser();
- var tab = browser.addTab("http://www.codepad.de/en/download/firefox-add-ons/telify.html");
- browser.selectedTab = tab;
-},
-
-msgNumberTemplateMissing: function()
-{
- return "Your template does not contain a placeholder for the phone number (i.e. '$0') and will therefore not transmit a phone number. "
- + "Do you really want to continue?";
-},
-
-msgUnknownProtocol: function()
-{
- return "No application is installed which registered itself for the used protocol. "
- + "Please configure a suitable protocol in the Telify preferences or install a suitable application.";
-}
-
-}
diff --git a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/telify/defaults/preferences/preferences.js b/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/telify/defaults/preferences/preferences.js
deleted file mode 100644
index b0bb58542d7bf77b8af405187f0174a00a620341..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/telify/defaults/preferences/preferences.js
+++ /dev/null
@@ -1,18 +0,0 @@
-pref("telify.settings.blacklist", "");
-pref("telify.settings.highlight", 25);
-pref("telify.settings.debug", false);
-pref("telify.settings.active", true);
-pref("telify.settings.exclude", "a,applet,map,select,script,textarea");
-pref("telify.settings.statusicon", true);
-pref("telify.settings.linktype", 0);
-pref("telify.settings.colsortcc", 1);
-pref("telify.settings.num_history", 5);
-pref("telify.settings.idd_prefix", "");
-pref("telify.settings.custom_url", "");
-pref("telify.settings.custom_tmpl", 0);
-pref("telify.settings.custom_param1", "");
-pref("telify.settings.custom_param2", "");
-pref("telify.settings.custom_param3", "");
-pref("telify.settings.custom_opentype", 3);
-pref("telify.settings.dont_escape_plus", false);
-pref("telify.settings.dial_cc_direct", false);
diff --git a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/telify/install.rdf b/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/telify/install.rdf
deleted file mode 100644
index 5128cb4f705a03d9e13df17e06a9472d8e09cc73..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/telify/install.rdf
+++ /dev/null
@@ -1,37 +0,0 @@
-<?xml version="1.0"?>
-<RDF xmlns="http://www.w3.org/1999/02/22-rdf-syntax-ns#" xmlns:em="http://www.mozilla.org/2004/em-rdf#">
- <Description about="urn:mozilla:install-manifest">
- <em:id>{6c5f349a-ddda-49ad-bdf0-326d3fe1f938}</em:id>
- <em:extension>true</em:extension>
- <em:iconURL>chrome://telify/content/icon32.png</em:iconURL>
- <em:version>0.4.7.3</em:version>
- <em:creator>Michael Koch</em:creator>
- <em:homepageURL>http://www.codepad.de/</em:homepageURL>
- <em:optionsURL>chrome://telify/content/config.xul</em:optionsURL>
-
- <em:localized>
- <Description>
- <em:locale>de-DE</em:locale>
- <em:name>Telify</em:name>
- <em:description>Erzeugt klickbare Links aus Telefonnummern</em:description>
- </Description>
- </em:localized>
-
- <em:localized>
- <Description>
- <em:locale>en-US</em:locale>
- <em:name>Telify</em:name>
- <em:description>Converts telephone numbers into clickable links</em:description>
- </Description>
- </em:localized>
-
- <!-- Firefox -->
- <em:targetApplication>
- <Description>
- <em:id>{ec8030f7-c20a-464f-9b0e-13a3a9e97384}</em:id>
- <em:minVersion>3.0</em:minVersion>
- <em:maxVersion>3.6.*</em:maxVersion>
- </Description>
- </em:targetApplication>
- </Description>
-</RDF>
\ No newline at end of file
diff --git a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/postinst b/tools/build-system/launchpad/mozilla-telify-sflphone/debian/postinst
deleted file mode 100755
index 1039df3268eae8fca20a728add6c4939f7494360..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/postinst
+++ /dev/null
@@ -1,16 +0,0 @@
-#!/bin/bash
-
-gconftool-2 --direct --config-source xml::/etc/gconf/gconf.xml.defaults -t string -s /desktop/gnome/url-handlers/tel/command "/usr/bin/sflphone-handler %s"
-gconftool-2 --direct --config-source xml::/etc/gconf/gconf.xml.defaults -s /desktop/gnome/url-handlers/tel/needs_terminal false -t bool
-gconftool-2 --direct --config-source xml::/etc/gconf/gconf.xml.defaults -t bool -s /desktop/gnome/url-handlers/tel/enabled true
-
-gconftool-2 --direct --config-source xml::/etc/gconf/gconf.xml.defaults -t string -s /desktop/gnome/url-handlers/callto/command "/usr/bin/sflphone-handler %s"
-gconftool-2 --direct --config-source xml::/etc/gconf/gconf.xml.defaults -s /desktop/gnome/url-handlers/callto/needs_terminal false -t bool
-gconftool-2 --direct --config-source xml::/etc/gconf/gconf.xml.defaults -t bool -s /desktop/gnome/url-handlers/callto/enabled true
-
-gconftool-2 --direct --config-source xml::/etc/gconf/gconf.xml.defaults -t string -s /desktop/gnome/url-handlers/sip/command "/usr/bin/sflphone-handler %s"
-gconftool-2 --direct --config-source xml::/etc/gconf/gconf.xml.defaults -s /desktop/gnome/url-handlers/sip/needs_terminal false -t bool
-gconftool-2 --direct --config-source xml::/etc/gconf/gconf.xml.defaults -t bool -s /desktop/gnome/url-handlers/sip/enabled true
-
-exit 0
-
diff --git a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/rules b/tools/build-system/launchpad/mozilla-telify-sflphone/debian/rules
deleted file mode 100755
index d002bc28395f49919b2bf8c9569e39f26f727577..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/rules
+++ /dev/null
@@ -1,56 +0,0 @@
-#!/usr/bin/make -f
-
-# Uncomment this to turn on verbose mode.
-export DH_VERBOSE=1
-
-VERSION="1.0"
-
-configure: configure-stamp
-configure-stamp:
- dh_testdir
- touch configure-stamp
-
-build: configure-stamp build-stamp
-build-stamp:
- dh_testdir
-
- echo ${DIR}
- umask 0022
- mkdir -p tmp/telify
- unzip telify-${VERSION}-fx.xpi -d tmp/telify
- touch build-stamp
-
-clean:
- dh_testdir
- dh_testroot
- rm -f build-stamp configure-stamp
- dh_clean
-
- rm -rf tmp
-
-install: build
- dh_testdir
- dh_testroot
- dh_prep
- dh_installdirs
- dh_install
-
-# Build architecture-independent files here.
-binary-indep: build install
- dh_testdir
- dh_testroot
- dh_installchangelogs -i
- dh_link -i
- dh_compress -XMPL -i
- dh_fixperms -i
- dh_installdeb -i
- dh_gencontrol -i
- dh_md5sums -i
- dh_builddeb -i
-
-# Build architecture-dependent files here.
-binary-arch: build install
-# We have nothing to do by default.
-
-binary: binary-indep binary-arch
-.PHONY: build clean binary-indep binary-arch binary install configure
diff --git a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/watch b/tools/build-system/launchpad/mozilla-telify-sflphone/debian/watch
deleted file mode 100644
index 5836cddd953dce1409e8c7a0794adc530a246362..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/watch
+++ /dev/null
@@ -1,2 +0,0 @@
-version=3
-http://www.codepad.de/en/download/firefox-add-ons/telify.html /download/Telify-(.*)-fx-tb.xpi
diff --git a/tools/build-system/launchpad/sflphone-daemon-video/debian/changelog b/tools/build-system/launchpad/sflphone-daemon-video/debian/changelog
deleted file mode 100644
index 4d80576014aa9c544610b616d4eddaa8ae4534b9..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/sflphone-daemon-video/debian/changelog
+++ /dev/null
@@ -1,3585 +0,0 @@
-sflphone-daemon-video (1.1.0-rc20120607~ppa1~SYSTEM) SYSTEM; urgency=low
-
- ** SNAPSHOT 1.1.0-rc20120607~ppa1~SYSTEM **
-
- * * #12071: audiopreferences: fix make check
- * * #12071: cleanup
- * #12071: Lower expat dependency version
- * * #12085: alsa: fix ringtone update bug, cleanup
- * #12071: Fix pulseaudio compilation error
- * #12071: Make pulseaudio optional at configuration time
- * * #12091: daemon: issue warning if falling back to ALSA
- * video: fixed make check
- * *#12085: alsa: don't segfault on snd_pcm_avail_update error
- * #12070: Add --without-pulse option
- * * #12055: video: fix build for older libav
- * [ #11886 ] cleanup
- * [ #11886 ] Add basic reverse peer naming support
- * [ #12008 ] Implement GUI part
- * * #12012: video: fix some regressions
- * * #12002: yamlparser: don't wipe out config if going from normal
- build to --enable-video
- * [ #12008 ] Add ConfigurationManager::getRingtoneList()
- * * #12002: video: fix config file serialization/deserialization
- * * #11987: managerimpl: fix bugs in conference when removing
- participants
- * * #11987: manager: fixed transfer from conference
- * * #11987: mainbuffer: cleanup logging
- * * #11979: pulse: fixed mismatched device list
- * * #11971: audiolayer: fix bugs with getDeviceList
- * * #11960: manager: validate conference earlier when processing
- participants
- * * #11960: manager: fixed segfault on transfer from conference
- * * #11966: IP2IP: make alias consistently IP2IP
- * * #11965: sipvoiplink: add more error checking in SDP negotiation
- * * #11964: mainbuffer/ringbuffer: cleanup API
- * sdp: remove unused variable warning
- * * #11941: video: fix deprecated libav_api warnings
- * * #11949: pulselayer: fix bug in getDeviceList
- * video: whitespace fixes
- * * #11951: video: fixed threading issues for ucommon Thread
- * [#11848] Properly disable testPulseConnect
- * [#11848] Disable pulseConnect test
- * sdp: cleanup
- * sdp: cleanup
- * * #11860: mainbuffer: remove dead and/or buggy code
- * * #11851: sdp: fixed gcc type narrowing warnings
- * * #11851: audiostream: fixed gcc type narrowing warnings
- * * #11841: don't put code with side effects in assert()
- * * #11840: audiortp: remove some global symbols/variables
- * * #11828: audiofile: fix broken build
- * * #11828: audioloop: don't shadow sampleRate variable in derived
- classes
- * * #11818: gnome: stop daemon on SIGTERM, SIGINT or SIGHUP
- * * #11499: daemon should also quit gracefully on SIGHUP
- * * #11813: daemon configure should fail if expat is not installed
- * #10304: updatePlaybackScale dbus method uses int 32 bit for size and
- position (allow for 24 days long recording playback)
- * #10304: Add time lable for seekslider
- * * #11780: sip: don't use abort or leak calls on error and don't
- restrict SDP size to 1000 bytes in transaction_request_cb
- * * #11735: daemon: added timestamp start to call details
- * * #10304: historyitem: added operator > defined in terms of operator
- <
- * * #11252: historyitem: added missing unistd.h header
- * * #11252: daemon: removed deprecated zrtp code
- * * #11728: yaml: check that nodes are valid before using them.
- * * #10797: send DTMF over RTP as per RFC2833
- * * #11706: managerimpl: added unsetCurrentCall method
- * * #11698: daemon: fix build for c++11
- * * #10304: historyitem: file_exists need not be a member method
- * * #11499: managerimpl: don't crash if signal and dbus try and finish
- the manager at the same time
- * #10304: Prevent from storing removed files in history
- * * #11499: daemon: Exit cleanly on SIGINT or SIGTERM
- * * #10226: audiocodecfactory: use array instead of vector for codec
- name lookup
- * * #10226: audiocodecfactory: make codec loading stricter
- * #10304: RCecale positions and size values for playback recording
- * #11530: Make sure that only appropriate configuration option are
- parsed for IP2IP calls
- * * 11480: video: disabled by default
- * * #11459: history: protect historyitems vector with mutex
- * * #11448: fix video preferences for empty camera list
- * #10304: Implemented playback seek in gnome client
- * * #11269: video: fix codec per account management
- * #10304: Implemented playback scale in gnome client
- * Fix includes for gcc 4.7
- * * #11269: make clearer distinction between codecs and audiocodecs
- * * #11269: merged master into video
- * * #10296: managerimpl: more usage of getCallFromCallID
- * * #10296: verify that calls exists before trying to join them in a
- conference
- * * #11208: bump version numbers for release 1.1.0
- * managerimpl: rename ManagerImpl::serialize/unserialize ->
- join_string/split_string
- * Fix warnings in resampler test
- * * #10732: sipvoiplink: fix code that validates IP address
- * Add historyChanged signal, better than managing it client side
- * #10795: fix sipaccount deserialisation broken
- * #10736: implement getConferenceId dbus method given a call id
- * #10736: do not use iterator in daemon when joining conferences
- * #10736: Fix joining conferences in daemon
- * * #10736: gnome: fix crash on restart with active conference
- * managerimpl: removed unused pulselayer.h header
- * Save history everytime it change, prevent the file never to be saved
- in some senario (SIG, crash, ASSERT, etc)
- * * #10320: manager: check that participants are unique before joining
- * #10335: Add a noise suppressor for incoming rtp streams
- * * #10322: sip: registration state should not be always set to
- ErrorAuth on error
- * #10220: Fix recording thread does not exit when hanging up while
- recording
- * * #9903: fix includes for new ccrtp
- * #10230: Get back default mainbuffer sampling rate to 8kHz, no need
- of decoding noise suppressor
- * #10230: Use a different samplerate converter for rtp encoding and
- decoding
- * * #9903: create DynamicPayloadFormat on stack, initialize earlier
- * * #9903: audiorecorder: initialize buffer to silence, not random
- data
- * #10230: Test for triangular and sine signals
- * #10230: Add resampling unit test
- * * #10230: DTMF sample rate should come from main buffer, it should
- not be hardcoded
- * * #10213: audiolayer: create samplerateconverter on the stack
- * * #10213: audiolayer: cleanup
- * * #10213: increase resample buffer size, and check output size when
- resampling
- * * #10095: sipvoiplink: check pointers before using them
- * #9981: IP2IP calls based on ip address instead of sip:
- * * #10213: speex codecs should initialize their own parameters
- * * #10213: account: removed redundant cast
- * * #9832: removed extra printf
- * * #10172: include -sflphone in recording file name
- * * #9832: cleanup logging in tests
- * * #10096: srtp: use vectors to simplify key/salt manipulation
- * #10096: add case for non-srtp calls
- * [ #10121 ] Sync the KDE with daemon, fix a few issues and implement
- a recorded call player
- * #9980: make keep registration optional as there is different
- behavior on different registrar
- * #10096: use c++ arrays to store keys in srtp sesssion
- * * #10018: renamed registration related keys in dbus
- * #10096: Fix onhold/offhold srtp
- * * #8586: fixed make distcheck
- * * #9832: logger: don't hide logging if NDEBUG is present
- * #10096: Reinit crypto context when required on INVITE request
- * * #10111: Fixes segfault on empty config file
- * #100096: Set in/out queue crypto context at initialization, not when
- starting the thread
- * #10096: Update srtp key generation when holding/unholding a call
- * * #9831: logger: removed extraneous carriage-return character
- * * #10095: sipvoiplink: validate pointers before using them
- * * #10094: renamed config/config.{h,cpp} config/sfl_config.{h,cpp}
- * * #10090: fix segfault in transaction_state_changed_cb
- * * #9832: audio_rtp_record_handler: cleanup logging
- * * #9832: pulse: cleanup logging
- * * #9832: cleanup logging
- * * #9832: cleanup logging
- * * #9832: dbus: fix logging
- * * #9832: config: cleanup logging
- * * #9832: remove unused header
- * * #9832: manager: fix logging
- * * #9832: audio: fix logging
- * * #9832: audio: fix logging
- * * #9832: zrtp: cleanup logging
- * * #9832: AudioZRTPSession: cleanup logging
- * * #9832: AudioSRTPSession: fix logging
- * * #9832: cleanup logging
- * * #9832: AudioRtpSession: cleanup logging
- * * #9832: AudioRtpFactory: cleanup logging
- * * #9832: codecs: fix logging
- * * #9832: alsa: fix logging
- * * #9832: audio: clean up logging
- * * #9832: AudioRecord: cleanup logging
- * * #9832: Fix logging in Manager
- * * #9832: new logging macros
- * * #9979: ulaw: fixed unused var warning
- * * #10039: sipvoiplink: use references to avoid unnecessary parameter
- validation
- * * #10039: Fixed segfault on failed registration
- * * #9979: codecs: fixed unused variable warnings
- * * #9979: Don't do runtime assertions on data.
- * * #10016: SDP: removed verbose debuggin
- * #10016: Crypto context deletion are now managed inside the library
- * * #10016: srtp: cleanup
- * * #10016: SDES: fix uninitialized value bug, use const char*
- * * 100016: don't double free crypto contexts, and don't improperly
- copy CryptoSuiteDefinitions
- * * #100016: cleanup crypto contexts in audio_srtp_session
- * * #9979: removed unused methods from audicodec
- * * #9979: ulaw: normalize types
- * * #9979: cleanup
- * * #9979: Alaw: cleanup
- * * #9979: removed duplicate/superfluous code and type issues from
- g722
- * * #9979: AudioRtpRecord: let AudioRtpRecord handle fadeIn internally
- * #9980: Fix registration timer and transport shutdown on 401, default
- registration timer to 3600
- * * #9979: use std::tr1::array instead of plain array for audio
- buffers
- * * #9969: set loose routing param when creating route set
- * #9975: Fix account registration status display
- * * #9969: SIP: initialize body earlier
- * * #9969: sip: get received and rport fields if present in OK
- * * #9968: fixed segfault in transaction callback
- * #9898: make sure account are unregistered when sflphone quit, add
- timeout on pending transaction
- * #9910: fix contact header in outgoing request if via parameter are
- present
- * * #9910: SIP: use rport from VIA header if present
- * * #9910: SipTransport: pass parameters by const reference
- * #9910: fix sending call with new transport
- * yaml: remove verbose debug messages
- * * #9911: sipvoiplink: fixed "unused variable" warning
- * #9910: create new udp transport to fix registration failure with 606
- error & received parameter
- * * #9910: SIP: use pjsip error codes instead of magic numbers
- * * #9911: SIP Transports must be cached by IP:port
- * * #9910: SIP: cleanup
- * * #9905: SipTransport: address has to stay on stack to be valid
- * #9910: Update parse received parameter on 606 registration error
- * * #9911: simplify network manager state reporting
- * #9902: Fix SIPTest for IP to IP call
- * #9911: Fix network manager crashes
- * #9902: Move logic for ip2ip call in SIPVoIPLink
- * #9902: Move logic for ip2ip call in SIPVoIPLink
- * * #9910: fix 606 error code nomenclature
- * * #9905: fixed address initialization in createUdpTransport
- * * #9903: cleanup
- * #9902: Log failure cause when new outgoing call fail
- * * #9898: properly initialize ports
- * #9898: Unregister account when leaving sflphone
- * iax: create iaxvoiplink on stack
- * account: removed unused methods
- * * #9847: don't use assertions for input coming from DBus
- * * #9897: audiorecord cleanup
- * * #9897: audiorecord: cleanup, removed unused methods
- * #9897: Initialize and fallback recording path in home directory if
- not valid
- * * #9871: SipTransport: hide more implementation
- * * #9871: SipTransport: refactor SIP transport creation
- * * #9871: disable STUN for account if STUN setup failed
- * * #9847: check pointer before using it
- * #9871: Fallback on normal upd transport when stun resolution fails
- * Revert "#9871: Fallback on normal upd transport when stun resolution
- fails"
- * #9871: Fallback on normal upd transport when stun resolution fails
- * pulse: cleanup
- * * #9847: removed outdated README file
- * * #9847: use references instead of pointers where possible
- * * #9847: pass call by reference where possible
- * * #9847: audiolayer: fixed typo
- * * #9847: SIPVoipLink: gracefully handle invalid pointers
- * * #9847: check that transport is initialized
- * * #9847: SDP: avoid buffer overflow
- * * #9847: fixed segfault on bad call invite
- * * #9847: SDP: don't use assertions for runtime errors
- * * #9847: handle invalid remote session gracefully
- * * #9851: fixed segfault on stun socket cleanup
- * * #8586: fixed warnings
- * * #9849: added missing sstream header
- * #9623: add required TLS certificates for testing purpose
- * #9623: fixed tls registration
- * #9623: fix storing tls port in config for normal account
- * #9623: Allow all account to change tls listener port (not only
- IP2IP)
- * #9623: Allow for changing interface / port for tls transport
- * #9623: Open TLS listener on selected interface
- * #9833: remove unused debug
- * #9833: handlingEvents_ must be initialized to true when starting iax
- thread
- * #9830: Remove create_route_set from sipvoiplink
- * #9831: Fix sip transport port number
- * #9830: move sip header parsing function in sip_utils
- * Revert "* #8586: don't restore and save test files"
- * * #8586: fixed make distcheck
- * #9623: fixed tls registration
- * * #8586: don't restore and save test files
- * * #8586: refactored yaml code
- * #9623: fix storing tls port in config for normal account
- * #9623: Allow all account to change tls listener port (not only
- IP2IP)
- * #9623: Allow for changing interface / port for tls transport
- * #9623: Open TLS listener on selected interface
- * * #8586: added missing tests
- * #9833: remove unused debug
- * #9833: handlingEvents_ must be initialized to true when starting iax
- thread
- * #9830: Remove create_route_set from sipvoiplink
- * #9831: Fix sip transport port number
- * #9830: move sip header parsing function in sip_utils
- * * #8977: removed unnecessary AC_CANONICAL macros from configure.ac
- * * #8977: use actual PJSIP linking flags from pjproject/build.mak
- * * #9774: sipvoiplink's destructor should not be public
- * dbus: cleanup
- * * #9774: make sure sipvoiplink is destroyed before accounts are
- unloaded
- * * #9777: don't use deprecated auto_ptr
- * * #9778: removed AC_CHECK_FUNCS calls
- * * #9782: fix warnings in tests
- * * #9782: sip/sdp: fix emptiness checks
- * * #9782: sdes_negotiator: fix iterator usage and set dangling
- pointers to 0
- * * #9782: initialize all vars in iaxvoiplink
- * * #9782: use fstreams instead of fscanf
- * * #9782: yamlnode: fixed iterator usage
- * * #9782: yamlemitter: fix iterator usage
- * * #9782: yamlnode: make some methods const
- * * #9782: initialize all member vars in constructor
- * * #9782: Tone::interpolate should be const
- * * #9782: mainbuffer: get rid of unused vars
- * * #9782: GainControl::limit should be const
- * * #9782: fix ARRAYSIZE check
- * * #9782: use nanosleep instead of usleep
- * * #9782: fixed "inefficient emptiness test" cppcheck warning
- * * #9782: initialize dcblockers vars in constructor
- * * #9779: dropped CELT support
- * * #9750: moved sfl_data_format.h -> sfl_types.h
- * * #9750: refactored global.h
- * * #9736: restored command line options to daemon
- * tests: cleanup
- * * #8586: make distcheck was missing a header
- * tests: cleanup
- * * #9731: use all caps for application-wide constants
- * tests: cleanup
- * * #9730: cc++: enforce better checks in headers
- * * #9730: builds against libccrtp1
- * * #9572: sipvoiplink: fixed typo
- * * #9572: fixed threading issues with ccrtp2
- * * #9572: manager: pass config filename by const reference
- * * #9572: Replace utilspp singleton implementation
- * * #9571: regenerated config.{guess,sub} file to fix FTBFS on
- armel/armhf.
- * * #9665: siptransport: fixed udp_transport_start calls
- * #9620 Add test SIP account in configuration sample (test/sflphoned-
- sample.yml)
- * * #9641: audiortp: Fixed CryptoContext management
- * * #9641: fixed another memory leak in audio_srtp_session
- * * #9641: audiosrtpsession: fixed memory leak, simplified memory
- management
- * * #9641: avoid dynamic memory allocs/raw pointer usage in audio rtp
- stack
- * * #9641: get rid of getType/RtpMethod logic
- * fixed typo
- * #9572: make sflphone compile with libccrtp 2
- * * #9490: fixed registration state change callback that was crashing
- client
- * * #9547: fixed warnings in SipTransport header
- * #9547: Add SipTransport class
- * #9547: Extract all the transport layer from SIPVoIPLink to new
- SipTransport Class
- * #9547: Destroy the STUN resolver in Transport shutdown
- * sipvoiplink: removed erroneous FIXME
- * sipvoiplink: cleanup
- * #9547: Destroy the STUN resolver if server name change
- * sipvoiplink: fix warning about variable shadowing
- * #8320: Rename declared exception to avoid parameter shadowing
- * #8320: Send signal to client on stun failure
- * #8320: Use the same API for all transport creation (UDP, STUN, TLS)
- * * #9509: use vector for credential info
- * * #9508: fixes segfault in manager by changing order in which
- destructors are called
- * #8320: add dbus signal for stun failure
- * #8320: Use two different variables for status and return statement
- in stun's on_status_cb
- * * #9490: removed resolve_once parameter that was causing a segfault
- * #8320: make the retransmission callback to be rescheduled on error
- * HookPreference: cleanup
- * daemon: hookpreference: cleanup
- * iaxvoiplink: terminate() doesn't have to be virtual
- * sipvoiplink: functions need not be static if they are in an
- anonymous namespace
- * * #9037: moved CHECK macro into separate header
- * * #9037: cleanup error handling/checking in video threads
- * * #9037: video: cleanup
- * * #9037: only signal receiving_video_event for rtp sessions
- * * #9037: shared memory moved out of video_receive_thread
- * * #9381: daemon: fixed make check for video
- * * #9381: YAML_LIBS must be explicitly set in AC_SEARCH_LIBS macro
- * * #9381: reverted yaml check
- * * #9381: fix celt plugin compilation on fedora
- * * #9381: use PKG_CHECK_MODULES to test for yaml
- * * #9381: use autoconf macros and AC_SEARCH_LIBS
- * * #9381: use AC_SEARCH_LIBS, AC_CHECK_LIB
- * ringtonetest: cleanup
- * configurationtest: cleanup
- * instantmessagingtest.cpp: cleanup
- * mainbuffertest: cleanup
- * tests: cleanup
- * #8320: Make sure stun keep alive is enabled
- * call: push answer logic into call classes
- * sipaccount: simplify IP2IP code
- * sipaccount: avoid segfault if sipaccount is NULL
- * sipaccount: cleanup
- * #8084: Fix get sip header segfault when stun transport selected
- * * #9037: created shared_memory class
- * #8084: Init stun port with default valueas defined by RFC 3489
- * #9046: Move IP2IP_PROFILE global definition inside SIPAccount class
- * #9045: fix Changing the account expire is not taken applied in
- daemon
- * vidoe_receive_thread: cleanup
- * * #8968: suppress unusedFunction warnings for functions that are
- actually used
- * * #8821: fixed unit tests
- * #8821: Renamed account map keys for consistency
- * * #8968: audiorecord: added debug, clarified wave header creation
- * * #8968: added debug message to get rid of "unused struct member"
- warning
- * * #8968: manager: create History on the stack
- * * #9026: sfl::InstantMessaging is now a namespace
- * * #8698: managerimpl: removed unused method isWaitingCall
- * * #9008: don't include yaml headers in serializable.h
- * * #9008: cleanup account map initialization
- * refactor accountmap initialization
- * #9020: fix config file not generated when no account created
- * * #8968: audiorecord: removed unused getSndSamplingRate
- * * #8968: config: removed getConfigTreeItemIntValue
- * * #8968: recordable: removed unused getRecFileId
- * * #8968: removed unused Codec::getMimeType method
- * * #8968: manage lifetime of IMModule with auto_ptr
- * * #8968: history: removed unused method
- * * #8968: managerimpl: removed unused method
- * * #8968: config: removed unused methods
- * * #8968: managerimpl: removed unused getConfigBool/Int methods
- * * #8968: networkmanager: cleanup
- * * #8968: managerimpl: removed unused getConfig
- * * #8968: managerimpl: Manage telephoneTone_ with auto_ptr.
- * * #8968: history: fix memory leak upon exception
- * * #8968: AudioFile: initialize filepath earlier
- * * #8968: audiofile: fix memory leak on exception
- * * #8968: audiocodec: removed unused getChannel method
- * * #8968: use auto_ptr for dtmfKey
- * * #8968: use vector instead of dynamically allocated int array
- * * #8968: sdp.h: pass paramter by reference
- * * #8968: sipvoiplink: avoid C-style pointer casting
- * * #8968: yaml: avoid C-style pointer casts
- * * #8968: pulselayer: avoid C-style pointer casting
- * * #8968: recordable: removed unused getRecordingSmplRate method
- * * #8968: alsalayer: use preincrement for iterators
- * * #8968: config: removed unused method saveConfigTree
- * * #8968: mainbuffer: preincrement iterators
- * * #8977: history: added #include <fstream>
- * * #8968: don't leak memory on exception
- * * #8968: pulselayer: avoid C-style pointer casting
- * * #8968: Member variables must be initialized in AudioSrtpSession
- constructor
- * * #8968: fix potential memory leak in audiorecord
- * * #8968: Pass function parameter 'item' by const reference.
- * * #8969: fixed memory leaks in sdes_negotiator
- * video: cleanup
- * * #8940: removed video test source for now
- * #8763 Fix doxygen generation
- * #8763 Generate Doxygen with Hudson
- * * #8940: videosendthread: cleanup
- * sipvoiplink: cleanup
- * fileutils: cleanup
- * #8335 Fix default transport initialization on 5062, 5064
- * #8762: update mute for mic only, fix remove slide for pulseaudio
- * #8672: Add linear to decibel conversion functions in audio layer
- * #8672: Implement audio gain management in pulseaudio
- * #8671: Move audio gain management in audiolayer
- * * #8542: create symbolic link properly
- * * #8613: make check should fail early if another sflphone is running
- * #8449: Update version 1.0.2
- * * #8545: fixed error case
- * * #8545: fixed broken ringtone
- * * #8586: fixed make dist
- * sipvoiplink: use static_cast instead of reinterpret_cast if possible
- * * #8542: test for .git existence before moving pre-commit hook
- * * #8542: autogen.sh should not require git
- * eventthread: cleanup
- * * #8542: removed trailing whitespace from tree
- * * #8357: added disable video option to client
- * siptest: cleanup
- * * #8521: use avcodec_open2 instead of deprecated avcodec_open, if
- available
-
- -- Julien Bonjean <julien.bonjean@savoirfairelinux.com> Thu, 07 Jun 2012 16:08:15 -0400
-
-sflphone-common (1.0.0-rc20110930~ppa1~SYSTEM) SYSTEM; urgency=low
-
- ** SNAPSHOT 1.0.0-rc20110930~ppa1~SYSTEM **
-
- * update kde .gitignore
- * Fix bug in volume widget
- * More polishing for release
- * Bump version to 1.0.0
- * [#7023] Add the ability to load an abstract contact backend in the
- library to resolve more data, polish code
- * [#7021] More cleanup for release
- * Cleanup
- * [#7021] Refactor KDE client dbus handling, add a missing call in
- daemon and port the DataEngine to the new API
- * Remove some annoying debug
- * merge language scripts
- * remove obsolete 'VERSION' files
- * update install instructions
- * Add missing translations to gnome
- * language update
- * Revert "Don't reference count DBus clients, exit core immediately
- when one of them request it"
- * Don't reference count DBus clients, exit core immediately when one
- of them request it
- * [7021] Add contact abstraction support
- * [#7121] Polishing library (over). Indentation, spacing and naming
- are now consistent
- * codecs: link to libccrtp, don't use logger
- * Fix a daemon bug
- * [#7038] Fix adding contact
- * * #7037 : stop audio stream after all calls have been hanged up
- * [#7025] Add full support for bookmark
- * SFLPhone KDE do not destroy history anymore
- * Fix config skeleton
- * Close the daemon once and for all, no more automatic respawning
- * Fix "unregistered account" bug (I hope so)
- * Close SFLPhone at the right place, it still respawn, I don't know
- why
- * Remove dead code
- * Fix regressions introduced in the last commit
- * Dead code elimination 1/3
- * Fix bug, add "add contact" option, fix warning
- * * #7019: Fix IAX codec negociation
- * Remove or comment unnecessary/unhelpful debug output
- * Fix "same as local" account setting, fix IP2IP LED color
- * Add support for some more advanced config options and add missing
- config dialog icons
- * Fix crash with noise suppressor
- * Alternative can now be selected from the call view context menu
- * Add drag and drop support, initial context menu and fix 3 bugs in
- the account dialog
- * Add basic history drag and drop support
- * Complete contact support is back
- * * #6991 : fix IAX problems
- * Fix IAX accounts being disabled by default
- * Revert "deb: forge -g flags for pjsip"
- * * #5884: Disable debug code in pjsip
- * echo suppressor : more assertions
- * Don't let the daemon think crypto is enabled when it's not
- * Simplify ToneList
- * Some progress on contact support
- * Remove unused getRegistrationCount()
- * remove annoying debug
- * revert SIP bit of e27e5c39bad27bae28f574eb2cba7717e8956229
- * Simplify CallManager::placeCallFirstAccount
- * Fix crash on hold
- * * #6905 : SIP refactor
- * gnome client: be sure key exchange is set correctly
- * Move code into createSipTransport
- * Fix account registration on start
- * ManagerImpl::registerAccounts(): simplify
- * * #5884: don't mess with pjsip threads in echo suppressor
- * * #6905 : simplify udp/stun/tls pjsip transport creation
- * Restore and improve support for Call history
- * fix launchpad build
- * SIPVoIPLink: simplify / refactor
- * Fix libwidget linking
- * SIP: simplify
- * IM : simplify
- * gnome: remove some debug
- * AudioRtpFactory::stop() cannot fail
- * * #6905: simplify SIP code
- * pjlib: fix build without SSLv2, fix warnings
- * Port history to the new syntax
- * Test a dock widget based implementation for contact and history
- * Disable SSLv2 support from pjsip and sflphone
- * deb: forge -g flags for pjsip
- * Fix deb packaging to get debug symbols
- * remove debug
- * pjproject: update to last stable release (1.10)
- * Require gtk >= 2.20 and glib >= 2.24
- * tlsadvanceddialog: simplify
- * * #6902 : fix errors spotted by -DGSEAL_ENABLE
- * Update daemon dbus XML and port KDE config backend from dbus to
- local
- * Remove unused but set variables
- * * #6929 : fix IM widget, cleanup
- * Unconditionally enable debug symbols
- * Should fix many KDE issues
- * * #6886 : hitting backspace on empty number have no side effects
- * * #6905 : fix AudioCodecFactory access in optimized builds (-O > 0)
- * Remove unsupported and broken jaunty/karmic packages
- * * #6902 : avoid using some gtk deprecated functions
- * Update dbus introspection files
- * * #6904: removed unused contactmanager
- * * #6903 : use correct dbus-cxx package name
- * * #6902: don't use individual gtk headers
- * Fix a segfault when config is not present
- * Merge latest (0.9.13) KDE code. This version is not yet ready for
- git master, but better than the previous one
- * addressbook : simplify
- * * #5659 : sflphone-plugins doesn't depend on libedataserverui
- * * #5659 : addressbook doesn't use libedataserverui
- * gnome client doesn't depend on evolution
- * * #5695: addressbook: simplify
- * * #5695: addressbook : remove AddrBookHandle from plugin
- * * #5695 : addressbook : remove unused stuff in the client
- * * #5695 : addressbook : remove unused stuff, use static mutex
- * gnome client doesn't use evolution
- * gnome: use proper API to set GTK_CAN_FOCUS
- * * #6897: removed unused focus state vars/callbacks
- * gnome: fix calls to sflphone_fill_codec_list_per_account
- * * #6623: gnome: don't leak in mainwindow
- * gnome: mainwindow whitespace cleanup
- * gnome: actions.c parameter doesn't have to be a double pointer
- * * #6895: fix memleaks, cleanup in accountconfigdialog
- * * #6893: fixes segfault in client on clean history
- * * #6894: fix leaks, cleanup in sflnotify
- * daemon: fixed prints in main
- * * #6892: simplify, fix leaks in dialpad
- * * #6887: audiopreference creates audio layer
- * * #6660: use const char * const, not std::string for globally
- visible constants
- * * #6852: Preferences now solely responsible for audiolayer creation.
- * * #6860: refactor uimanager, also fixes #6865
- * * #6853: hangup as soon as all digits have been deleted
- * * #6852: alsa: retry if device is busy
- * * #6852: audiolayer creation depends only on preference.audioApi
- * * #6850: gnome: fix build for gtk < 2.22.0
- * cleanup in iax
- * alsa: typo
- * pulse: if we can't peek in audio input, we can't drop samples
- * * #6849: show error window if codecs are missing, instead of dying
- * EchoCancel: unused, remove
- * * #6629 : use number of samples as arguments for audio filters
- * * #6629 : remove unused Algorithm interface
- * * #6629 : use helper to call alsa functions and display error msgs
- * Remove unused type
- * * #6841: fix some error handling
- * * #6629: simplify AlsaLayer::alsa_set_params()
- * Get gdk key definition from header
- * * #6828: Replace raw key codes by gdk defines
- * remove some debug, enhance some other
- * mainbuffer: simplify
- * * #6561 : fix phantom call after transfer
- * Conference Participant set : simplify
- * SIPCall: remove unused functions, make invite session public
- * * #6229 : remove malloc/free from pulse audio loop
- * * #6629 : simplify pulse callbacks
- * * #6629
- * Simplify widgets
- * * #6629 : keep the correct audio module when frequency changes
- * * #6751: fixed erroneous debug msgs
- * callable_obj.h: removed unneeded pthread header
- * alsalayer: cleanup
- * * #6629: Always restart audio driver when changing parameters (ALSA
- only)
- * gnome GUI: don't block in DBus signal errorAlert()
- * * #6629 : simplify AudioLayer creation
- * * #6629 : remove unused and unconfigurable frameSize from audiolayer
- * * #6629 : remove unused error message from audio layer
- * Fix logic error when switching audio API
- * Remove unused AudioProcessing class
- * AudioRtpRecordHandler::initNoiseSuppress() : use noiseSuppress
- directly
- * * #6629 : use DC blocker directly in audio layers
- * * #6629 : clean AudioLayer
- * * #6629 : don't store mainbuffer inside audiolayer
- * * #6629 : correct AudioLayer::notifyincomingCall()
- * * #6554: cleanup, refactoring in sipvoiplink
- * * #6554: cleanup in iaxvoiplink
- * * #6554: throw exception in getSIPCall if pointer is NULL
- * * #6554: make some methods of sipvoiplink static
- * * #6655: cleanup in managerimpl
- * * #6554: refactoring, fix memleaks in sipvoiplink
- * * #6478: remove throw specs, cleanup in voiplink
- * * #6629 : remove unused AudioDevice
- * * #6655: removed more dependencies from managerimpl
- * * #6744: simplified numbercleaner
- * conference : remove one prototype
- * * #6743: fix ip2ip
- * Don't give glib warnings if icons are not found
- * gnome: fixed includes
- * Codec.h: removed unused function
- * * #6742 : clean dbus & icons
- * * #6699: refactor/cleanup accounts
- * icons: cleanup
- * timer : use second precision, not millisecond
- * calltree_update_clock : use correct type, returns something
- * * #6737: fixed typo in dbus call
- * * #6737: removed tests for removed API
- * * #6737: dbus: fixed bug from merge
- * * #6737: cleanup in accountlist
- * * #6737: cleanup in dbus
- * * #6740 : fix history double free
- * * #6740 : remove time updating thread from calls
- * * #6737 : use c99 for client
- * * #6738 : make history loading faster
- * sipvoiplink : don't crash on transfers
- * fixed typo
- * Remove unused file
- * Don't build networkmanager.cpp at all if NM is disabled
- * _debug* -> _debug
- * * #6554 : simplify sipvoiplink
- * hudson: added -x to git clean command
- * added git clean to hudson script
- * audiocodecfactory: cleanup
- * * #6718: refactored setTlsSettings into SIPAccount
- * * #6718: removed more unused methods
- * * #6718: refactored confmanager code into sipaccount
- * remove unused functions
- * * #6718: confmanager: removed more unused methods
- * AudioCodecFactory : cleanup
- * #6697 : Turn callableElement struct into union
- * * #6718: confmanager: removed more unused methods
- * * #6718: confmanager: removed more unused methods
- * * #6718: removed unused dbus methods, refactoring
- * * #6699: accounts: cleanup/refactoring
- * * #6699: refactoring, cleanup in accounts
- * * #6699: more account cleanup
- * remove unused autoconf variable
- * * #6714: fixed hudson script
- * make distclean in hudson
- * added || exit 1 to run_tests.sh call
- * * #6714: fixed make distcheck for sflphone-plugins
- * * #6714: fixed make distcheck for gnome client
- * * #6714: fixed make distcheck for daemon
- * git: #6698 split the main .gitignore file
- * gnome: gpointer is already a pointer
- * gnome: calltab_init: use calloc instead of malloc
- * * #6699: more account cleanup
- * * #6699: cleanup account
- * * #6554 : more *voiplink cleanup
- * * #6558 : more sipvoiplink simplification
- * * #6558: saner loadSIPLocalIP prototype
- * gnome: #6623 clean calllists
- * * #6692: more audiolayer cleanup
- * * #6692: cleanup/refactoring in audiolayers
- * * #6692: more forward declarations, AudioThread->AlsaThread
- * * #6692: audiolayer cleanup
- * * #6692: alsalayer cleanup
- * * #6558 : remove account creator
- * * #6558 : clean sipvoiplink
- * * #6554 : cleanup sipvoiplink
- * audiortp: cleanup
- * * #6657 : fix launchpad builds for good
- * * #6675 : send RTP dtmf events only once
- * * #6655: more cleanup
- * AudioRtpSession::updateSessionMedia() : simplify
- * * #6655: more cleanup in managerimpl
- * * #6655: removed more code, cleanup
- * * #6655: more cleanup, fixed infinite loop
- * * #6655: removed more unused files
- * * #6655: removed unused mutex
- * * #6655 removed more unused code
- * * #6655: removed unused methods
- * * #6655: cleanup in main
- * * #6663: fixed segfault when off hold from transfer
- * * #6658: user's active codec selection is respected
- * * #6660: static global string should be static const char* const
- class member
- * * #6659: use g_strcmp0, not strcmp for vals that may be null
- * callable_obj: fix double free
- * calltree_display_call_info() : simplify
- * * #6657: Fix launchpad builds
- * Logger::log() : simplify
- * AudioRtpSession : privatize members
- * * #6655: more constness, cleaned up/simplified methods
- * * #6654: call DBus::_init_threading so that dbus-c++ to make it
- threadaware
- * set default credentials on account creation
- * AudioCodecFactory::scanCodecDirectory() : simplify and correct
- * * #6623: fixed typos
- * * #6623: fixed more leaks
- * * #6623: fixed more leaks
- * * #6623: fixed more leaks, don't print codec name if null
- * * #6623: more leaks fixed in client
- * * #6623: fix more leaks, fixed some warnings
- * * #6623: fixed leak in history
- * updated gitignore
- * initialize dbus dispatcher correctly
- * Fix tests, hudson doesn't have a dbus daemon running
- * remove unused code
- * removeCall() : simplify , fix leak
- * stopRtpThread() : simplify
- * *CurrentCall : simplify
- * Fix memleak
- * fix serialization of audio api (pulse / alsa)
- * account map : simplify
- * remove call from callmap before terminating it, avoid use after free
- * * #6630 : don't make DBusManager a singleton
- * call: return confID by value
- * add back history code deleted by error
- * history : reverse logic
- * simplify history serialization and remove some debug
- * remove annoying debug
- * * #6464 : replace cerr with _error
- * * #6464: replace cout with logger macros
- * replace printf() with logger macros
- * update .gitignore
- * remove unused function
- * update eclipse projects
- * uimanager_new() : simplify
- * rename directories
- * celt: simplify a bit
- * Fix CELT configure.ac test
- * * #6612 : template speex codecs
- * * #6623: refactored conference obj
- * * #6623: refactored callable object, removed leaks
- * * #6623: more cleanup, fix leaks, make global vars static and rename
- them
- * * #6623: calltree: fixed memleaks, simplified code.
- * audiolayer: init pointer members
- * manager: catch exception on invalid hangup
- * * #6623: don't leak on calls to create_new_call
- * * #6611 : clarify codecs prototypes
- * ringtones : .au and .ul files are both ulaw
- * * #6611 : make sure samplerate converters are called correctly
- * ManagerImpl::switchAudioManager() : simplify
- * * #6623: fixed more leaks
- * * #6623: fixed more leaks
- * * #6623: fixed more leaks
- * * #6623: fixed leak, line-endings in imwidget
- * * #6627: zero-initialize pointers if they're going to be deleted
- * * #6628: don't leak calls on exceptions
- * Revert "audiortp: call join after calling stop on RtpThread"
- * sflphone-client: more constness
- * audiortp: call join after calling stop on RtpThread
- * * #6625: return 0 on successful completion
- * * #6624: fix segfault on servercallfailure
- * * #6621: Fixed double free, unlock mutex in ManagerImpl::terminate
- * * #6220: remove audio stream when peer hangs up
- * * #6596: AudioSymmetricSession shouldn't self-delete
- * resampler: grow internal buffers dynamically
- * merge up and down sampling => resampling
- * Leave test directory unchanged when running make check
- * audio algorithms : remove unused prototype
- * ringtone: detect codec from file extension
- * *AudioFile : simplify
- * * #6596: create local SDP on the stack, not the heap
- * * #6596: don't call Ost::Thread::terminate from dtor
- * audiofile: cleanup (samplerate -> unsigned)
- * remove unused func
- * samplerateconverter: cleanup
- * RingBuffer::Put() : remove unused return value
- * MainBuffer::putData() : remove unused return argument
- * audiolayer::putMain() : remove unused func
- * AudioLayer::putUrgent() : remove unused return value
- * * #6618: delete any remaining ringbuffers in destructor
- * RingBuffer::availForPut() : remove
- * * #6617: return from main rather than calling exit
- * MainBuffer::availForPut(): remove
- * RingBuffer: simplify
- * alsa : remove write only variable
- * fix memcpy declaration
- * bcopy(src, dst) -> memcpy(dst, src)
- * RingBuffer::Get() : remove constant volume argument
- * return a copy of the call ID, not just a reference.
- * MainBuffer::getDataById() : remove volume argument (always 100)
- * MainBuffer::getData() : remove constant volume argument
- * RingBuffer::Put() : remove constant volume argument
- * MainBuffer::putData() : remove constant (=100) volume argument
- * audiolayer: remove constant _defaultvolume
- * AudioRtpRecordHandler / AudioRtpSession : simplify
- * mainbuffer: fix test
- * iaxvoiplink : simplify
- * sip registration callback: fix a dbus crash
- * MainBuffer: simplify
- * AudioRtpFactory: return cached type of rtp session. The rtp session
- can have disappeared if the call was put on hold
- * AudioRtpFactory: remove unused setters
- * Fix launchpad builds
- * * #6611 : remove unused bandwidth codec information
- * * #6611: AudioCodec: remove useless/unused setters
- * make sure buffer string is initialized correctly
- * * #6596: declare certain destructors virtual
- * audiolayer : cleanup
- * Simplify doc build rules
- * * #6270: don't build dbus-api doc with make, should require make all
- * configure.ac: cleanup
- * Remove copy of dbus-c++ from libs/
- * * #6596: stop clock thread when peer hangs up
- * removed unused Fmtp.h
- * * #6595: more logical initialization order
- * * #6600 : fix account creation
- * * #6601 : fix configure.ac tests
- * remove unused variable
- * Don't mix stack and heap based allocations
- * Fix copyright (2009, 2008, 2009 -> 2008, 2009)
- * Fix warnings found by clang
- * * #6595: fix initialization order for AudioRTP
- * * #6592: removed typedef std::string CallID
- * * #6586: implement local g_slist_free_full for older glib versions
- * * #6579: fix memory leaks in client (there's a lot left)
- * ShortcutPreferences::setShortcuts() : simplify
- * Fix merge
- * * #6548: remove call to non thread-safe strerror()
- * AudioRtpFactory: each instance is associated to exactly one SipCall
- * create_audiocodecs_configuration() : make static
- * * #6269 : refactor AudioRtpSession
- * Fix AudioSymmetricRtpSession.h inclusion guard (cherry picked from
- commit c3081dce1cc1370d6d3558a4c4ef5cfac0d21caf)
- * * #6269: Rename AudioRtpSession to AudioSymmetricRtpSession
- * * #6574: Don't exit when connection to pulseaudio server fails
- * accountconfigdialog.h : remove some stuff from header
- * * #6560: fix configuration test
- * Fix warning in test
- * * #6560: don't hide password entry in security tab
- * * #6560: set initial password for SIP accounts
- * * #6506: remove useless pointer indirection
- * * 6560: password is now specific to IAX accounts
- * * #6560 : actually use, store, restore, transmit SIP credentials
- * * #6560: YamlEmitter: serialize sequences
- * YamlEmitterException: typo
- * ManagerImpl::computeMd5HashFromCredential() : simplify, fix memleak
- * * #6561: invite_session_state_changed_cb() : simplify
- * * #6561: More useful debug in VoIPLink::removeCall
- * * #6561 : fix ghost call reappearing in GUI after transfer
- * while -> for (make the code smaller)
- * * #6558 : Account::loadConfig() : move IAX code to IAXAccount
- * IAXVoIPLink::getAccountPtr : simplify
- * * #6554 : access the SIPVoIPLink directly, not per account
- * SIPVoIPLink is instanciated only once and is not associated to a
- single account
- * yamlnode: use const references when possible (still some left to do)
- * Account::_accountID: constify
- * VoIPLink: simplify, remove unused method
- * hudson test : no need to call run_tests.sh anymore
- * Remove AccountID type and AccountNULL define
- * Make check runs the test (no need to call run_tests.sh manually
- anymore)
- * gnome GUI: Fix tests
- * Revert "Move registration information from SIPAccount to
- SIPVoIPLink"
- * * #6392: pluginmanagertest: fix warnings reported by valgrind
- * * #6547 : remove unused exceptions
- * * #6547: CallManagerException: use runtime exceptions
- * * #6547: InstantMessageException: use runtime exceptions
- * * #6547: do not throw exceptions if some settings are not present in
- config file
- * * #6547: YamlParserException: use runtime exceptions
- * * #6547: VoipLinkException: use runtime exceptions
- * * #6547: YamlEmitterException: use runtime exceptions
- * * #6547: DTMFException: use runtime exceptions
- * * #6547: AudioFile: use runtime exceptions
- * * 6547: AudioZRtpSession: remove impossible error case
- * * #6547 : AudioRtpSession: remove impossible error case
- * * #6547: AudioZrtp: use runtime exceptions
- * * #6408 : send authenticationUsername to GUI
- * * #6408 : store/restore authenticationUsername from config file
- * SIPAccount: simplify
- * Move registration information from SIPAccount to SIPVoIPLink
- * SIPAccount::getAccountDetails : simplify
- * * #6540: yaml parser: simplify
- * sdp.cpp : fix a warning
- * * #6540: yaml parser : remove std::string typedefs
- * * #6540: Simplify yaml unserialization
- * * #6540 : add a Conf::ScalarNode constructor for booleans
- * setAccountDetails(): simplify
- * * #6408: store authentication username in daemon
- * * #6408: Be able to set the authentication username in the GUI
- * * #6507 : do not crash if the program is not sflphoned
- * Fix tests
- * macroify SIPAccount::unserialize()
- * Move all .cpp files from sflphoned target to libsflphone.la, except
- main.c
- * main() : simplify, return positive error codes
- * * #6507 : find codecs dir in build directory
- * * #6392: Sdp: move clean functions to destructor
- * AlsaLayer::adjustVolume() : simplify
- * alsalayer : reduce indentation
- * malloc/free -> new/delete
- * malloc/free -> new[]/delete[]
- * malloc/free -> new/delete
- * AudioSrtpSession: simplify base64 encoding
- * * #6392: Initialize std::string from pj_str_t correctly
- * * #6392: AudioRtpSession: Initialize remote port
- * Audio settings : Initialize _echoCancelTailLength and
- _echoCancelDelay(0)
- * Initialize variable
- * YamlParserException : fix use of stack variable after it has been
- deallocated
- * * #6392: fix memory leak in history
- * * #6392 AudioCodec : fix memory leak
- * * #6392 : fix memory leak in sip account
- * * #6408: clean up sipaccount (cosmetics mostly)
- * sipaccount.cpp serialize() : reduce number of lines
- * * #6392: invalid memory access
- * * #6392 : fix invalid memory access
- * * #6479: merged useful code from MimeParameters into Codec interface
- * * #6462: fixed hangup on IP2IP call
- * added run_daemon.sh script
- * test: remove unused variable
- * Remove functions only used by a failing test (cherry picked from
- commit fcf718cb75de7f1882dc61c07bb8d300dfa10f85)
- * * #6360 : make client tests build (cherry picked from commit
- 028b2835f040e51ab8ab979b32732b07b8798fce)
- * * #6360 : fix warnings in check_global test (cherry picked from
- commit 9e2bd6a7496dd64f6f48595e385760019aab1193)
- * * 6360: updated API calls in tests, but they're not building yet
- (cherry picked from commit 548f6f0f919b43772a3e9c667e5e292791281795)
- * Fixed include in tests (cherry picked from commit
- aeadc7525c1e31f936670ac8b02f0bcf387c38a8)
- * Remove unused variables and functions
- * IAX: fix warnings (cherry picked from commit
- fd7a113a11cac2cd9a7c36929e88ad28195c4c35)
- * Remove unused DEBUG define which interferes with logger.h (cherry
- picked from commit b2f72b91d0f43cb1dd94d138882a8caa9c841c24)
- * * #6392: no need to check for account NULLity since it is
- dereferenced above
- * * #6392: fix a memory leak, replace by stack allocation
- * * #6392: remove a variable assignement which confuses cppcheck
- * process_conference_participant_from_serialized() : remove unused
- function
- * * #6392: s/free/g_free/
- * * #6392: fix a memory leak in abookfactory_load_module()
- * * #6392: remove generate_call_id() used only once
- * * #6392: fix memory leak (opendir() without closedir())
- * * #6392: AudioRecorder(): ensures mbuffer is set
- * Remove SFLPHONED_VERSION from global.h, use autoconf PACKAGE_VERSION
- * #6298: Cleanup
- * #6331: Fix deleting ringtone file after call have been answered
- * * #6330: merged user_cfg into headers
- * #6298: Fix conference recording file update at conference end
- * #6298: Fix record file name serialization for conference
- * * #6295: cleanup of codec hierarchy
- * #6298: Fix gtk warnings
- * * #6300: added script to run tests
- * #6109: Add recording playback for conference
- * * #6300: tests do not require an installed sflphone
- * * #6295: re-removed clone methods
- * #6109: Fix gtk_critical warnings for incoming calls
- * #6109: Fix GTK_CRITICAL warning
- * #6109: Fix icons when history is not activated
- * #6109: Fix warnings
- * #6109: Implement stop recorded file playback signal
- * Revert "* #6295: removed unused clone method"
- * * #6295: removed unused clone method
- * * #6296: removed non existant file from Makefile.am
- * #6109: Stop fileplayback for outgoing call
- * #6109: Implement stop recording playback button
- * Fix binding names errors in dbus introspection file
- * #6109: Implement playback recorded file callback in client
- * #6109: Store recorded file path on client side
- * #6109: Add dbus methods for call recording playback
- * * #6290: remove unused classes from utilspp
- * * #6288: cleanup sdp
- * * #6288: fix exception usage
- * * #6288: simplify SdpException
- * * #6288: cleanup in sdp.cpp/h
- * #6109: Only display playback button if record file is set and valid
- * * 6290: updated configure.ac to remove functor Makefile
- * * #6290, #6289: removed unused classes from utilspp, fixed make
- check
- * #6109: Add button for history playback of recorded file
- * * #6289: removed unused observer class
- * * #6282: forward declare sdpMedia in sdp.h
- * * #6281: renamed setCallAudioLocal->setCallMediaLocal
- * #6183: Handle conference with more tahn two calls
- * #6183: Fix history icons when calling back a conference from history
- * #6183: Fix icons inconsistencies in history for conference hang up
- * #6183: Fix toolbar actions when selecting a conference in history
- * #6183: Fix conference serialization
- * #6268: Serialize only calls
- * * #6269: removed useless type testing
- * ignore some files in test/
- * * #6268: Remove dead class AudioSymmetricRtpSession
- * #6251: Do not had history calls in calllist when loading history
- file
- * #6251: Fix insertion in history map in before saving history file in
- daemon
- * #6251: Fix history unit tests
- * #6251: Order the list before serailization, get rid of the hashtable
- in history
- * #6251: Implement history serialization using a list wether than a
- map
- * * #6253: remove external audioport from header, make all members
- private
- * * #6253: don't store external local audio port (used for NAT) in
- Call
- * #6251: Add start_time timestamp in history serialization
- * #6251: Fix call insertion in conference items
- * #6233: Fix serialized account list terminated with a ";" character
- * #6238: Fix draggable history calls into current calls
- * #6233: Fix toolbar updates
- * #6233: Fix history
- * * #6235: remove pyc files from git tree
- * #6233: Handle cases when one or manuy calls are unreachable in
- createConfFomrParticipantList
- * #6233: Handle wrong numbers in createConferenceFromParticipantList
- * #6231: Fix drag-n-drop issue
- * * #6173 : move sippxml in tools
- * #6231: Fix merging issue
- * #6183: Implement conference unserialize
- * * #6212: remove extraneous flags from globals.mak
- * #6183: Unserialize conference data in conference
- * #6183: Add account information in request for conference call from
- history
- * #5755: Add -ldl to liker in sflphone-client-gnome
- * #5755: Fix fedora 15 compilation issue
- * #6183: Serialize conference participant phone number and account
- * #6183: Add conference timestamp in serialization
- * * #6186: don't include global.h, just logger.h
- * #6183: Fix saving history to file
- * #6183: Fix removing call from calllist
- * * #6184: remove pointers to Manager from AudioRtpSessions
- * #6183: Calling calltree_add_call explicitely for history
- * #6183: Ability to store conference inside history tab queue
- * * 6181: remove unused API from sipcall
- * #6171: Implment nreCallCreated callback
- * #6167: Fix participant list NULL ending
- * #6149: First draft of conference creation from history
- * #6149: Fix multiple call/conf selection callbacks ...
- * #6129: Fix place_call function called twice for pressing enter
- action
- * #6129: Fix double click action for history
- * #6149: Add dbus call for creating conference from history
- * #6129: Fix placing call from history and addressbook (still need to
- fix icon)
- * * #6148: removed unused AudioRtpFactory constructor
- * * #6145: remove unused isAudioStarted
- * * #6145: remove unused isAudioStarted
- * #6129: Add conference into history, fix call/conference selection
- * * #6143: don't use getType outside of serialization methods
- * * #6132: forward declarations instead of includes
- * * #6132: add constness, remove redundant "inline" keywords
- * #6129: Add timestamp to conference object to order history entries
- * * #6128: remove unused forward declarations from header
- * * #6127: make noncopyable class actually noncopyable
- * * #6125: don't include AudioRtpFactory in sipcall.h
- * #6123: Fix alsa ringback audio file
- * #6123: Fix raw audio file loading problem
- * #6109: Fix daemon plugin manager unit test
- * #6109: Fix history manager unit tests
- * #6109: Recording filename in daemon and client for history items +
- serialization
- * #6109: Refactor AudioFile to play recorded call
- * * #6104: AudioCodec moved to sfl namespace
- * * #6099: remove active flags from codec classes
- * #6095: Add notification-daemon as a runtime dependencies for rpm
- packages
- * #6095: Fix fedora 15 compilation in MineParameters.h
- * #6095: Declare static variable explicitely for client
- * #6095: Add logs to build OSC build machine
- * * #6098: global variables should have file-scope to avoid name
- conflicts
- * #6095: Fix compilation error for Fedora 15
- * #6095: Update SFLphone version to 0.9.14
- * #6095: Add specification file in opensusse build service for
- sflphone-plugins
- * #6073: Fix sflphone-plugins build on launchpad
- * #6093: Rename CodecDescriptor for AudioCodecFactory
- * * #6089: fix warnings in make check
- * * #6086: renamed codecs methods to audio_codecs
- * * #6085: renamed codec related dbus calls to audio_codec
- * #6065: Remove g_print from client, use DEBUG instead
- * #6065: Add actions name for addressbook
- * * #6085: renamed codecs* widgets/functions audiocodecs*
- * #6065: Fix Addressbook runtime warnings
- * #6065: Replace Codecs tab for Audio in account preference dialog
- * #6065: Fix "transfert" typo
- * #6065: Fix addressbook action runtime warning in uimanager
- * * #6082: fixes make check by adding libcrypto libs to test
- dependencies
- * #6073: Rename plugin/addressbook folders for addressbook/evolution
- in sflphone-plugins
- * #6074: Removed AC_SUBST from configure.ac when using
- PKG_CHECK_MODULE
- * #6073: Fix sflphone-plugins package build
- * #6073: Fix sflphone-common build
- * #6065: Fix runtime gtk warning when initializing searchbar without
- addressbook
- * #6063: Fix mozilla-tellify gitignore
- * #6063: Remove stream copy file using ifdef macro
- * * #6012: fix make dist for sflphone-common
- * #6063: Update .gitignore file
- * #6058: Fix base64 encoding related warnings
- * #6056: Fix SdpException handling
- * #6055: Fix unknown pargma warning for gcc <= 4.5
- * * #5949: test gcc version before disabling unused-but-set warning
- * #6054: Fix addressbook plugin compilation warning
- * #6048: Fix uimanager static initialization
- * #6046: Fix addressbook factory static initialization of member
- addrbook
- * #5979: Fix implicit function declaration warning
- * #6042: Fixed discarding qualifier warnings in client
- * #6041: Fix instant messaging unhandled case warning
- * #5994: Implement set current addressbook name and search type in
- addressbook plugin
- * #5994: add rules for launchpad packaging of addressbook plugin
- * #5994: Fix addressbook plugin configuration loading
- * #6027: Fix addressbook enabled test from configuration
- * #6027: No need of gnomedoc related macros in addressbook plugin
- * #6027: Add NEWS file required for build
- * #6027: Add addressbook plugin autogen.sh script
- * #6027: Remove plugins from client
- * #6027: Add sflphone-plugins folder at project's root level
- * #5994: Move addressbook folder from contacts to plugin folder
- * * #6011: removed unused Makefiles
- * * #6010: remove unused headers
- * * #5952: fix "string constant to char*" warnings
- * * #6009 fixed warnings
- * * #6003: finished cleanup of account classes
- * * #6003, #6004: cleanup of account classes, defaultAccount no longer
- global
- * * #6000: fix memory leak of args object
- * * #5998: removed using namespace std from networkmanager
- * * #5998: removed "using namespace std" from ZrtpSessionCallback
- * * #5998: removed using namespacestd from AudioZrtpSession.h
- * * #5998: remove "using namespace std" from auriorecord.h and
- MimeParameters.h
- * * #5998: remove using namespace std in main
- * * #5998: removed "using namespace std" from logger
- * * #5949: test gcc version before disabling unused-but-set warning
- * #5994: Installation of addressbook plugin
- * #5979: Implement codec full addressbook search from plugin
- * #5979: Implement addressbook factory and plugin
- * * #5981: unused webwidget removed
- * #5966: Account config synchronization fix (for stun)
- * #5954: Handle media name exception
- * #5954: Fix audio codec name display in client
- * #5954: Clean up getSessionMedia methods
- * * #5957: getRecordingSmplRate returns a value
- * #5954: Clean up getCurrentCodec methods
- * * #5950: remove "converting to non-pointer type 'int' from NULL"
- warnings
- * #5915: Full gain control version
- * * #5949: remove more unused variable warnings
- * * #5949: remove unused/unused-but-set variable warnings
- * * #5949: show_preferences_dialog returns a success value
- * * #5946: cleanup of include directives, undefined function
- * * #5515: comment out SSLv2 calls in pjsip
- * #5915: Implement different slope for attack tme and release time for
- gain control
- * #5915: use only one input signal for gain control (removed output
- buffer)
- * #5921: Fix no audio after holding a conference
- * #5916: Add gaincontrol files
- * #5916: Implement FFMPEG/CCRTP video streaming prototype
- * #5903: Fix call transfer during a conference
- * #5915: implement rms detector, first order averager, limiter for
- gain control
- * #5914: Fix call transfer when no notification request is required
- * #5899: Fix conference right-click segfault
- * #5884: temporary fix segfault in pjsip memory pool
- * #5883: Fix compilation issues on maverick and lucid
- * #5755: Fix fedora 15 compilation without patching ccrtp
- * [#5855] Make echo canceller optional
- * #5855: Fix echo suppression activation/deactivation
- * #5855: Implement pjsip echo canceller
- * #5814: Speex initialization function uses samples, not bytes
- * #5814: Test using more unbalanced signals
- * #5814: Fix buffer size for long echo length or long echo delay
- * #5814: Adjust level for echo cancellation at runtime
- * #5814: Process noise reduction before echo cancelling
- * #5814: Implement speex post echo canceller processing
- * #5814: Dump echo cancel file to disk
- * #5814: Add parameters for echo cancel
- * #5809: Add configuration parameters
- * #5809: Implement speex echo canceller in audio rtp session
- * #5814: Code cleanup
- * #5814: Fix conf creation with several incomming ringing calls
- * #5814: Fix conf creation segfault when dragging a call on hold on a
- ringing call
- * #5809: Added unit test for echo cancellation and implemented
- "process" virtual method
- * #5709: Add always recording option in configuration
- * #5709: Add always recording option in audio conference panel
- * #5709: Add core functionnality for always recording (missing config
- options)
- * #5769: Fix conference participant handling (detach/attach) and hold
- actions
- * #5747: Fix recording icons and state for conference when adding new
- participant
- * #5769: Code cleanup
- * #5769: Fix hangup unsent calls
- * #5769: Fix remove/add additional participant to conference
- * 5769: Several fixes concerning confererence handling
- * #5769: Fix compilation error
- * [#5769] Fix audio streams binding in main buffer
- * #5769: Removed access to audio mixer from audio layer
- * #5765: Fix audio crash for illformated wavefiles
- * #5765: Add maximum iteration for finding fmt and data "chunck"
- * #5589: Fix compilation of libnotify under
- * #5757: Fix abort signal when receiving INFO
- * #5747: Add usersDetached.svg
- * #5747: Handle offhold action for recording conference
- * #5747: Fix off hold action for conferences
- * #5747: Implement update conference in record action in calltree
- * #5747: Add new icons for recording conferences
- * #5747: Add recording state for conferences
- * [#5738] Remove getAudioDriver call from manager (replace by
- _audiodriver var)
- * [#5738] Refactor mutex protecting audiolayer
- * [#5737] Fix HD conference recording
- * [#5730] Fix start audio session after changing sampling rate
- * [#5714] Fix enter keyboard event for addressbbok and history
- * [5695] Fix addressbook combo box update when no addressbook selected
- * [#5695] Fix addressbook initialization and search bar update
- * [#5695] Add mutex for books_data in addressbook to protect async
- calls
- * [#5695] Get back addressbook open from uri
- * [#5695] Fix absolute addressbook URI for local addressbooks
- * [#5695] Implement libebook 3.0 interface
- * [#5571] Better logic for hangup (for case where call have not been
- sent yet)
- * [#5571] Update error handling in voip links
- * [#5571] Fix compile time warnings
- * [#5696] Fix installation dependencies for Natty
- * [#5669] Add mention that sflphone.org is for testing only
- * [#5693] Add natty in teh dput.conf file
- * [#5690] Remove not useful logs
- * [#5670] Use dynamic payload type for rtp dtmf
- * [#5668] Clean up sflphone configuration logging
- * [#5668] Fix hook checkbox configuration update
- * [#5666] Fix unit tests
- * [#5666] Manage event subscription
- * [#5666] Emit bye request when subscription is terminated
- * [#5666] Bye request should be sent after event subscription
- notification is done on transfer
- * [#5666] Make reinvite method static (to be called in pjsip
- callbacks)
- * [#5666] Hangup Call in manager for AccountNULL and IP2IP
- * [#5589] Use PKG_CHECK_MODULE for every client's dependencies
- * [#5623] Enlarge initial size of pjsip memory pool for calls (16k)
- * [#5564] Fix audio recording resampling for g722
- * [#5571] Move attribute handling for onhold/offhold actions in SDP
- session
- * [#5571] Codec negotiation refactored and unittested
- * [#5571] Implement tests
- * [#5571] Implement pjsip negociator
- * [#5571] Fix unit tests
- * [#5571] Add Fmtp.h to repository
- * [#5571] Integrate mime types and codec factory
- * [#5571] Handle exception when SDP negotiation fails
- * [#5570] Add sflphoned-sample.yml in repository
- * [#5564]: Implement stereo to mono mixing for rigntone
- * [#5342] Update audio stream initialization
- * [#5514] Restore test ni historytest suite
- * [#5514] Fix
- * [#5514] Disable test_create_history_path
- * [#5514] use pulseaudio in sample config file
- * [#5514] Fix test: load history from file
- * [#5514] Do not use X
- * [#5513] Make unit tests compile successfully
- * [#3947] Enable unit tests in Jenkins
- * [#5454] Fix build system to handle new version number
- * [#5454] Update languages from launchpad
- * [#5454] Add --without-celt in OpenSuse build service
- * [#5454] Change version number
- * [#5331] Added first SDP session tests
- * [#5273] Update nightly build version tags to conform dpkg rules
- * [#5211] Refactor send register method for iaxvoiplink and
- sipvoiplink
- * [#3950] Remove call being transfered from calltree
- * [#5211] Use appropriate memory pool for transport selector
- * [#5211] Fix strict aliasing rules warning in pjsip
- * [#5211] Bring back pjsip shutting down sleep to 1000 ms
- * [#5211] Fix registration callback segfault when closing the
- application
- * [#5211] Use the dialog memory pool for Route header in INVITE
- request
- * [#5211] Add temporary memory pool for findLocalAddressFromUri and
- findLocalPortFromUri
- * [#5211] Use individual memory pool for dtmfs
- * [#5211] SipVoipLink refactoring
- * [#3950] Attended transfer for conference calls
- * [#5284] Fix DNS resolution for Route with specified port number
- * [#5284] Some code cleanup
- * [#3947] Fix typo in hudson script
- * [#5284] Added sip route to REGISTER, INVITE, BYE request, plus DNS
- resolution
- * [#5266] Use RTP dtmf as default
- * [#5284] Added pjsip_process_route_set after setting routes in regc
- structure
- * [#5286] Fix parsing error due to long configuration file (removed
- max event)
- * [#5286] Fix false test in configuration emmiter
- * [#5286] Code cleanup
- * [#5286] Updated exception handling in configuration system
- * [#4969] Fix put SRTP call on hold
- * [#3950] Add debug messages
- * [#3950] Ability to perform an attended transfer
- * [#5276] Fix initialization problem in g722
- * [#3950] Add replace header in SIPVoIPLink::transferWithReplaces
- method
- * [#3950] Implemented attended method in SIPVoIPLink
- * [#3950] Cleanup transaction request received callback
- * [#3950] Implement dummy attended transfer in gnome-client
- * [#5249] Fix audio samplerate update algorithm for g722
- * [#5249] Fix uninitialized variable used in conditional jumps
- * [#5249] Fix conditional jump error in audiolayer (uninitialized
- value)
- * [#5267] Use autoconf 2.65 as a requirement (instead of 2.67)
- * [#5267] Restore manual pjsip configuration and compilation
- * [#5267] Autodetect celt version (0.9.1, 0.7.1)
- * [#5267] Fix deprecated macros in gnome client configure.ac
- * [#5267] Update configuration for libcelt-dev
- * [#5267] Fix build autoconf and automake
- * [#5227] Deactivate automatic call to astyle after compilation
- * [#5242] Hangup every calls before leaving
- * [#5237] Will now nightly-build for natty, Karmic deprecated
- * [#5229] Use inner class for rtp thread instead of inheritance
- * [#5211] Move mainbuffer unbind call in rtp final method
- * [#5211] Initialize sip call memory pool using 16 kb
- * [#5211] Use call memory pool in session reinvite
- * [#5211] Add debug messages
- * [#5211] Use and internal pool for calls
- * [#5211] Reduce pjsip memory pool usage for stateless error messages
- * [#5211] Refactor call deletion
- * [#5212]
- * [#5208] Refactor codec management for accounts
- * [#5168] Remove printf from codec's encode & decode method
- * [#5168] Fix celt compilation on launchpad
- * [#5168] Fix sflphoned compilation warnings in audiocodec.h
- * [#[#5168] Must keep the g722 specific RTP rate to avoid incoming
- packet timeout
- * [#5168] Fix static/dynamic payload rtp session update
- * [#5168] Throw SIPVoipLink Error if codec not instantiated in new
- outgoing call
- * [#5168] Fix dynamic/static codec payload type ambiguity
- * [#5169] Fix doubled IP2IP profile when no config file
- * [#4867] Add gtkinfobar in configuration panel
- * [#4867] Disable input/output/ringtone selection when using default
- alsa plugin
- * [#4952] Patches for possible buffer overflows
- * [$4885] Fix schemas problem
- * [#4885] sflphone-client-gnome.schemas not present during build
- * [#4885] Add gconf shemas directories in opensuse build system
- * [#4885] Add file/folder ownership for opensuse-factory build system
- * [#4906] Fix opensuse-factory build
- * [#4885] Update name dependency for libedataserver
- * [#4885] Fix non-void function without return in dbus-c++
- * [#4895] Update language translation
- * [#4896] Update session timestamp when updating media
- * [#4896] Reapply RTP hack for G722 payload type
- * [#4896] Update recording sampling rate when updating codec
- * [#4897] Save codecs in config for each configuration changes
- * [#4895] Do not save config when sflphone quit
- * [#4885] Update date for copyright
- * [#4885] Deactivate siptest that require more than one sipp instance
- * [#4879] Remove inmcoming call notification from IAX
- * [#4885] Some cleanup
- * [#4874] Add setCancel immediate/deffered for ost::Thread
- * [#4879] Fix incoming call notification
- * [#4878] Set keyboard focus on searchbar when selecting addressbook
- * [#4874] Fixed compilation warning
- * [#4874] Fixed compilation warning in sipvoiplink
- * [#4874] Fix compile time warning in RTP record handler
- * [#4874] Fix conditional jump in SDP
- * [#4874] Fix conditional jump based on uninitialized value
- * [#4874] Store call id within rtp thread context
- * [#4874] Fixed conditional jump based on uninitialised value in
- conference
- * [#4871] Fix default account fetching
- * [#4870] Delete RTP session when Refusing an incoming call
- * Restore IP to IP call
- * [#4857] Fix audio codec negotiation problem
- * [#3947] Adjust ressources allocated to compilation
- * [#3947] Disable unit tests in Hudson
- * [#4305] Free mutex only when really quiting SFLphone
- * [#4859] Update copyright to 2011 in every source file
- * [#3218] Character '.' stripped by the caller engine
- * [#4854] Fix typos, desktop entry
- * [#4847] Apply RTP modification to ZRTP session
- * [#4852] Update Karmic and Lucid dependencies
- * [#4852] Add Libedataserver and libedataserverui as gnome client
- dependencies
- * [#4852] Add authentication mechanism for EDS
- * [#4851] Fix segfault when closing pulseaudio layer too rapidly
- * [#4808] Some otehr cleanup
- * [#4808] Made some cleanup
- * [#4808] Added mutex in rtp session for codecs and noise process
- * [#4847] Update audio processing when updating RTP media
- * [#4842] Add support for linking with gold/ld --no-add-needed
- * [#4808] Make update g722 related static/dynamic payload logic
- * [#4827] Upper limit on the number of contacts to import from EDS is
- hard-coded to 500
- * [#4808] Fix put call on/off hold
- * [#4808] Implement early RTP start for incoming calls
- * [#4808] Audio stream is no longer start within RTP session.
- * [#4808] Removed coupling between audio layer and and RTP session
- * [#4702] Start audio rtp session as soon as it is created
- * [#4702] Init timestamp to 0
- * #4702: Send RTP packets immediately, no need of outgoing queue
- * [#4784] Update dbus-c++ version from gitorious
- * [#4702] Update RTP timeouts
- * [#4702] Lengthen RTP timeouts
- * [PATCH] Fixed compatibility with old libtool versions.
- * [PATCH] Accept older libebook (Maemo 5 has 1.4.2)
- * [PATCH] Fixed double-free error in preferences dialog
- * [PATCH] Fixed building of sflphone-common on Maemo5
- * [PATCH] Improved Gnome client initialization error handling. 1. It
- no longer segfaults when sflphoned isn't available. 2. User is
- provided with GUI error dialog.
- * [PATCH] Improved autogen.sh scripts 1. They do not require bash
- anymore 2. Added workaround for Debian bug #565663 3. Replaced
- manual autotools invocations with single autoreconf call 4. Non-zero
- return status on failure
- * Revert "[#4468] libtool <= 2.2 doesn't have LT_INIT macro so
- AC_PROG_LIBTOOL should be used instead."
- * Revert "[#4468] Libebook 1.4 is sufficient"
- * Revert "[#4468] Apply big path on dbus communication system"
- * [#4468] Apply big path on dbus communication system
- * [#4468] Libebook 1.4 is sufficient
- * [#4468] libtool <= 2.2 doesn't have LT_INIT macro so AC_PROG_LIBTOOL
- should be used instead.
- * [#4639] Fix determining default addressbook if this property is not
- set in gconf
- * [#4639] Fix memory leaks in Addressbook
- * [#4637] Fix opening default addressbook at sflphone init
- * [#4622] Free yaml events while parsing configuration file
- * [#4623] Fix conditional jumps based on uninitialized variable
- * [#4622] Fix leaks in yaml serialization engine
- * [#4616] Fix addressbook warnings
- * [#4514] Adjust RTP timestamp
- * #4527: Rename Karmic libyaml and Celt package in debian control file
- * #4495: Rework addressbook opening loop
- * [#4524] Increment RTP count when sending data
- * [#4524] DO NOT start RTP session twice
- * [#4367] Use PKG_CHECK_MODULE for celt
- * [#4367] Fedora package celt as celt (not libcelt)
- * [#4367] Astyling
- * [#4367] Update .po files
- * [#4367] Fix segfault in gensin
- * [#4354] Make celt a direct dependency on launchpad opensuse build
- service
- * [#4367] Make celt a required package, option --without-celt valid
- * [#4367] Fix zrtp timestamping error
- * [#4367] Fix audio zrtp timing
- * [#4367] Dispatch ZRTP packets
- * [#4367] Fix segfault when unloading account map
- * [#4367] Fix zrtp session
- * [#4367] Implement on packet receive
- * [#4367] use symetric audio rtp session, not dual
- * [#4367] Reduce packet receive/sent timeout
- * [#4367] Reduce RTP timeouts
- * [#4367] Move speaker data receive
- * [#4367] Move speaker data receive
- * [#4367] Move receive speaker data method
- * [#4367] Remove debug in rtp session
- * [#4367] Fix g722 codec clock rate
- * [#4367] Fix noise suppression initialization
- * [#4367] Fix segfault in RTP mic fadein method
- * [#4367] Refactor mic data encoding in rtp session
- * [#4367] Implement RTP main loop
- * [#4367] Fix compilation problem
- * [#4367] Fix AudioRtpclass using TRTPSessionBase
- * [#4367] Fix AudioRtpSession putDtmfEvent shadowing
- * [#4367] Fix AudioRtpSession putDtmfEvent shadowing
- * [#4367] Refactor RTP session (phase 2)
- * [#4367] Refactor RTP session (phase 1)
- * [#4367] Remove Redeclaration of SymetricAudioRtpSession in
- rtpfactory
- * [#4265] Add continue statement in for loop for invalid addressbook
- * [#4261] Makes addressbook initialization more robust
- * [#4257] Add maverick in build system
- * [#4233] Add sdp related unit tests
- * [#4233] Add condition and signal in two incoming call test
- * [#4243] Fix segfault in AudioSrtpSession
- * [#4243] Fix memory leak in AudioSrtpSession
- * [#4243] Make audio srtp optional in for incoming call
- * [#4243] Add boolean variable to make sure remote crypto context
- initialized only once
- * [#4243] Add documentation to AudioSrtpSession
- * [#4243] Use 80 bits authentication tags by default
- * [#4243] Init audio srtp remote crypto context in
- call_on_media_update
- * [#4243] Move SDP negotiastion in mod_on_rx_request
- * [#4243] Implement initLocalCryptoInfo to be called at different
- momment
- * [#4243] Init init local crypto context in when initializing audiortp
- * [#4243] Change key length according to sdes negociation
- * [#4243] Associate callid to accountid for incoming calls
- * [#4242] Fix no SDES keys in IP2IP calls
- * [#4242] Fix no SDES keys in IP2IP calls
- * [#4233] Test for call on/off hold
- * [#4233] Add two incoming call test
- * [#4233]
- * [#4233] Add 2 outgoing simultaneous call unit tests
-
- -- Julien Bonjean <julien.bonjean@savoirfairelinux.com> Fri, 30 Sep 2011 13:51:04 -0400
-
-sflphone-common (0.9.7~rc1~ppa1~SYSTEM) SYSTEM; urgency=low
-
- ** 0.9.7~rc1~ppa1~SYSTEM **
-
- * [#2462] Set explicitly the transport on incoming call too
- * [#2462] fix typo
- * [#2462] Use different address for SDP and call IP
- * [#2462] Use published address in SIP-SDP
- * [#2181] Fixed changelog files
- * [#2181] Updated spec file
- * [#2402] Fix pointer to int conversion warning (atoi)
- * [#2402] Remove daemon warnings, make indent
- * [#2459] Make sure the stream is opened when the call is answered
- * [#2402] Add conference related picture in documentation
- * [#2443] Not much ...
- * [#2399] Fix dialing display problem
- * [#2450] Fix incoming call already in conference crash
- * [#2399] Display peer name on the first line and peer number on the
- second
- * [#2450] Handle 403 FORBIDDEN when refused
- * [#2447] Bind offHold/onHold actions to button in gtk client
- * [#2447] Bind hangup action to button for conference
- * [#2447] Add conference action in gtk client's ToolBar
- * [#2381] Disable the password hashing in config file
- * [#2402] Cleanup
- * [#2366] Set callback to null when deleting Pulseaudio streams
- * [#1313] Fix main buffer unit test
- * [#1313] Fix audio layer unit test
- * [#2315] Hide pw in security tab, display when editing, sync with
- basic tab
- * [#1313] UnitTest change AudioRtpSession for AudioSymetricRtpSession
- instance
- * [#2402] Code cleanup
- * [#2444] Add debug to catch occasional crash when loading client's
- config
- * [#2444] Add debug info to catch occasional crash when loading config
- dialog
- * [#2402] Restore Call menu translations
- * [#2403] Use the published address if checked in GUI
- * [#2442] Add protection test in sdp
- * [#1841] Reapply pjsip patch concerning DNS SRV resolution
- * [#2384] Tags incoming call as direct SIP call, if applicable
- * [#2402] Change the monkey face
- * [#2315] Enable user to display password in clear text
- * [#2434] Force optimization level at 2
- * [#2284] Fix dbus_get_all_ip_interface compilation warnings
- * [#2431] Popup main window on incoming if applicable
- * [$2402] Fix simple warnings
- * [#2402] Fix implicit variable init order in LibraryManagerException
- * [#2402] Fixing implicit variable initialization warnings in
- AudioRtpSession
- * [#2402] Revert atoi change, fixing codec list doubled entries
- * [#2402] Fix gpointer to gint conversion
- * [#2402] Fix pointer casting to integer different size warning in
- codec list
- * [#2402] Fix warning discarting qualifiers from pointer target
- * [#2402] Fix gtk tree view assignement from incompatible type warning
- * [#1669] Fix audio recording folder utf-8 non compatibility issue
- * [#2414] Clean up debugs
- * [#2414] Use transport set in iptoip Account and update it frm
- preference
- * [#2348] Use macro N_() to mark ui.xml strings as translatable
- * [#2414] Rename getSipAddress/setSipAddress functions
- * [#2407] Fix volume controls display
- * [#2407] Fixes dialpad
- * [#2383] Set ip to ip config when clicking apply button
- * [#2404] Update call-to script - Maxime Chambreuil
- * [#2405] Client handles unknown call in current state as well
- * [#2383] Add DBUS signal to send IPtoIP local address and port as
- string
- * [#2383] Add Ip to IP config change apply call back
- * Clonflict
- * [#2402] Code cleanup
- * [#2383] Do the same for IPtoIP (init localn ip with first in the
- list)
- * [#2383] Use first interface in the list if local addresss is not
- defined
- * [#2403] Clean up unuseful addresses/ports
- * [#2403] Use the IP profile SIP port as global SIP port
- * [#2383] Fix dbus_get_all_ip_interface warnings
- * [#2383] Take into account sameAsLocal when loading published address
- * [#2383] Tsake into account sameAsLocal option when saving published
- address
- * [#2383] Update local ip address in ip to ip config
- * [#2383] Save ip 2 ip local port in config
- * [#2406] Update toolbar at startup
- * [#2284] Remove redefinition warnings + speex warnings
- * [#2383] Fix security table in account config
- * [#2383] Save ip 2 ip network interface parameters in config
- * [#2403] Restore sip transport selector
- * [#2383] Fix filling the Localt IP Address on account creation
- * [#2383] Fix Gtk-Critical when checking STUN
- * [#2383] Fix reopening account configuration display issue
- * [#2383] Load IPtoIP local address and port in preference iptoiptab
- * [#2383] Add LocalAddress and Localport in Preference IpToIp tab
- * [#2403] Use the address and port associated to the account as often
- as possible
- * [#1753] Removed pjsip generated files
- * [#1753] Removed remaining milenage lib references
- * [#2383] Add _publishedSameasLocal variable in sipaccount
- * [#2383] Add PUBLISHED_SAMEAS_LOCAL variable in config
- * [#2383] Fix stun set active or not when opening config
- * [#2181] Added RPM 64bits dbus patch
- * [#2402] Code indentation
- * [#2313] Force $(HOME).cache directory creation at startup
- * [#2383] Separate network interface and published address in account
- config
- * [#2400] Change dbus service installation path to libdir
- * [#2382] Move TLS related published address options in security tab
- * [#2382] Indent accountconfigdialog.c
- * [#2181] Install libdbus-c++ in $pkglib instead of $lib
- * [#1753] Remove ILBC code and disable it by default in the configure
- * [#1753] Remove milenage directory
- * [#2382] Fix switching interaface instabilities
- * [#2396] Save local ip in account creation wizard
- * [#2284] Remove warning on hold
- * [#2387] Fixes history searching and filtering
- * [#1215] Add samplerate display in the GUI
- * [#1663] Voicemail icon reflects voice messages
- * [#2395] Fix account registration ( specifically with callcentric)
- * [#2386] Strip "sip:" on incoming call, fixing history call back
- * [#2181] Updated spec files
- * [#1215] Display codec name in calltree instead of status bar
- * [#2390] Move back nbCalls and stopStream higher in refuseCall
- * [#2392] Fix ringtone during call in IAX
- * [#2391] Stop audio streams when there is 0 calls only
- * [#2391] Add debug when call state is not valid
- * [#2390] Clear returns in IAXvoipLink::sendAudioFromMic() method
- * [#2380] Fixing IncomingCallNotification not regular
- * [#2339] Query conference at client startup
- * [#2339] Working conference querying at startup
- * [#2339] Add conference in call tree
- * [#2339] Primitives to query conferences at client startup
- * [#2320] Add account selection in history
- * [#2355] Temporary solution: do not delete pointer when removing
- account
- * [#2380] Change algorithm in AudioRtp to trigger an
- IncomingCallNotification
- * [#2274] Comment sdebug in MainBuffer flush method
- * [#2274] Add flushMain() in ManagerImpl::addStream
- * [#2274] Add getBufferID() method in ring buffer
- * [#2274] Fix warning, comment debug in ringbuffer's flush method
- * [#2274] Use AudioLayer flushMain() and flushUrgent() in ALSA
- * [#2274] Clean up unused variable warning
- * [#2274] Protect minbudffer pointer on flushing
- * [#2274] Fix playATone method which writing empty buffer in urgent
- ringbuffer
- * [#2274] Use audio layer flushUrgent and flushMain in createStreams
- * [#2274] Use flush audio calls from audiolayer
- * [#2274] Flush when peer answered call
- * [#2375] Flush main buffer in iax when answering a call
- * [#2274] Parse displayname using c++ string method
- * [#2375] Flush main buffer when off holding calls
- * [#2375] Flush main buffer mon RTP startup
- * [#2376] Use now Pulseaudio module-cork-music-on-phone
- * Updated OSC packaging
-
- -- Julien Bonjean <julien.bonjean@savoirfairelinux.com> Fri, 20 Nov 2009 14:00:02 -0500
-
-sflphone-common (0.9.7~beta~ppa1~SYSTEM) SYSTEM; urgency=low
-
- ** 0.9.7~beta~ppa1~SYSTEM **
-
- * [#1933] Cleanup debug
- * [#1933] Clean up debug
- * Fix mic
- * [#1933] Set the IAx format earlier
- * [#1933] Move IAX sendAudioFromMic outside if (call) statement
- * [#1933] Fix startstream when offhold in iax and add debug concerning
- codec neg.
- * [#2371] sflphone_notify_voice_mail: minor gettext message formatting
- cleanup
- * [#2371] select_account_cb: properly gettextize status message
- * [#2371] show_account_list_config_dialog: properly gettextize status
- message
- * INSTALL: Minor tidyup of core install guide
- * Add /sflphone-client-gnome/src/icons/Makefile to .gitignore
- * [#2181] Updated OpenSUSE files (tmp)
- * [#1933] Add debug for codec negociation for iax
- * [#1933] Get rid of getMicAvail and getMicData in audiolayer (not
- used anymore)
- * [#1933] Add "audio codec not determined" error in IAX
- * [#1933] Test flush data
- * [#1933] Do not need to start audio stream in iax anymore
- * [#1933] Protecting pointer
- * [#2284] Remove more compilation/execution warnings
- * [#2284] Cleanup debug in client, use DEBUG instead of g_print
- * [#2284] Clean up uimanager
- * [#2370] Remove warnings
- * [#2366] Clean up other debug
- * [#2366] Clean up debug
- * [#2366] Call pa_xfree explicitely in writeToSpeaker
- * [#2284] Remove address book warnings
- * [#2365] Fixes bad cast
- * [#2352] Fix continuous ringing when peer hangup and call not yet
- answered
- * [#2181] Added version support
- * [#2181] Fixed some minor issues
- * [#2360] Moved MainBuffer from AudioLayer to ManagerImpl
- * [#2352] Makes getMainBuffer() everywhere
- * [#2352] Use 50 sec latency on pulseaudio stream creation
- * [#2352] Add alsa debug
- * [#2359] Update repository documentation
- * [#2354] Move pulseaudio disconnectAudioStream after stopping main
- loop
- * [#2352] Adjust nb byte copied in pulseaudio according to
- writeableSize
- * [#2352] Specify pulseaudio tlength parameters using pa_usec_to_bytes
- * [#2322] Convert italian translation to UTF-8
- * [#2357] Fixes window size
- * [#2357] Display only actionnable tool item
- * [#2333] Update streams parameters
- * [#2347] Use GNOME user settings for Menu and Toolbar appareance
- * [#2349] Load/Save properly audio params
- * [#2322] Update translations from Launchpad
- * [#2181] Added Francois Marier script
- * [#2350] Remove non-valid test
- * [#2181] Updated launchpad packaging
- * [#2333] Fix Pulseaudio Capture
- * [#2333] Use pulseaudio ADJUST_LATENCY flag and ALSA RT-SCHEDULING
- * [#2333] Pulseaudio Interpolate timing
- * [#2333] Change (again) Pulseaudio settings to fit logiteck usb hdw
- requirement
- * [#2333] Adjust pulseaudio fragment size to 4096 (max sflphone's
- frames per buffer)
- * [#2284] Remove recurrent compilation warning (g++ linker problem)
- * [#2333] Safer Audiostream parameters
- * [#2333] Fix alsa playback to reduce underrun
- * [#2333] Better audiostream parameters
- * [#2181] Updated version management
- * [#2333] Exclusive test in playback loop
- * [#2181] Updated build system
- * [#2333] Less underrun with these value
- * [#2333] Update playback audiostream parameters
- * [#2333] Lengthen the audio buffer reduce number of underrun in
- pulseaudio
- * [#2333] Add ALSA recovery functions for underrun (begin)
- * [#2333] Add pa_stream_trigger in pulse audio underrun callabck
- * [#2048] Reduce prebuffering in pulseaudio (which affect incomming
- calls' plbck)
- * [#2316] Do not display any icons to the right on the history tab
- * [#2333] Comment pa_stream_trigger in pulseaudio underrun
- * [#2333] Modify pulseaudio streams parameters
- * [#2318] Fix transfer tool button double signal
- * [#2181] Updated
- * [#2333] Fix ALSA ringtone
- * [#2333] Flush all main buffer before starting audio
- * [#2333] Open/Close Alsa thread between calls while there is no audio
- * [#2333] Add debug message and test condition on starting playback
- and capture
- * [#2181] Fixed gnome client makefile
- * [#2181] Updated
- * [#2308] Remove getTelephoneTone debug
- * [#2308] Change plughw for default in ALSA
- * [#2308] Oups, forgot to change function name in audiolayertest.cpp
- * [#2308] Cleanup in pulseaudio code (debug, function name)
- * [#2308] Fix pulseaudio stream closing assertion failure
- * [#2308] Moved pulseaudio mainloop locking from AudioStream
- disconnect stream
- * [2308] Fix latency at the beginning of a call, when playing DTMF and
- wehn starting tone
- * [#2181] Updated karmic
- * [#2317] [#2319] Fix address book toggle button contextual behaviour
- * [#2308] Stop stream when refusing a call
- * [#2308] Stop pulseaudio stream when peer hungup
- * [#2308] Fix tone and ringtone
- * [#2312] Display the STUN entry widget when opening the tab
- * [#2308] Implement two different callbacks for capture/playback in
- pulseaudio
- * [#2309] Open/close pulseaudio connections in startStream/stopStream
- * [2308] Leave pulseaudio stream running, do not cork/uncork them
- anymore
- * [#2295] Set gtk file chooser to None if nothing is set in
- configuration
- * [#1976] Add codec and conference documentation
- * [#2209] Fix recording in regard of resamling
- * [#2297] Update .gitignore
- * [#2297] Update translation files
- * [#2297] Add reference to our coding standards
- * [#2297] Remove old docbook code
- * [#2296] Reinit tls account settings after modification
- * [#2253] Add DcBlocker class to remove capture's dc offset
- * [#2034] Fixes for TLS transport to initialize
- * [#2284] Add silent build rule + client clean warnings
- * [#2274] Fix unserialize history items in cilent at startup
- * [#2274] Complete display name parsing and displaying
- * [#2274] Parse the Display Name in sip INVITE message
- * [#2050] Fix capture volume control in ALSA
- * [#1970] Volume controls disable when using pulseaudio
- * [#1970] Disable volume controls when using pulseaudio
- * [#2277] Fix direct ip2ip ZRTP enabling/disabling in ip2ip
- preferences
- * [#2181] Added launchpad debian files
- * [#2181] Added spec files for OSC
- * [#2274] Set display name for "Contact" sip header as the hostname
- * [#2181] Fixed daemon issues
- * [#2181] Fixed gnome client issues
- * [#1976] Remove warnings - need to fix the transfer
- * [#2006] Add init is_rec variable in ManagerImpl
- * [#2006] Update codec display on call selection
- * [#2006] Restore double click actions in history and contact calltree
- (GTK)
- * [#2176] use XDG_CACHE_HOME when initializing sfl.zid file
- * [#1976] Fix calltree switching from history
- * [#2209] (Re)Fix cache for zid
- * [#2209] Clean up debug messages
- * [#2209] Clean debug messages
- * [#2209] Fix trasnfering a call during a conference
- * [#2209] Speex decode must return the number of bytes
- * [#2209] Change frameSize speex 32kHz
- * [#2209] Fix speex codec framesize
- * [#2209] Reinit converterSamplingRate in RTP sessions
- * [#2209] Change speex ultra wide band framesize
- * [#1747] Add pixmap data
- * [#2252] Fix Receiving a server error 488 crashes the callee
- * [#2209] Fix iax low rate packate sending
- * [#2209] Clean up debug messages
- * [#2209] Add resampling changes for IAX
- * [#2209] Clean up resampling code
- * [#2209] Fix latency introduced by pulseaudio
- * [#2209] Fix initialization of mainbuffer's internal sampling rate
- * [#2176] Fix upsampling buffer size in audiolayer
- * [#2209] Add dynamic converter sampling rate in audiortp sessions
- * [#1747] Fixes runtime warnings
- * [#1747] Remove from repo
- * [#1747] register our icons to be used as stock icons
- * [#2209] Fix number of byte in alsa's write to speaker
- * [#2209] Fix putting non-resampled data in RTP's mainbuffer
- * [#2209] Add alsa resampler
- * [#2209] Add a samplerate converter in PulseLayer
- * [#2209] Add mainbuffer's internal sampling rate and flushall method
- * [#2176] Add mainbuffer stateInfo debug method
- * [#2209] Resampling is optimal using SRC_LINEAR not SRC_FASTEST
- * [#2176] Remove debug recordings
- * [#2176] Fix Holding a conference participant on new calls
- * [#2224] Add confID in callable object
- * [#2176] Fix putting onhold a call participating to a conference when
- pressing new call
- * [#2176] Reset auidio buffers when adding streams (rtp, audiolayer)
- * [#1976] Use xml to describe toolbars - Add a naviguation toolbar
- * [#2176] Remove conference default_id in joinParticipant
- * [#2176] Display error message in alsa's snd_pcm_avail_update call
- * [#2176] Alsa mic avail data debug
- * [#2176] Add some debug message for mic loss problem
- * [#2176] Flush mic ring buffer when offholding a call
- * [#2176] Reset ringbuffers' readpointer when adding main participant
- * [#2176] Fix getAvailData algorithm
- * [#2176] Reset ringbuffer's readpointer when adding a new participant
- to a conference
- * [#1744] Regex object renamed to Pattern. Previous attempt at
- providing
- * [#2176] Fix detach main participant problem when adding new one
- * [#1976] Use right domain to translate
- * [#1976] Add xml menu description
- * [#2176] Store a list of confernece participant in client
- * [#2176] Fix add participant, joinparticipant methods
- * [#2181] Do not install dbus-c++ headers + add return value
- * [#2176] Fix minor call handling instabilities
- * [#2174] Fix incoming IP call contact address
- * [#2211] Add test to protect NULL pointer
- * [#1163] Add Advanced account configuration section
- * [#2176] Add some usefull comments and debugging info
- * [#2176] Add conditions to display security icons in conference
- * [#2176] Fix detaching one participant while keeping communication to
- others
- * [#2176] Reenable userActive.svg in call tree
- * [#2176] Make user active blue (not red)
- * [#2176] Fix user active picture
- * [#2176] Fix "hidden" merge conflict in sipvoiplink
- * [#2176] Remove iax audio stream on peer hungup
- * [#2174] Multiple UDP transports functional (TESTED with 2 accounts
- and 3 calls)
- * [#2176] Fix fix audio stream binding in iax
- * [#2174] Create a default UDP transport + use tp selector for dialogs
- also
- * [#2176] Register iax audio stream in mainbuffer
- * [#2176] Fix getAudioCodecName in IAXvoipLink
- * [#2176] Fix iax account init
- * [#2176] Handle multiple account using the same sip transport
- * [#2165] Add .png files
- * [#2176] Small fixes concerning dtmf
- * [#2176] Fix make uninstall in codecs
- * [#2174] remove stund makefile generation
- * [#2176] Add conference lock
- * [#2174] Add transport selector for multiple accounts
- * [#2176] Change userActive picture from red to blue
- * [#2176] Fix security pixbuff in calltree
- * [#2176] Replace sfl.zid in .cache/sflphone instead of .sflphone
- * [#2176] Fix add call description
- * [#2176] Remove detach button from toolbar
- * [#2176] Fix calltree call description state and state code in
- conferences
- * [#2176] Fix pulse audio double free
- * [#2176] Fix conference selection
- * [#2174] Clean up - remove stun settings in client network
- configuration panel
- * [#2174] Remove voviva stun code
- * [#2174] Rsolve STUN with pjsip - DO NOT WORK
- * [#2165] Add user svg
- * [#2165] Debugging sip call failed
- * [#929] Link against uuid if installed
- * Oops
- * Fixed bugs related to libsexy (with GTK < 2.16)
- * [#929] Remove uuid-dev dependency in the core
- * [#2165] Debugging no negociated codecs at communicatio start
- * [#2165] Fix calltree bug (gtktreestore instead of gtkliststore)
- * [#2165] Fix several merge problems
- * Updated opensuse packaging script
- * [#1163] Add missing figures
- * [#1163] Update INSTALL file
- * [#2165] Fix IAX
- * [#2165] Add recordabe interface
- * [#2165] Finish recording refactoring for call (not for conference)
- * [#2165] Enable speaker recording for two different calls
- simultanously
- * [#2165] Implement call recording using the Recordable interface
- * [#2165] Add get and set to AudioLayer's audio recorder
- * [#2165] Add class recordable from which inherit call and conference
- * [#2006] Fix G722 and Speex 8khz codec conferencing
- * [#2006] add recording of audio buffers
- * [#1163] Add general settings section
- * [#1163] Fixes makefile error
- * [#2006] Fix some minor issues
- * [#2006] Drag a conference call on another conference call
- (difference conferences)
- * [#2006] Fix dragging a conference on itself
- * [#1744] Integrating some of the needed regular expression patterns
- in order
- * COmplete call features
- * [#1744] Added support for named subgroup in the Regex object. Also,
- new
- * [#1744] Adds thread safety features, compile() and setPattern()
- methods to the Regex class.
- * [#1744] Fix inconsistency in the finditer method from the last
- commit.
- * [#1744] Added regex pattern object built on top of libpcre. To be
- used
- * [#1744] Initial commit towards implementing RFC4568. Unimplemented
- in the
- * [#2157] Hide "security" and "advanced" tabs for IAX under account
- * [#1163] Add call features section
- * [#2006] Add joinConference capabilities
- * [#2006] Add dbus joinConference signal
- * [#2006] Drag a conference call onto a conference to add it
- * [#1163] Add addressbook section
- * [#2006] Drag a conference call onto a single call to create a
- conference
- * [#2006] Expand rows automatically
- * [#2006] Add minimal multiple conference handling
- * [#2006] Add atached/detached conference icons
- * [#2006] Add function processRemainingParticipant
- * [#2006] Deep refactoring, fix hangup bug
- * [#1163] Update documentation - Accounts part
- * [#1976] Integrate user doc to gnome client build system
- * [#2122] Remove double inclusion in dbus-c++/src/Makefile.am
- * Remove pjproject version number
- * [#2006] Fix peerHungup
- * [#1976] Make Yelp accessible from the GNOME client (need to install
- the sflphone.xml first)
- * [#2006] Fix multiconferencing hangup
- * [#2006] Fix hangup calls in a conference
- * [#2150] Make IAx2 reappear
- * [#2006] Fix detach participant on multiple call
- * [#2006] Can remove rining call from a conference
- * [#2006] Reinit confID when removing a participant
- * [#2006] Remove get isCurrentCAll in hangup/peerhungup (SipVoipLink)
- * [#2006] Fix refuse call
- * [#2006] Fix answerring incoming call
- * [#2006] Refactor conference's participant list
- * [#2101] Re-integrate test compilation in main build system
- * [#2101] Make the test directory compile
- * [#2136] Restore history functionality
- * [#2006] Fix binding main participant to himself
- * [#2006] Fix add current/incoming/onHold participant to an existing
- conference
- * [#2006] Fix add incoming calls to an already created conference
- * [#2006] Fix remove stream
- * [#2006] Fix detachParticipant/removeParticipant switchCall ids
- * [#2006] Fix adding a call in conference having state "CURRENT"
- * [#2006] Remove/add main participant from conferences
- * [#2006] Hold/unHold conference
- * [#2006] Detach a partcipant from drag n drop
- * [#2006] Hangup a conference
- * [#2006] Add hold/unhold conference dbus messages
- * [#2034] gtk-ui fix under the "basic" tab.
- * [#2006] Fix dragging calls on conference calls
- * [#2006] Fix detach participant from a conference
- * [#2034] Added default message is status bar under the account config
- dialog
- * [#2112] Fix a crashed caused when a non-md5 password was sent to
- pjsip.
- * [#2006] Detach participant by ID
- * [#2006] Fix addParticipant method in managerImpl to handle
- incoming/answered calls
- * [#2006] Add addParticipant method in managerimpl and related dbus
- messages
- * [#2111] Added the ability to configure zrtp on sip.sflphone.org from
- * [#2106] Fixed problem in the account assistant under gtk-ui. Also,
- assistant.c
- * [#2006] Fix dragging a conference call on another conference call
- (same conference)
- * [#1904] Small UI fix. Assistant was moved from "Call" to "Edit"
- menu.
- * [#1904] Fix a wrong label under gtk-ui.
- * [#2034] Renaming and source code splitting.
- * [#2034] Status bar added to account window to better reflect the
- registration
- * [#2006] Make calltree_remove_call recursive (for GtkTreeStore)
- * [#1110] Small gtk-UI fix in the account window (alignment).
- * [#2006] Fix remove conference, display children which are still
- active
- * [#2006] Recursive function call in calltree_update_call
- * [#2006] Add multilayered capabilities to calltree (GtkTreeStore)
- * [#2006] Implement remove conference in calltree
- * [#2034] Now useless as Direct Ip calls settings moved under
- Preferences.
- * [#2034] Edit/add buttons were set insensitive all the time under
- gtk-ui.
- * [#1887] Information about the state of the current SIP call is
- displayed
- * [#2006] Add call tree remove callback
- * [#2006] Fix create_conference function
- * [#2006] Update conference_added_cb to add new conference to the list
- * [#812] Added new tab under GTK-ui Preferences. Moving Direct Ip
- Calls from
- * [#2121] Disable temporarily test compilation
- * [#2006] Fix conferencelist to handle conference_obj_t instead of
- gchar
- * [#2006] Add conference_obj structure
- * [#2121] Update version
- * [#2006] Fix conference selection
- * [#2101] Use the new source tree to fetch the right object files
- * [#2006] Add conference in calltree
- * [#2006] Add Dbus signal conference added/removed/changed
- * [#2006] Add getConferenceDetails call on dbus
- * [#1904] Registration expire now appears as a spin box under gtk-ui.
- * [#812] Fixing a segmentation fault caused by a non-existing account
- ID
- * [#2006] Add getConfList method over dbus
- * [#2006] Add a conferencelist data structure in client-gnome
- * [#812] Defaults value are now sent if a non-existing account is
- requested
- * [#2006] Add sflphone action sflphone_join_participant
- * [#2006] Fix buffer read pointer problem deletion
- * [pjsip] Attempt at fixing via header incompatibility with
- Freeswitch.
- * [#1797] forget something
- * [#2006] Add call new state conferencing in deamon
- * [#2006] Remove addParticipant method for conference, use
- joinParticipant only
- * [#1163] Update INSTALL documentation
- * [#812] Msec/sec values were not taken into account.
- * [#1797] Make pjproject-1.4 compile
- * [#2006] Add Detach participant method
- * [#2006] Dragndrop fully functional with INCOMING and HOLD call
- * [#1797] Add pjproject-1.4
- * [#1797] Remove pjproject-1.0.3
- * [#2006] Get call state in conference related function
- * [#2006] Add joinParticipant (conference) method in ManagerImpl
- * [#2006] Add joinConference DBUS message
- * [#2006] Store the previously selected call_id on dragndrop
- * [#2006] Fix GValue pointer unref in selection callback
- * [#2006] Store dragged call_id
- * [#2006] Update drag_data_received_cb callback to manipulate CallIDs
- * [#2006] Add dragndrop signals
- * [#2006] Set calltree reordable
- * [#812] Adds the ability to create a TLS listener in case the user
- requests
- * [#812] Adds the ability to configure local/published address from
- * [#1883] Move switchCall in onHoldCall function
- * [#812] Deals with the published address/port problem when
- integrating TLS.
- * [#1883] Switch call id in managerimpl when peerHungUp
- * [#1883] Switch call id before hangup
- * [#1883] Add usefull and permanent debug info for conference
- cretion/deletion
- * [#812] Fix various segmentation faults related to Direct IP kind of
- calls.
- * [#1883] Fix deletion of std::map elements using iterators
- * [#2014] Add libzrtpcpp build dependency
- * [#1883] Still some for loop test ambiguity (while loop instead)
- * [#1883] Fix for loop initial test ambiguity (use while loop instead)
- * [#1883] We must discard data in urgent ring buffer if data is get in
- mainbuf
- * [#1883] Fix availForGet same id for ringbuffer and readpointer
- * [#812] Match "sips" as a Direct IP Call when the user enter a sip
- uri
- * [#812] Fix segmentation fault related to SIP URI creation.
- * [#812] Towards integrating multiple tls listeners at the same time.
- This
- * [#1883] Add debug messages in conference and fix mainbufferTest
- * [#812] gkt-ui fix. Private key must be fed as a filename and not as-
- is.
- * [#812] TLS integration within sipvoiplink and pjsip. Also,
- configure.ac
- * [#1883] Fix Alsa/Pulse mallocation
- * [#1883] Fix data corruption in AudioRtp's micData buffer
- * [#812] Full dbus integration for all the tls related options under
- gtk-ui.
- * [#1883] Fix memory leaks in audiortp session
- * [#1883] Fix mem leaks in audio rtp
- * [#812] Fix setAccountDetails where TLS_ENABLE was set to the value
- * [#812] Small gtk-ui fix.
- * [#811][#812] Small gtk-ui fix.
- * [#812] Introduced a mechanism for configuration files that makes
- possible
- * [#812] New dbus bindings added. Also, configuration compliance was
- enforced
- * [#1881] Remove default buffer from MainBuffer (update unit-tests)
- * [#1881] Add ring buffer read pointer tests
- * [#1883] Fix issues in ringbuffer reader pointers
- * [#2034] Implementing a new configuration dialogue for TLS transport
- settings
- * [#1883] Add some usefull debug and safety checks
- * [#2028] Notify the client with libnotify when the zrtp negotiation
- failed.
- * [#811] Harmless no to throw an exception, an makes the application
- less
- * [#2028] A minidialog is showed to the user under sflphone-client-
- gnome
- * Removed useless file.
- * Ignoring Makefile in src/widget
- * [#2027] Fix segmentation fault when showMessage callback is called
- after
- * [#2026] keyExchange was set to ZRTP instead of "1"
- * [#2024] Fix the wrong summary at the end of the assistant.
- * [#1883] Fix mnagerimpl conference map insertion
- * [#1883] Add Mutexes in MainBuffer
- * [#811] Gtk ui was not presenting the right information about zrtp
- for
- * [#2023] security icons were not installed in sflphone-client-gnome.
- * [#2021] Fix a mistake in the readme from sflphone-common that gives
- wrong
- * [#811] The current SRTP mode was not properly displayed for the
- IP2IP
- * [#1743] Re-implementation of the "automatically remove error dialogs
- [...]"
- * [#2017] [#2019] Fix the inability to dial a number and place a
- registered
- * [#811] Final re-integration of ZRTP support in the main branch from
- 0.9.6
- * [#1883] Fix map insertion methods
- * [#811] Combo box now is now set to the active key exchange method
- * [#811] ZRTP options now configurable back again from the Gtk UI.
- IP2IP
- * Updated hostname for git clone
- * [#1883] Add minimal functionalities to create a conference
- * [#811] re-integration of all the methods and signals on dbus.
- ManagerImpl
- * [#811] Got out of a precarious position were nothing would compile.
- * [#1976] Build documentation squeleton with docbook
- * [#1883] Add sflphone-client "addParticipant" button for conference
- * [#1994] Better organize the source directory structure. New
- subdirectories
- * [#1883] Add a simple Conference class
- * [#1882] Use static audio buffer in Pulse and ALSA layer (instead of
- malloc)
- * [#811] First commit toward re-integration and refactoring of ZRTP
- * [#1882] Flush RTP ring buffer before entering mainloop
- * [#1882] Fixed MainBuffer::UnBinCallID() in case there is no
- ringbuffer
- * [#1882] Test (and fixe) high level conference and mixing
- functionalities
- * [#1772] Apply patch to compile on fedora (sent by Marcin
- Zajączkowski <mszpak@wp.pl>)
- * [#1882] Update Bind, unBind call_id in MainBuffer
- * [#1959] This adds the ability to store password as an MD5 Hash in
- the
- * [#1538] Fixes rules compilation
- * [#1930][#1931] Fixed a mistake (again) related to index and
- credential count
- * [#1753] Remove ILBC from pjproject - Hacks in pjsip
- * [#1930][#1931] Credential was not selected properly using realm
- * [#1882] Finilize multiple reading pointer in RingBuffer
- * [#1538] Remove configure from autogen.sh to respect debian upstream
- authors policy
- * [#1773] Remove generated files from repo
- * [#1791] Use XDG_CACHE_HOME to save pid file
- * [#1791] Fixes path to save history
- * [#1791] Fix debian installation scripts
- * [#1930][#1931] Settings are now taken into account in the server.
- * [#1882] Add ringbuffer default ring buffer pointer in methods
- involving mStart
- * [#1882] Add default ringbuffer pointer
- * [#1882] Add RingBuffer multiple read pointer basic functionnalities
- * [#1882] Fix MainBuffer flushData unit test
- * [#1930][#1931] Ability to save and retreive the configuration from
- * [#1882] Added Multiple CallID mapping to MainBuffer
- * [#1791] Not much
- * [#1791] If XDG env variables are not null but empty, use default
- ones
- * [#1791] Make XDG_CONFIG_HOME writable
- * [#1930][#1931] Partial commit. Not working yet. Cannot delete
- account
- * [#1881] Fixed alsa capture latency problem
- * [#1881] Fixed Alsa capture temporarily
- * [#1930] [#1931] Partial unbroken commit providing the ability to
- * [#1881] MainBuffer implemented in AudioLayer/AudioRTP
- * [#1881] Add discard and flush unit-tests
- * [#1881] Add discard and flush functionnalites to MainRingBuffer
- * [#1881] Add availForGet in MainBuffer
- * [#1881] Add availForPut function to MainBuffer
- * [#1880] Remove AudioRTP* pointer from SipVoIP (reapered while
- merging master)
- * [#1881] Add a map between call id and coresponding ring buffer
- * [#1855] Refresh pot file and upload on Launchpad
- * [#1881] MainBuffe now robust to false ids on getData and putData
- * [#1881] Fix big big big memory leak
- * [#1881] Add getData and putData to mainBuffer
- * [#1881] Unit-test basic ring buffer functionnaities
- * [#1881] Add class MainBuffer and basic buffer creation unit-tests
- * [#1880] Fix call transfer (step2) issues
- * [#1880] Moved AudioRtp* pointer from SIPVoIPLink to SIPCall class
- * [#1791] Add postinst script to keep user data when migrating
- config/history file
- * [#1797] Make pjsip compile
- * [#1777] Code indentation
- * [#1791] Use XDG_DATA_HOME and XDG_CONFIG_HOME for sflphonedrc and
- history + unit tests
- * [#1746] Useless space does not appear anymore when volume sliders
- and
- * [#1643] GtkCheckMenuItem is used instead of icons for elements in
- the
- * [#1110] [#1668] STUN parameters are now located in the preferences,
- under
-
- -- Julien Bonjean <julien.bonjean@savoirfairelinux.com> Fri, 06 Nov 2009 11:23:15 -0500
-
-sflphone-common (0.9.6-SYSTEM) SYSTEM; urgency=low
-
- ** 0.9.6 **
-
- * Documentation on echo test
- * [redmine_down] codec names not displayed in total
- * [redmine_down] crash when hanging up a dialing call because tries to
- add it to history whereas no starttime
- * [#1927] alternate every time screen changed to call history
- * [#1886] clean code
- * [#1886] debug messages when loading history removed
- * [redmine_down] sflphone-kde icons
- * [#1855] Update language files
- * [#1502] Update version number
- * [redmine_down] setHistory at close
- * [#redmine_down] Handle PJ_DECLINE_SC as failure
- * [#1923] Fix segmentation fault when adding a new account
- * [#1923] Check on iterator before setting the config
- * [#1904] Added mnemonic to tabs in sflphone-client-gnome.
- * [#1905] The daemon was not sending the currentSelectedCodec signal
- on dbus when answering a call.
- * [#1922] Default values set to all account details
- * [#1886] Spinbox reg expire enables apply, and address book is not
- visible when disabled
- * [#1905] Bug fix for segmentation fault caused by an empty string,
- * [#1910] Warnings in test directory
- * [#1919] Error fixed
- * [#1855] Update russian translation - Hussein Abdallah
- * [#1910] Remove files
- * [#1919] fixed
- * [#1777] Code indentation
- * [#1918] fixed
- * [#1917] fixed
- * [#1910] Remove warnings compilation in src
- * [#1886] removed AccountListModel in configskeleton
- * [#1914]
- * [#1911] check previous and new port
- * [#1910] Remove compilation warnings in src/dbus and src/history
- * [#1910] Remove compilation warnings in src/audio
- * [1855] Update german translation - Sven Werlen
- * [#1909] removed
- * [#1906] Done
- * [#1904] The registration expire value is now configurable from the
- * Cleaned up debug messages.
- * [#1886] separated initCallItem in two functions
- * [#1886] reversed error in commit
- * [#1886] clean debug
- * [#1886] changed Name of classes and files
- * [#1886] clean
- * [#1870] In call_state_cb (dbus.c:126), _time_stop was overridden by
- the actual time.
- * [#1884] Added some new gpg flags to prevent tty warnings
- * [#1886] Clean audio config dialog
- * [#1886] No more compile warnings. + 1 comm
- * [#1872] Check if the user input is smaller than PJ_MAX_HOSTNAME.
- * [#1886]
- * [#1785] Fixed build when no new commit
- * [#1852] If chosen by the user, the hostname can now be solved and
- used
- * [#1871] * and # inverted back
- * [#1869] Conditional compilation that checks if
- * [#1309] removed test in main
- * [#1425] Put actions in SFLPhone window class instead of ui view,
- made a separate toolbar for screens.
-
- -- SFLphone Automatic Build System <team@sflphone.org> Mon, 27 Jul 2009 09:53:00 -0400
-
-sflphone-common (0.9.6~rc2-SYSTEM) SYSTEM; urgency=low
-
- ** 0.9.6~rc2 **
-
- * [#1755] Remove generated file
- * [#1753] restore ilbc ...
- * [#1866] Methods getSipPort and setSipPort now have an effect on the
- * [#1753] make pjsip compile without ilbc. Use ./autogen.sh --disable-
- ilbc-codec
- * [#1855] Fix error in russian translation
- * [#1805] Remove the old flawed signal mechanism which was failing in
- * [#1855] Refresh translation
- * Spanish translation finished + po README files updated + echo's in
- copy-in-clients
- * [#1850] Yun made the chinese HK-CN translation
- * [#1848] Fix transfer interface bug
- * [#1862] At install, kde client installs only french translation file
- * [#1841] A new fallback mechanism was added to the internal resolver
- in PJSIP.
- * Started AccountList model/view
- * [#1855] Remove po subdir in Makefile.am
- * [#1855] Fix typo error in sflphone-client-gnome
- * [#1855] Do not generate Makefile in sflphone-common/po
- * [#1855] Copy translation files into both clients dirs
- * [#1855] Remove po dir from sflphone-common
- * Comments added
- * [#1860] mailbox->voicemail...
- * make scripts executable
- * [#1855] French translation
- * [#1855] Chinese zh_HK partially filled...
- * [#1859] An unnamed pipe monitored by poll() was added. When we want
- to
- * [#1855] Sven completed the first part of the german translation
- * [#1855] Cantonese manually filled for already translated, almost
- equal strings
- * [#1855] Merge russian translation
- * [#1855] Spanish manually filled for already translated, almost equal
- strings
- * [#1855] Update german translation in ./lang/de
- * [#1858] This problem was fixed by removing a useless line in
- * [#1855] merged existing translations in lang/ sflphone.po's
- * [#1842] [#1843] An attempt at improving the expected behaviour that
- can't
- * [#1855] added po folder in gnome client and scripts for copying from
- common lang folder to clients
- * [#1853] Edit before call does nothing on call history
- * Put most language entries possible in common. From 300 to 250
- entries. Stays underscores problem. Scripts for copy in clients.
- * commit to merge master
- * [#1825] Changed "Bad authentification" to "Authentication Failed".
- * common po files
- * [#1753] Remove ILBC from pjproject
-
- -- SFLphone Automatic Build System <team@sflphone.org> Fri, 17 Jul 2009 19:12:44 -0400
-
-sflphone-common (0.9.6~rc1-SYSTEM) SYSTEM; urgency=low
-
- ** 0.9.6~rc1 **
-
- * Update some version number
- * [#1792] Creates .sflphone directory with permission 600. Also,
- "chmod 600" after
- * [#1810] GUI is now notified that the call failed. Also, a segfault
- was
- * [#1816] Address book search disabled when disabled address book and
- enabled it back plus button stays triggered
- * codeclistmodel + asynchronous loading of address book +
- enable/disable address book
- * [#1810] Now checking SDP answer after 200 OK. Still need to
- implement full
- * [#1794] Can't use the interface during a call
- * Updated translation files
- * Russian translation integrated
- * Codec list model/view started.
- * [#1807] Add configure.ac in pjproject-1.0.3
- * [#1787] closeRtpSession added in some places where it should have
- been
- * Use Item class for contacts and accounts
- * Comments + clean code
- * [#1794] Improved debug messages
- * [#1805] Replaced the old and unreliable mecanism that was was
- waiting for
- * [#1794] Can't use the interface during a call
- * [#1787] For those cases where no registered SIP account is
- configured
- * [#1797] Make pjsip compile
- * [#1787] Minor changes. Removed useless commented line. Changed order
- of
- * [#1777] Code indentation
- * [#1797] Update package generation with new pjsip version
- * [#1798] Does not hang up when the call is building up
- * [#1797] Update .gitignore with new pjsip version
- * [#1797] Remove generated files from repo
- * [#1797] Main build system now uses pjproject-1.0.3
- * [#1797] Add pjproject-1.0.3
- * [#1797] Remove pjproject-1.0.2
- * [#1796] Computing time optimization (samplerate conversion)
- * [#1787] _audiortp->start() moved away from offhold(),
- SIPCallAnswered()
- * [#1312] Added new states for calls initialized by other clients
- * [#1795] Crashes when adding a new account, checking it and applying
- * [#1782] Missing icons
- * [#1793] KDE client compilation problem
- * Fake ringtone files can no longer be set.
- * indentation
- * [#1312] Able to fetch to differentiate incoming/ringing call state
- * [#1784] Use DESTDIR variable in po Makefile - fix language file
- installation
- * [#1785] Fixed typo
- * [#1785] Fixed changelog update
- * [#1759] ./autogen.sh --prefix=/usr --with-debug to use optimization
- level 0
- * [#1773] Changed snapshot naming convention
- * [#1773] Removed gpg agent use, added repository cache cleaning
- * [#1759] Use optimization level 0 for repository, 2 for packages
- * [#1777] Code indentation/formatting
- * Translated new features in french
- * [#1785] Added missing changelog entry
- * [#1781] Window title is SFLPhone
- * [#1777] Add code indentation/formatting in the buil system
- * [#1774] Can't set voicemail number in KDE account creation wizard
- * [#1775] Can't modify account information for account created with
- the wizard
- * [#1771] Add a "Default" button in context menu to disable chosen
- prior account
- * [#1705]
- * [#1224] Remove generated file from the repo
- * [#1224] Remove generated file from the repo
- * [#1762] distclean target should remove kconfig generated files
- (settings.h, settings.cpp). Rename them?
- * [#1761] clear history button should really clear history
- * Dialpad works.
- * Implemented Dialpad widget instead of building it in main view.
- * Removed last occurence of the old config dialog, that made the build
- crash.
- * [#1755] Do not consider G722 as a dynamic payload elsewhere than in
- RTP layer
- * [#1753] Remove ilbc Makefile generation
- * [#1756] Implement a kde configuration dialog with kconfig xt and
- kconfigdialog class
- * [#1755] fix audiocodec folder parsing problem
- * [#1450] Reinit timestamp comparison in RTP, create session in
- newOutgoingCall
- * [#1753] Remove milenage third party code from pjsip
- * New Config Dialog integrated in GUI.(without codecs)
- * [#1753] Remove ILBC codec
- * kconfig started, tr2i18n -> i18n, icons folder, accountList changed
- * [#1705] Fixed Audio RTP thread creation/start
- * [#1714] Fix codec negociation result handling
- * [#1678] Fix audiortp payload setting
- * [#1678] Put bac putData method in rtp
- * [#1669] gtk_file_chooser_get_filename() support UTF-8 by default
- * [#1735] Add conditions to sdp update call if call declined
- * [#1737] substr of recordings destination folder to remove "file://"
- should be done in client rather than in daemon
- * [#1731] Enlarge audio stream buffer size
- * [#1714] Missing true
- * [#1317] Fixed Mandriva timeout
- * [#1317] Changed tag convention
- * [#1317] Cleaned git-dch
-
- -- SFLphone Automatic Build System <team@sflphone.org> Fri, 10 Jul 2009 15:49:56 -0400
-
-sflphone-common (0.9.6~beta-SYSTEM) SYSTEM; urgency=low
-
- ** 0.9.6~beta **
-
- * spec files for mandriva and opensuse updated with buildrequires
- libqt4-dev >=4.3
- * [#1700] Cannot build on ubuntu 8.10 and a few other distribs
- * [#1502] Update version number where applicable
- * [#1642] Update client icons
- * [#1450] Clean up useless debug and comments in sipvoiplink and
- audiortp
- * [#1450] Remove Semaphore object in AudioRtp thread deletion
- * [#1450] Audio RTP init now synchronized with Sip/SDP
- * [#1693] kde client crashes when changing codecs order/activation
- * [#1450] Deep refactoring of audiortp
- * [#1450] setRtpSessionRemoteIp
- * [#1689] getCallList at start
- * [#1224] Change path in package files
- * [#1450] Audio RTP initialized only once, payload and remote ip set
- at runtime
- * [#1450] Add setRtpSessionMedia and setRtpSessionRemoteIp address
- * [#1642] Make GNOME GUI fresher and younger ;)
- * [#1686] Status bar displaying used account
- * added sflphone-kde icon so that it compiles
- * [#1659] Ending a call causes the daemon to crash
- * corrected introspection XMLs, po files...
- * [#1211] g722 media descriptor in codecDescriptor
- * [#1310] Install sflphoned in $(prefix)/lib/sflphone
- * [#1502] Do not install test binaries and dbus utilitaries
- * [#1224] hack for pjsip build system!
- * [#1224] Remove pjsip binaries from repo
- * [#1224] Upgrade to pjsip 1.0.2
- * [#1658] About SFLphone (bugs)
- * [#1658] About SFLphone
- * [#1660] Displaying all dialed numbers in a call
- * Tested status bar.
- * [#790] Optimize pulse audio streams parameters
- * [#1678] Some usefull debug messages for mutex/semaphore deadlock
- problem
- * [#1669] Add/remove some usefull/unusefull debug
- * [#1665] Fix latency related to pulse audio stream openning/closing
- * [#1457] Make the menus and panels accessible in french
- * [#1457] Improve broken keyboard accessibility in menus and conf
- panels
- * [#961] Instanciate only once the searchbar icons
- * [#961] Restore transfer fonction
- * [#961] Filter on the history type OK
- * [#961] Fix compilation problems on hardy/intrepid
- * [#1157] Commit missing files
- * [#790] Reduce number of start/stop streams call on pulse audio
- * [#1639] kde client crashes when no account registered
- * [#1620] Fix the searchbar
- * [#1620] Get back caltree as it was during gtkcritical area
- * [#1620] Add history filter reinit function
- * [#1335] Add a missing label in address book preferences
- * [#1561] Update russian translation - Hussein Abdallah
- * [#1605] Fix edit menu french translation
- * [#961] Enable to search in the history according to the call type
- * [#1449] Searchbar does not work anymore
- * [#961] Add popup menu on the entry primary icon for history
- * [#1317] Fixed KDE client package dependency
- * [#936] speex 32 khz integration completed
- * [#936] Use 320 frame size
- * [#936] Test using a frame size at 320 smpls
- * [#1214] Enable / Disable history
- * [#1607] Fix compilation problem for ubuntu 8.10 (libsexy)
- * [#1313] Implement processDataEncode processDataDecode in audiortp
- * [#1613] codec list order can't be set
- * Better handling of localisation + added languages + corrected
- warnings + begginning of new config dialog with kconfig + 14px
- account leds
- * [#1214] Save and load history according to the limit timestamp +
- unit tests
- * [1609] Fix call number copy/paste feature
- * [1607] Restore clear action icon in searchbar
- * [#936] Try to decode using 1280 samples
- * [#936] Add some debug
- * [#936] Add .cpp file
- * [#936] Oops Forgot speex 32 khz
- * [#1214] Add configuration panel for history + D-Bus calls
- * [#1313] Test rtp thread function, frame size, nbbytes, resampling
- * [#790] Flush audio data before closing audio streams
- * [#1214] History displays local time
- * [#1214] Skip empty field on display
- * [#1214] Associate an account to an history entry
- * [#1342] Get addressbook options sensitive/non-sensitive
- * [#1211] Clean up and comments
- * [#1211] Get back to 20 ms framesize
- * [#1211] Use sendImmediate instead of putData in RTP
- * [#1211] Fix nb byte available in RTP
- * [#1211] Clear condition on maxNbSamples in RTP
- * [#1211] Fix max byte available in RTP session
- * [#1211] G722: Use 160 samples per frame instead of 320
- * [#1211] Test using a dynamic payload
- * [#1211] Test using a dynamic payload type
- * [#1211] Rename size variable (nb_samples, nb_bytes)
- * [#1211] Test g722 ip-to-ip sending twice the data lenth
- * [#1211] Test g722 ip-to-ip
- * [#1214] Do not select an history item by default at startup
- * [#1214] Remove some compilation warnings
- * [#1214] Handle empty field - remove g_print
- * [#1214] Add each history item only once
- * [#1214] Handle call timestamps properlier
- * [#1214] Do not need timestamp files anymore
- * [#1214] Use the saved date for history entry
- * Clean up
- * [#1214] Client doesn't crash if the D-Bus call fails
- * [#1214] Client is able to save its history - still some glitches
- * [#1211] Forgot 16000 for g722
- * [#1211] G722 initialization
- * [#1214] Save name/number, successfully load the history if no fields
- are empty
- * [#1499] Fixed destination directory bug
- * [#1214] Restore all the functionalities; peer name/number way more
- easy to handle !!
- * [#1214] Add callable_object instead of call_t, refactoring
- * [#1211] Test with polycom soundstation 16000
- * [#1211] Remove C like inline function in g722 codec
- * [#1342] Finalize gnome client preference window formating
- * [#1214] Retrieve the history when the gnome client startsup
- * [#1306] Implement localization for KDE client
- * [#1593] enable accounts apply button when account checked/unchecked
- * [#1214] Implement the dbus calls on server side
- * [#1214] Add serialized/unserialized functions to pass data on DBUS
- * [#1342] Formating gnome client configuration windows
- * [#1214] Save sucessfully a map of history items
- * [#1499] Removed multiple jobs compilation for KDE client (2)
- * [#1214] Load history from file into memory, add unit tests
- * [#1534] Throws a length_error exception in case URL exceeds
- std::string max_size
- * [#1499] Removed multiple jobs compilation for KDE client
- * [#1565] make account leds smaller
- * [1430] Fix dbus debug
- * [#1562] crashes when trying to change item of a call of state "OVER"
- * [#1116] Fix compilation bug
- * [#1317] Added mandriva and opensuse-11 64 bits
- * [#1108] Add messges in main window concerning transfer success
- failure
- * [#1116] Fix compilation problems
- * [#1211] g722 Makefile
- * [#1108] Client side transferFailed/trasferSucceded signals handling
- * [#1211] G722 mostly completed,
- * [#1555] make bigger toolbar (24x24)
- * [#1551] remove default mailbox number in wizard and disable mailbox
- button when first account doesn't have mailbox number
- * [#1342] Re-add sflphone manpages
- * [#1116] Fix compilation on non-jaunty distros
- * [#1317] Fixed opensuse startup sleep
- * [#1108] Add a signal in the client to notify successful or failed
- transfer
- * [#1108] Dbus signals concerning call transfer success/failure
- * [#1317] Added opensuse to automatic build system
- * [#1223] Fix manpages bug
- * [#1060] german translation glitch
- * Clean up some gnome client warnings
- * [#1547] replace ugly account leds by beautiful icons
- * [#1548] add close button that hides windowand just hide on clicking
- the cross
- * [#1549] put introspec XMLs in the client's source
- * [#1312] Implement getCallList D-BUS method
- * [#1116] Clear text in history and contacts
- * [#1499] KDE integration
- * [#1469] Modify header linkers in dbus-c++'s Makefile.am's
- * [#1469] Remove examples folder from dbus-c++
- * [#1214] History integration in build system; unit test squeleton
- * [#1317] Cleaning
- * [#1469] Remove configure stuff in dbus-c++
- * [#1469] Add unofficial mainline dbus-c++
- * [#1469] Remove dbus-c++ from freedesktop
- * [#1430] Bring account changed signal/callback back to normal
- * [#1060] Update german translation - Sven Werlen
- * [#1430] Add marshaller one string define
- * [#1430] Send account change signal broadcast using account id
- * [#1430] Remove condition on setRegistrationState, cause stun to
- crash
- * [#1317] Centralized version handling
- * [#1317] Fixed version number on sfl-git-dch
- * [#1317] Refactoring for new distributions
- * [#1215] Fix account order at startup if latency
- * [#1088] Restore sip dns srv
- * [#1214] Add squeleton for history manager
- * [#1430] Add accout id to accout changed method
- * [#1430] No connectionStatusNotification (account changed) if no
- changes
- * [#1538] Add COPYING file
- * [#1430] Add audio rtp thread tests
- * [#1317] Changed version detection
- * [#1538] Document license in libs/stund
- * [#1317] Added version files
- * [#1538] Apply François patches - debian packages
- * [#1317] Updated spec files
- * add files
- * [#1538] Apply François patches - debian packages
- * [#1535] Change program file structure (directory src...)
- * [#1317] Updated build system scripts
- * [#1317] Cleaning
- * [#1317] Copied introspect files to gnome client
- * [#1317] Added opensuse to build-system : first-shot
- * [#1317] Remove spec files from configure
- * [#1317] Added missing prefix
- * removed debug for daemon account fix
- * [#1430] Add a connection reference which most likely belong to
- libdbus
- * [#1430] Use shared connection instead of private
- * make daemon find the account, added userMatch
- * Clean code, add comments...
- * [#1317] Fixed packaging rules
- * [#1317] Updated autogen
- * Updated autogen.sh for pjsip
- * [#1526] Set accounts order
- * [#1317] Fixed pjsip lib dirs
- * [#1317] Updated debian packaging for new pjsip configuration script
- * [#1317] Switch to autogenerated guess and sub files
- * [#1317] Updated pjsip inclusion in build system
- * [#1317] Replaced pjsip guess and sub files
- * [#1317] Fixed compilation issues on opensuse 11
- * [#1505] account list seem to crash the application when clicking
- Apply very fast...
- * [#1456] Add a flag to be replaced in the control files
- * [#1456] Added version dependancy handling
- * put account alias in AccountWidgetItem rather than in the item with
- " " before.
- * [#1034] The KDE client should start sflphoned if it is not started
- * [#1500] Handle options for notifications and display on incoming
- call.
- * [#1443] Client should not crash when receive an unexpected
- stateChanged signal
- * [#1403] Do not stop the notification anymore
- * [#1456] Added version dependancy handling
- * [#1426] Daemon crashes when get alsa plugin
- * [#1422] Improved error messages
- * commit for merge
- * [#1424] Change logo in tray icon and put a different one when
- incoming call
- * [#1425] first part done, window title...
- * [#1413] add manpages creating and installing in build system
- * [#1417] The client should start the account creation wizard if
- started for the first time (if config file doesn't exist)
- * [#1421] Make volume bars horizontal when dialpad is hidden.
- * Changed main window title and fixed a mistake in sflphone_const.h
- * [#1412] make debian package building work
- * changelog changed.
- * Changed addAccount method in gnome client.
- * Debian and man folders added.
- * [#1388] Change project name from sflphone_kde to sflphone-client-kde
- * Better handle of kabc check.
- * [#1351] Automatic generation of dbus interfaces in makefile
- generated by cmake
- * [#1307] Implement "edit before call" in history and address book.
- * [#1344] change action_call label in call history from "call" to
- "call back".
- * [#1308] Implement Hook feature in kde client
- * Improved build system.
- * #1219 : Add address book configuration page
- * Better handling of registration to the daemon.
- * #1039 : Add tray icon in kde.
- * Issue no 1216 : Double click on item in history or address book
- causes call.
- * display peer name in call list and call history when called from
- address book.
- * Address book functionnal with photo displayed.
- * Help menu kde available but actions disappeared. All fonctions in
- view.
- * Address book functionnal but ugly and making its own sort in the
- complete address book.
- * Account choice on right click, clean out includes, page address
- book, fixed bugs...
- * Wizard, double click, context menu...
- * Removed sflphone_kde.kdevelop.filelist
- * Added account creation wizard and translated interface in english.
- * Transfer functionnal but ugly.
- * transfer not functionnal
- * Bug fixed : unholding (UNHOLD_CURRENT, UNHOLD_RECORD)
- * Commit functional for push. With install.sh
- * Before merge.
- * Problem with enable accounts. Account display increased.
- * Functional with codec order working , playDTMF.
- * Commit functional.
- * sflphone_kde/build added in .gitignore.
- * complete commit for checkout previous.
- * Commit before checkout previous version to check the display
- bug(little font everywhere...)
- * Functionnal client. Rest : history icons, config icons and
- functionalities
- * commit before merge asavard for isRecording.
- * Call and Automate fusion done and seems to work.
- * Commiting before putting Automate class in Call class.
- * Functionnal main window without recording, history, voicemail, kio
- widgets.
- * client kde avec kdevelop.
- * Config Dialog almost finished.
- * Base of QT client
-
- -- SFLphone Automatic Build System <team@sflphone.org> Tue, 23 Jun 2009 11:12:06 -0400
-
-sflphone-common (0.9.5-SYSTEM) SYSTEM; urgency=low
-
- ** 0.9.5 release **
-
- * [#1060] FIx bug in chinese translation
- * [#1313] git add rtpTest.cpp rtpTest.h
- * [#1313] Add init/close rtp tests
- * [#1313] Basic instanciation of the rtp layer
- * [#1449] Gtk-Critical concerning history filters and new calls
- * [#1400] Make the match with the hostname instead of username
- * [#1324] Change status bar label for "Using %s (%s)"
- * [#1403] Icon size: 60x60 px
- * [#1403] Do not remove notification, improve icon quality
- * [#1403] Add smaller icon for gnome notifications
- * [#1403] Prevent crash when hangup && no notification
- * [#1403] Remove all actions on notifications; code refactoring
- * [#1451] Use stun.sflphone.org as default STUN server
- * [#1060] New po files - need to be translated
- * [#1060] Update french translation - Rebuild template file
- * [#1456] Add a flag to be replaced in the control files
- * [#1454] Make cppunit optional; remove from build deps in control
- files
- * [#1401] Add libexpat1-dev dependency in control files
- * [#1448] Take off these ugly debug messages
- * [#1448] fixed getTelephoneTone and getTelephoneFile() called
- repeatedly
- * [#1406] add liblog4c-dev in build-depends
- * [#1409] Restore .desktop icon
-
- -- SFLphone Automatic Build System <team@sflphone.org> Mon, 25 May 2009 11:34:40 -0400
-
-sflphone-common (0.9.5-SYSTEM~rc2) SYSTEM; urgency=low
-
- ** 0.9.5 rc2 **
-
- * [#1422] Improved error message
- * [#1402] Fix pjsip build
- * [#1404] Clear GTK-Critical Bug at client startup
- * [#1422] Added automatic VM shutdown when building on more than one
- VM
- * [#1422] Fixed some issues with new changelog generation script
- * [#1422] Moved distribution update to specific file
- * [#1422] Dropped git-dch, replace by home made implementation
- * [#1402] Fix pjsip build
- * [#1404] Clear GTK-Critical Bug at client startup
- * Changes for name based dbus connection
- * Clean changelogs
- * [#1343] Gnome: Implement a callback system to handle focus on
- different widgets
- * Debus Session
- * Refactoring Python code, PEP8
- * [#1430] Get back dbus_g_proxy_new_for_name
- * [#1430] Get back DBUS_BUS_SESSION type
- * [#1430] Dbus fixed owner message binding
- * Second test with DBUS owner
- * [#1404] Gnome -> Preferences -> Hooks
- * [#1404] Gnome -> Preferences -> Recordings
- * [#1404] Call History
- * [#1404] Gnome -> Preferences -> Address Book
- * [#1404] IF the first notification option disable the second
- notification
- * Dbus with fixed owner does not automatically start the deamon
- * Add codec debug tests in pysflphone
- * [#1407] Some print info
- * [#1407] Add a scenario to pick_up action
- * Test client dbus connection to a fixed owner
- * Add python dbus test suite
- * [#1161] Modified version handling in build system
- * [#1314] Test pulse audio and audio streams connect and disconnect
- * [#1402] Add info message after configure
- * [#1402] Build the daemon with the local pjsip library (vs the
- installed one)
- * [#1009] Fix Codec Sampling Rate set to zeros
- * [#1314] Add mutex to pulse layer audio streams
- * [#1314] Refactoring pulseaudio stream to test connect disconnect
- * [#1314] Refactoring of pulselayer to test conect/disconnect
- * Add debug messages in debus calls concerning account
- * [#1314] Add some return values to audio init functions
- * [#1406] add liblog4c-dev in build-depends
- * [#1409] Restore .desktop icon
- * Bug #1405: Fix strings as requested.
- * Bug #1404: Fix strings in preferences panel.
-
- -- SFLphone Automatic Build System <team@sflphone.org> Tue, 19 May 2009 12:08:03 -0400
-
-sflphone-common (0.9.5-0ubuntu1~rc1) SYSTEM; urgency=low
-
- [ SFLphone Project ]
- * [#1262] Updated changelogs for version 0.9.5-0ubuntu1 Snapshot 2009-
- 05-05
-
- [ Emmanuel Milou ]
- * Add some python CLI client code; not really functional
- * [#1108] Fix peerHungup method for IP to IP call
-
- [ Alexandre Savard ]
- * [#1108] Correct setting of SIP contact for direct IP call
- * [#1108] SIP user agent handles incoming REFER
-
- [ Emmanuel Milou ]
- * Remove website from repository
- * Update translation
-
- [ Alexandre Savard ]
- * Sflphone icon's tooltip changed for "configured" instead of
- "registered"
-
- [ Emmanuel Milou ]
- * Update translation
-
- [ Sflphone Project ]
-
- -- Sflphone Project <sflphone@mtl.savoirfairelinux.net> Tue, 05 May 2009 19:16:09 -0400
-
-sflphone-common (0.9.5-0ubuntu1~beta) SYSTEM; urgency=low
-
- [ Julien Bonjean ]
- * Updated Eclipse stuff
- * Improved addressbook config window
- * Added sflphone Eclipse stuff
- * Implemented addressbook list server side
- * Moved dbus stuff in dbus directory
- * Updated addressbook configuration
-
- [ Emmanuel Milou ]
- * Remove unuseful installation scripts. Use apt-get build-dep sflphone
- instead
- * fix bug #1090
-
- [ Alexandre Savard ]
- * defining speex 16khz
-
- [ Emmanuel Milou ]
- * Remove unuseful file from build system
- * Start dns srv resolver
-
- [ Alexandre Savard ]
- * Basic ogg/vorbis initialization
-
- [ Emmanuel Milou ]
- * Handle incoming IP-to-IP invite correctly
-
- [ Alexandre Savard ]
- * speex wideband 16000
-
- [ Emmanuel Milou ]
- * Better handling of incoming IP to IP call
- * DNS SRV resolution functional
- * Implement IAX2 incoming URL
- * Allow user to make IP call without any accounts configured
- * Add a contextual menu to edit a number from the contacts tab
- * Add comments, tooltip and new button to the contextual menu
- * add delete event, migrate to GTK 2.16 for sexy icons
- * Resolve ticket #1118
- * Update suse spec file
- * Add phone number cleanup functions, unit tests and panel
- configuration
- * Add pertinent test that fails
- * fix dependencies for suse package
- * Add contextual edit menu in history - #1120
-
- [ Alexandre Savard ]
- * Temporary comit: make speex wideband (16 khz)
- * Temporary: shared object for speex narrow band
- * Temporary: speex narrowband and wideband coexist
-
- [ Julien Bonjean ]
- * Fixed bug when no book selected
- * Fixed addressbook related compilation warnings
- * Fixed GTK client remaining compilation warnings
- * Fixed segfault when book removed since last sflphone run
- * Fixed bug when book is unreachable (ldap error)
-
- [ Alexandre Savard ]
- * Fix codec list in audio config window
- * Active/inactive speex codec by payload
-
- [ Julien Bonjean ]
- * Updated gitignore
- * Added some comments
-
- [ Emmanuel Milou ]
- * Add callto: handler script for browsers and al.
- * Integrate test compilation in the daemon build-system
-
- [ Julien Bonjean ]
- * Fixed g_object_unref warning for pixbuf
- * Cleaned too verbose output
- * Fixed toolbar update warning
- * Added support for asynchornous books open (first shot)
-
- [ Emmanuel Milou ]
- * Add a DBus call to fetch the call details from a call ID - Ticket
- #928
-
- [ Julien Bonjean ]
- * Improved async open books
- * Fixed bug #1139
-
- [ Emmanuel Milou ]
- * Add a way to save account order
- * commit missing files
-
- [ Julien Bonjean ]
- * Introduced log4c (ticket #1162)
-
- [ Emmanuel Milou ]
- * Load/save account order functionnal - ticket #813
-
- [ Alexandre Savard ]
- * Add CELT codec (#1143)
- * Make celt frame size 256 (*1143)
-
- [ Julien Bonjean ]
- * Switched everything to log4c (ticket #1162)
- * Updated eclipse settings
-
- [ Emmanuel Milou ]
- * Restore adding account - ticket #1172
- * Add liblog4c dependecy - ticket #1179
-
- [ Alexandre Savard ]
- * Double maxAvailByte for frame size in rtp (#1143)
-
- [ Emmanuel Milou ]
- * Add User-Agent SIP header - Ticket #1173
-
- [ Julien Bonjean ]
- * Fixed autoresize issue (#708)
-
- [ Emmanuel Milou ]
- * Remove libcppuint dependency for the debian packages
- * Look for libsexy only if gtk version < 2.16 - Ticket #1116
- * Remove libsexy dependency for jaunty. ticket #1116
-
- [ Julien Bonjean ]
- * Introduced unit tests (#1146)
- * Updated gitignore
- * Fixed Makefile (#1146)
-
- [ Emmanuel Milou ]
- * [TICKET #1112] Add a test on the voice buffer to send through iax
- packets
- * Remove doublon in dependencies
- * Remove warnings from the client test framework
- * Update version number to 0.9.5~beta
- * Update build-package script
- * Add check dependency in build-deps control file field
- * Create debian files for the new sflphone-client-gnome
- * [TICKET #1212] Add Replaces field in control files
- * [TICKET #1212] Fix manpages installation path
- * [TICKET #1212] Add maintainer scripts to create alternatives
- * [#1212] Update the manpages generation - edit preinst maintainer
- script
- * [#1212] Fix reference error in manpage
- * [#1212] Add missing files on the client side
- * [#1212] Fix debian docs files - no TODO file
- * [1212] Fix manpage creation problem
- * [#1220] Generate client-side glue files and marshaller at
- compilation time
- * [#1220] Generate server-side glue files at compilation time
- * [#1212] Change binary name to sflphone-client-gnome
- * [#1212] Update .gitignore to fit the new working tree
- * [#1220] Explicitly generate glue files before building the library
- * [#1220] Compile dbus directory before audio
- * [#1212] Create sflphone-common at the root of the repository
- * [#1212] Re-add pjproject
- * [#1212] Remove Makefile from repo
- * [#1220] Fix Makefile.am
- * [#1212] New working directory functional
- * [#1212] Update .gitignore
- * [#1212] Hack to make pjsip compile..
- * [#1220] Use non-installed binary for dbusxx-xml2cpp
- * [#1212] Add descriptive files, remove unuseful scripts from tools/
-
- [ Alexandre Savard ]
- * Restore speex codecs
- * add frame size for celt (#1143)
- * add framesize to codec, independant from audiolayer (#1143)
- * use codec frame size in rtp (#1143)
- * compute fixed_codec_framesize (#1143)
- * do not resample if not required (#1143)
- * add condition on resampling for decoder (#1143)
- * add a condition on bytesAvail == 0 from mic data
- * no maximum in rtp decode (#1143)
- * compute maximum for decoding (#1143)
-
- [ Emmanuel Milou ]
- * [#1146] Implement unitary tests on the client-side
-
- [ Alexandre Savard ]
- * use float instead of int to compute max nb of sample (#1143)
- * add nbSampleMax for unresampled data (#1143)
- * make thread sleep during 5 ms insead of 20 (#1143)
- * use unix usleep (#1143)
- * 50 usecond thread!!!!! (#1143)
- * try with the smallest compression (#1143)
- * use timer set at framesize (#1143)
-
- [ Emmanuel Milou ]
- * [#1161] Restore changelog version
-
- [ Alexandre Savard ]
- * Remove celt stuff
-
- [ Emmanuel Milou ]
- * [#1161] Update changelog
- * [#1220] Add Conflicts: sflphone in debian control files
- * [#1179] Add liblog4c3 runtime dependency
- * [#1212] FIx typo error in dependency list for itnrepid
- * [#1212] FIx .desktop file to point on the right exec
- * [#1212] Modify changelog replacing tag
-
- [ Sflphone Project ]
- * "[#1262] Updated changelogs for version 0.9.5-0ubuntu1~beta"
-
- [ Emmanuel Milou ]
- * [#1212] restore changelogs
-
- [ Sflphone Project ]
- * [#1262] Updated changelogs for version 0.9.5-0ubuntu1 Snapshot 2009-
- 04-27
-
- [ Emmanuel Milou ]
- * [#1212] restore changelogs
-
- [ Sflphone Project ]
- * [#1262] Updated changelogs for version 0.9.5-0ubuntu1~beta
-
- [ Emmanuel Milou ]
- * [#1212] restore changelogs
-
- [ Sflphone Project ]
-
- -- Sflphone Project <sflphone@mtl.savoirfairelinux.net> Mon, 27 Apr 2009 16:57:00 -0400
-
-sflphone-common (0.9.4-0ubuntu2) SYSTEM; urgency=low
-
- [ Alexandre Savard ]
- * Restore speex and GSM detection
-
- [ Emmanuel Milou ]
- * Fix bug #1090
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Wed, 8 Apr 2009 11:29:15 -0500
-
-sflphone (0.9.4-0ubuntu1) SYSTEM; urgency=low
-
- [ Emmanuel Milou ]
- * Integrate DBus-c++ and libiax2 in the main build system
- * Clean up in the working repository
- * Reorder hooks configuration panel
- * Protect case when no codecs are active
- * Fix some return values
- * Add unitary tests for the hook manager (premisces)
-
- [Yun Liu]
- * Update chinese translation
-
- [Sven Werlen]
- * Update german translation
-
- [Hussein Abdallah]
- * Update russian translation
-
- [Maxime Chambreuil]
- * Update spanish translation
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Fri, 3 Apr 2009 18:29:15 -0500
-
-
-sflphone (0.9.4-rc1) SYSTEM; urgency=low
-
- [ Emmanuel Milou ]
- * Fix bug while trying to hold/unhold several simultaneous call
- * Improve address book build system
- * Implement SIP url popup on incoming call
- * Improve GTK+ panel configuration
- [ Julien Bonjean ]
- * GTK+ client refactoring
- * GTK+ clean up
- * Address book improvment
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Fri, 27 Mar 2009 18:29:15 -0500
-
-sflphone (0.9.4-0beta1) SYSTEM; urgency=low
-
- [ Alexandre Savard ]
- * Display codec used during conversation on the GUI
- * Enable/disable STUN parameters at runtime
- * Refactor search bar use
- [ Emmanuel Milou ]
- * Build system fixes
- * Implement SIP re-invite
- * Implement IP to IP call
- [ Julien Bonjean ]
- * Integrate GNOME address book based on evolution data server
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Fri, 20 Mar 2009 18:29:15 -0500
-
-
-sflphone (0.9.3-0ubuntu3) SYSTEM; urgency=low
-
- [ Alexandre Savard ]
- * Both playback and record streams in PA_STREAM_CORKED (pulseaudio)
- * Use PLUGHW device for ALSA capture
- * Functional IAX and SIP recording for voicemail
- * Use the less CPU-consuming interpolator algorithm for resampling
- * Display in GTK GUI the codec used in conversation
- * GTK GUI use ASCII instread of utf-8
- * Add record menus in GTK GUI
- * Put on hold when dialing a new number
- * AccountID's are saved in the history
-
- [ Emmanuel Milou ]
- * Integrate DBUS C++, libiax2 in the git repository
- * Update website
- * Use libspeexdsp only if available on the system
- * Updated .gitignore file
-
- [Cyrille Béraud]
- * Account assistant manager improvment
- * Add an email request when creating a new account to receive voicemails
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Sat, 14 Feb 2009 13:29:15 -0500
-
-sflphone (0.9.3-0ubuntu2) SYSTEM; urgency=low
-
- [ Emmanuel Milou ]
- * Add compilation note in README
- * Use default ALSA plugin for capture
- * Fix the ALSA capture problem one more time
- * Clean up debug messages in dbus.c
- * Add libspeexdsp dependency
- * Remove implicit declaration compilation warnings
- * Fix links in the website, add release note
- * Change capture for the website front page
- * Add alsa devel dependency in build-depends control file field
- * Clean up, indentation, try to handle latency problems in iax/pulseaudio
- * Remove pjsip generated files from the repo
- * Use the previous declared curAlias function in accountwindow
- * Fix bug in history call duration when the call fails
- * Remove runtime warning in the GTK+ client
- * Add librsvg2-common dependency to load SVG under KDE
- * Refresh .gitignore
- * Update locales files + french translation
- * Add configuration panel for future noise reduction
- * Add configuration panel for audio record module
- * Daemon less verbose; accounts don't try to access STUn options anymore
- * Fix typo in configwindow
- * Add content in the official website
- * use a GTK_STOCK icon for the record button
- * Complete description text in the assistant manager
- * Add libtool flags in client configure.ac
- * Remove unuseful dependency (snd)
- * Fix SIP transfer problems
- * Remove previous version of PJSIP from the repo
- * Upgrade PJSIP to version 1.0.1
- * Add the new website source in the repository
- * Use libspeexdsp for silence detection only if available
-
- [ Loïc Faure-Lacroix ]
- * Ajout du logo gpl3
- * Ajout des images
- * Ajout de la section screenshot pour le site
- * Ajout du favicon dans le header
- * Modification des cartes
-
- [ Alexandre Savard ]
- * Clean up <speex/libspeexdsp>
- * Small cleanup
- * Save Wave fixed
- * Fix new call button when recording
- * libspeexdsp added
- * Recording: default home folder at startup
- * Minor changes to config window
- * IAX recording fixed
- * Set / get recording path, still need some GTK for client
- * AudioRecord file name format
- * Now recording in HOME folder
-
- [ Cyrille Béraud ]
- * Fix bug in reqaccount.c
-
- [ Maxime Chambreuil ]
- * Update spanish translation
-
- [Yun Liu ]
- * Update chinese translation
-
- [ Hussein Abdallah ]
- * Update russian translation
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Sat, 14 Feb 2009 13:29:15 -0500
-
-sflphone (0.9.3-0ubuntu1) SYSTEM; urgency=low
-
- * Remove debug
- * Join thread before leaving
- * Fix implicit declaration in reqaccount
- * Add REST code to build the request to server
- * Fix GValue initialization warnings
- * Update version number, fix implicit declaration, fix GTK markup
- warnings
- * Apply patch to create custom SIP account from our own server
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Fri, 06 Feb 2009 19:17:32 -0500
-
-sflphone (0.9.2-2ubuntu9) SYSTEM; urgency=low
-
- [ Alexandre Savard ]
- * Speex audio codec preprocessing initialization
- * peer hung up segmentation fault solved
- * Stop recording when transfering
- * Terminate only one call
- * Add isRecording() function
- * Fix call_icon GTK client
- * Fix SIPCallClose() function, recorded file now close properly
- * Function terminateSIPCall added in sipvoiplink and managerimpl
- * Fix thread destructor
- * setRecordingOption function implement in audiorecord
- * Record now implemented in Call class
- * Record interface complete (on hold erase previous recording)
- * Added recButton in client
- * Added: record button related icons
- * Record button added
- * Overload AudioRecord::recData to get mic and speaker data mixed
- * Recording now in audiortp::run() method
- * Audio recording working in AudioRTP: receiveSessionForSpeaker
- * Open/close a wave file when pulse audio stream start/stop
-
- [ Emmanuel Milou ]
- * Fix path for GTK+ icons; clean up
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Thu, 05 Feb 2009 18:27:53 -0500
-
-sflphone (0.9.2-2ubuntu8) SYSTEM; urgency=low
-
- [ Emmanuel Milou ]
- * Update changelogs
- * Fix bug in merge and in Makefile.am
- * Terminate only one call
- * Disable PJsip shutdown when changing STUN parameters
- * Function terminateSIPCall added in sipvoiplink and managerimpl
- * Add a timer to the alsa thread to not jam the CPU load
- * Fix bug in sipvoiplink.cpp
- * Clean shutdown of pulseaudio on quiting
- * Fix DTMF at first start with Pulseaudio
- * Remove zeroconf from the build system
- * Add a library manager + exception handling
- * Clean up in the working directory
- * Better handling of capture XRUNs
- * Restore mic adjust volume on ALSA layer
- * Protect device ALSA operation if not opened
- * Fix the switching layer bug
- * Use dynamic_cast<> to use audiolayer-specific methods
- * Open the audio devices only once at startup
- * Refactoring of the ALSA part
- * Functional plug-in manager
- * Use a C++ thread to handle tones and DTMF in ALSA
- * Restore IAXVoIPLink, restore Mutex
- * Make the plugins registering against the plugin manager
- * Migrate to 1->N relationship between voiplink and accounts
- * API plugin for registration
- * Use C++ thread in SIP, move everything in sipvoiplink
- * Complete singleton pattern for the plugin manager
- * Add -Wno-return-type compilation flag to remove warnings; Update
- version number in configure.ac
- * Add the dynamic loading for the plugin framework; integate unittest
-
- [ Yun Liu ]
- * Update rpm spec file
- * modify build package script and spec file for suse
-
- [ Alexandre Savard ]
- * Add audiorecorder plugin and testaudiorecorder
- * Add audio Recording class, edit global.h
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Wed, 04 Feb 2009 14:00:30 -0500
-
-sflphone (0.9.2-2ubuntu7) SYSTEM; urgency=low
-
- [ Emmanuel Milou ]
- * Update changelog to 0.9.2-6
- * Fix some dbus-glib implementation details on the client side
- * Init history after dbus initialization
- * Add error checking in useragent; Clean sipvoiplink
- * Prevent crash when trying to call an empty number
- * Set the volume of the playback stream to PA_VOLUME_NORM at startup
- * Fix GTK+ generic value double initialization
- * Fix jaunty control file dependency problems
- * Fix jaunty control file dependency problems
-
- [ Yun Liu ]
- * Fix bug ticket # 137
- * Tolerant to gsm library of OpenSuse 11
-
- [ Sven Werlen ]
- * Update german translation
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Fri, 23 Jan 2009 17:48:13 -0500
-
-sflphone (0.9.2-2ubuntu6) SYSTEM; urgency=low
-
- [ Emmanuel Milou ]
- * Migrate STUN configuration to the main config window
- * Update french translation
- * Other tiny memory leaks
- * Fix memory leak in sampleconverter.cpp
- * Generate packages from the release branch
- * update the build package script
- * modify the control files with architecture=any
- * Remove valgring uninitialized value
- * IAX and SIP use the same global variables to set account
- configuration ; fix broken code
-
- [ Maxime Chambreuil ]
- * Update spanish translation
-
- [ Hussein Abdallah ]
- * Update russian translation
-
- [ Yun Liu ]
- * Update translation files
- * Fix the bug when user uncheck the account which fails in the
- previous registration
- * Add stun error status
- * Fix bug ticket #143
- * Script for auto-install dependencies
- * Fix bug ticket #140
- * Fix bug ticket 141
- * Fix the reregister process when user change the details of an
- account
-
- -- Emmanuel Milou <manu@sulfur.inside.savoirfairelinux.net> Fri, 16 Jan 2009 18:19:05 -0500
-
-sflphone (0.9.2-2ubuntu5) SYSTEM; urgency=low
-
- * Fix memory leak in the pulseaudio callback
- * Update debian package generation script
- * Warnings removal in GTK+ client
- * Clean adjust volume method in alsalayer
- * Plug the sflphone playback volume control to the pulseaudio volume
- manager
- * Display the date in history according to the current locale
- * Generate the changelog according to the git commit messages
- * Complete header in chinese translation file
- * Use the right gpg key to sign the packages
- * add debian jaunty jackalope support
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Wed, 14 Jan 2009 21:17:20 -0500
-
-sflphone (0.9.2-2ubuntu4) SYSTEM; urgency=low
-
- [ Emmanuel Milou ]
- * add german translation
-
- [ Yun Liu ]
- * Fix GUI crash in Ubuntu8.10 64bit system
-
- -- Yun Liu <yun.liu@savoirfairelinux.com> Thu, 08 Jan 2009 13:08:51 -0500
-
-sflphone (0.9.2-2ubuntu3) SYSTEM; urgency=low
-
- [ Emmanuel Milou ]
- * The main thread synchronizes the ringtone thread
- * disable custom ringtone for the ALSA layer
- * Fix the Makefile.am in man directory, add a SEE ALSO section
-
- [ Yun Liu ]
- * Fix daemon crash caused by the previous patch ( for bug ticket #129)
-
- -- Yun Liu <yun.liu@savoirfairelinux.com> Tue, 06 Jan 2009 16:18:38 -0500
-
-sflphone (0.9.2-2ubuntu2) SYSTEM; urgency=low
-
- * Fix bug ticket #129
-
- -- Yun Liu <yun.liu@savoirfairelinux.com> Wed, 5 Jan 2009 15:54:53 -0500
-
-sflphone (0.9.2-2ubuntu1) SYSTEM; urgency=low
-
- * Migrate from eXosip library to pjsip
- * Add multiple SIP accounts support
- * Fix ringtones problems
- * Add a pulseaudio support
- * Improve audio quality with ALSA
- * Add chinese translation
- * Improve spanish translation
- * Migrate to a maintained C++ DBus bindings
- * Clean and improve the build system
- * Add build-dependency on Perl because we need pod2man to generate manpages
-
- -- Yun Liu <yun.liu@savoirfairelinux.com> Wed, 26 Nov 2008 09:47:53 -0500
-
-sflphone (0.9.1) unstable; urgency=low
- * Add a search tool in the history
- * Migrate some gtk_entry_new to sexy_icon_entry_new
- * Bug fix (Ticket #78): The voicemail password isn't displayed anymore in
- the history tab
- * Add the SIP registration expire value in the user file.
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Thu, 22 May 2008 11:14:25 -0500
-
-sflphone (0.9.0) unstable; urgency=low
- * Add history features
- * Call date
- * Call duration
- * Mouse events in the history tab
- * Smooth switch from the history tab to the calls tab
- * Remove most of GTK-Critical warnings
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Wed, 13 May 2008 16:58:25 -0500
-
-sflphone (0.9-2008-06-06) unstable; urgency=low
- * Audio bug correction: capture stopped after a few minutes of conversation
- with USB Plantronics sound card
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Tue, 06 May 2008 16:58:25 -0500
-
-sflphone (0.9-2008-05-06) unstable; urgency=low
- * Bug correction: account creation with the assistant
- * GTK+ warnings removal
- * libnotify warnings removal
- * Remove aliasing on the SFLphone logo
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Mon, 05 May 2008 16:58:25 -0500
-
-sflphone (0.9) unstable; urgency=low
- * Clean dependencies ( removal of libboost )
- * Several GTK improvement and updates
- -account window
- -configuration window
- * Migrate from GtkCheckMenuItem to GtkImageMenuItem
- * ALSA standard I/O transfers: MMAP instead of R/W
- * Fix speex audio quality
- * IAX2 protocol
- -Fix hold/unhold situation
- -Add on hold music
- * SIP protocol
- -Ringtone on incoming call
- -Fix transfer situation
- * Add desktop notification ( libnotify )
- * Improve the system tray icon behaviour
- * Improve registration error handling
- * Register/unregister from the account window takes effect without starting back SFLphone
- * Compilation warnings removal
- * Call history
- * Add an account configuration wizard
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Wed, 30 Apr 2008 16:58:25 -0500
-
-sflphone (0.8.2) unstable; urgency=low
- * Internationalization of the GTK GUI
- * English / French
- * STUN support
- * Slight modifications of the graphical interface ( tooltips, dialpad, ...)
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Fri, 21 Mar 2008 11:37:53 -0500
diff --git a/tools/build-system/launchpad/sflphone-daemon-video/debian/compat b/tools/build-system/launchpad/sflphone-daemon-video/debian/compat
deleted file mode 100644
index 7ed6ff82de6bcc2a78243fc9c54d3ef5ac14da69..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/sflphone-daemon-video/debian/compat
+++ /dev/null
@@ -1 +0,0 @@
-5
diff --git a/tools/build-system/launchpad/sflphone-daemon-video/debian/control b/tools/build-system/launchpad/sflphone-daemon-video/debian/control
deleted file mode 100644
index a42e722c2b84b89095f5e2775da18694b9ef1f6b..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/sflphone-daemon-video/debian/control
+++ /dev/null
@@ -1,20 +0,0 @@
-Source: sflphone-daemon-video
-Maintainer: SavoirFaireLinux Inc <julien.bonjean@savoirfairelinux.com>
-Section: gnome
-Priority: optional
-Build-Depends: debhelper (>= 7.0.50), libgcc1, autoconf, automake, libpulse-dev, libsamplerate0-dev, libccrtp-dev, libgsm1-dev, libspeex-dev, libtool, libdbus-1-dev, libasound2-dev, libopus-dev, libspeexdsp-dev, libexpat1-dev, libzrtpcpp-dev, libssl-dev, libgnutls-dev, libpcre3-dev, libyaml-cpp-dev, libboost-dev, libdbus-c++-dev, libsndfile1-dev, libavcodec-dev, libavformat-dev, libswscale-dev, libavdevice-dev, libavutil-dev, libudev-dev, libpjproject-dev, libsrtp-dev, libjack-dev, libvpx-dev
-Standards-Version: 3.7.3
-
-Package: sflphone-daemon-video
-Priority: optional
-Architecture: any
-Depends: ${shlibs:Depends}, ${misc:Depends}, libavcodec56 (>= 6:11~beta1) | libavcodec-extra-56 (>= 6:11~beta1) | libavcodec54 | libavcodec-extra-54, libavdevice55 (>= 6:11~beta1) | libavdevice53 | libavdevice-extra-53, libavformat56 (>= 6:11~beta1) | libavformat54 | libavformat-extra-54, libswscale3 (>= 6:11~beta1) | libswscale2 | libswscale-extra-2, libavutil54 (>= 6:11~beta1) | libavutil52 | libavutil-extra-52
-Replaces: sflphone, sflphone-common-video
-Conflicts: sflphone-common, sflphone-daemon, sflphone-data
-Provides: sflphone-common-video
-Homepage: http://www.sflphone.org
-Description: SIP and IAX2 compatible softphone - Core with video support
- SFLphone is meant to be a robust enterprise-class desktop phone.
- SFLphone is released under the GNU General Public License.
- SFLphone is being developed by the global community, and maintained by
- Savoir-faire Linux, a Montreal, Quebec, Canada-based Linux consulting company.
diff --git a/tools/build-system/launchpad/sflphone-daemon-video/debian/copyright b/tools/build-system/launchpad/sflphone-daemon-video/debian/copyright
deleted file mode 100644
index 7b3bdc5eebd9324f7616980b85dd03f0354d32a5..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/sflphone-daemon-video/debian/copyright
+++ /dev/null
@@ -1,28 +0,0 @@
-This package was debianized by Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> on
-Fri, 3 Apr 2009 09:47:53 -0500.
-
-It was downloaded from the git repository of SFLphone: git://sflphone.org/git/sflphone.git
-
-Upstream Author: Emmanuel Milou <emmanuel.milou@savoirfairelinux.com>
-
-Copyright:
-
-Savoir-faire Linux Inc.
-
-License:
-
-This software is copyright (c) 2004-2016 Savoir-faire Linux Inc.
-
-You are free to distribute this software under the terms of
-the GNU General Public License version 3.
-On Debian systems, the complete text of the GNU General Public
-License can be found in the file `/usr/share/common-licenses/GPL'.
-
-This program is distributed in the hope that it will be useful,
-but WITHOUT ANY WARRANTY; without even the implied warranty of
-MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
-GNU General Public License for more details.
-
-You should have received a copy of the GNU General Public License
-along with this program; if not, write to the Free Software
-Foundation, Inc., 51 Franklyn St, Fifth Floor, Boston, MA 02110-1301, USA.
diff --git a/tools/build-system/launchpad/sflphone-daemon-video/debian/cron.d b/tools/build-system/launchpad/sflphone-daemon-video/debian/cron.d
deleted file mode 100644
index d11e61177739b56bce3aac6de6483b48e797a258..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/sflphone-daemon-video/debian/cron.d
+++ /dev/null
@@ -1,4 +0,0 @@
-#
-# Regular cron jobs for the sflphone package
-#
-0 4 * * * root sflphone_maintenance
diff --git a/tools/build-system/launchpad/sflphone-daemon-video/debian/dirs b/tools/build-system/launchpad/sflphone-daemon-video/debian/dirs
deleted file mode 100644
index 93e7926139602286c03695199324cd3d2bcdcb39..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/sflphone-daemon-video/debian/dirs
+++ /dev/null
@@ -1,9 +0,0 @@
-usr/bin
-usr/lib
-usr/lib/sflphone
-usr/share/applications
-usr/share/dbus-1/services
-usr/share/sflphone/ringtones
-usr/share/locale
-usr/share/doc
-usr/share/man
diff --git a/tools/build-system/launchpad/sflphone-daemon-video/debian/docs b/tools/build-system/launchpad/sflphone-daemon-video/debian/docs
deleted file mode 100644
index 0f8394ba76ddb960f65b85e14e6519f6dd462aa5..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/sflphone-daemon-video/debian/docs
+++ /dev/null
@@ -1,5 +0,0 @@
-NEWS
-README
-TODO
-ChangeLog
-AUTHORS
diff --git a/tools/build-system/launchpad/sflphone-daemon-video/debian/manpages b/tools/build-system/launchpad/sflphone-daemon-video/debian/manpages
deleted file mode 100644
index 0b7e5f1c26cb4203d1b4d6dcfd0c761be11e9a09..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/sflphone-daemon-video/debian/manpages
+++ /dev/null
@@ -1 +0,0 @@
-debian/sflphone-daemon/usr/share/man/man1/sflphoned.1
diff --git a/tools/build-system/launchpad/sflphone-daemon-video/debian/postinst b/tools/build-system/launchpad/sflphone-daemon-video/debian/postinst
deleted file mode 100644
index 5ac10f4f256c9f3d9fa280eabd191c3d750f2a0c..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/sflphone-daemon-video/debian/postinst
+++ /dev/null
@@ -1,56 +0,0 @@
-#!/bin/bash
-# postinst script for sflphone-common
-#
-# see: dh_installdeb(1)
-
-# Script to copy and move, if exists, configuration file sflphonedrc and history in the XDG directory
-# Freedesktop specifications: http://standards.freedesktop.org/basedir-spec/latest/
-
-set -e
-
-INST_CONFIG="$HOME/.sflphone/sflphonedrc";
-INST_DATA="$HOME/.sflphone/history";
-INST_CACHE="$HOME/.sflphone/sfl.pid";
-
-NEW_INST_CONFIG=
-NEW_INST_DATA=
-NEW_INST_CACHE=
-
-# Set the XDG CONFIG directory to the default one or to the path set in the environment variable
-if [ -z $XDG_CONFIG_HOME ]; then
- NEW_INST_CONFIG=$HOME"/.config/sflphone/"; # This is the standard path
-else
- NEW_INST_CONFIG=$XDG_CONFIG_HOME;
-fi;
-
-# Set the XDG DATA directory to the default one or to the path set in the environment variable
-if [ -z $XDG_DATA_HOME ]; then
- NEW_INST_DATA=$HOME"/.local/share/sflphone/"; # This is the standard path
-else
- NEW_INST_DATA=$XDG_DATA_HOME;
-fi;
-
-# Move the configuration file
-if [ -f $INST_CONFIG ] ; then
- echo "Moving the configuration file into $NEW_INST_CONFIG directory";
- if [ ! -d $NEW_INST_CONFIG ]; then
- mkdir $NEW_INST_CONFIG;
- fi
- mv $INST_CONFIG $NEW_INST_CONFIG;
-fi
-
-# Move the history
-if [ -f $INST_DATA ] ; then
- echo "Moving the history file into $NEW_INST_DATA directory";
- if [ ! -d $NEW_INST_DATA ]; then
- mkdir $NEW_INST_DATA;
- fi
- mv $INST_DATA $NEW_INST_DATA;
-fi
-
-# Remove the directory
-# rmdir $HOME"/.sflphone";
-
-echo "You may remove the $HOME/.sflphone, the application won't use it anymore, but the XDG directories instead. Thank you.";
-
-exit 0
diff --git a/tools/build-system/launchpad/sflphone-daemon-video/debian/postrm b/tools/build-system/launchpad/sflphone-daemon-video/debian/postrm
deleted file mode 100644
index e6107444fa259e7d87e9d0d0fcfff80ca8ff144d..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/sflphone-daemon-video/debian/postrm
+++ /dev/null
@@ -1,34 +0,0 @@
-#!/bin/sh
-# postrm script for sflphone
-#
-# see: dh_installdeb(1)
-
-set -e
-
-# summary of how this script can be called:
-# * <postrm> `remove'
-# * <postrm> `purge'
-# * <old-postrm> `upgrade' <new-version>
-# * <new-postrm> `failed-upgrade' <old-version>
-# * <new-postrm> `abort-install'
-# * <new-postrm> `abort-install' <old-version>
-# * <new-postrm> `abort-upgrade' <old-version>
-# * <disappearer's-postrm> `disappear' <overwriter>
-# <overwriter-version>
-# for details, see http://www.debian.org/doc/debian-policy/ or
-# the debian-policy package
-
-if [ "$1" = "purge" ]
-then
-
- # remove the user config file
- rm -f $HOME/.sflphone/sflphonedrc
-
-fi
-
-# dh_installdeb will replace this with shell code automatically
-# generated by other debhelper scripts.
-
-#DEBHELPER#
-
-exit 0
diff --git a/tools/build-system/launchpad/sflphone-daemon-video/debian/preinst b/tools/build-system/launchpad/sflphone-daemon-video/debian/preinst
deleted file mode 100644
index 6d04e97b452606720e35f07523b60c32ad7f9d6b..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/sflphone-daemon-video/debian/preinst
+++ /dev/null
@@ -1,16 +0,0 @@
-#!/bin/sh
-# postrm script for sflphone
-#
-# see: dh_installdeb(1)
-
-set -e
-
-package=sflphone
-
-case "$1" in
- install|upgrade)
- # Clear the old dbus-c++ and iax2 if presents
- ;;
-esac
-
-exit 0
diff --git a/tools/build-system/launchpad/sflphone-daemon-video/debian/rules b/tools/build-system/launchpad/sflphone-daemon-video/debian/rules
deleted file mode 100755
index 03eb67e044de504a6b8e87a4e828732eff5daea1..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/sflphone-daemon-video/debian/rules
+++ /dev/null
@@ -1,91 +0,0 @@
-#!/usr/bin/make -f
-# -*- makefile -*-
-# Sample debian/rules that uses debhelper.
-# This file was originally written by Joey Hess and Craig Small.
-# As a special exception, when this file is copied by dh-make into a
-# dh-make output file, you may use that output file without restriction.
-# This special exception was added by Craig Small in version 0.37 of dh-make.
-
-# Uncomment this to turn on verbose mode.
-#export DH_VERBOSE=1
-export DH_OPTIONS
-
-package=sflphone-daemon-video
-
-CXX = g++-4.0
-CFLAGS = -Wall -g
-DEB_INSTALL_MANPAGES_sflphone_daemon_video = sflphoned.1
-
-configure: configure-stamp
-configure-stamp:
- dh_testdir
- # Add here commands to configure the package.
- # build iax and opendht with contrib since they are not packaged
- cd contrib && mkdir -p native && cd native && ../bootstrap && make .iax && make .dht && cd ../..
- ./autogen.sh
- ./configure --prefix=/usr
- touch configure-stamp
-
-#Architecture
-build: build-arch
-
-build-arch: build-arch-stamp
-build-arch-stamp: configure-stamp
-
- # Add here commands to compile the arch part of the package.
- $(MAKE)
- touch $@
-
-clean:
- dh_testdir
- dh_testroot
- rm -f build-arch-stamp configure-stamp
- # Add here commands to clean up after the build process.
- [ ! -f Makefile ] || $(MAKE) distclean
-
-ifneq "$(wildcard /usr/share/misc/config.sub)" ""
- cp -f /usr/share/misc/config.sub config.sub
-endif
-ifneq "$(wildcard /usr/share/misc/config.guess)" ""
- cp -f /usr/share/misc/config.guess config.guess
-endif
- dh_clean
-
-install: install-arch
-
-install-arch:
- dh_testdir
- dh_testroot
- dh_clean -k -s
- dh_installdirs -s
- # Add here commands to install the arch part of the package into
- # debian/tmp.
- $(MAKE) DESTDIR=$(CURDIR)/debian/$(package) install
- rm -rf $(CURDIR)/debian/$(package)/usr/include
- dh_install -s
-
-binary-common:
- dh_testdir
- dh_testroot
- dh_installchangelogs ChangeLog
- dh_installdocs
- dh_installexamples
-# dh_installman
- dh_link
- dh_compress
- dh_fixperms
- dh_makeshlibs
- dh_installdeb
- dh_shlibdeps
- dh_gencontrol
- dh_md5sums
- dh_builddeb
-
-# Build architecture dependant packages using the common target.
-binary-arch: build-arch install-arch
- $(MAKE) -f debian/rules DH_OPTIONS=-s binary-common
-
-override_dh_strip:
-
-binary: binary-arch
-.PHONY: build clean binary-arch binary install install-arch configure override_dh_strip
diff --git a/tools/build-system/launchpad/sflphone-daemon/debian/changelog b/tools/build-system/launchpad/sflphone-daemon/debian/changelog
deleted file mode 100644
index a6cb2908c9264387a41e265172f460de5e43b4a8..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/sflphone-daemon/debian/changelog
+++ /dev/null
@@ -1,3585 +0,0 @@
-sflphone-daemon (1.1.0-rc20120607~ppa1~SYSTEM) SYSTEM; urgency=low
-
- ** SNAPSHOT 1.1.0-rc20120607~ppa1~SYSTEM **
-
- * * #12071: audiopreferences: fix make check
- * * #12071: cleanup
- * #12071: Lower expat dependency version
- * * #12085: alsa: fix ringtone update bug, cleanup
- * #12071: Fix pulseaudio compilation error
- * #12071: Make pulseaudio optional at configuration time
- * * #12091: daemon: issue warning if falling back to ALSA
- * video: fixed make check
- * *#12085: alsa: don't segfault on snd_pcm_avail_update error
- * #12070: Add --without-pulse option
- * * #12055: video: fix build for older libav
- * [ #11886 ] cleanup
- * [ #11886 ] Add basic reverse peer naming support
- * [ #12008 ] Implement GUI part
- * * #12012: video: fix some regressions
- * * #12002: yamlparser: don't wipe out config if going from normal
- build to --enable-video
- * [ #12008 ] Add ConfigurationManager::getRingtoneList()
- * * #12002: video: fix config file serialization/deserialization
- * * #11987: managerimpl: fix bugs in conference when removing
- participants
- * * #11987: manager: fixed transfer from conference
- * * #11987: mainbuffer: cleanup logging
- * * #11979: pulse: fixed mismatched device list
- * * #11971: audiolayer: fix bugs with getDeviceList
- * * #11960: manager: validate conference earlier when processing
- participants
- * * #11960: manager: fixed segfault on transfer from conference
- * * #11966: IP2IP: make alias consistently IP2IP
- * * #11965: sipvoiplink: add more error checking in SDP negotiation
- * * #11964: mainbuffer/ringbuffer: cleanup API
- * sdp: remove unused variable warning
- * * #11941: video: fix deprecated libav_api warnings
- * * #11949: pulselayer: fix bug in getDeviceList
- * video: whitespace fixes
- * * #11951: video: fixed threading issues for ucommon Thread
- * [#11848] Properly disable testPulseConnect
- * [#11848] Disable pulseConnect test
- * sdp: cleanup
- * sdp: cleanup
- * * #11860: mainbuffer: remove dead and/or buggy code
- * * #11851: sdp: fixed gcc type narrowing warnings
- * * #11851: audiostream: fixed gcc type narrowing warnings
- * * #11841: don't put code with side effects in assert()
- * * #11840: audiortp: remove some global symbols/variables
- * * #11828: audiofile: fix broken build
- * * #11828: audioloop: don't shadow sampleRate variable in derived
- classes
- * * #11818: gnome: stop daemon on SIGTERM, SIGINT or SIGHUP
- * * #11499: daemon should also quit gracefully on SIGHUP
- * * #11813: daemon configure should fail if expat is not installed
- * #10304: updatePlaybackScale dbus method uses int 32 bit for size and
- position (allow for 24 days long recording playback)
- * #10304: Add time lable for seekslider
- * * #11780: sip: don't use abort or leak calls on error and don't
- restrict SDP size to 1000 bytes in transaction_request_cb
- * * #11735: daemon: added timestamp start to call details
- * * #10304: historyitem: added operator > defined in terms of operator
- <
- * * #11252: historyitem: added missing unistd.h header
- * * #11252: daemon: removed deprecated zrtp code
- * * #11728: yaml: check that nodes are valid before using them.
- * * #10797: send DTMF over RTP as per RFC2833
- * * #11706: managerimpl: added unsetCurrentCall method
- * * #11698: daemon: fix build for c++11
- * * #10304: historyitem: file_exists need not be a member method
- * * #11499: managerimpl: don't crash if signal and dbus try and finish
- the manager at the same time
- * #10304: Prevent from storing removed files in history
- * * #11499: daemon: Exit cleanly on SIGINT or SIGTERM
- * * #10226: audiocodecfactory: use array instead of vector for codec
- name lookup
- * * #10226: audiocodecfactory: make codec loading stricter
- * #10304: RCecale positions and size values for playback recording
- * #11530: Make sure that only appropriate configuration option are
- parsed for IP2IP calls
- * * 11480: video: disabled by default
- * * #11459: history: protect historyitems vector with mutex
- * * #11448: fix video preferences for empty camera list
- * #10304: Implemented playback seek in gnome client
- * * #11269: video: fix codec per account management
- * #10304: Implemented playback scale in gnome client
- * Fix includes for gcc 4.7
- * * #11269: make clearer distinction between codecs and audiocodecs
- * * #11269: merged master into video
- * * #10296: managerimpl: more usage of getCallFromCallID
- * * #10296: verify that calls exists before trying to join them in a
- conference
- * * #11208: bump version numbers for release 1.1.0
- * managerimpl: rename ManagerImpl::serialize/unserialize ->
- join_string/split_string
- * Fix warnings in resampler test
- * * #10732: sipvoiplink: fix code that validates IP address
- * Add historyChanged signal, better than managing it client side
- * #10795: fix sipaccount deserialisation broken
- * #10736: implement getConferenceId dbus method given a call id
- * #10736: do not use iterator in daemon when joining conferences
- * #10736: Fix joining conferences in daemon
- * * #10736: gnome: fix crash on restart with active conference
- * managerimpl: removed unused pulselayer.h header
- * Save history everytime it change, prevent the file never to be saved
- in some senario (SIG, crash, ASSERT, etc)
- * * #10320: manager: check that participants are unique before joining
- * #10335: Add a noise suppressor for incoming rtp streams
- * * #10322: sip: registration state should not be always set to
- ErrorAuth on error
- * #10220: Fix recording thread does not exit when hanging up while
- recording
- * * #9903: fix includes for new ccrtp
- * #10230: Get back default mainbuffer sampling rate to 8kHz, no need
- of decoding noise suppressor
- * #10230: Use a different samplerate converter for rtp encoding and
- decoding
- * * #9903: create DynamicPayloadFormat on stack, initialize earlier
- * * #9903: audiorecorder: initialize buffer to silence, not random
- data
- * #10230: Test for triangular and sine signals
- * #10230: Add resampling unit test
- * * #10230: DTMF sample rate should come from main buffer, it should
- not be hardcoded
- * * #10213: audiolayer: create samplerateconverter on the stack
- * * #10213: audiolayer: cleanup
- * * #10213: increase resample buffer size, and check output size when
- resampling
- * * #10095: sipvoiplink: check pointers before using them
- * #9981: IP2IP calls based on ip address instead of sip:
- * * #10213: speex codecs should initialize their own parameters
- * * #10213: account: removed redundant cast
- * * #9832: removed extra printf
- * * #10172: include -sflphone in recording file name
- * * #9832: cleanup logging in tests
- * * #10096: srtp: use vectors to simplify key/salt manipulation
- * #10096: add case for non-srtp calls
- * [ #10121 ] Sync the KDE with daemon, fix a few issues and implement
- a recorded call player
- * #9980: make keep registration optional as there is different
- behavior on different registrar
- * #10096: use c++ arrays to store keys in srtp sesssion
- * * #10018: renamed registration related keys in dbus
- * #10096: Fix onhold/offhold srtp
- * * #8586: fixed make distcheck
- * * #9832: logger: don't hide logging if NDEBUG is present
- * #10096: Reinit crypto context when required on INVITE request
- * * #10111: Fixes segfault on empty config file
- * #100096: Set in/out queue crypto context at initialization, not when
- starting the thread
- * #10096: Update srtp key generation when holding/unholding a call
- * * #9831: logger: removed extraneous carriage-return character
- * * #10095: sipvoiplink: validate pointers before using them
- * * #10094: renamed config/config.{h,cpp} config/sfl_config.{h,cpp}
- * * #10090: fix segfault in transaction_state_changed_cb
- * * #9832: audio_rtp_record_handler: cleanup logging
- * * #9832: pulse: cleanup logging
- * * #9832: cleanup logging
- * * #9832: cleanup logging
- * * #9832: dbus: fix logging
- * * #9832: config: cleanup logging
- * * #9832: remove unused header
- * * #9832: manager: fix logging
- * * #9832: audio: fix logging
- * * #9832: audio: fix logging
- * * #9832: zrtp: cleanup logging
- * * #9832: AudioZRTPSession: cleanup logging
- * * #9832: AudioSRTPSession: fix logging
- * * #9832: cleanup logging
- * * #9832: AudioRtpSession: cleanup logging
- * * #9832: AudioRtpFactory: cleanup logging
- * * #9832: codecs: fix logging
- * * #9832: alsa: fix logging
- * * #9832: audio: clean up logging
- * * #9832: AudioRecord: cleanup logging
- * * #9832: Fix logging in Manager
- * * #9832: new logging macros
- * * #9979: ulaw: fixed unused var warning
- * * #10039: sipvoiplink: use references to avoid unnecessary parameter
- validation
- * * #10039: Fixed segfault on failed registration
- * * #9979: codecs: fixed unused variable warnings
- * * #9979: Don't do runtime assertions on data.
- * * #10016: SDP: removed verbose debuggin
- * #10016: Crypto context deletion are now managed inside the library
- * * #10016: srtp: cleanup
- * * #10016: SDES: fix uninitialized value bug, use const char*
- * * 100016: don't double free crypto contexts, and don't improperly
- copy CryptoSuiteDefinitions
- * * #100016: cleanup crypto contexts in audio_srtp_session
- * * #9979: removed unused methods from audicodec
- * * #9979: ulaw: normalize types
- * * #9979: cleanup
- * * #9979: Alaw: cleanup
- * * #9979: removed duplicate/superfluous code and type issues from
- g722
- * * #9979: AudioRtpRecord: let AudioRtpRecord handle fadeIn internally
- * #9980: Fix registration timer and transport shutdown on 401, default
- registration timer to 3600
- * * #9979: use std::tr1::array instead of plain array for audio
- buffers
- * * #9969: set loose routing param when creating route set
- * #9975: Fix account registration status display
- * * #9969: SIP: initialize body earlier
- * * #9969: sip: get received and rport fields if present in OK
- * * #9968: fixed segfault in transaction callback
- * #9898: make sure account are unregistered when sflphone quit, add
- timeout on pending transaction
- * #9910: fix contact header in outgoing request if via parameter are
- present
- * * #9910: SIP: use rport from VIA header if present
- * * #9910: SipTransport: pass parameters by const reference
- * #9910: fix sending call with new transport
- * yaml: remove verbose debug messages
- * * #9911: sipvoiplink: fixed "unused variable" warning
- * #9910: create new udp transport to fix registration failure with 606
- error & received parameter
- * * #9910: SIP: use pjsip error codes instead of magic numbers
- * * #9911: SIP Transports must be cached by IP:port
- * * #9910: SIP: cleanup
- * * #9905: SipTransport: address has to stay on stack to be valid
- * #9910: Update parse received parameter on 606 registration error
- * * #9911: simplify network manager state reporting
- * #9902: Fix SIPTest for IP to IP call
- * #9911: Fix network manager crashes
- * #9902: Move logic for ip2ip call in SIPVoIPLink
- * #9902: Move logic for ip2ip call in SIPVoIPLink
- * * #9910: fix 606 error code nomenclature
- * * #9905: fixed address initialization in createUdpTransport
- * * #9903: cleanup
- * #9902: Log failure cause when new outgoing call fail
- * * #9898: properly initialize ports
- * #9898: Unregister account when leaving sflphone
- * iax: create iaxvoiplink on stack
- * account: removed unused methods
- * * #9847: don't use assertions for input coming from DBus
- * * #9897: audiorecord cleanup
- * * #9897: audiorecord: cleanup, removed unused methods
- * #9897: Initialize and fallback recording path in home directory if
- not valid
- * * #9871: SipTransport: hide more implementation
- * * #9871: SipTransport: refactor SIP transport creation
- * * #9871: disable STUN for account if STUN setup failed
- * * #9847: check pointer before using it
- * #9871: Fallback on normal upd transport when stun resolution fails
- * Revert "#9871: Fallback on normal upd transport when stun resolution
- fails"
- * #9871: Fallback on normal upd transport when stun resolution fails
- * pulse: cleanup
- * * #9847: removed outdated README file
- * * #9847: use references instead of pointers where possible
- * * #9847: pass call by reference where possible
- * * #9847: audiolayer: fixed typo
- * * #9847: SIPVoipLink: gracefully handle invalid pointers
- * * #9847: check that transport is initialized
- * * #9847: SDP: avoid buffer overflow
- * * #9847: fixed segfault on bad call invite
- * * #9847: SDP: don't use assertions for runtime errors
- * * #9847: handle invalid remote session gracefully
- * * #9851: fixed segfault on stun socket cleanup
- * * #8586: fixed warnings
- * * #9849: added missing sstream header
- * #9623: add required TLS certificates for testing purpose
- * #9623: fixed tls registration
- * #9623: fix storing tls port in config for normal account
- * #9623: Allow all account to change tls listener port (not only
- IP2IP)
- * #9623: Allow for changing interface / port for tls transport
- * #9623: Open TLS listener on selected interface
- * #9833: remove unused debug
- * #9833: handlingEvents_ must be initialized to true when starting iax
- thread
- * #9830: Remove create_route_set from sipvoiplink
- * #9831: Fix sip transport port number
- * #9830: move sip header parsing function in sip_utils
- * Revert "* #8586: don't restore and save test files"
- * * #8586: fixed make distcheck
- * #9623: fixed tls registration
- * * #8586: don't restore and save test files
- * * #8586: refactored yaml code
- * #9623: fix storing tls port in config for normal account
- * #9623: Allow all account to change tls listener port (not only
- IP2IP)
- * #9623: Allow for changing interface / port for tls transport
- * #9623: Open TLS listener on selected interface
- * * #8586: added missing tests
- * #9833: remove unused debug
- * #9833: handlingEvents_ must be initialized to true when starting iax
- thread
- * #9830: Remove create_route_set from sipvoiplink
- * #9831: Fix sip transport port number
- * #9830: move sip header parsing function in sip_utils
- * * #8977: removed unnecessary AC_CANONICAL macros from configure.ac
- * * #8977: use actual PJSIP linking flags from pjproject/build.mak
- * * #9774: sipvoiplink's destructor should not be public
- * dbus: cleanup
- * * #9774: make sure sipvoiplink is destroyed before accounts are
- unloaded
- * * #9777: don't use deprecated auto_ptr
- * * #9778: removed AC_CHECK_FUNCS calls
- * * #9782: fix warnings in tests
- * * #9782: sip/sdp: fix emptiness checks
- * * #9782: sdes_negotiator: fix iterator usage and set dangling
- pointers to 0
- * * #9782: initialize all vars in iaxvoiplink
- * * #9782: use fstreams instead of fscanf
- * * #9782: yamlnode: fixed iterator usage
- * * #9782: yamlemitter: fix iterator usage
- * * #9782: yamlnode: make some methods const
- * * #9782: initialize all member vars in constructor
- * * #9782: Tone::interpolate should be const
- * * #9782: mainbuffer: get rid of unused vars
- * * #9782: GainControl::limit should be const
- * * #9782: fix ARRAYSIZE check
- * * #9782: use nanosleep instead of usleep
- * * #9782: fixed "inefficient emptiness test" cppcheck warning
- * * #9782: initialize dcblockers vars in constructor
- * * #9779: dropped CELT support
- * * #9750: moved sfl_data_format.h -> sfl_types.h
- * * #9750: refactored global.h
- * * #9736: restored command line options to daemon
- * tests: cleanup
- * * #8586: make distcheck was missing a header
- * tests: cleanup
- * * #9731: use all caps for application-wide constants
- * tests: cleanup
- * * #9730: cc++: enforce better checks in headers
- * * #9730: builds against libccrtp1
- * * #9572: sipvoiplink: fixed typo
- * * #9572: fixed threading issues with ccrtp2
- * * #9572: manager: pass config filename by const reference
- * * #9572: Replace utilspp singleton implementation
- * * #9571: regenerated config.{guess,sub} file to fix FTBFS on
- armel/armhf.
- * * #9665: siptransport: fixed udp_transport_start calls
- * #9620 Add test SIP account in configuration sample (test/sflphoned-
- sample.yml)
- * * #9641: audiortp: Fixed CryptoContext management
- * * #9641: fixed another memory leak in audio_srtp_session
- * * #9641: audiosrtpsession: fixed memory leak, simplified memory
- management
- * * #9641: avoid dynamic memory allocs/raw pointer usage in audio rtp
- stack
- * * #9641: get rid of getType/RtpMethod logic
- * fixed typo
- * #9572: make sflphone compile with libccrtp 2
- * * #9490: fixed registration state change callback that was crashing
- client
- * * #9547: fixed warnings in SipTransport header
- * #9547: Add SipTransport class
- * #9547: Extract all the transport layer from SIPVoIPLink to new
- SipTransport Class
- * #9547: Destroy the STUN resolver in Transport shutdown
- * sipvoiplink: removed erroneous FIXME
- * sipvoiplink: cleanup
- * #9547: Destroy the STUN resolver if server name change
- * sipvoiplink: fix warning about variable shadowing
- * #8320: Rename declared exception to avoid parameter shadowing
- * #8320: Send signal to client on stun failure
- * #8320: Use the same API for all transport creation (UDP, STUN, TLS)
- * * #9509: use vector for credential info
- * * #9508: fixes segfault in manager by changing order in which
- destructors are called
- * #8320: add dbus signal for stun failure
- * #8320: Use two different variables for status and return statement
- in stun's on_status_cb
- * * #9490: removed resolve_once parameter that was causing a segfault
- * #8320: make the retransmission callback to be rescheduled on error
- * HookPreference: cleanup
- * daemon: hookpreference: cleanup
- * iaxvoiplink: terminate() doesn't have to be virtual
- * sipvoiplink: functions need not be static if they are in an
- anonymous namespace
- * * #9037: moved CHECK macro into separate header
- * * #9037: cleanup error handling/checking in video threads
- * * #9037: video: cleanup
- * * #9037: only signal receiving_video_event for rtp sessions
- * * #9037: shared memory moved out of video_receive_thread
- * * #9381: daemon: fixed make check for video
- * * #9381: YAML_LIBS must be explicitly set in AC_SEARCH_LIBS macro
- * * #9381: reverted yaml check
- * * #9381: fix celt plugin compilation on fedora
- * * #9381: use PKG_CHECK_MODULES to test for yaml
- * * #9381: use autoconf macros and AC_SEARCH_LIBS
- * * #9381: use AC_SEARCH_LIBS, AC_CHECK_LIB
- * ringtonetest: cleanup
- * configurationtest: cleanup
- * instantmessagingtest.cpp: cleanup
- * mainbuffertest: cleanup
- * tests: cleanup
- * #8320: Make sure stun keep alive is enabled
- * call: push answer logic into call classes
- * sipaccount: simplify IP2IP code
- * sipaccount: avoid segfault if sipaccount is NULL
- * sipaccount: cleanup
- * #8084: Fix get sip header segfault when stun transport selected
- * * #9037: created shared_memory class
- * #8084: Init stun port with default valueas defined by RFC 3489
- * #9046: Move IP2IP_PROFILE global definition inside SIPAccount class
- * #9045: fix Changing the account expire is not taken applied in
- daemon
- * vidoe_receive_thread: cleanup
- * * #8968: suppress unusedFunction warnings for functions that are
- actually used
- * * #8821: fixed unit tests
- * #8821: Renamed account map keys for consistency
- * * #8968: audiorecord: added debug, clarified wave header creation
- * * #8968: added debug message to get rid of "unused struct member"
- warning
- * * #8968: manager: create History on the stack
- * * #9026: sfl::InstantMessaging is now a namespace
- * * #8698: managerimpl: removed unused method isWaitingCall
- * * #9008: don't include yaml headers in serializable.h
- * * #9008: cleanup account map initialization
- * refactor accountmap initialization
- * #9020: fix config file not generated when no account created
- * * #8968: audiorecord: removed unused getSndSamplingRate
- * * #8968: config: removed getConfigTreeItemIntValue
- * * #8968: recordable: removed unused getRecFileId
- * * #8968: removed unused Codec::getMimeType method
- * * #8968: manage lifetime of IMModule with auto_ptr
- * * #8968: history: removed unused method
- * * #8968: managerimpl: removed unused method
- * * #8968: config: removed unused methods
- * * #8968: managerimpl: removed unused getConfigBool/Int methods
- * * #8968: networkmanager: cleanup
- * * #8968: managerimpl: removed unused getConfig
- * * #8968: managerimpl: Manage telephoneTone_ with auto_ptr.
- * * #8968: history: fix memory leak upon exception
- * * #8968: AudioFile: initialize filepath earlier
- * * #8968: audiofile: fix memory leak on exception
- * * #8968: audiocodec: removed unused getChannel method
- * * #8968: use auto_ptr for dtmfKey
- * * #8968: use vector instead of dynamically allocated int array
- * * #8968: sdp.h: pass paramter by reference
- * * #8968: sipvoiplink: avoid C-style pointer casting
- * * #8968: yaml: avoid C-style pointer casts
- * * #8968: pulselayer: avoid C-style pointer casting
- * * #8968: recordable: removed unused getRecordingSmplRate method
- * * #8968: alsalayer: use preincrement for iterators
- * * #8968: config: removed unused method saveConfigTree
- * * #8968: mainbuffer: preincrement iterators
- * * #8977: history: added #include <fstream>
- * * #8968: don't leak memory on exception
- * * #8968: pulselayer: avoid C-style pointer casting
- * * #8968: Member variables must be initialized in AudioSrtpSession
- constructor
- * * #8968: fix potential memory leak in audiorecord
- * * #8968: Pass function parameter 'item' by const reference.
- * * #8969: fixed memory leaks in sdes_negotiator
- * video: cleanup
- * * #8940: removed video test source for now
- * #8763 Fix doxygen generation
- * #8763 Generate Doxygen with Hudson
- * * #8940: videosendthread: cleanup
- * sipvoiplink: cleanup
- * fileutils: cleanup
- * #8335 Fix default transport initialization on 5062, 5064
- * #8762: update mute for mic only, fix remove slide for pulseaudio
- * #8672: Add linear to decibel conversion functions in audio layer
- * #8672: Implement audio gain management in pulseaudio
- * #8671: Move audio gain management in audiolayer
- * * #8542: create symbolic link properly
- * * #8613: make check should fail early if another sflphone is running
- * #8449: Update version 1.0.2
- * * #8545: fixed error case
- * * #8545: fixed broken ringtone
- * * #8586: fixed make dist
- * sipvoiplink: use static_cast instead of reinterpret_cast if possible
- * * #8542: test for .git existence before moving pre-commit hook
- * * #8542: autogen.sh should not require git
- * eventthread: cleanup
- * * #8542: removed trailing whitespace from tree
- * * #8357: added disable video option to client
- * siptest: cleanup
- * * #8521: use avcodec_open2 instead of deprecated avcodec_open, if
- available
-
- -- Julien Bonjean <julien.bonjean@savoirfairelinux.com> Thu, 07 Jun 2012 16:08:15 -0400
-
-sflphone-daemon (1.0.0-rc20110930~ppa1~SYSTEM) SYSTEM; urgency=low
-
- ** SNAPSHOT 1.0.0-rc20110930~ppa1~SYSTEM **
-
- * update kde .gitignore
- * Fix bug in volume widget
- * More polishing for release
- * Bump version to 1.0.0
- * [#7023] Add the ability to load an abstract contact backend in the
- library to resolve more data, polish code
- * [#7021] More cleanup for release
- * Cleanup
- * [#7021] Refactor KDE client dbus handling, add a missing call in
- daemon and port the DataEngine to the new API
- * Remove some annoying debug
- * merge language scripts
- * remove obsolete 'VERSION' files
- * update install instructions
- * Add missing translations to gnome
- * language update
- * Revert "Don't reference count DBus clients, exit core immediately
- when one of them request it"
- * Don't reference count DBus clients, exit core immediately when one
- of them request it
- * [7021] Add contact abstraction support
- * [#7121] Polishing library (over). Indentation, spacing and naming
- are now consistent
- * codecs: link to libccrtp, don't use logger
- * Fix a daemon bug
- * [#7038] Fix adding contact
- * * #7037 : stop audio stream after all calls have been hanged up
- * [#7025] Add full support for bookmark
- * SFLPhone KDE do not destroy history anymore
- * Fix config skeleton
- * Close the daemon once and for all, no more automatic respawning
- * Fix "unregistered account" bug (I hope so)
- * Close SFLPhone at the right place, it still respawn, I don't know
- why
- * Remove dead code
- * Fix regressions introduced in the last commit
- * Dead code elimination 1/3
- * Fix bug, add "add contact" option, fix warning
- * * #7019: Fix IAX codec negociation
- * Remove or comment unnecessary/unhelpful debug output
- * Fix "same as local" account setting, fix IP2IP LED color
- * Add support for some more advanced config options and add missing
- config dialog icons
- * Fix crash with noise suppressor
- * Alternative can now be selected from the call view context menu
- * Add drag and drop support, initial context menu and fix 3 bugs in
- the account dialog
- * Add basic history drag and drop support
- * Complete contact support is back
- * * #6991 : fix IAX problems
- * Fix IAX accounts being disabled by default
- * Revert "deb: forge -g flags for pjsip"
- * * #5884: Disable debug code in pjsip
- * echo suppressor : more assertions
- * Don't let the daemon think crypto is enabled when it's not
- * Simplify ToneList
- * Some progress on contact support
- * Remove unused getRegistrationCount()
- * remove annoying debug
- * revert SIP bit of e27e5c39bad27bae28f574eb2cba7717e8956229
- * Simplify CallManager::placeCallFirstAccount
- * Fix crash on hold
- * * #6905 : SIP refactor
- * gnome client: be sure key exchange is set correctly
- * Move code into createSipTransport
- * Fix account registration on start
- * ManagerImpl::registerAccounts(): simplify
- * * #5884: don't mess with pjsip threads in echo suppressor
- * * #6905 : simplify udp/stun/tls pjsip transport creation
- * Restore and improve support for Call history
- * fix launchpad build
- * SIPVoIPLink: simplify / refactor
- * Fix libwidget linking
- * SIP: simplify
- * IM : simplify
- * gnome: remove some debug
- * AudioRtpFactory::stop() cannot fail
- * * #6905: simplify SIP code
- * pjlib: fix build without SSLv2, fix warnings
- * Port history to the new syntax
- * Test a dock widget based implementation for contact and history
- * Disable SSLv2 support from pjsip and sflphone
- * deb: forge -g flags for pjsip
- * Fix deb packaging to get debug symbols
- * remove debug
- * pjproject: update to last stable release (1.10)
- * Require gtk >= 2.20 and glib >= 2.24
- * tlsadvanceddialog: simplify
- * * #6902 : fix errors spotted by -DGSEAL_ENABLE
- * Update daemon dbus XML and port KDE config backend from dbus to
- local
- * Remove unused but set variables
- * * #6929 : fix IM widget, cleanup
- * Unconditionally enable debug symbols
- * Should fix many KDE issues
- * * #6886 : hitting backspace on empty number have no side effects
- * * #6905 : fix AudioCodecFactory access in optimized builds (-O > 0)
- * Remove unsupported and broken jaunty/karmic packages
- * * #6902 : avoid using some gtk deprecated functions
- * Update dbus introspection files
- * * #6904: removed unused contactmanager
- * * #6903 : use correct dbus-cxx package name
- * * #6902: don't use individual gtk headers
- * Fix a segfault when config is not present
- * Merge latest (0.9.13) KDE code. This version is not yet ready for
- git master, but better than the previous one
- * addressbook : simplify
- * * #5659 : sflphone-plugins doesn't depend on libedataserverui
- * * #5659 : addressbook doesn't use libedataserverui
- * gnome client doesn't depend on evolution
- * * #5695: addressbook: simplify
- * * #5695: addressbook : remove AddrBookHandle from plugin
- * * #5695 : addressbook : remove unused stuff in the client
- * * #5695 : addressbook : remove unused stuff, use static mutex
- * gnome client doesn't use evolution
- * gnome: use proper API to set GTK_CAN_FOCUS
- * * #6897: removed unused focus state vars/callbacks
- * gnome: fix calls to sflphone_fill_codec_list_per_account
- * * #6623: gnome: don't leak in mainwindow
- * gnome: mainwindow whitespace cleanup
- * gnome: actions.c parameter doesn't have to be a double pointer
- * * #6895: fix memleaks, cleanup in accountconfigdialog
- * * #6893: fixes segfault in client on clean history
- * * #6894: fix leaks, cleanup in sflnotify
- * daemon: fixed prints in main
- * * #6892: simplify, fix leaks in dialpad
- * * #6887: audiopreference creates audio layer
- * * #6660: use const char * const, not std::string for globally
- visible constants
- * * #6852: Preferences now solely responsible for audiolayer creation.
- * * #6860: refactor uimanager, also fixes #6865
- * * #6853: hangup as soon as all digits have been deleted
- * * #6852: alsa: retry if device is busy
- * * #6852: audiolayer creation depends only on preference.audioApi
- * * #6850: gnome: fix build for gtk < 2.22.0
- * cleanup in iax
- * alsa: typo
- * pulse: if we can't peek in audio input, we can't drop samples
- * * #6849: show error window if codecs are missing, instead of dying
- * EchoCancel: unused, remove
- * * #6629 : use number of samples as arguments for audio filters
- * * #6629 : remove unused Algorithm interface
- * * #6629 : use helper to call alsa functions and display error msgs
- * Remove unused type
- * * #6841: fix some error handling
- * * #6629: simplify AlsaLayer::alsa_set_params()
- * Get gdk key definition from header
- * * #6828: Replace raw key codes by gdk defines
- * remove some debug, enhance some other
- * mainbuffer: simplify
- * * #6561 : fix phantom call after transfer
- * Conference Participant set : simplify
- * SIPCall: remove unused functions, make invite session public
- * * #6229 : remove malloc/free from pulse audio loop
- * * #6629 : simplify pulse callbacks
- * * #6629
- * Simplify widgets
- * * #6629 : keep the correct audio module when frequency changes
- * * #6751: fixed erroneous debug msgs
- * callable_obj.h: removed unneeded pthread header
- * alsalayer: cleanup
- * * #6629: Always restart audio driver when changing parameters (ALSA
- only)
- * gnome GUI: don't block in DBus signal errorAlert()
- * * #6629 : simplify AudioLayer creation
- * * #6629 : remove unused and unconfigurable frameSize from audiolayer
- * * #6629 : remove unused error message from audio layer
- * Fix logic error when switching audio API
- * Remove unused AudioProcessing class
- * AudioRtpRecordHandler::initNoiseSuppress() : use noiseSuppress
- directly
- * * #6629 : use DC blocker directly in audio layers
- * * #6629 : clean AudioLayer
- * * #6629 : don't store mainbuffer inside audiolayer
- * * #6629 : correct AudioLayer::notifyincomingCall()
- * * #6554: cleanup, refactoring in sipvoiplink
- * * #6554: cleanup in iaxvoiplink
- * * #6554: throw exception in getSIPCall if pointer is NULL
- * * #6554: make some methods of sipvoiplink static
- * * #6655: cleanup in managerimpl
- * * #6554: refactoring, fix memleaks in sipvoiplink
- * * #6478: remove throw specs, cleanup in voiplink
- * * #6629 : remove unused AudioDevice
- * * #6655: removed more dependencies from managerimpl
- * * #6744: simplified numbercleaner
- * conference : remove one prototype
- * * #6743: fix ip2ip
- * Don't give glib warnings if icons are not found
- * gnome: fixed includes
- * Codec.h: removed unused function
- * * #6742 : clean dbus & icons
- * * #6699: refactor/cleanup accounts
- * icons: cleanup
- * timer : use second precision, not millisecond
- * calltree_update_clock : use correct type, returns something
- * * #6737: fixed typo in dbus call
- * * #6737: removed tests for removed API
- * * #6737: dbus: fixed bug from merge
- * * #6737: cleanup in accountlist
- * * #6737: cleanup in dbus
- * * #6740 : fix history double free
- * * #6740 : remove time updating thread from calls
- * * #6737 : use c99 for client
- * * #6738 : make history loading faster
- * sipvoiplink : don't crash on transfers
- * fixed typo
- * Remove unused file
- * Don't build networkmanager.cpp at all if NM is disabled
- * _debug* -> _debug
- * * #6554 : simplify sipvoiplink
- * hudson: added -x to git clean command
- * added git clean to hudson script
- * audiocodecfactory: cleanup
- * * #6718: refactored setTlsSettings into SIPAccount
- * * #6718: removed more unused methods
- * * #6718: refactored confmanager code into sipaccount
- * remove unused functions
- * * #6718: confmanager: removed more unused methods
- * AudioCodecFactory : cleanup
- * #6697 : Turn callableElement struct into union
- * * #6718: confmanager: removed more unused methods
- * * #6718: confmanager: removed more unused methods
- * * #6718: removed unused dbus methods, refactoring
- * * #6699: accounts: cleanup/refactoring
- * * #6699: refactoring, cleanup in accounts
- * * #6699: more account cleanup
- * remove unused autoconf variable
- * * #6714: fixed hudson script
- * make distclean in hudson
- * added || exit 1 to run_tests.sh call
- * * #6714: fixed make distcheck for sflphone-plugins
- * * #6714: fixed make distcheck for gnome client
- * * #6714: fixed make distcheck for daemon
- * git: #6698 split the main .gitignore file
- * gnome: gpointer is already a pointer
- * gnome: calltab_init: use calloc instead of malloc
- * * #6699: more account cleanup
- * * #6699: cleanup account
- * * #6554 : more *voiplink cleanup
- * * #6558 : more sipvoiplink simplification
- * * #6558: saner loadSIPLocalIP prototype
- * gnome: #6623 clean calllists
- * * #6692: more audiolayer cleanup
- * * #6692: cleanup/refactoring in audiolayers
- * * #6692: more forward declarations, AudioThread->AlsaThread
- * * #6692: audiolayer cleanup
- * * #6692: alsalayer cleanup
- * * #6558 : remove account creator
- * * #6558 : clean sipvoiplink
- * * #6554 : cleanup sipvoiplink
- * audiortp: cleanup
- * * #6657 : fix launchpad builds for good
- * * #6675 : send RTP dtmf events only once
- * * #6655: more cleanup
- * AudioRtpSession::updateSessionMedia() : simplify
- * * #6655: more cleanup in managerimpl
- * * #6655: removed more code, cleanup
- * * #6655: more cleanup, fixed infinite loop
- * * #6655: removed more unused files
- * * #6655: removed unused mutex
- * * #6655 removed more unused code
- * * #6655: removed unused methods
- * * #6655: cleanup in main
- * * #6663: fixed segfault when off hold from transfer
- * * #6658: user's active codec selection is respected
- * * #6660: static global string should be static const char* const
- class member
- * * #6659: use g_strcmp0, not strcmp for vals that may be null
- * callable_obj: fix double free
- * calltree_display_call_info() : simplify
- * * #6657: Fix launchpad builds
- * Logger::log() : simplify
- * AudioRtpSession : privatize members
- * * #6655: more constness, cleaned up/simplified methods
- * * #6654: call DBus::_init_threading so that dbus-c++ to make it
- threadaware
- * set default credentials on account creation
- * AudioCodecFactory::scanCodecDirectory() : simplify and correct
- * * #6623: fixed typos
- * * #6623: fixed more leaks
- * * #6623: fixed more leaks
- * * #6623: fixed more leaks, don't print codec name if null
- * * #6623: more leaks fixed in client
- * * #6623: fix more leaks, fixed some warnings
- * * #6623: fixed leak in history
- * updated gitignore
- * initialize dbus dispatcher correctly
- * Fix tests, hudson doesn't have a dbus daemon running
- * remove unused code
- * removeCall() : simplify , fix leak
- * stopRtpThread() : simplify
- * *CurrentCall : simplify
- * Fix memleak
- * fix serialization of audio api (pulse / alsa)
- * account map : simplify
- * remove call from callmap before terminating it, avoid use after free
- * * #6630 : don't make DBusManager a singleton
- * call: return confID by value
- * add back history code deleted by error
- * history : reverse logic
- * simplify history serialization and remove some debug
- * remove annoying debug
- * * #6464 : replace cerr with _error
- * * #6464: replace cout with logger macros
- * replace printf() with logger macros
- * update .gitignore
- * remove unused function
- * update eclipse projects
- * uimanager_new() : simplify
- * rename directories
- * celt: simplify a bit
- * Fix CELT configure.ac test
- * * #6612 : template speex codecs
- * * #6623: refactored conference obj
- * * #6623: refactored callable object, removed leaks
- * * #6623: more cleanup, fix leaks, make global vars static and rename
- them
- * * #6623: calltree: fixed memleaks, simplified code.
- * audiolayer: init pointer members
- * manager: catch exception on invalid hangup
- * * #6623: don't leak on calls to create_new_call
- * * #6611 : clarify codecs prototypes
- * ringtones : .au and .ul files are both ulaw
- * * #6611 : make sure samplerate converters are called correctly
- * ManagerImpl::switchAudioManager() : simplify
- * * #6623: fixed more leaks
- * * #6623: fixed more leaks
- * * #6623: fixed more leaks
- * * #6623: fixed leak, line-endings in imwidget
- * * #6627: zero-initialize pointers if they're going to be deleted
- * * #6628: don't leak calls on exceptions
- * Revert "audiortp: call join after calling stop on RtpThread"
- * sflphone-client: more constness
- * audiortp: call join after calling stop on RtpThread
- * * #6625: return 0 on successful completion
- * * #6624: fix segfault on servercallfailure
- * * #6621: Fixed double free, unlock mutex in ManagerImpl::terminate
- * * #6220: remove audio stream when peer hangs up
- * * #6596: AudioSymmetricSession shouldn't self-delete
- * resampler: grow internal buffers dynamically
- * merge up and down sampling => resampling
- * Leave test directory unchanged when running make check
- * audio algorithms : remove unused prototype
- * ringtone: detect codec from file extension
- * *AudioFile : simplify
- * * #6596: create local SDP on the stack, not the heap
- * * #6596: don't call Ost::Thread::terminate from dtor
- * audiofile: cleanup (samplerate -> unsigned)
- * remove unused func
- * samplerateconverter: cleanup
- * RingBuffer::Put() : remove unused return value
- * MainBuffer::putData() : remove unused return argument
- * audiolayer::putMain() : remove unused func
- * AudioLayer::putUrgent() : remove unused return value
- * * #6618: delete any remaining ringbuffers in destructor
- * RingBuffer::availForPut() : remove
- * * #6617: return from main rather than calling exit
- * MainBuffer::availForPut(): remove
- * RingBuffer: simplify
- * alsa : remove write only variable
- * fix memcpy declaration
- * bcopy(src, dst) -> memcpy(dst, src)
- * RingBuffer::Get() : remove constant volume argument
- * return a copy of the call ID, not just a reference.
- * MainBuffer::getDataById() : remove volume argument (always 100)
- * MainBuffer::getData() : remove constant volume argument
- * RingBuffer::Put() : remove constant volume argument
- * MainBuffer::putData() : remove constant (=100) volume argument
- * audiolayer: remove constant _defaultvolume
- * AudioRtpRecordHandler / AudioRtpSession : simplify
- * mainbuffer: fix test
- * iaxvoiplink : simplify
- * sip registration callback: fix a dbus crash
- * MainBuffer: simplify
- * AudioRtpFactory: return cached type of rtp session. The rtp session
- can have disappeared if the call was put on hold
- * AudioRtpFactory: remove unused setters
- * Fix launchpad builds
- * * #6611 : remove unused bandwidth codec information
- * * #6611: AudioCodec: remove useless/unused setters
- * make sure buffer string is initialized correctly
- * * #6596: declare certain destructors virtual
- * audiolayer : cleanup
- * Simplify doc build rules
- * * #6270: don't build dbus-api doc with make, should require make all
- * configure.ac: cleanup
- * Remove copy of dbus-c++ from libs/
- * * #6596: stop clock thread when peer hangs up
- * removed unused Fmtp.h
- * * #6595: more logical initialization order
- * * #6600 : fix account creation
- * * #6601 : fix configure.ac tests
- * remove unused variable
- * Don't mix stack and heap based allocations
- * Fix copyright (2009, 2008, 2009 -> 2008, 2009)
- * Fix warnings found by clang
- * * #6595: fix initialization order for AudioRTP
- * * #6592: removed typedef std::string CallID
- * * #6586: implement local g_slist_free_full for older glib versions
- * * #6579: fix memory leaks in client (there's a lot left)
- * ShortcutPreferences::setShortcuts() : simplify
- * Fix merge
- * * #6548: remove call to non thread-safe strerror()
- * AudioRtpFactory: each instance is associated to exactly one SipCall
- * create_audiocodecs_configuration() : make static
- * * #6269 : refactor AudioRtpSession
- * Fix AudioSymmetricRtpSession.h inclusion guard (cherry picked from
- commit c3081dce1cc1370d6d3558a4c4ef5cfac0d21caf)
- * * #6269: Rename AudioRtpSession to AudioSymmetricRtpSession
- * * #6574: Don't exit when connection to pulseaudio server fails
- * accountconfigdialog.h : remove some stuff from header
- * * #6560: fix configuration test
- * Fix warning in test
- * * #6560: don't hide password entry in security tab
- * * #6560: set initial password for SIP accounts
- * * #6506: remove useless pointer indirection
- * * 6560: password is now specific to IAX accounts
- * * #6560 : actually use, store, restore, transmit SIP credentials
- * * #6560: YamlEmitter: serialize sequences
- * YamlEmitterException: typo
- * ManagerImpl::computeMd5HashFromCredential() : simplify, fix memleak
- * * #6561: invite_session_state_changed_cb() : simplify
- * * #6561: More useful debug in VoIPLink::removeCall
- * * #6561 : fix ghost call reappearing in GUI after transfer
- * while -> for (make the code smaller)
- * * #6558 : Account::loadConfig() : move IAX code to IAXAccount
- * IAXVoIPLink::getAccountPtr : simplify
- * * #6554 : access the SIPVoIPLink directly, not per account
- * SIPVoIPLink is instanciated only once and is not associated to a
- single account
- * yamlnode: use const references when possible (still some left to do)
- * Account::_accountID: constify
- * VoIPLink: simplify, remove unused method
- * hudson test : no need to call run_tests.sh anymore
- * Remove AccountID type and AccountNULL define
- * Make check runs the test (no need to call run_tests.sh manually
- anymore)
- * gnome GUI: Fix tests
- * Revert "Move registration information from SIPAccount to
- SIPVoIPLink"
- * * #6392: pluginmanagertest: fix warnings reported by valgrind
- * * #6547 : remove unused exceptions
- * * #6547: CallManagerException: use runtime exceptions
- * * #6547: InstantMessageException: use runtime exceptions
- * * #6547: do not throw exceptions if some settings are not present in
- config file
- * * #6547: YamlParserException: use runtime exceptions
- * * #6547: VoipLinkException: use runtime exceptions
- * * #6547: YamlEmitterException: use runtime exceptions
- * * #6547: DTMFException: use runtime exceptions
- * * #6547: AudioFile: use runtime exceptions
- * * 6547: AudioZRtpSession: remove impossible error case
- * * #6547 : AudioRtpSession: remove impossible error case
- * * #6547: AudioZrtp: use runtime exceptions
- * * #6408 : send authenticationUsername to GUI
- * * #6408 : store/restore authenticationUsername from config file
- * SIPAccount: simplify
- * Move registration information from SIPAccount to SIPVoIPLink
- * SIPAccount::getAccountDetails : simplify
- * * #6540: yaml parser: simplify
- * sdp.cpp : fix a warning
- * * #6540: yaml parser : remove std::string typedefs
- * * #6540: Simplify yaml unserialization
- * * #6540 : add a Conf::ScalarNode constructor for booleans
- * setAccountDetails(): simplify
- * * #6408: store authentication username in daemon
- * * #6408: Be able to set the authentication username in the GUI
- * * #6507 : do not crash if the program is not sflphoned
- * Fix tests
- * macroify SIPAccount::unserialize()
- * Move all .cpp files from sflphoned target to libsflphone.la, except
- main.c
- * main() : simplify, return positive error codes
- * * #6507 : find codecs dir in build directory
- * * #6392: Sdp: move clean functions to destructor
- * AlsaLayer::adjustVolume() : simplify
- * alsalayer : reduce indentation
- * malloc/free -> new/delete
- * malloc/free -> new[]/delete[]
- * malloc/free -> new/delete
- * AudioSrtpSession: simplify base64 encoding
- * * #6392: Initialize std::string from pj_str_t correctly
- * * #6392: AudioRtpSession: Initialize remote port
- * Audio settings : Initialize _echoCancelTailLength and
- _echoCancelDelay(0)
- * Initialize variable
- * YamlParserException : fix use of stack variable after it has been
- deallocated
- * * #6392: fix memory leak in history
- * * #6392 AudioCodec : fix memory leak
- * * #6392 : fix memory leak in sip account
- * * #6408: clean up sipaccount (cosmetics mostly)
- * sipaccount.cpp serialize() : reduce number of lines
- * * #6392: invalid memory access
- * * #6392 : fix invalid memory access
- * * #6479: merged useful code from MimeParameters into Codec interface
- * * #6462: fixed hangup on IP2IP call
- * added run_daemon.sh script
- * test: remove unused variable
- * Remove functions only used by a failing test (cherry picked from
- commit fcf718cb75de7f1882dc61c07bb8d300dfa10f85)
- * * #6360 : make client tests build (cherry picked from commit
- 028b2835f040e51ab8ab979b32732b07b8798fce)
- * * #6360 : fix warnings in check_global test (cherry picked from
- commit 9e2bd6a7496dd64f6f48595e385760019aab1193)
- * * 6360: updated API calls in tests, but they're not building yet
- (cherry picked from commit 548f6f0f919b43772a3e9c667e5e292791281795)
- * Fixed include in tests (cherry picked from commit
- aeadc7525c1e31f936670ac8b02f0bcf387c38a8)
- * Remove unused variables and functions
- * IAX: fix warnings (cherry picked from commit
- fd7a113a11cac2cd9a7c36929e88ad28195c4c35)
- * Remove unused DEBUG define which interferes with logger.h (cherry
- picked from commit b2f72b91d0f43cb1dd94d138882a8caa9c841c24)
- * * #6392: no need to check for account NULLity since it is
- dereferenced above
- * * #6392: fix a memory leak, replace by stack allocation
- * * #6392: remove a variable assignement which confuses cppcheck
- * process_conference_participant_from_serialized() : remove unused
- function
- * * #6392: s/free/g_free/
- * * #6392: fix a memory leak in abookfactory_load_module()
- * * #6392: remove generate_call_id() used only once
- * * #6392: fix memory leak (opendir() without closedir())
- * * #6392: AudioRecorder(): ensures mbuffer is set
- * Remove SFLPHONED_VERSION from global.h, use autoconf PACKAGE_VERSION
- * #6298: Cleanup
- * #6331: Fix deleting ringtone file after call have been answered
- * * #6330: merged user_cfg into headers
- * #6298: Fix conference recording file update at conference end
- * #6298: Fix record file name serialization for conference
- * * #6295: cleanup of codec hierarchy
- * #6298: Fix gtk warnings
- * * #6300: added script to run tests
- * #6109: Add recording playback for conference
- * * #6300: tests do not require an installed sflphone
- * * #6295: re-removed clone methods
- * #6109: Fix gtk_critical warnings for incoming calls
- * #6109: Fix GTK_CRITICAL warning
- * #6109: Fix icons when history is not activated
- * #6109: Fix warnings
- * #6109: Implement stop recorded file playback signal
- * Revert "* #6295: removed unused clone method"
- * * #6295: removed unused clone method
- * * #6296: removed non existant file from Makefile.am
- * #6109: Stop fileplayback for outgoing call
- * #6109: Implement stop recording playback button
- * Fix binding names errors in dbus introspection file
- * #6109: Implement playback recorded file callback in client
- * #6109: Store recorded file path on client side
- * #6109: Add dbus methods for call recording playback
- * * #6290: remove unused classes from utilspp
- * * #6288: cleanup sdp
- * * #6288: fix exception usage
- * * #6288: simplify SdpException
- * * #6288: cleanup in sdp.cpp/h
- * #6109: Only display playback button if record file is set and valid
- * * 6290: updated configure.ac to remove functor Makefile
- * * #6290, #6289: removed unused classes from utilspp, fixed make
- check
- * #6109: Add button for history playback of recorded file
- * * #6289: removed unused observer class
- * * #6282: forward declare sdpMedia in sdp.h
- * * #6281: renamed setCallAudioLocal->setCallMediaLocal
- * #6183: Handle conference with more tahn two calls
- * #6183: Fix history icons when calling back a conference from history
- * #6183: Fix icons inconsistencies in history for conference hang up
- * #6183: Fix toolbar actions when selecting a conference in history
- * #6183: Fix conference serialization
- * #6268: Serialize only calls
- * * #6269: removed useless type testing
- * ignore some files in test/
- * * #6268: Remove dead class AudioSymmetricRtpSession
- * #6251: Do not had history calls in calllist when loading history
- file
- * #6251: Fix insertion in history map in before saving history file in
- daemon
- * #6251: Fix history unit tests
- * #6251: Order the list before serailization, get rid of the hashtable
- in history
- * #6251: Implement history serialization using a list wether than a
- map
- * * #6253: remove external audioport from header, make all members
- private
- * * #6253: don't store external local audio port (used for NAT) in
- Call
- * #6251: Add start_time timestamp in history serialization
- * #6251: Fix call insertion in conference items
- * #6233: Fix serialized account list terminated with a ";" character
- * #6238: Fix draggable history calls into current calls
- * #6233: Fix toolbar updates
- * #6233: Fix history
- * * #6235: remove pyc files from git tree
- * #6233: Handle cases when one or manuy calls are unreachable in
- createConfFomrParticipantList
- * #6233: Handle wrong numbers in createConferenceFromParticipantList
- * #6231: Fix drag-n-drop issue
- * * #6173 : move sippxml in tools
- * #6231: Fix merging issue
- * #6183: Implement conference unserialize
- * * #6212: remove extraneous flags from globals.mak
- * #6183: Unserialize conference data in conference
- * #6183: Add account information in request for conference call from
- history
- * #5755: Add -ldl to liker in sflphone-client-gnome
- * #5755: Fix fedora 15 compilation issue
- * #6183: Serialize conference participant phone number and account
- * #6183: Add conference timestamp in serialization
- * * #6186: don't include global.h, just logger.h
- * #6183: Fix saving history to file
- * #6183: Fix removing call from calllist
- * * #6184: remove pointers to Manager from AudioRtpSessions
- * #6183: Calling calltree_add_call explicitely for history
- * #6183: Ability to store conference inside history tab queue
- * * 6181: remove unused API from sipcall
- * #6171: Implment nreCallCreated callback
- * #6167: Fix participant list NULL ending
- * #6149: First draft of conference creation from history
- * #6149: Fix multiple call/conf selection callbacks ...
- * #6129: Fix place_call function called twice for pressing enter
- action
- * #6129: Fix double click action for history
- * #6149: Add dbus call for creating conference from history
- * #6129: Fix placing call from history and addressbook (still need to
- fix icon)
- * * #6148: removed unused AudioRtpFactory constructor
- * * #6145: remove unused isAudioStarted
- * * #6145: remove unused isAudioStarted
- * #6129: Add conference into history, fix call/conference selection
- * * #6143: don't use getType outside of serialization methods
- * * #6132: forward declarations instead of includes
- * * #6132: add constness, remove redundant "inline" keywords
- * #6129: Add timestamp to conference object to order history entries
- * * #6128: remove unused forward declarations from header
- * * #6127: make noncopyable class actually noncopyable
- * * #6125: don't include AudioRtpFactory in sipcall.h
- * #6123: Fix alsa ringback audio file
- * #6123: Fix raw audio file loading problem
- * #6109: Fix daemon plugin manager unit test
- * #6109: Fix history manager unit tests
- * #6109: Recording filename in daemon and client for history items +
- serialization
- * #6109: Refactor AudioFile to play recorded call
- * * #6104: AudioCodec moved to sfl namespace
- * * #6099: remove active flags from codec classes
- * #6095: Add notification-daemon as a runtime dependencies for rpm
- packages
- * #6095: Fix fedora 15 compilation in MineParameters.h
- * #6095: Declare static variable explicitely for client
- * #6095: Add logs to build OSC build machine
- * * #6098: global variables should have file-scope to avoid name
- conflicts
- * #6095: Fix compilation error for Fedora 15
- * #6095: Update SFLphone version to 0.9.14
- * #6095: Add specification file in opensusse build service for
- sflphone-plugins
- * #6073: Fix sflphone-plugins build on launchpad
- * #6093: Rename CodecDescriptor for AudioCodecFactory
- * * #6089: fix warnings in make check
- * * #6086: renamed codecs methods to audio_codecs
- * * #6085: renamed codec related dbus calls to audio_codec
- * #6065: Remove g_print from client, use DEBUG instead
- * #6065: Add actions name for addressbook
- * * #6085: renamed codecs* widgets/functions audiocodecs*
- * #6065: Fix Addressbook runtime warnings
- * #6065: Replace Codecs tab for Audio in account preference dialog
- * #6065: Fix "transfert" typo
- * #6065: Fix addressbook action runtime warning in uimanager
- * * #6082: fixes make check by adding libcrypto libs to test
- dependencies
- * #6073: Rename plugin/addressbook folders for addressbook/evolution
- in sflphone-plugins
- * #6074: Removed AC_SUBST from configure.ac when using
- PKG_CHECK_MODULE
- * #6073: Fix sflphone-plugins package build
- * #6073: Fix sflphone-daemon build
- * #6065: Fix runtime gtk warning when initializing searchbar without
- addressbook
- * #6063: Fix mozilla-tellify gitignore
- * #6063: Remove stream copy file using ifdef macro
- * * #6012: fix make dist for sflphone-daemon
- * #6063: Update .gitignore file
- * #6058: Fix base64 encoding related warnings
- * #6056: Fix SdpException handling
- * #6055: Fix unknown pargma warning for gcc <= 4.5
- * * #5949: test gcc version before disabling unused-but-set warning
- * #6054: Fix addressbook plugin compilation warning
- * #6048: Fix uimanager static initialization
- * #6046: Fix addressbook factory static initialization of member
- addrbook
- * #5979: Fix implicit function declaration warning
- * #6042: Fixed discarding qualifier warnings in client
- * #6041: Fix instant messaging unhandled case warning
- * #5994: Implement set current addressbook name and search type in
- addressbook plugin
- * #5994: add rules for launchpad packaging of addressbook plugin
- * #5994: Fix addressbook plugin configuration loading
- * #6027: Fix addressbook enabled test from configuration
- * #6027: No need of gnomedoc related macros in addressbook plugin
- * #6027: Add NEWS file required for build
- * #6027: Add addressbook plugin autogen.sh script
- * #6027: Remove plugins from client
- * #6027: Add sflphone-plugins folder at project's root level
- * #5994: Move addressbook folder from contacts to plugin folder
- * * #6011: removed unused Makefiles
- * * #6010: remove unused headers
- * * #5952: fix "string constant to char*" warnings
- * * #6009 fixed warnings
- * * #6003: finished cleanup of account classes
- * * #6003, #6004: cleanup of account classes, defaultAccount no longer
- global
- * * #6000: fix memory leak of args object
- * * #5998: removed using namespace std from networkmanager
- * * #5998: removed "using namespace std" from ZrtpSessionCallback
- * * #5998: removed using namespacestd from AudioZrtpSession.h
- * * #5998: remove "using namespace std" from auriorecord.h and
- MimeParameters.h
- * * #5998: remove using namespace std in main
- * * #5998: removed "using namespace std" from logger
- * * #5949: test gcc version before disabling unused-but-set warning
- * #5994: Installation of addressbook plugin
- * #5979: Implement codec full addressbook search from plugin
- * #5979: Implement addressbook factory and plugin
- * * #5981: unused webwidget removed
- * #5966: Account config synchronization fix (for stun)
- * #5954: Handle media name exception
- * #5954: Fix audio codec name display in client
- * #5954: Clean up getSessionMedia methods
- * * #5957: getRecordingSmplRate returns a value
- * #5954: Clean up getCurrentCodec methods
- * * #5950: remove "converting to non-pointer type 'int' from NULL"
- warnings
- * #5915: Full gain control version
- * * #5949: remove more unused variable warnings
- * * #5949: remove unused/unused-but-set variable warnings
- * * #5949: show_preferences_dialog returns a success value
- * * #5946: cleanup of include directives, undefined function
- * * #5515: comment out SSLv2 calls in pjsip
- * #5915: Implement different slope for attack tme and release time for
- gain control
- * #5915: use only one input signal for gain control (removed output
- buffer)
- * #5921: Fix no audio after holding a conference
- * #5916: Add gaincontrol files
- * #5916: Implement FFMPEG/CCRTP video streaming prototype
- * #5903: Fix call transfer during a conference
- * #5915: implement rms detector, first order averager, limiter for
- gain control
- * #5914: Fix call transfer when no notification request is required
- * #5899: Fix conference right-click segfault
- * #5884: temporary fix segfault in pjsip memory pool
- * #5883: Fix compilation issues on maverick and lucid
- * #5755: Fix fedora 15 compilation without patching ccrtp
- * [#5855] Make echo canceller optional
- * #5855: Fix echo suppression activation/deactivation
- * #5855: Implement pjsip echo canceller
- * #5814: Speex initialization function uses samples, not bytes
- * #5814: Test using more unbalanced signals
- * #5814: Fix buffer size for long echo length or long echo delay
- * #5814: Adjust level for echo cancellation at runtime
- * #5814: Process noise reduction before echo cancelling
- * #5814: Implement speex post echo canceller processing
- * #5814: Dump echo cancel file to disk
- * #5814: Add parameters for echo cancel
- * #5809: Add configuration parameters
- * #5809: Implement speex echo canceller in audio rtp session
- * #5814: Code cleanup
- * #5814: Fix conf creation with several incomming ringing calls
- * #5814: Fix conf creation segfault when dragging a call on hold on a
- ringing call
- * #5809: Added unit test for echo cancellation and implemented
- "process" virtual method
- * #5709: Add always recording option in configuration
- * #5709: Add always recording option in audio conference panel
- * #5709: Add core functionnality for always recording (missing config
- options)
- * #5769: Fix conference participant handling (detach/attach) and hold
- actions
- * #5747: Fix recording icons and state for conference when adding new
- participant
- * #5769: Code cleanup
- * #5769: Fix hangup unsent calls
- * #5769: Fix remove/add additional participant to conference
- * 5769: Several fixes concerning confererence handling
- * #5769: Fix compilation error
- * [#5769] Fix audio streams binding in main buffer
- * #5769: Removed access to audio mixer from audio layer
- * #5765: Fix audio crash for illformated wavefiles
- * #5765: Add maximum iteration for finding fmt and data "chunck"
- * #5589: Fix compilation of libnotify under
- * #5757: Fix abort signal when receiving INFO
- * #5747: Add usersDetached.svg
- * #5747: Handle offhold action for recording conference
- * #5747: Fix off hold action for conferences
- * #5747: Implement update conference in record action in calltree
- * #5747: Add new icons for recording conferences
- * #5747: Add recording state for conferences
- * [#5738] Remove getAudioDriver call from manager (replace by
- _audiodriver var)
- * [#5738] Refactor mutex protecting audiolayer
- * [#5737] Fix HD conference recording
- * [#5730] Fix start audio session after changing sampling rate
- * [#5714] Fix enter keyboard event for addressbbok and history
- * [5695] Fix addressbook combo box update when no addressbook selected
- * [#5695] Fix addressbook initialization and search bar update
- * [#5695] Add mutex for books_data in addressbook to protect async
- calls
- * [#5695] Get back addressbook open from uri
- * [#5695] Fix absolute addressbook URI for local addressbooks
- * [#5695] Implement libebook 3.0 interface
- * [#5571] Better logic for hangup (for case where call have not been
- sent yet)
- * [#5571] Update error handling in voip links
- * [#5571] Fix compile time warnings
- * [#5696] Fix installation dependencies for Natty
- * [#5669] Add mention that sflphone.org is for testing only
- * [#5693] Add natty in teh dput.conf file
- * [#5690] Remove not useful logs
- * [#5670] Use dynamic payload type for rtp dtmf
- * [#5668] Clean up sflphone configuration logging
- * [#5668] Fix hook checkbox configuration update
- * [#5666] Fix unit tests
- * [#5666] Manage event subscription
- * [#5666] Emit bye request when subscription is terminated
- * [#5666] Bye request should be sent after event subscription
- notification is done on transfer
- * [#5666] Make reinvite method static (to be called in pjsip
- callbacks)
- * [#5666] Hangup Call in manager for AccountNULL and IP2IP
- * [#5589] Use PKG_CHECK_MODULE for every client's dependencies
- * [#5623] Enlarge initial size of pjsip memory pool for calls (16k)
- * [#5564] Fix audio recording resampling for g722
- * [#5571] Move attribute handling for onhold/offhold actions in SDP
- session
- * [#5571] Codec negotiation refactored and unittested
- * [#5571] Implement tests
- * [#5571] Implement pjsip negociator
- * [#5571] Fix unit tests
- * [#5571] Add Fmtp.h to repository
- * [#5571] Integrate mime types and codec factory
- * [#5571] Handle exception when SDP negotiation fails
- * [#5570] Add sflphoned-sample.yml in repository
- * [#5564]: Implement stereo to mono mixing for rigntone
- * [#5342] Update audio stream initialization
- * [#5514] Restore test ni historytest suite
- * [#5514] Fix
- * [#5514] Disable test_create_history_path
- * [#5514] use pulseaudio in sample config file
- * [#5514] Fix test: load history from file
- * [#5514] Do not use X
- * [#5513] Make unit tests compile successfully
- * [#3947] Enable unit tests in Jenkins
- * [#5454] Fix build system to handle new version number
- * [#5454] Update languages from launchpad
- * [#5454] Add --without-celt in OpenSuse build service
- * [#5454] Change version number
- * [#5331] Added first SDP session tests
- * [#5273] Update nightly build version tags to conform dpkg rules
- * [#5211] Refactor send register method for iaxvoiplink and
- sipvoiplink
- * [#3950] Remove call being transfered from calltree
- * [#5211] Use appropriate memory pool for transport selector
- * [#5211] Fix strict aliasing rules warning in pjsip
- * [#5211] Bring back pjsip shutting down sleep to 1000 ms
- * [#5211] Fix registration callback segfault when closing the
- application
- * [#5211] Use the dialog memory pool for Route header in INVITE
- request
- * [#5211] Add temporary memory pool for findLocalAddressFromUri and
- findLocalPortFromUri
- * [#5211] Use individual memory pool for dtmfs
- * [#5211] SipVoipLink refactoring
- * [#3950] Attended transfer for conference calls
- * [#5284] Fix DNS resolution for Route with specified port number
- * [#5284] Some code cleanup
- * [#3947] Fix typo in hudson script
- * [#5284] Added sip route to REGISTER, INVITE, BYE request, plus DNS
- resolution
- * [#5266] Use RTP dtmf as default
- * [#5284] Added pjsip_process_route_set after setting routes in regc
- structure
- * [#5286] Fix parsing error due to long configuration file (removed
- max event)
- * [#5286] Fix false test in configuration emmiter
- * [#5286] Code cleanup
- * [#5286] Updated exception handling in configuration system
- * [#4969] Fix put SRTP call on hold
- * [#3950] Add debug messages
- * [#3950] Ability to perform an attended transfer
- * [#5276] Fix initialization problem in g722
- * [#3950] Add replace header in SIPVoIPLink::transferWithReplaces
- method
- * [#3950] Implemented attended method in SIPVoIPLink
- * [#3950] Cleanup transaction request received callback
- * [#3950] Implement dummy attended transfer in gnome-client
- * [#5249] Fix audio samplerate update algorithm for g722
- * [#5249] Fix uninitialized variable used in conditional jumps
- * [#5249] Fix conditional jump error in audiolayer (uninitialized
- value)
- * [#5267] Use autoconf 2.65 as a requirement (instead of 2.67)
- * [#5267] Restore manual pjsip configuration and compilation
- * [#5267] Autodetect celt version (0.9.1, 0.7.1)
- * [#5267] Fix deprecated macros in gnome client configure.ac
- * [#5267] Update configuration for libcelt-dev
- * [#5267] Fix build autoconf and automake
- * [#5227] Deactivate automatic call to astyle after compilation
- * [#5242] Hangup every calls before leaving
- * [#5237] Will now nightly-build for natty, Karmic deprecated
- * [#5229] Use inner class for rtp thread instead of inheritance
- * [#5211] Move mainbuffer unbind call in rtp final method
- * [#5211] Initialize sip call memory pool using 16 kb
- * [#5211] Use call memory pool in session reinvite
- * [#5211] Add debug messages
- * [#5211] Use and internal pool for calls
- * [#5211] Reduce pjsip memory pool usage for stateless error messages
- * [#5211] Refactor call deletion
- * [#5212]
- * [#5208] Refactor codec management for accounts
- * [#5168] Remove printf from codec's encode & decode method
- * [#5168] Fix celt compilation on launchpad
- * [#5168] Fix sflphoned compilation warnings in audiocodec.h
- * [#[#5168] Must keep the g722 specific RTP rate to avoid incoming
- packet timeout
- * [#5168] Fix static/dynamic payload rtp session update
- * [#5168] Throw SIPVoipLink Error if codec not instantiated in new
- outgoing call
- * [#5168] Fix dynamic/static codec payload type ambiguity
- * [#5169] Fix doubled IP2IP profile when no config file
- * [#4867] Add gtkinfobar in configuration panel
- * [#4867] Disable input/output/ringtone selection when using default
- alsa plugin
- * [#4952] Patches for possible buffer overflows
- * [$4885] Fix schemas problem
- * [#4885] sflphone-client-gnome.schemas not present during build
- * [#4885] Add gconf shemas directories in opensuse build system
- * [#4885] Add file/folder ownership for opensuse-factory build system
- * [#4906] Fix opensuse-factory build
- * [#4885] Update name dependency for libedataserver
- * [#4885] Fix non-void function without return in dbus-c++
- * [#4895] Update language translation
- * [#4896] Update session timestamp when updating media
- * [#4896] Reapply RTP hack for G722 payload type
- * [#4896] Update recording sampling rate when updating codec
- * [#4897] Save codecs in config for each configuration changes
- * [#4895] Do not save config when sflphone quit
- * [#4885] Update date for copyright
- * [#4885] Deactivate siptest that require more than one sipp instance
- * [#4879] Remove inmcoming call notification from IAX
- * [#4885] Some cleanup
- * [#4874] Add setCancel immediate/deffered for ost::Thread
- * [#4879] Fix incoming call notification
- * [#4878] Set keyboard focus on searchbar when selecting addressbook
- * [#4874] Fixed compilation warning
- * [#4874] Fixed compilation warning in sipvoiplink
- * [#4874] Fix compile time warning in RTP record handler
- * [#4874] Fix conditional jump in SDP
- * [#4874] Fix conditional jump based on uninitialized value
- * [#4874] Store call id within rtp thread context
- * [#4874] Fixed conditional jump based on uninitialised value in
- conference
- * [#4871] Fix default account fetching
- * [#4870] Delete RTP session when Refusing an incoming call
- * Restore IP to IP call
- * [#4857] Fix audio codec negotiation problem
- * [#3947] Adjust ressources allocated to compilation
- * [#3947] Disable unit tests in Hudson
- * [#4305] Free mutex only when really quiting SFLphone
- * [#4859] Update copyright to 2011 in every source file
- * [#3218] Character '.' stripped by the caller engine
- * [#4854] Fix typos, desktop entry
- * [#4847] Apply RTP modification to ZRTP session
- * [#4852] Update Karmic and Lucid dependencies
- * [#4852] Add Libedataserver and libedataserverui as gnome client
- dependencies
- * [#4852] Add authentication mechanism for EDS
- * [#4851] Fix segfault when closing pulseaudio layer too rapidly
- * [#4808] Some otehr cleanup
- * [#4808] Made some cleanup
- * [#4808] Added mutex in rtp session for codecs and noise process
- * [#4847] Update audio processing when updating RTP media
- * [#4842] Add support for linking with gold/ld --no-add-needed
- * [#4808] Make update g722 related static/dynamic payload logic
- * [#4827] Upper limit on the number of contacts to import from EDS is
- hard-coded to 500
- * [#4808] Fix put call on/off hold
- * [#4808] Implement early RTP start for incoming calls
- * [#4808] Audio stream is no longer start within RTP session.
- * [#4808] Removed coupling between audio layer and and RTP session
- * [#4702] Start audio rtp session as soon as it is created
- * [#4702] Init timestamp to 0
- * #4702: Send RTP packets immediately, no need of outgoing queue
- * [#4784] Update dbus-c++ version from gitorious
- * [#4702] Update RTP timeouts
- * [#4702] Lengthen RTP timeouts
- * [PATCH] Fixed compatibility with old libtool versions.
- * [PATCH] Accept older libebook (Maemo 5 has 1.4.2)
- * [PATCH] Fixed double-free error in preferences dialog
- * [PATCH] Fixed building of sflphone-daemon on Maemo5
- * [PATCH] Improved Gnome client initialization error handling. 1. It
- no longer segfaults when sflphoned isn't available. 2. User is
- provided with GUI error dialog.
- * [PATCH] Improved autogen.sh scripts 1. They do not require bash
- anymore 2. Added workaround for Debian bug #565663 3. Replaced
- manual autotools invocations with single autoreconf call 4. Non-zero
- return status on failure
- * Revert "[#4468] libtool <= 2.2 doesn't have LT_INIT macro so
- AC_PROG_LIBTOOL should be used instead."
- * Revert "[#4468] Libebook 1.4 is sufficient"
- * Revert "[#4468] Apply big path on dbus communication system"
- * [#4468] Apply big path on dbus communication system
- * [#4468] Libebook 1.4 is sufficient
- * [#4468] libtool <= 2.2 doesn't have LT_INIT macro so AC_PROG_LIBTOOL
- should be used instead.
- * [#4639] Fix determining default addressbook if this property is not
- set in gconf
- * [#4639] Fix memory leaks in Addressbook
- * [#4637] Fix opening default addressbook at sflphone init
- * [#4622] Free yaml events while parsing configuration file
- * [#4623] Fix conditional jumps based on uninitialized variable
- * [#4622] Fix leaks in yaml serialization engine
- * [#4616] Fix addressbook warnings
- * [#4514] Adjust RTP timestamp
- * #4527: Rename Karmic libyaml and Celt package in debian control file
- * #4495: Rework addressbook opening loop
- * [#4524] Increment RTP count when sending data
- * [#4524] DO NOT start RTP session twice
- * [#4367] Use PKG_CHECK_MODULE for celt
- * [#4367] Fedora package celt as celt (not libcelt)
- * [#4367] Astyling
- * [#4367] Update .po files
- * [#4367] Fix segfault in gensin
- * [#4354] Make celt a direct dependency on launchpad opensuse build
- service
- * [#4367] Make celt a required package, option --without-celt valid
- * [#4367] Fix zrtp timestamping error
- * [#4367] Fix audio zrtp timing
- * [#4367] Dispatch ZRTP packets
- * [#4367] Fix segfault when unloading account map
- * [#4367] Fix zrtp session
- * [#4367] Implement on packet receive
- * [#4367] use symetric audio rtp session, not dual
- * [#4367] Reduce packet receive/sent timeout
- * [#4367] Reduce RTP timeouts
- * [#4367] Move speaker data receive
- * [#4367] Move speaker data receive
- * [#4367] Move receive speaker data method
- * [#4367] Remove debug in rtp session
- * [#4367] Fix g722 codec clock rate
- * [#4367] Fix noise suppression initialization
- * [#4367] Fix segfault in RTP mic fadein method
- * [#4367] Refactor mic data encoding in rtp session
- * [#4367] Implement RTP main loop
- * [#4367] Fix compilation problem
- * [#4367] Fix AudioRtpclass using TRTPSessionBase
- * [#4367] Fix AudioRtpSession putDtmfEvent shadowing
- * [#4367] Fix AudioRtpSession putDtmfEvent shadowing
- * [#4367] Refactor RTP session (phase 2)
- * [#4367] Refactor RTP session (phase 1)
- * [#4367] Remove Redeclaration of SymetricAudioRtpSession in
- rtpfactory
- * [#4265] Add continue statement in for loop for invalid addressbook
- * [#4261] Makes addressbook initialization more robust
- * [#4257] Add maverick in build system
- * [#4233] Add sdp related unit tests
- * [#4233] Add condition and signal in two incoming call test
- * [#4243] Fix segfault in AudioSrtpSession
- * [#4243] Fix memory leak in AudioSrtpSession
- * [#4243] Make audio srtp optional in for incoming call
- * [#4243] Add boolean variable to make sure remote crypto context
- initialized only once
- * [#4243] Add documentation to AudioSrtpSession
- * [#4243] Use 80 bits authentication tags by default
- * [#4243] Init audio srtp remote crypto context in
- call_on_media_update
- * [#4243] Move SDP negotiastion in mod_on_rx_request
- * [#4243] Implement initLocalCryptoInfo to be called at different
- momment
- * [#4243] Init init local crypto context in when initializing audiortp
- * [#4243] Change key length according to sdes negociation
- * [#4243] Associate callid to accountid for incoming calls
- * [#4242] Fix no SDES keys in IP2IP calls
- * [#4242] Fix no SDES keys in IP2IP calls
- * [#4233] Test for call on/off hold
- * [#4233] Add two incoming call test
- * [#4233]
- * [#4233] Add 2 outgoing simultaneous call unit tests
-
- -- Julien Bonjean <julien.bonjean@savoirfairelinux.com> Fri, 30 Sep 2011 13:51:04 -0400
-
-sflphone-daemon (0.9.7~rc1~ppa1~SYSTEM) SYSTEM; urgency=low
-
- ** 0.9.7~rc1~ppa1~SYSTEM **
-
- * [#2462] Set explicitly the transport on incoming call too
- * [#2462] fix typo
- * [#2462] Use different address for SDP and call IP
- * [#2462] Use published address in SIP-SDP
- * [#2181] Fixed changelog files
- * [#2181] Updated spec file
- * [#2402] Fix pointer to int conversion warning (atoi)
- * [#2402] Remove daemon warnings, make indent
- * [#2459] Make sure the stream is opened when the call is answered
- * [#2402] Add conference related picture in documentation
- * [#2443] Not much ...
- * [#2399] Fix dialing display problem
- * [#2450] Fix incoming call already in conference crash
- * [#2399] Display peer name on the first line and peer number on the
- second
- * [#2450] Handle 403 FORBIDDEN when refused
- * [#2447] Bind offHold/onHold actions to button in gtk client
- * [#2447] Bind hangup action to button for conference
- * [#2447] Add conference action in gtk client's ToolBar
- * [#2381] Disable the password hashing in config file
- * [#2402] Cleanup
- * [#2366] Set callback to null when deleting Pulseaudio streams
- * [#1313] Fix main buffer unit test
- * [#1313] Fix audio layer unit test
- * [#2315] Hide pw in security tab, display when editing, sync with
- basic tab
- * [#1313] UnitTest change AudioRtpSession for AudioSymetricRtpSession
- instance
- * [#2402] Code cleanup
- * [#2444] Add debug to catch occasional crash when loading client's
- config
- * [#2444] Add debug info to catch occasional crash when loading config
- dialog
- * [#2402] Restore Call menu translations
- * [#2403] Use the published address if checked in GUI
- * [#2442] Add protection test in sdp
- * [#1841] Reapply pjsip patch concerning DNS SRV resolution
- * [#2384] Tags incoming call as direct SIP call, if applicable
- * [#2402] Change the monkey face
- * [#2315] Enable user to display password in clear text
- * [#2434] Force optimization level at 2
- * [#2284] Fix dbus_get_all_ip_interface compilation warnings
- * [#2431] Popup main window on incoming if applicable
- * [$2402] Fix simple warnings
- * [#2402] Fix implicit variable init order in LibraryManagerException
- * [#2402] Fixing implicit variable initialization warnings in
- AudioRtpSession
- * [#2402] Revert atoi change, fixing codec list doubled entries
- * [#2402] Fix gpointer to gint conversion
- * [#2402] Fix pointer casting to integer different size warning in
- codec list
- * [#2402] Fix warning discarting qualifiers from pointer target
- * [#2402] Fix gtk tree view assignement from incompatible type warning
- * [#1669] Fix audio recording folder utf-8 non compatibility issue
- * [#2414] Clean up debugs
- * [#2414] Use transport set in iptoip Account and update it frm
- preference
- * [#2348] Use macro N_() to mark ui.xml strings as translatable
- * [#2414] Rename getSipAddress/setSipAddress functions
- * [#2407] Fix volume controls display
- * [#2407] Fixes dialpad
- * [#2383] Set ip to ip config when clicking apply button
- * [#2404] Update call-to script - Maxime Chambreuil
- * [#2405] Client handles unknown call in current state as well
- * [#2383] Add DBUS signal to send IPtoIP local address and port as
- string
- * [#2383] Add Ip to IP config change apply call back
- * Clonflict
- * [#2402] Code cleanup
- * [#2383] Do the same for IPtoIP (init localn ip with first in the
- list)
- * [#2383] Use first interface in the list if local addresss is not
- defined
- * [#2403] Clean up unuseful addresses/ports
- * [#2403] Use the IP profile SIP port as global SIP port
- * [#2383] Fix dbus_get_all_ip_interface warnings
- * [#2383] Take into account sameAsLocal when loading published address
- * [#2383] Tsake into account sameAsLocal option when saving published
- address
- * [#2383] Update local ip address in ip to ip config
- * [#2383] Save ip 2 ip local port in config
- * [#2406] Update toolbar at startup
- * [#2284] Remove redefinition warnings + speex warnings
- * [#2383] Fix security table in account config
- * [#2383] Save ip 2 ip network interface parameters in config
- * [#2403] Restore sip transport selector
- * [#2383] Fix filling the Localt IP Address on account creation
- * [#2383] Fix Gtk-Critical when checking STUN
- * [#2383] Fix reopening account configuration display issue
- * [#2383] Load IPtoIP local address and port in preference iptoiptab
- * [#2383] Add LocalAddress and Localport in Preference IpToIp tab
- * [#2403] Use the address and port associated to the account as often
- as possible
- * [#1753] Removed pjsip generated files
- * [#1753] Removed remaining milenage lib references
- * [#2383] Add _publishedSameasLocal variable in sipaccount
- * [#2383] Add PUBLISHED_SAMEAS_LOCAL variable in config
- * [#2383] Fix stun set active or not when opening config
- * [#2181] Added RPM 64bits dbus patch
- * [#2402] Code indentation
- * [#2313] Force $(HOME).cache directory creation at startup
- * [#2383] Separate network interface and published address in account
- config
- * [#2400] Change dbus service installation path to libdir
- * [#2382] Move TLS related published address options in security tab
- * [#2382] Indent accountconfigdialog.c
- * [#2181] Install libdbus-c++ in $pkglib instead of $lib
- * [#1753] Remove ILBC code and disable it by default in the configure
- * [#1753] Remove milenage directory
- * [#2382] Fix switching interaface instabilities
- * [#2396] Save local ip in account creation wizard
- * [#2284] Remove warning on hold
- * [#2387] Fixes history searching and filtering
- * [#1215] Add samplerate display in the GUI
- * [#1663] Voicemail icon reflects voice messages
- * [#2395] Fix account registration ( specifically with callcentric)
- * [#2386] Strip "sip:" on incoming call, fixing history call back
- * [#2181] Updated spec files
- * [#1215] Display codec name in calltree instead of status bar
- * [#2390] Move back nbCalls and stopStream higher in refuseCall
- * [#2392] Fix ringtone during call in IAX
- * [#2391] Stop audio streams when there is 0 calls only
- * [#2391] Add debug when call state is not valid
- * [#2390] Clear returns in IAXvoipLink::sendAudioFromMic() method
- * [#2380] Fixing IncomingCallNotification not regular
- * [#2339] Query conference at client startup
- * [#2339] Working conference querying at startup
- * [#2339] Add conference in call tree
- * [#2339] Primitives to query conferences at client startup
- * [#2320] Add account selection in history
- * [#2355] Temporary solution: do not delete pointer when removing
- account
- * [#2380] Change algorithm in AudioRtp to trigger an
- IncomingCallNotification
- * [#2274] Comment sdebug in MainBuffer flush method
- * [#2274] Add flushMain() in ManagerImpl::addStream
- * [#2274] Add getBufferID() method in ring buffer
- * [#2274] Fix warning, comment debug in ringbuffer's flush method
- * [#2274] Use AudioLayer flushMain() and flushUrgent() in ALSA
- * [#2274] Clean up unused variable warning
- * [#2274] Protect minbudffer pointer on flushing
- * [#2274] Fix playATone method which writing empty buffer in urgent
- ringbuffer
- * [#2274] Use audio layer flushUrgent and flushMain in createStreams
- * [#2274] Use flush audio calls from audiolayer
- * [#2274] Flush when peer answered call
- * [#2375] Flush main buffer in iax when answering a call
- * [#2274] Parse displayname using c++ string method
- * [#2375] Flush main buffer when off holding calls
- * [#2375] Flush main buffer mon RTP startup
- * [#2376] Use now Pulseaudio module-cork-music-on-phone
- * Updated OSC packaging
-
- -- Julien Bonjean <julien.bonjean@savoirfairelinux.com> Fri, 20 Nov 2009 14:00:02 -0500
-
-sflphone-daemon (0.9.7~beta~ppa1~SYSTEM) SYSTEM; urgency=low
-
- ** 0.9.7~beta~ppa1~SYSTEM **
-
- * [#1933] Cleanup debug
- * [#1933] Clean up debug
- * Fix mic
- * [#1933] Set the IAx format earlier
- * [#1933] Move IAX sendAudioFromMic outside if (call) statement
- * [#1933] Fix startstream when offhold in iax and add debug concerning
- codec neg.
- * [#2371] sflphone_notify_voice_mail: minor gettext message formatting
- cleanup
- * [#2371] select_account_cb: properly gettextize status message
- * [#2371] show_account_list_config_dialog: properly gettextize status
- message
- * INSTALL: Minor tidyup of core install guide
- * Add /sflphone-client-gnome/src/icons/Makefile to .gitignore
- * [#2181] Updated OpenSUSE files (tmp)
- * [#1933] Add debug for codec negociation for iax
- * [#1933] Get rid of getMicAvail and getMicData in audiolayer (not
- used anymore)
- * [#1933] Add "audio codec not determined" error in IAX
- * [#1933] Test flush data
- * [#1933] Do not need to start audio stream in iax anymore
- * [#1933] Protecting pointer
- * [#2284] Remove more compilation/execution warnings
- * [#2284] Cleanup debug in client, use DEBUG instead of g_print
- * [#2284] Clean up uimanager
- * [#2370] Remove warnings
- * [#2366] Clean up other debug
- * [#2366] Clean up debug
- * [#2366] Call pa_xfree explicitely in writeToSpeaker
- * [#2284] Remove address book warnings
- * [#2365] Fixes bad cast
- * [#2352] Fix continuous ringing when peer hangup and call not yet
- answered
- * [#2181] Added version support
- * [#2181] Fixed some minor issues
- * [#2360] Moved MainBuffer from AudioLayer to ManagerImpl
- * [#2352] Makes getMainBuffer() everywhere
- * [#2352] Use 50 sec latency on pulseaudio stream creation
- * [#2352] Add alsa debug
- * [#2359] Update repository documentation
- * [#2354] Move pulseaudio disconnectAudioStream after stopping main
- loop
- * [#2352] Adjust nb byte copied in pulseaudio according to
- writeableSize
- * [#2352] Specify pulseaudio tlength parameters using pa_usec_to_bytes
- * [#2322] Convert italian translation to UTF-8
- * [#2357] Fixes window size
- * [#2357] Display only actionnable tool item
- * [#2333] Update streams parameters
- * [#2347] Use GNOME user settings for Menu and Toolbar appareance
- * [#2349] Load/Save properly audio params
- * [#2322] Update translations from Launchpad
- * [#2181] Added Francois Marier script
- * [#2350] Remove non-valid test
- * [#2181] Updated launchpad packaging
- * [#2333] Fix Pulseaudio Capture
- * [#2333] Use pulseaudio ADJUST_LATENCY flag and ALSA RT-SCHEDULING
- * [#2333] Pulseaudio Interpolate timing
- * [#2333] Change (again) Pulseaudio settings to fit logiteck usb hdw
- requirement
- * [#2333] Adjust pulseaudio fragment size to 4096 (max sflphone's
- frames per buffer)
- * [#2284] Remove recurrent compilation warning (g++ linker problem)
- * [#2333] Safer Audiostream parameters
- * [#2333] Fix alsa playback to reduce underrun
- * [#2333] Better audiostream parameters
- * [#2181] Updated version management
- * [#2333] Exclusive test in playback loop
- * [#2181] Updated build system
- * [#2333] Less underrun with these value
- * [#2333] Update playback audiostream parameters
- * [#2333] Lengthen the audio buffer reduce number of underrun in
- pulseaudio
- * [#2333] Add ALSA recovery functions for underrun (begin)
- * [#2333] Add pa_stream_trigger in pulse audio underrun callabck
- * [#2048] Reduce prebuffering in pulseaudio (which affect incomming
- calls' plbck)
- * [#2316] Do not display any icons to the right on the history tab
- * [#2333] Comment pa_stream_trigger in pulseaudio underrun
- * [#2333] Modify pulseaudio streams parameters
- * [#2318] Fix transfer tool button double signal
- * [#2181] Updated
- * [#2333] Fix ALSA ringtone
- * [#2333] Flush all main buffer before starting audio
- * [#2333] Open/Close Alsa thread between calls while there is no audio
- * [#2333] Add debug message and test condition on starting playback
- and capture
- * [#2181] Fixed gnome client makefile
- * [#2181] Updated
- * [#2308] Remove getTelephoneTone debug
- * [#2308] Change plughw for default in ALSA
- * [#2308] Oups, forgot to change function name in audiolayertest.cpp
- * [#2308] Cleanup in pulseaudio code (debug, function name)
- * [#2308] Fix pulseaudio stream closing assertion failure
- * [#2308] Moved pulseaudio mainloop locking from AudioStream
- disconnect stream
- * [2308] Fix latency at the beginning of a call, when playing DTMF and
- wehn starting tone
- * [#2181] Updated karmic
- * [#2317] [#2319] Fix address book toggle button contextual behaviour
- * [#2308] Stop stream when refusing a call
- * [#2308] Stop pulseaudio stream when peer hungup
- * [#2308] Fix tone and ringtone
- * [#2312] Display the STUN entry widget when opening the tab
- * [#2308] Implement two different callbacks for capture/playback in
- pulseaudio
- * [#2309] Open/close pulseaudio connections in startStream/stopStream
- * [2308] Leave pulseaudio stream running, do not cork/uncork them
- anymore
- * [#2295] Set gtk file chooser to None if nothing is set in
- configuration
- * [#1976] Add codec and conference documentation
- * [#2209] Fix recording in regard of resamling
- * [#2297] Update .gitignore
- * [#2297] Update translation files
- * [#2297] Add reference to our coding standards
- * [#2297] Remove old docbook code
- * [#2296] Reinit tls account settings after modification
- * [#2253] Add DcBlocker class to remove capture's dc offset
- * [#2034] Fixes for TLS transport to initialize
- * [#2284] Add silent build rule + client clean warnings
- * [#2274] Fix unserialize history items in cilent at startup
- * [#2274] Complete display name parsing and displaying
- * [#2274] Parse the Display Name in sip INVITE message
- * [#2050] Fix capture volume control in ALSA
- * [#1970] Volume controls disable when using pulseaudio
- * [#1970] Disable volume controls when using pulseaudio
- * [#2277] Fix direct ip2ip ZRTP enabling/disabling in ip2ip
- preferences
- * [#2181] Added launchpad debian files
- * [#2181] Added spec files for OSC
- * [#2274] Set display name for "Contact" sip header as the hostname
- * [#2181] Fixed daemon issues
- * [#2181] Fixed gnome client issues
- * [#1976] Remove warnings - need to fix the transfer
- * [#2006] Add init is_rec variable in ManagerImpl
- * [#2006] Update codec display on call selection
- * [#2006] Restore double click actions in history and contact calltree
- (GTK)
- * [#2176] use XDG_CACHE_HOME when initializing sfl.zid file
- * [#1976] Fix calltree switching from history
- * [#2209] (Re)Fix cache for zid
- * [#2209] Clean up debug messages
- * [#2209] Clean debug messages
- * [#2209] Fix trasnfering a call during a conference
- * [#2209] Speex decode must return the number of bytes
- * [#2209] Change frameSize speex 32kHz
- * [#2209] Fix speex codec framesize
- * [#2209] Reinit converterSamplingRate in RTP sessions
- * [#2209] Change speex ultra wide band framesize
- * [#1747] Add pixmap data
- * [#2252] Fix Receiving a server error 488 crashes the callee
- * [#2209] Fix iax low rate packate sending
- * [#2209] Clean up debug messages
- * [#2209] Add resampling changes for IAX
- * [#2209] Clean up resampling code
- * [#2209] Fix latency introduced by pulseaudio
- * [#2209] Fix initialization of mainbuffer's internal sampling rate
- * [#2176] Fix upsampling buffer size in audiolayer
- * [#2209] Add dynamic converter sampling rate in audiortp sessions
- * [#1747] Fixes runtime warnings
- * [#1747] Remove from repo
- * [#1747] register our icons to be used as stock icons
- * [#2209] Fix number of byte in alsa's write to speaker
- * [#2209] Fix putting non-resampled data in RTP's mainbuffer
- * [#2209] Add alsa resampler
- * [#2209] Add a samplerate converter in PulseLayer
- * [#2209] Add mainbuffer's internal sampling rate and flushall method
- * [#2176] Add mainbuffer stateInfo debug method
- * [#2209] Resampling is optimal using SRC_LINEAR not SRC_FASTEST
- * [#2176] Remove debug recordings
- * [#2176] Fix Holding a conference participant on new calls
- * [#2224] Add confID in callable object
- * [#2176] Fix putting onhold a call participating to a conference when
- pressing new call
- * [#2176] Reset auidio buffers when adding streams (rtp, audiolayer)
- * [#1976] Use xml to describe toolbars - Add a naviguation toolbar
- * [#2176] Remove conference default_id in joinParticipant
- * [#2176] Display error message in alsa's snd_pcm_avail_update call
- * [#2176] Alsa mic avail data debug
- * [#2176] Add some debug message for mic loss problem
- * [#2176] Flush mic ring buffer when offholding a call
- * [#2176] Reset ringbuffers' readpointer when adding main participant
- * [#2176] Fix getAvailData algorithm
- * [#2176] Reset ringbuffer's readpointer when adding a new participant
- to a conference
- * [#1744] Regex object renamed to Pattern. Previous attempt at
- providing
- * [#2176] Fix detach main participant problem when adding new one
- * [#1976] Use right domain to translate
- * [#1976] Add xml menu description
- * [#2176] Store a list of confernece participant in client
- * [#2176] Fix add participant, joinparticipant methods
- * [#2181] Do not install dbus-c++ headers + add return value
- * [#2176] Fix minor call handling instabilities
- * [#2174] Fix incoming IP call contact address
- * [#2211] Add test to protect NULL pointer
- * [#1163] Add Advanced account configuration section
- * [#2176] Add some usefull comments and debugging info
- * [#2176] Add conditions to display security icons in conference
- * [#2176] Fix detaching one participant while keeping communication to
- others
- * [#2176] Reenable userActive.svg in call tree
- * [#2176] Make user active blue (not red)
- * [#2176] Fix user active picture
- * [#2176] Fix "hidden" merge conflict in sipvoiplink
- * [#2176] Remove iax audio stream on peer hungup
- * [#2174] Multiple UDP transports functional (TESTED with 2 accounts
- and 3 calls)
- * [#2176] Fix fix audio stream binding in iax
- * [#2174] Create a default UDP transport + use tp selector for dialogs
- also
- * [#2176] Register iax audio stream in mainbuffer
- * [#2176] Fix getAudioCodecName in IAXvoipLink
- * [#2176] Fix iax account init
- * [#2176] Handle multiple account using the same sip transport
- * [#2165] Add .png files
- * [#2176] Small fixes concerning dtmf
- * [#2176] Fix make uninstall in codecs
- * [#2174] remove stund makefile generation
- * [#2176] Add conference lock
- * [#2174] Add transport selector for multiple accounts
- * [#2176] Change userActive picture from red to blue
- * [#2176] Fix security pixbuff in calltree
- * [#2176] Replace sfl.zid in .cache/sflphone instead of .sflphone
- * [#2176] Fix add call description
- * [#2176] Remove detach button from toolbar
- * [#2176] Fix calltree call description state and state code in
- conferences
- * [#2176] Fix pulse audio double free
- * [#2176] Fix conference selection
- * [#2174] Clean up - remove stun settings in client network
- configuration panel
- * [#2174] Remove voviva stun code
- * [#2174] Rsolve STUN with pjsip - DO NOT WORK
- * [#2165] Add user svg
- * [#2165] Debugging sip call failed
- * [#929] Link against uuid if installed
- * Oops
- * Fixed bugs related to libsexy (with GTK < 2.16)
- * [#929] Remove uuid-dev dependency in the core
- * [#2165] Debugging no negociated codecs at communicatio start
- * [#2165] Fix calltree bug (gtktreestore instead of gtkliststore)
- * [#2165] Fix several merge problems
- * Updated opensuse packaging script
- * [#1163] Add missing figures
- * [#1163] Update INSTALL file
- * [#2165] Fix IAX
- * [#2165] Add recordabe interface
- * [#2165] Finish recording refactoring for call (not for conference)
- * [#2165] Enable speaker recording for two different calls
- simultanously
- * [#2165] Implement call recording using the Recordable interface
- * [#2165] Add get and set to AudioLayer's audio recorder
- * [#2165] Add class recordable from which inherit call and conference
- * [#2006] Fix G722 and Speex 8khz codec conferencing
- * [#2006] add recording of audio buffers
- * [#1163] Add general settings section
- * [#1163] Fixes makefile error
- * [#2006] Fix some minor issues
- * [#2006] Drag a conference call on another conference call
- (difference conferences)
- * [#2006] Fix dragging a conference on itself
- * [#1744] Integrating some of the needed regular expression patterns
- in order
- * COmplete call features
- * [#1744] Added support for named subgroup in the Regex object. Also,
- new
- * [#1744] Adds thread safety features, compile() and setPattern()
- methods to the Regex class.
- * [#1744] Fix inconsistency in the finditer method from the last
- commit.
- * [#1744] Added regex pattern object built on top of libpcre. To be
- used
- * [#1744] Initial commit towards implementing RFC4568. Unimplemented
- in the
- * [#2157] Hide "security" and "advanced" tabs for IAX under account
- * [#1163] Add call features section
- * [#2006] Add joinConference capabilities
- * [#2006] Add dbus joinConference signal
- * [#2006] Drag a conference call onto a conference to add it
- * [#1163] Add addressbook section
- * [#2006] Drag a conference call onto a single call to create a
- conference
- * [#2006] Expand rows automatically
- * [#2006] Add minimal multiple conference handling
- * [#2006] Add atached/detached conference icons
- * [#2006] Add function processRemainingParticipant
- * [#2006] Deep refactoring, fix hangup bug
- * [#1163] Update documentation - Accounts part
- * [#1976] Integrate user doc to gnome client build system
- * [#2122] Remove double inclusion in dbus-c++/src/Makefile.am
- * Remove pjproject version number
- * [#2006] Fix peerHungup
- * [#1976] Make Yelp accessible from the GNOME client (need to install
- the sflphone.xml first)
- * [#2006] Fix multiconferencing hangup
- * [#2006] Fix hangup calls in a conference
- * [#2150] Make IAx2 reappear
- * [#2006] Fix detach participant on multiple call
- * [#2006] Can remove rining call from a conference
- * [#2006] Reinit confID when removing a participant
- * [#2006] Remove get isCurrentCAll in hangup/peerhungup (SipVoipLink)
- * [#2006] Fix refuse call
- * [#2006] Fix answerring incoming call
- * [#2006] Refactor conference's participant list
- * [#2101] Re-integrate test compilation in main build system
- * [#2101] Make the test directory compile
- * [#2136] Restore history functionality
- * [#2006] Fix binding main participant to himself
- * [#2006] Fix add current/incoming/onHold participant to an existing
- conference
- * [#2006] Fix add incoming calls to an already created conference
- * [#2006] Fix remove stream
- * [#2006] Fix detachParticipant/removeParticipant switchCall ids
- * [#2006] Fix adding a call in conference having state "CURRENT"
- * [#2006] Remove/add main participant from conferences
- * [#2006] Hold/unHold conference
- * [#2006] Detach a partcipant from drag n drop
- * [#2006] Hangup a conference
- * [#2006] Add hold/unhold conference dbus messages
- * [#2034] gtk-ui fix under the "basic" tab.
- * [#2006] Fix dragging calls on conference calls
- * [#2006] Fix detach participant from a conference
- * [#2034] Added default message is status bar under the account config
- dialog
- * [#2112] Fix a crashed caused when a non-md5 password was sent to
- pjsip.
- * [#2006] Detach participant by ID
- * [#2006] Fix addParticipant method in managerImpl to handle
- incoming/answered calls
- * [#2006] Add addParticipant method in managerimpl and related dbus
- messages
- * [#2111] Added the ability to configure zrtp on sip.sflphone.org from
- * [#2106] Fixed problem in the account assistant under gtk-ui. Also,
- assistant.c
- * [#2006] Fix dragging a conference call on another conference call
- (same conference)
- * [#1904] Small UI fix. Assistant was moved from "Call" to "Edit"
- menu.
- * [#1904] Fix a wrong label under gtk-ui.
- * [#2034] Renaming and source code splitting.
- * [#2034] Status bar added to account window to better reflect the
- registration
- * [#2006] Make calltree_remove_call recursive (for GtkTreeStore)
- * [#1110] Small gtk-UI fix in the account window (alignment).
- * [#2006] Fix remove conference, display children which are still
- active
- * [#2006] Recursive function call in calltree_update_call
- * [#2006] Add multilayered capabilities to calltree (GtkTreeStore)
- * [#2006] Implement remove conference in calltree
- * [#2034] Now useless as Direct Ip calls settings moved under
- Preferences.
- * [#2034] Edit/add buttons were set insensitive all the time under
- gtk-ui.
- * [#1887] Information about the state of the current SIP call is
- displayed
- * [#2006] Add call tree remove callback
- * [#2006] Fix create_conference function
- * [#2006] Update conference_added_cb to add new conference to the list
- * [#812] Added new tab under GTK-ui Preferences. Moving Direct Ip
- Calls from
- * [#2121] Disable temporarily test compilation
- * [#2006] Fix conferencelist to handle conference_obj_t instead of
- gchar
- * [#2006] Add conference_obj structure
- * [#2121] Update version
- * [#2006] Fix conference selection
- * [#2101] Use the new source tree to fetch the right object files
- * [#2006] Add conference in calltree
- * [#2006] Add Dbus signal conference added/removed/changed
- * [#2006] Add getConferenceDetails call on dbus
- * [#1904] Registration expire now appears as a spin box under gtk-ui.
- * [#812] Fixing a segmentation fault caused by a non-existing account
- ID
- * [#2006] Add getConfList method over dbus
- * [#2006] Add a conferencelist data structure in client-gnome
- * [#812] Defaults value are now sent if a non-existing account is
- requested
- * [#2006] Add sflphone action sflphone_join_participant
- * [#2006] Fix buffer read pointer problem deletion
- * [pjsip] Attempt at fixing via header incompatibility with
- Freeswitch.
- * [#1797] forget something
- * [#2006] Add call new state conferencing in deamon
- * [#2006] Remove addParticipant method for conference, use
- joinParticipant only
- * [#1163] Update INSTALL documentation
- * [#812] Msec/sec values were not taken into account.
- * [#1797] Make pjproject-1.4 compile
- * [#2006] Add Detach participant method
- * [#2006] Dragndrop fully functional with INCOMING and HOLD call
- * [#1797] Add pjproject-1.4
- * [#1797] Remove pjproject-1.0.3
- * [#2006] Get call state in conference related function
- * [#2006] Add joinParticipant (conference) method in ManagerImpl
- * [#2006] Add joinConference DBUS message
- * [#2006] Store the previously selected call_id on dragndrop
- * [#2006] Fix GValue pointer unref in selection callback
- * [#2006] Store dragged call_id
- * [#2006] Update drag_data_received_cb callback to manipulate CallIDs
- * [#2006] Add dragndrop signals
- * [#2006] Set calltree reordable
- * [#812] Adds the ability to create a TLS listener in case the user
- requests
- * [#812] Adds the ability to configure local/published address from
- * [#1883] Move switchCall in onHoldCall function
- * [#812] Deals with the published address/port problem when
- integrating TLS.
- * [#1883] Switch call id in managerimpl when peerHungUp
- * [#1883] Switch call id before hangup
- * [#1883] Add usefull and permanent debug info for conference
- cretion/deletion
- * [#812] Fix various segmentation faults related to Direct IP kind of
- calls.
- * [#1883] Fix deletion of std::map elements using iterators
- * [#2014] Add libzrtpcpp build dependency
- * [#1883] Still some for loop test ambiguity (while loop instead)
- * [#1883] Fix for loop initial test ambiguity (use while loop instead)
- * [#1883] We must discard data in urgent ring buffer if data is get in
- mainbuf
- * [#1883] Fix availForGet same id for ringbuffer and readpointer
- * [#812] Match "sips" as a Direct IP Call when the user enter a sip
- uri
- * [#812] Fix segmentation fault related to SIP URI creation.
- * [#812] Towards integrating multiple tls listeners at the same time.
- This
- * [#1883] Add debug messages in conference and fix mainbufferTest
- * [#812] gkt-ui fix. Private key must be fed as a filename and not as-
- is.
- * [#812] TLS integration within sipvoiplink and pjsip. Also,
- configure.ac
- * [#1883] Fix Alsa/Pulse mallocation
- * [#1883] Fix data corruption in AudioRtp's micData buffer
- * [#812] Full dbus integration for all the tls related options under
- gtk-ui.
- * [#1883] Fix memory leaks in audiortp session
- * [#1883] Fix mem leaks in audio rtp
- * [#812] Fix setAccountDetails where TLS_ENABLE was set to the value
- * [#812] Small gtk-ui fix.
- * [#811][#812] Small gtk-ui fix.
- * [#812] Introduced a mechanism for configuration files that makes
- possible
- * [#812] New dbus bindings added. Also, configuration compliance was
- enforced
- * [#1881] Remove default buffer from MainBuffer (update unit-tests)
- * [#1881] Add ring buffer read pointer tests
- * [#1883] Fix issues in ringbuffer reader pointers
- * [#2034] Implementing a new configuration dialogue for TLS transport
- settings
- * [#1883] Add some usefull debug and safety checks
- * [#2028] Notify the client with libnotify when the zrtp negotiation
- failed.
- * [#811] Harmless no to throw an exception, an makes the application
- less
- * [#2028] A minidialog is showed to the user under sflphone-client-
- gnome
- * Removed useless file.
- * Ignoring Makefile in src/widget
- * [#2027] Fix segmentation fault when showMessage callback is called
- after
- * [#2026] keyExchange was set to ZRTP instead of "1"
- * [#2024] Fix the wrong summary at the end of the assistant.
- * [#1883] Fix mnagerimpl conference map insertion
- * [#1883] Add Mutexes in MainBuffer
- * [#811] Gtk ui was not presenting the right information about zrtp
- for
- * [#2023] security icons were not installed in sflphone-client-gnome.
- * [#2021] Fix a mistake in the readme from sflphone-daemon that gives
- wrong
- * [#811] The current SRTP mode was not properly displayed for the
- IP2IP
- * [#1743] Re-implementation of the "automatically remove error dialogs
- [...]"
- * [#2017] [#2019] Fix the inability to dial a number and place a
- registered
- * [#811] Final re-integration of ZRTP support in the main branch from
- 0.9.6
- * [#1883] Fix map insertion methods
- * [#811] Combo box now is now set to the active key exchange method
- * [#811] ZRTP options now configurable back again from the Gtk UI.
- IP2IP
- * Updated hostname for git clone
- * [#1883] Add minimal functionalities to create a conference
- * [#811] re-integration of all the methods and signals on dbus.
- ManagerImpl
- * [#811] Got out of a precarious position were nothing would compile.
- * [#1976] Build documentation squeleton with docbook
- * [#1883] Add sflphone-client "addParticipant" button for conference
- * [#1994] Better organize the source directory structure. New
- subdirectories
- * [#1883] Add a simple Conference class
- * [#1882] Use static audio buffer in Pulse and ALSA layer (instead of
- malloc)
- * [#811] First commit toward re-integration and refactoring of ZRTP
- * [#1882] Flush RTP ring buffer before entering mainloop
- * [#1882] Fixed MainBuffer::UnBinCallID() in case there is no
- ringbuffer
- * [#1882] Test (and fixe) high level conference and mixing
- functionalities
- * [#1772] Apply patch to compile on fedora (sent by Marcin
- Zajączkowski <mszpak@wp.pl>)
- * [#1882] Update Bind, unBind call_id in MainBuffer
- * [#1959] This adds the ability to store password as an MD5 Hash in
- the
- * [#1538] Fixes rules compilation
- * [#1930][#1931] Fixed a mistake (again) related to index and
- credential count
- * [#1753] Remove ILBC from pjproject - Hacks in pjsip
- * [#1930][#1931] Credential was not selected properly using realm
- * [#1882] Finilize multiple reading pointer in RingBuffer
- * [#1538] Remove configure from autogen.sh to respect debian upstream
- authors policy
- * [#1773] Remove generated files from repo
- * [#1791] Use XDG_CACHE_HOME to save pid file
- * [#1791] Fixes path to save history
- * [#1791] Fix debian installation scripts
- * [#1930][#1931] Settings are now taken into account in the server.
- * [#1882] Add ringbuffer default ring buffer pointer in methods
- involving mStart
- * [#1882] Add default ringbuffer pointer
- * [#1882] Add RingBuffer multiple read pointer basic functionnalities
- * [#1882] Fix MainBuffer flushData unit test
- * [#1930][#1931] Ability to save and retreive the configuration from
- * [#1882] Added Multiple CallID mapping to MainBuffer
- * [#1791] Not much
- * [#1791] If XDG env variables are not null but empty, use default
- ones
- * [#1791] Make XDG_CONFIG_HOME writable
- * [#1930][#1931] Partial commit. Not working yet. Cannot delete
- account
- * [#1881] Fixed alsa capture latency problem
- * [#1881] Fixed Alsa capture temporarily
- * [#1930] [#1931] Partial unbroken commit providing the ability to
- * [#1881] MainBuffer implemented in AudioLayer/AudioRTP
- * [#1881] Add discard and flush unit-tests
- * [#1881] Add discard and flush functionnalites to MainRingBuffer
- * [#1881] Add availForGet in MainBuffer
- * [#1881] Add availForPut function to MainBuffer
- * [#1880] Remove AudioRTP* pointer from SipVoIP (reapered while
- merging master)
- * [#1881] Add a map between call id and coresponding ring buffer
- * [#1855] Refresh pot file and upload on Launchpad
- * [#1881] MainBuffe now robust to false ids on getData and putData
- * [#1881] Fix big big big memory leak
- * [#1881] Add getData and putData to mainBuffer
- * [#1881] Unit-test basic ring buffer functionnaities
- * [#1881] Add class MainBuffer and basic buffer creation unit-tests
- * [#1880] Fix call transfer (step2) issues
- * [#1880] Moved AudioRtp* pointer from SIPVoIPLink to SIPCall class
- * [#1791] Add postinst script to keep user data when migrating
- config/history file
- * [#1797] Make pjsip compile
- * [#1777] Code indentation
- * [#1791] Use XDG_DATA_HOME and XDG_CONFIG_HOME for sflphonedrc and
- history + unit tests
- * [#1746] Useless space does not appear anymore when volume sliders
- and
- * [#1643] GtkCheckMenuItem is used instead of icons for elements in
- the
- * [#1110] [#1668] STUN parameters are now located in the preferences,
- under
-
- -- Julien Bonjean <julien.bonjean@savoirfairelinux.com> Fri, 06 Nov 2009 11:23:15 -0500
-
-sflphone-daemon (0.9.6-SYSTEM) SYSTEM; urgency=low
-
- ** 0.9.6 **
-
- * Documentation on echo test
- * [redmine_down] codec names not displayed in total
- * [redmine_down] crash when hanging up a dialing call because tries to
- add it to history whereas no starttime
- * [#1927] alternate every time screen changed to call history
- * [#1886] clean code
- * [#1886] debug messages when loading history removed
- * [redmine_down] sflphone-kde icons
- * [#1855] Update language files
- * [#1502] Update version number
- * [redmine_down] setHistory at close
- * [#redmine_down] Handle PJ_DECLINE_SC as failure
- * [#1923] Fix segmentation fault when adding a new account
- * [#1923] Check on iterator before setting the config
- * [#1904] Added mnemonic to tabs in sflphone-client-gnome.
- * [#1905] The daemon was not sending the currentSelectedCodec signal
- on dbus when answering a call.
- * [#1922] Default values set to all account details
- * [#1886] Spinbox reg expire enables apply, and address book is not
- visible when disabled
- * [#1905] Bug fix for segmentation fault caused by an empty string,
- * [#1910] Warnings in test directory
- * [#1919] Error fixed
- * [#1855] Update russian translation - Hussein Abdallah
- * [#1910] Remove files
- * [#1919] fixed
- * [#1777] Code indentation
- * [#1918] fixed
- * [#1917] fixed
- * [#1910] Remove warnings compilation in src
- * [#1886] removed AccountListModel in configskeleton
- * [#1914]
- * [#1911] check previous and new port
- * [#1910] Remove compilation warnings in src/dbus and src/history
- * [#1910] Remove compilation warnings in src/audio
- * [1855] Update german translation - Sven Werlen
- * [#1909] removed
- * [#1906] Done
- * [#1904] The registration expire value is now configurable from the
- * Cleaned up debug messages.
- * [#1886] separated initCallItem in two functions
- * [#1886] reversed error in commit
- * [#1886] clean debug
- * [#1886] changed Name of classes and files
- * [#1886] clean
- * [#1870] In call_state_cb (dbus.c:126), _time_stop was overridden by
- the actual time.
- * [#1884] Added some new gpg flags to prevent tty warnings
- * [#1886] Clean audio config dialog
- * [#1886] No more compile warnings. + 1 comm
- * [#1872] Check if the user input is smaller than PJ_MAX_HOSTNAME.
- * [#1886]
- * [#1785] Fixed build when no new commit
- * [#1852] If chosen by the user, the hostname can now be solved and
- used
- * [#1871] * and # inverted back
- * [#1869] Conditional compilation that checks if
- * [#1309] removed test in main
- * [#1425] Put actions in SFLPhone window class instead of ui view,
- made a separate toolbar for screens.
-
- -- SFLphone Automatic Build System <team@sflphone.org> Mon, 27 Jul 2009 09:53:00 -0400
-
-sflphone-daemon (0.9.6~rc2-SYSTEM) SYSTEM; urgency=low
-
- ** 0.9.6~rc2 **
-
- * [#1755] Remove generated file
- * [#1753] restore ilbc ...
- * [#1866] Methods getSipPort and setSipPort now have an effect on the
- * [#1753] make pjsip compile without ilbc. Use ./autogen.sh --disable-
- ilbc-codec
- * [#1855] Fix error in russian translation
- * [#1805] Remove the old flawed signal mechanism which was failing in
- * [#1855] Refresh translation
- * Spanish translation finished + po README files updated + echo's in
- copy-in-clients
- * [#1850] Yun made the chinese HK-CN translation
- * [#1848] Fix transfer interface bug
- * [#1862] At install, kde client installs only french translation file
- * [#1841] A new fallback mechanism was added to the internal resolver
- in PJSIP.
- * Started AccountList model/view
- * [#1855] Remove po subdir in Makefile.am
- * [#1855] Fix typo error in sflphone-client-gnome
- * [#1855] Do not generate Makefile in sflphone-daemon/po
- * [#1855] Copy translation files into both clients dirs
- * [#1855] Remove po dir from sflphone-daemon
- * Comments added
- * [#1860] mailbox->voicemail...
- * make scripts executable
- * [#1855] French translation
- * [#1855] Chinese zh_HK partially filled...
- * [#1859] An unnamed pipe monitored by poll() was added. When we want
- to
- * [#1855] Sven completed the first part of the german translation
- * [#1855] Cantonese manually filled for already translated, almost
- equal strings
- * [#1855] Merge russian translation
- * [#1855] Spanish manually filled for already translated, almost equal
- strings
- * [#1855] Update german translation in ./lang/de
- * [#1858] This problem was fixed by removing a useless line in
- * [#1855] merged existing translations in lang/ sflphone.po's
- * [#1842] [#1843] An attempt at improving the expected behaviour that
- can't
- * [#1855] added po folder in gnome client and scripts for copying from
- common lang folder to clients
- * [#1853] Edit before call does nothing on call history
- * Put most language entries possible in common. From 300 to 250
- entries. Stays underscores problem. Scripts for copy in clients.
- * commit to merge master
- * [#1825] Changed "Bad authentification" to "Authentication Failed".
- * common po files
- * [#1753] Remove ILBC from pjproject
-
- -- SFLphone Automatic Build System <team@sflphone.org> Fri, 17 Jul 2009 19:12:44 -0400
-
-sflphone-daemon (0.9.6~rc1-SYSTEM) SYSTEM; urgency=low
-
- ** 0.9.6~rc1 **
-
- * Update some version number
- * [#1792] Creates .sflphone directory with permission 600. Also,
- "chmod 600" after
- * [#1810] GUI is now notified that the call failed. Also, a segfault
- was
- * [#1816] Address book search disabled when disabled address book and
- enabled it back plus button stays triggered
- * codeclistmodel + asynchronous loading of address book +
- enable/disable address book
- * [#1810] Now checking SDP answer after 200 OK. Still need to
- implement full
- * [#1794] Can't use the interface during a call
- * Updated translation files
- * Russian translation integrated
- * Codec list model/view started.
- * [#1807] Add configure.ac in pjproject-1.0.3
- * [#1787] closeRtpSession added in some places where it should have
- been
- * Use Item class for contacts and accounts
- * Comments + clean code
- * [#1794] Improved debug messages
- * [#1805] Replaced the old and unreliable mecanism that was was
- waiting for
- * [#1794] Can't use the interface during a call
- * [#1787] For those cases where no registered SIP account is
- configured
- * [#1797] Make pjsip compile
- * [#1787] Minor changes. Removed useless commented line. Changed order
- of
- * [#1777] Code indentation
- * [#1797] Update package generation with new pjsip version
- * [#1798] Does not hang up when the call is building up
- * [#1797] Update .gitignore with new pjsip version
- * [#1797] Remove generated files from repo
- * [#1797] Main build system now uses pjproject-1.0.3
- * [#1797] Add pjproject-1.0.3
- * [#1797] Remove pjproject-1.0.2
- * [#1796] Computing time optimization (samplerate conversion)
- * [#1787] _audiortp->start() moved away from offhold(),
- SIPCallAnswered()
- * [#1312] Added new states for calls initialized by other clients
- * [#1795] Crashes when adding a new account, checking it and applying
- * [#1782] Missing icons
- * [#1793] KDE client compilation problem
- * Fake ringtone files can no longer be set.
- * indentation
- * [#1312] Able to fetch to differentiate incoming/ringing call state
- * [#1784] Use DESTDIR variable in po Makefile - fix language file
- installation
- * [#1785] Fixed typo
- * [#1785] Fixed changelog update
- * [#1759] ./autogen.sh --prefix=/usr --with-debug to use optimization
- level 0
- * [#1773] Changed snapshot naming convention
- * [#1773] Removed gpg agent use, added repository cache cleaning
- * [#1759] Use optimization level 0 for repository, 2 for packages
- * [#1777] Code indentation/formatting
- * Translated new features in french
- * [#1785] Added missing changelog entry
- * [#1781] Window title is SFLPhone
- * [#1777] Add code indentation/formatting in the buil system
- * [#1774] Can't set voicemail number in KDE account creation wizard
- * [#1775] Can't modify account information for account created with
- the wizard
- * [#1771] Add a "Default" button in context menu to disable chosen
- prior account
- * [#1705]
- * [#1224] Remove generated file from the repo
- * [#1224] Remove generated file from the repo
- * [#1762] distclean target should remove kconfig generated files
- (settings.h, settings.cpp). Rename them?
- * [#1761] clear history button should really clear history
- * Dialpad works.
- * Implemented Dialpad widget instead of building it in main view.
- * Removed last occurence of the old config dialog, that made the build
- crash.
- * [#1755] Do not consider G722 as a dynamic payload elsewhere than in
- RTP layer
- * [#1753] Remove ilbc Makefile generation
- * [#1756] Implement a kde configuration dialog with kconfig xt and
- kconfigdialog class
- * [#1755] fix audiocodec folder parsing problem
- * [#1450] Reinit timestamp comparison in RTP, create session in
- newOutgoingCall
- * [#1753] Remove milenage third party code from pjsip
- * New Config Dialog integrated in GUI.(without codecs)
- * [#1753] Remove ILBC codec
- * kconfig started, tr2i18n -> i18n, icons folder, accountList changed
- * [#1705] Fixed Audio RTP thread creation/start
- * [#1714] Fix codec negociation result handling
- * [#1678] Fix audiortp payload setting
- * [#1678] Put bac putData method in rtp
- * [#1669] gtk_file_chooser_get_filename() support UTF-8 by default
- * [#1735] Add conditions to sdp update call if call declined
- * [#1737] substr of recordings destination folder to remove "file://"
- should be done in client rather than in daemon
- * [#1731] Enlarge audio stream buffer size
- * [#1714] Missing true
- * [#1317] Fixed Mandriva timeout
- * [#1317] Changed tag convention
- * [#1317] Cleaned git-dch
-
- -- SFLphone Automatic Build System <team@sflphone.org> Fri, 10 Jul 2009 15:49:56 -0400
-
-sflphone-daemon (0.9.6~beta-SYSTEM) SYSTEM; urgency=low
-
- ** 0.9.6~beta **
-
- * spec files for mandriva and opensuse updated with buildrequires
- libqt4-dev >=4.3
- * [#1700] Cannot build on ubuntu 8.10 and a few other distribs
- * [#1502] Update version number where applicable
- * [#1642] Update client icons
- * [#1450] Clean up useless debug and comments in sipvoiplink and
- audiortp
- * [#1450] Remove Semaphore object in AudioRtp thread deletion
- * [#1450] Audio RTP init now synchronized with Sip/SDP
- * [#1693] kde client crashes when changing codecs order/activation
- * [#1450] Deep refactoring of audiortp
- * [#1450] setRtpSessionRemoteIp
- * [#1689] getCallList at start
- * [#1224] Change path in package files
- * [#1450] Audio RTP initialized only once, payload and remote ip set
- at runtime
- * [#1450] Add setRtpSessionMedia and setRtpSessionRemoteIp address
- * [#1642] Make GNOME GUI fresher and younger ;)
- * [#1686] Status bar displaying used account
- * added sflphone-kde icon so that it compiles
- * [#1659] Ending a call causes the daemon to crash
- * corrected introspection XMLs, po files...
- * [#1211] g722 media descriptor in codecDescriptor
- * [#1310] Install sflphoned in $(prefix)/lib/sflphone
- * [#1502] Do not install test binaries and dbus utilitaries
- * [#1224] hack for pjsip build system!
- * [#1224] Remove pjsip binaries from repo
- * [#1224] Upgrade to pjsip 1.0.2
- * [#1658] About SFLphone (bugs)
- * [#1658] About SFLphone
- * [#1660] Displaying all dialed numbers in a call
- * Tested status bar.
- * [#790] Optimize pulse audio streams parameters
- * [#1678] Some usefull debug messages for mutex/semaphore deadlock
- problem
- * [#1669] Add/remove some usefull/unusefull debug
- * [#1665] Fix latency related to pulse audio stream openning/closing
- * [#1457] Make the menus and panels accessible in french
- * [#1457] Improve broken keyboard accessibility in menus and conf
- panels
- * [#961] Instanciate only once the searchbar icons
- * [#961] Restore transfer fonction
- * [#961] Filter on the history type OK
- * [#961] Fix compilation problems on hardy/intrepid
- * [#1157] Commit missing files
- * [#790] Reduce number of start/stop streams call on pulse audio
- * [#1639] kde client crashes when no account registered
- * [#1620] Fix the searchbar
- * [#1620] Get back caltree as it was during gtkcritical area
- * [#1620] Add history filter reinit function
- * [#1335] Add a missing label in address book preferences
- * [#1561] Update russian translation - Hussein Abdallah
- * [#1605] Fix edit menu french translation
- * [#961] Enable to search in the history according to the call type
- * [#1449] Searchbar does not work anymore
- * [#961] Add popup menu on the entry primary icon for history
- * [#1317] Fixed KDE client package dependency
- * [#936] speex 32 khz integration completed
- * [#936] Use 320 frame size
- * [#936] Test using a frame size at 320 smpls
- * [#1214] Enable / Disable history
- * [#1607] Fix compilation problem for ubuntu 8.10 (libsexy)
- * [#1313] Implement processDataEncode processDataDecode in audiortp
- * [#1613] codec list order can't be set
- * Better handling of localisation + added languages + corrected
- warnings + begginning of new config dialog with kconfig + 14px
- account leds
- * [#1214] Save and load history according to the limit timestamp +
- unit tests
- * [1609] Fix call number copy/paste feature
- * [1607] Restore clear action icon in searchbar
- * [#936] Try to decode using 1280 samples
- * [#936] Add some debug
- * [#936] Add .cpp file
- * [#936] Oops Forgot speex 32 khz
- * [#1214] Add configuration panel for history + D-Bus calls
- * [#1313] Test rtp thread function, frame size, nbbytes, resampling
- * [#790] Flush audio data before closing audio streams
- * [#1214] History displays local time
- * [#1214] Skip empty field on display
- * [#1214] Associate an account to an history entry
- * [#1342] Get addressbook options sensitive/non-sensitive
- * [#1211] Clean up and comments
- * [#1211] Get back to 20 ms framesize
- * [#1211] Use sendImmediate instead of putData in RTP
- * [#1211] Fix nb byte available in RTP
- * [#1211] Clear condition on maxNbSamples in RTP
- * [#1211] Fix max byte available in RTP session
- * [#1211] G722: Use 160 samples per frame instead of 320
- * [#1211] Test using a dynamic payload
- * [#1211] Test using a dynamic payload type
- * [#1211] Rename size variable (nb_samples, nb_bytes)
- * [#1211] Test g722 ip-to-ip sending twice the data lenth
- * [#1211] Test g722 ip-to-ip
- * [#1214] Do not select an history item by default at startup
- * [#1214] Remove some compilation warnings
- * [#1214] Handle empty field - remove g_print
- * [#1214] Add each history item only once
- * [#1214] Handle call timestamps properlier
- * [#1214] Do not need timestamp files anymore
- * [#1214] Use the saved date for history entry
- * Clean up
- * [#1214] Client doesn't crash if the D-Bus call fails
- * [#1214] Client is able to save its history - still some glitches
- * [#1211] Forgot 16000 for g722
- * [#1211] G722 initialization
- * [#1214] Save name/number, successfully load the history if no fields
- are empty
- * [#1499] Fixed destination directory bug
- * [#1214] Restore all the functionalities; peer name/number way more
- easy to handle !!
- * [#1214] Add callable_object instead of call_t, refactoring
- * [#1211] Test with polycom soundstation 16000
- * [#1211] Remove C like inline function in g722 codec
- * [#1342] Finalize gnome client preference window formating
- * [#1214] Retrieve the history when the gnome client startsup
- * [#1306] Implement localization for KDE client
- * [#1593] enable accounts apply button when account checked/unchecked
- * [#1214] Implement the dbus calls on server side
- * [#1214] Add serialized/unserialized functions to pass data on DBUS
- * [#1342] Formating gnome client configuration windows
- * [#1214] Save sucessfully a map of history items
- * [#1499] Removed multiple jobs compilation for KDE client (2)
- * [#1214] Load history from file into memory, add unit tests
- * [#1534] Throws a length_error exception in case URL exceeds
- std::string max_size
- * [#1499] Removed multiple jobs compilation for KDE client
- * [#1565] make account leds smaller
- * [1430] Fix dbus debug
- * [#1562] crashes when trying to change item of a call of state "OVER"
- * [#1116] Fix compilation bug
- * [#1317] Added mandriva and opensuse-11 64 bits
- * [#1108] Add messges in main window concerning transfer success
- failure
- * [#1116] Fix compilation problems
- * [#1211] g722 Makefile
- * [#1108] Client side transferFailed/trasferSucceded signals handling
- * [#1211] G722 mostly completed,
- * [#1555] make bigger toolbar (24x24)
- * [#1551] remove default mailbox number in wizard and disable mailbox
- button when first account doesn't have mailbox number
- * [#1342] Re-add sflphone manpages
- * [#1116] Fix compilation on non-jaunty distros
- * [#1317] Fixed opensuse startup sleep
- * [#1108] Add a signal in the client to notify successful or failed
- transfer
- * [#1108] Dbus signals concerning call transfer success/failure
- * [#1317] Added opensuse to automatic build system
- * [#1223] Fix manpages bug
- * [#1060] german translation glitch
- * Clean up some gnome client warnings
- * [#1547] replace ugly account leds by beautiful icons
- * [#1548] add close button that hides windowand just hide on clicking
- the cross
- * [#1549] put introspec XMLs in the client's source
- * [#1312] Implement getCallList D-BUS method
- * [#1116] Clear text in history and contacts
- * [#1499] KDE integration
- * [#1469] Modify header linkers in dbus-c++'s Makefile.am's
- * [#1469] Remove examples folder from dbus-c++
- * [#1214] History integration in build system; unit test squeleton
- * [#1317] Cleaning
- * [#1469] Remove configure stuff in dbus-c++
- * [#1469] Add unofficial mainline dbus-c++
- * [#1469] Remove dbus-c++ from freedesktop
- * [#1430] Bring account changed signal/callback back to normal
- * [#1060] Update german translation - Sven Werlen
- * [#1430] Add marshaller one string define
- * [#1430] Send account change signal broadcast using account id
- * [#1430] Remove condition on setRegistrationState, cause stun to
- crash
- * [#1317] Centralized version handling
- * [#1317] Fixed version number on sfl-git-dch
- * [#1317] Refactoring for new distributions
- * [#1215] Fix account order at startup if latency
- * [#1088] Restore sip dns srv
- * [#1214] Add squeleton for history manager
- * [#1430] Add accout id to accout changed method
- * [#1430] No connectionStatusNotification (account changed) if no
- changes
- * [#1538] Add COPYING file
- * [#1430] Add audio rtp thread tests
- * [#1317] Changed version detection
- * [#1538] Document license in libs/stund
- * [#1317] Added version files
- * [#1538] Apply François patches - debian packages
- * [#1317] Updated spec files
- * add files
- * [#1538] Apply François patches - debian packages
- * [#1535] Change program file structure (directory src...)
- * [#1317] Updated build system scripts
- * [#1317] Cleaning
- * [#1317] Copied introspect files to gnome client
- * [#1317] Added opensuse to build-system : first-shot
- * [#1317] Remove spec files from configure
- * [#1317] Added missing prefix
- * removed debug for daemon account fix
- * [#1430] Add a connection reference which most likely belong to
- libdbus
- * [#1430] Use shared connection instead of private
- * make daemon find the account, added userMatch
- * Clean code, add comments...
- * [#1317] Fixed packaging rules
- * [#1317] Updated autogen
- * Updated autogen.sh for pjsip
- * [#1526] Set accounts order
- * [#1317] Fixed pjsip lib dirs
- * [#1317] Updated debian packaging for new pjsip configuration script
- * [#1317] Switch to autogenerated guess and sub files
- * [#1317] Updated pjsip inclusion in build system
- * [#1317] Replaced pjsip guess and sub files
- * [#1317] Fixed compilation issues on opensuse 11
- * [#1505] account list seem to crash the application when clicking
- Apply very fast...
- * [#1456] Add a flag to be replaced in the control files
- * [#1456] Added version dependancy handling
- * put account alias in AccountWidgetItem rather than in the item with
- " " before.
- * [#1034] The KDE client should start sflphoned if it is not started
- * [#1500] Handle options for notifications and display on incoming
- call.
- * [#1443] Client should not crash when receive an unexpected
- stateChanged signal
- * [#1403] Do not stop the notification anymore
- * [#1456] Added version dependancy handling
- * [#1426] Daemon crashes when get alsa plugin
- * [#1422] Improved error messages
- * commit for merge
- * [#1424] Change logo in tray icon and put a different one when
- incoming call
- * [#1425] first part done, window title...
- * [#1413] add manpages creating and installing in build system
- * [#1417] The client should start the account creation wizard if
- started for the first time (if config file doesn't exist)
- * [#1421] Make volume bars horizontal when dialpad is hidden.
- * Changed main window title and fixed a mistake in sflphone_const.h
- * [#1412] make debian package building work
- * changelog changed.
- * Changed addAccount method in gnome client.
- * Debian and man folders added.
- * [#1388] Change project name from sflphone_kde to sflphone-client-kde
- * Better handle of kabc check.
- * [#1351] Automatic generation of dbus interfaces in makefile
- generated by cmake
- * [#1307] Implement "edit before call" in history and address book.
- * [#1344] change action_call label in call history from "call" to
- "call back".
- * [#1308] Implement Hook feature in kde client
- * Improved build system.
- * #1219 : Add address book configuration page
- * Better handling of registration to the daemon.
- * #1039 : Add tray icon in kde.
- * Issue no 1216 : Double click on item in history or address book
- causes call.
- * display peer name in call list and call history when called from
- address book.
- * Address book functionnal with photo displayed.
- * Help menu kde available but actions disappeared. All fonctions in
- view.
- * Address book functionnal but ugly and making its own sort in the
- complete address book.
- * Account choice on right click, clean out includes, page address
- book, fixed bugs...
- * Wizard, double click, context menu...
- * Removed sflphone_kde.kdevelop.filelist
- * Added account creation wizard and translated interface in english.
- * Transfer functionnal but ugly.
- * transfer not functionnal
- * Bug fixed : unholding (UNHOLD_CURRENT, UNHOLD_RECORD)
- * Commit functional for push. With install.sh
- * Before merge.
- * Problem with enable accounts. Account display increased.
- * Functional with codec order working , playDTMF.
- * Commit functional.
- * sflphone_kde/build added in .gitignore.
- * complete commit for checkout previous.
- * Commit before checkout previous version to check the display
- bug(little font everywhere...)
- * Functionnal client. Rest : history icons, config icons and
- functionalities
- * commit before merge asavard for isRecording.
- * Call and Automate fusion done and seems to work.
- * Commiting before putting Automate class in Call class.
- * Functionnal main window without recording, history, voicemail, kio
- widgets.
- * client kde avec kdevelop.
- * Config Dialog almost finished.
- * Base of QT client
-
- -- SFLphone Automatic Build System <team@sflphone.org> Tue, 23 Jun 2009 11:12:06 -0400
-
-sflphone-daemon (0.9.5-SYSTEM) SYSTEM; urgency=low
-
- ** 0.9.5 release **
-
- * [#1060] FIx bug in chinese translation
- * [#1313] git add rtpTest.cpp rtpTest.h
- * [#1313] Add init/close rtp tests
- * [#1313] Basic instanciation of the rtp layer
- * [#1449] Gtk-Critical concerning history filters and new calls
- * [#1400] Make the match with the hostname instead of username
- * [#1324] Change status bar label for "Using %s (%s)"
- * [#1403] Icon size: 60x60 px
- * [#1403] Do not remove notification, improve icon quality
- * [#1403] Add smaller icon for gnome notifications
- * [#1403] Prevent crash when hangup && no notification
- * [#1403] Remove all actions on notifications; code refactoring
- * [#1451] Use stun.sflphone.org as default STUN server
- * [#1060] New po files - need to be translated
- * [#1060] Update french translation - Rebuild template file
- * [#1456] Add a flag to be replaced in the control files
- * [#1454] Make cppunit optional; remove from build deps in control
- files
- * [#1401] Add libexpat1-dev dependency in control files
- * [#1448] Take off these ugly debug messages
- * [#1448] fixed getTelephoneTone and getTelephoneFile() called
- repeatedly
- * [#1406] add liblog4c-dev in build-depends
- * [#1409] Restore .desktop icon
-
- -- SFLphone Automatic Build System <team@sflphone.org> Mon, 25 May 2009 11:34:40 -0400
-
-sflphone-daemon (0.9.5-SYSTEM~rc2) SYSTEM; urgency=low
-
- ** 0.9.5 rc2 **
-
- * [#1422] Improved error message
- * [#1402] Fix pjsip build
- * [#1404] Clear GTK-Critical Bug at client startup
- * [#1422] Added automatic VM shutdown when building on more than one
- VM
- * [#1422] Fixed some issues with new changelog generation script
- * [#1422] Moved distribution update to specific file
- * [#1422] Dropped git-dch, replace by home made implementation
- * [#1402] Fix pjsip build
- * [#1404] Clear GTK-Critical Bug at client startup
- * Changes for name based dbus connection
- * Clean changelogs
- * [#1343] Gnome: Implement a callback system to handle focus on
- different widgets
- * Debus Session
- * Refactoring Python code, PEP8
- * [#1430] Get back dbus_g_proxy_new_for_name
- * [#1430] Get back DBUS_BUS_SESSION type
- * [#1430] Dbus fixed owner message binding
- * Second test with DBUS owner
- * [#1404] Gnome -> Preferences -> Hooks
- * [#1404] Gnome -> Preferences -> Recordings
- * [#1404] Call History
- * [#1404] Gnome -> Preferences -> Address Book
- * [#1404] IF the first notification option disable the second
- notification
- * Dbus with fixed owner does not automatically start the deamon
- * Add codec debug tests in pysflphone
- * [#1407] Some print info
- * [#1407] Add a scenario to pick_up action
- * Test client dbus connection to a fixed owner
- * Add python dbus test suite
- * [#1161] Modified version handling in build system
- * [#1314] Test pulse audio and audio streams connect and disconnect
- * [#1402] Add info message after configure
- * [#1402] Build the daemon with the local pjsip library (vs the
- installed one)
- * [#1009] Fix Codec Sampling Rate set to zeros
- * [#1314] Add mutex to pulse layer audio streams
- * [#1314] Refactoring pulseaudio stream to test connect disconnect
- * [#1314] Refactoring of pulselayer to test conect/disconnect
- * Add debug messages in debus calls concerning account
- * [#1314] Add some return values to audio init functions
- * [#1406] add liblog4c-dev in build-depends
- * [#1409] Restore .desktop icon
- * Bug #1405: Fix strings as requested.
- * Bug #1404: Fix strings in preferences panel.
-
- -- SFLphone Automatic Build System <team@sflphone.org> Tue, 19 May 2009 12:08:03 -0400
-
-sflphone-daemon (0.9.5-0ubuntu1~rc1) SYSTEM; urgency=low
-
- [ SFLphone Project ]
- * [#1262] Updated changelogs for version 0.9.5-0ubuntu1 Snapshot 2009-
- 05-05
-
- [ Emmanuel Milou ]
- * Add some python CLI client code; not really functional
- * [#1108] Fix peerHungup method for IP to IP call
-
- [ Alexandre Savard ]
- * [#1108] Correct setting of SIP contact for direct IP call
- * [#1108] SIP user agent handles incoming REFER
-
- [ Emmanuel Milou ]
- * Remove website from repository
- * Update translation
-
- [ Alexandre Savard ]
- * Sflphone icon's tooltip changed for "configured" instead of
- "registered"
-
- [ Emmanuel Milou ]
- * Update translation
-
- [ Sflphone Project ]
-
- -- Sflphone Project <sflphone@mtl.savoirfairelinux.net> Tue, 05 May 2009 19:16:09 -0400
-
-sflphone-daemon (0.9.5-0ubuntu1~beta) SYSTEM; urgency=low
-
- [ Julien Bonjean ]
- * Updated Eclipse stuff
- * Improved addressbook config window
- * Added sflphone Eclipse stuff
- * Implemented addressbook list server side
- * Moved dbus stuff in dbus directory
- * Updated addressbook configuration
-
- [ Emmanuel Milou ]
- * Remove unuseful installation scripts. Use apt-get build-dep sflphone
- instead
- * fix bug #1090
-
- [ Alexandre Savard ]
- * defining speex 16khz
-
- [ Emmanuel Milou ]
- * Remove unuseful file from build system
- * Start dns srv resolver
-
- [ Alexandre Savard ]
- * Basic ogg/vorbis initialization
-
- [ Emmanuel Milou ]
- * Handle incoming IP-to-IP invite correctly
-
- [ Alexandre Savard ]
- * speex wideband 16000
-
- [ Emmanuel Milou ]
- * Better handling of incoming IP to IP call
- * DNS SRV resolution functional
- * Implement IAX2 incoming URL
- * Allow user to make IP call without any accounts configured
- * Add a contextual menu to edit a number from the contacts tab
- * Add comments, tooltip and new button to the contextual menu
- * add delete event, migrate to GTK 2.16 for sexy icons
- * Resolve ticket #1118
- * Update suse spec file
- * Add phone number cleanup functions, unit tests and panel
- configuration
- * Add pertinent test that fails
- * fix dependencies for suse package
- * Add contextual edit menu in history - #1120
-
- [ Alexandre Savard ]
- * Temporary comit: make speex wideband (16 khz)
- * Temporary: shared object for speex narrow band
- * Temporary: speex narrowband and wideband coexist
-
- [ Julien Bonjean ]
- * Fixed bug when no book selected
- * Fixed addressbook related compilation warnings
- * Fixed GTK client remaining compilation warnings
- * Fixed segfault when book removed since last sflphone run
- * Fixed bug when book is unreachable (ldap error)
-
- [ Alexandre Savard ]
- * Fix codec list in audio config window
- * Active/inactive speex codec by payload
-
- [ Julien Bonjean ]
- * Updated gitignore
- * Added some comments
-
- [ Emmanuel Milou ]
- * Add callto: handler script for browsers and al.
- * Integrate test compilation in the daemon build-system
-
- [ Julien Bonjean ]
- * Fixed g_object_unref warning for pixbuf
- * Cleaned too verbose output
- * Fixed toolbar update warning
- * Added support for asynchornous books open (first shot)
-
- [ Emmanuel Milou ]
- * Add a DBus call to fetch the call details from a call ID - Ticket
- #928
-
- [ Julien Bonjean ]
- * Improved async open books
- * Fixed bug #1139
-
- [ Emmanuel Milou ]
- * Add a way to save account order
- * commit missing files
-
- [ Julien Bonjean ]
- * Introduced log4c (ticket #1162)
-
- [ Emmanuel Milou ]
- * Load/save account order functionnal - ticket #813
-
- [ Alexandre Savard ]
- * Add CELT codec (#1143)
- * Make celt frame size 256 (*1143)
-
- [ Julien Bonjean ]
- * Switched everything to log4c (ticket #1162)
- * Updated eclipse settings
-
- [ Emmanuel Milou ]
- * Restore adding account - ticket #1172
- * Add liblog4c dependecy - ticket #1179
-
- [ Alexandre Savard ]
- * Double maxAvailByte for frame size in rtp (#1143)
-
- [ Emmanuel Milou ]
- * Add User-Agent SIP header - Ticket #1173
-
- [ Julien Bonjean ]
- * Fixed autoresize issue (#708)
-
- [ Emmanuel Milou ]
- * Remove libcppuint dependency for the debian packages
- * Look for libsexy only if gtk version < 2.16 - Ticket #1116
- * Remove libsexy dependency for jaunty. ticket #1116
-
- [ Julien Bonjean ]
- * Introduced unit tests (#1146)
- * Updated gitignore
- * Fixed Makefile (#1146)
-
- [ Emmanuel Milou ]
- * [TICKET #1112] Add a test on the voice buffer to send through iax
- packets
- * Remove doublon in dependencies
- * Remove warnings from the client test framework
- * Update version number to 0.9.5~beta
- * Update build-package script
- * Add check dependency in build-deps control file field
- * Create debian files for the new sflphone-client-gnome
- * [TICKET #1212] Add Replaces field in control files
- * [TICKET #1212] Fix manpages installation path
- * [TICKET #1212] Add maintainer scripts to create alternatives
- * [#1212] Update the manpages generation - edit preinst maintainer
- script
- * [#1212] Fix reference error in manpage
- * [#1212] Add missing files on the client side
- * [#1212] Fix debian docs files - no TODO file
- * [1212] Fix manpage creation problem
- * [#1220] Generate client-side glue files and marshaller at
- compilation time
- * [#1220] Generate server-side glue files at compilation time
- * [#1212] Change binary name to sflphone-client-gnome
- * [#1212] Update .gitignore to fit the new working tree
- * [#1220] Explicitly generate glue files before building the library
- * [#1220] Compile dbus directory before audio
- * [#1212] Create sflphone-daemon at the root of the repository
- * [#1212] Re-add pjproject
- * [#1212] Remove Makefile from repo
- * [#1220] Fix Makefile.am
- * [#1212] New working directory functional
- * [#1212] Update .gitignore
- * [#1212] Hack to make pjsip compile..
- * [#1220] Use non-installed binary for dbusxx-xml2cpp
- * [#1212] Add descriptive files, remove unuseful scripts from tools/
-
- [ Alexandre Savard ]
- * Restore speex codecs
- * add frame size for celt (#1143)
- * add framesize to codec, independant from audiolayer (#1143)
- * use codec frame size in rtp (#1143)
- * compute fixed_codec_framesize (#1143)
- * do not resample if not required (#1143)
- * add condition on resampling for decoder (#1143)
- * add a condition on bytesAvail == 0 from mic data
- * no maximum in rtp decode (#1143)
- * compute maximum for decoding (#1143)
-
- [ Emmanuel Milou ]
- * [#1146] Implement unitary tests on the client-side
-
- [ Alexandre Savard ]
- * use float instead of int to compute max nb of sample (#1143)
- * add nbSampleMax for unresampled data (#1143)
- * make thread sleep during 5 ms insead of 20 (#1143)
- * use unix usleep (#1143)
- * 50 usecond thread!!!!! (#1143)
- * try with the smallest compression (#1143)
- * use timer set at framesize (#1143)
-
- [ Emmanuel Milou ]
- * [#1161] Restore changelog version
-
- [ Alexandre Savard ]
- * Remove celt stuff
-
- [ Emmanuel Milou ]
- * [#1161] Update changelog
- * [#1220] Add Conflicts: sflphone in debian control files
- * [#1179] Add liblog4c3 runtime dependency
- * [#1212] FIx typo error in dependency list for itnrepid
- * [#1212] FIx .desktop file to point on the right exec
- * [#1212] Modify changelog replacing tag
-
- [ Sflphone Project ]
- * "[#1262] Updated changelogs for version 0.9.5-0ubuntu1~beta"
-
- [ Emmanuel Milou ]
- * [#1212] restore changelogs
-
- [ Sflphone Project ]
- * [#1262] Updated changelogs for version 0.9.5-0ubuntu1 Snapshot 2009-
- 04-27
-
- [ Emmanuel Milou ]
- * [#1212] restore changelogs
-
- [ Sflphone Project ]
- * [#1262] Updated changelogs for version 0.9.5-0ubuntu1~beta
-
- [ Emmanuel Milou ]
- * [#1212] restore changelogs
-
- [ Sflphone Project ]
-
- -- Sflphone Project <sflphone@mtl.savoirfairelinux.net> Mon, 27 Apr 2009 16:57:00 -0400
-
-sflphone-daemon (0.9.4-0ubuntu2) SYSTEM; urgency=low
-
- [ Alexandre Savard ]
- * Restore speex and GSM detection
-
- [ Emmanuel Milou ]
- * Fix bug #1090
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Wed, 8 Apr 2009 11:29:15 -0500
-
-sflphone (0.9.4-0ubuntu1) SYSTEM; urgency=low
-
- [ Emmanuel Milou ]
- * Integrate DBus-c++ and libiax2 in the main build system
- * Clean up in the working repository
- * Reorder hooks configuration panel
- * Protect case when no codecs are active
- * Fix some return values
- * Add unitary tests for the hook manager (premisces)
-
- [Yun Liu]
- * Update chinese translation
-
- [Sven Werlen]
- * Update german translation
-
- [Hussein Abdallah]
- * Update russian translation
-
- [Maxime Chambreuil]
- * Update spanish translation
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Fri, 3 Apr 2009 18:29:15 -0500
-
-
-sflphone (0.9.4-rc1) SYSTEM; urgency=low
-
- [ Emmanuel Milou ]
- * Fix bug while trying to hold/unhold several simultaneous call
- * Improve address book build system
- * Implement SIP url popup on incoming call
- * Improve GTK+ panel configuration
- [ Julien Bonjean ]
- * GTK+ client refactoring
- * GTK+ clean up
- * Address book improvment
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Fri, 27 Mar 2009 18:29:15 -0500
-
-sflphone (0.9.4-0beta1) SYSTEM; urgency=low
-
- [ Alexandre Savard ]
- * Display codec used during conversation on the GUI
- * Enable/disable STUN parameters at runtime
- * Refactor search bar use
- [ Emmanuel Milou ]
- * Build system fixes
- * Implement SIP re-invite
- * Implement IP to IP call
- [ Julien Bonjean ]
- * Integrate GNOME address book based on evolution data server
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Fri, 20 Mar 2009 18:29:15 -0500
-
-
-sflphone (0.9.3-0ubuntu3) SYSTEM; urgency=low
-
- [ Alexandre Savard ]
- * Both playback and record streams in PA_STREAM_CORKED (pulseaudio)
- * Use PLUGHW device for ALSA capture
- * Functional IAX and SIP recording for voicemail
- * Use the less CPU-consuming interpolator algorithm for resampling
- * Display in GTK GUI the codec used in conversation
- * GTK GUI use ASCII instread of utf-8
- * Add record menus in GTK GUI
- * Put on hold when dialing a new number
- * AccountID's are saved in the history
-
- [ Emmanuel Milou ]
- * Integrate DBUS C++, libiax2 in the git repository
- * Update website
- * Use libspeexdsp only if available on the system
- * Updated .gitignore file
-
- [Cyrille Béraud]
- * Account assistant manager improvment
- * Add an email request when creating a new account to receive voicemails
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Sat, 14 Feb 2009 13:29:15 -0500
-
-sflphone (0.9.3-0ubuntu2) SYSTEM; urgency=low
-
- [ Emmanuel Milou ]
- * Add compilation note in README
- * Use default ALSA plugin for capture
- * Fix the ALSA capture problem one more time
- * Clean up debug messages in dbus.c
- * Add libspeexdsp dependency
- * Remove implicit declaration compilation warnings
- * Fix links in the website, add release note
- * Change capture for the website front page
- * Add alsa devel dependency in build-depends control file field
- * Clean up, indentation, try to handle latency problems in iax/pulseaudio
- * Remove pjsip generated files from the repo
- * Use the previous declared curAlias function in accountwindow
- * Fix bug in history call duration when the call fails
- * Remove runtime warning in the GTK+ client
- * Add librsvg2-common dependency to load SVG under KDE
- * Refresh .gitignore
- * Update locales files + french translation
- * Add configuration panel for future noise reduction
- * Add configuration panel for audio record module
- * Daemon less verbose; accounts don't try to access STUn options anymore
- * Fix typo in configwindow
- * Add content in the official website
- * use a GTK_STOCK icon for the record button
- * Complete description text in the assistant manager
- * Add libtool flags in client configure.ac
- * Remove unuseful dependency (snd)
- * Fix SIP transfer problems
- * Remove previous version of PJSIP from the repo
- * Upgrade PJSIP to version 1.0.1
- * Add the new website source in the repository
- * Use libspeexdsp for silence detection only if available
-
- [ Loïc Faure-Lacroix ]
- * Ajout du logo gpl3
- * Ajout des images
- * Ajout de la section screenshot pour le site
- * Ajout du favicon dans le header
- * Modification des cartes
-
- [ Alexandre Savard ]
- * Clean up <speex/libspeexdsp>
- * Small cleanup
- * Save Wave fixed
- * Fix new call button when recording
- * libspeexdsp added
- * Recording: default home folder at startup
- * Minor changes to config window
- * IAX recording fixed
- * Set / get recording path, still need some GTK for client
- * AudioRecord file name format
- * Now recording in HOME folder
-
- [ Cyrille Béraud ]
- * Fix bug in reqaccount.c
-
- [ Maxime Chambreuil ]
- * Update spanish translation
-
- [Yun Liu ]
- * Update chinese translation
-
- [ Hussein Abdallah ]
- * Update russian translation
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Sat, 14 Feb 2009 13:29:15 -0500
-
-sflphone (0.9.3-0ubuntu1) SYSTEM; urgency=low
-
- * Remove debug
- * Join thread before leaving
- * Fix implicit declaration in reqaccount
- * Add REST code to build the request to server
- * Fix GValue initialization warnings
- * Update version number, fix implicit declaration, fix GTK markup
- warnings
- * Apply patch to create custom SIP account from our own server
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Fri, 06 Feb 2009 19:17:32 -0500
-
-sflphone (0.9.2-2ubuntu9) SYSTEM; urgency=low
-
- [ Alexandre Savard ]
- * Speex audio codec preprocessing initialization
- * peer hung up segmentation fault solved
- * Stop recording when transfering
- * Terminate only one call
- * Add isRecording() function
- * Fix call_icon GTK client
- * Fix SIPCallClose() function, recorded file now close properly
- * Function terminateSIPCall added in sipvoiplink and managerimpl
- * Fix thread destructor
- * setRecordingOption function implement in audiorecord
- * Record now implemented in Call class
- * Record interface complete (on hold erase previous recording)
- * Added recButton in client
- * Added: record button related icons
- * Record button added
- * Overload AudioRecord::recData to get mic and speaker data mixed
- * Recording now in audiortp::run() method
- * Audio recording working in AudioRTP: receiveSessionForSpeaker
- * Open/close a wave file when pulse audio stream start/stop
-
- [ Emmanuel Milou ]
- * Fix path for GTK+ icons; clean up
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Thu, 05 Feb 2009 18:27:53 -0500
-
-sflphone (0.9.2-2ubuntu8) SYSTEM; urgency=low
-
- [ Emmanuel Milou ]
- * Update changelogs
- * Fix bug in merge and in Makefile.am
- * Terminate only one call
- * Disable PJsip shutdown when changing STUN parameters
- * Function terminateSIPCall added in sipvoiplink and managerimpl
- * Add a timer to the alsa thread to not jam the CPU load
- * Fix bug in sipvoiplink.cpp
- * Clean shutdown of pulseaudio on quiting
- * Fix DTMF at first start with Pulseaudio
- * Remove zeroconf from the build system
- * Add a library manager + exception handling
- * Clean up in the working directory
- * Better handling of capture XRUNs
- * Restore mic adjust volume on ALSA layer
- * Protect device ALSA operation if not opened
- * Fix the switching layer bug
- * Use dynamic_cast<> to use audiolayer-specific methods
- * Open the audio devices only once at startup
- * Refactoring of the ALSA part
- * Functional plug-in manager
- * Use a C++ thread to handle tones and DTMF in ALSA
- * Restore IAXVoIPLink, restore Mutex
- * Make the plugins registering against the plugin manager
- * Migrate to 1->N relationship between voiplink and accounts
- * API plugin for registration
- * Use C++ thread in SIP, move everything in sipvoiplink
- * Complete singleton pattern for the plugin manager
- * Add -Wno-return-type compilation flag to remove warnings; Update
- version number in configure.ac
- * Add the dynamic loading for the plugin framework; integate unittest
-
- [ Yun Liu ]
- * Update rpm spec file
- * modify build package script and spec file for suse
-
- [ Alexandre Savard ]
- * Add audiorecorder plugin and testaudiorecorder
- * Add audio Recording class, edit global.h
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Wed, 04 Feb 2009 14:00:30 -0500
-
-sflphone (0.9.2-2ubuntu7) SYSTEM; urgency=low
-
- [ Emmanuel Milou ]
- * Update changelog to 0.9.2-6
- * Fix some dbus-glib implementation details on the client side
- * Init history after dbus initialization
- * Add error checking in useragent; Clean sipvoiplink
- * Prevent crash when trying to call an empty number
- * Set the volume of the playback stream to PA_VOLUME_NORM at startup
- * Fix GTK+ generic value double initialization
- * Fix jaunty control file dependency problems
- * Fix jaunty control file dependency problems
-
- [ Yun Liu ]
- * Fix bug ticket # 137
- * Tolerant to gsm library of OpenSuse 11
-
- [ Sven Werlen ]
- * Update german translation
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Fri, 23 Jan 2009 17:48:13 -0500
-
-sflphone (0.9.2-2ubuntu6) SYSTEM; urgency=low
-
- [ Emmanuel Milou ]
- * Migrate STUN configuration to the main config window
- * Update french translation
- * Other tiny memory leaks
- * Fix memory leak in sampleconverter.cpp
- * Generate packages from the release branch
- * update the build package script
- * modify the control files with architecture=any
- * Remove valgring uninitialized value
- * IAX and SIP use the same global variables to set account
- configuration ; fix broken code
-
- [ Maxime Chambreuil ]
- * Update spanish translation
-
- [ Hussein Abdallah ]
- * Update russian translation
-
- [ Yun Liu ]
- * Update translation files
- * Fix the bug when user uncheck the account which fails in the
- previous registration
- * Add stun error status
- * Fix bug ticket #143
- * Script for auto-install dependencies
- * Fix bug ticket #140
- * Fix bug ticket 141
- * Fix the reregister process when user change the details of an
- account
-
- -- Emmanuel Milou <manu@sulfur.inside.savoirfairelinux.net> Fri, 16 Jan 2009 18:19:05 -0500
-
-sflphone (0.9.2-2ubuntu5) SYSTEM; urgency=low
-
- * Fix memory leak in the pulseaudio callback
- * Update debian package generation script
- * Warnings removal in GTK+ client
- * Clean adjust volume method in alsalayer
- * Plug the sflphone playback volume control to the pulseaudio volume
- manager
- * Display the date in history according to the current locale
- * Generate the changelog according to the git commit messages
- * Complete header in chinese translation file
- * Use the right gpg key to sign the packages
- * add debian jaunty jackalope support
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Wed, 14 Jan 2009 21:17:20 -0500
-
-sflphone (0.9.2-2ubuntu4) SYSTEM; urgency=low
-
- [ Emmanuel Milou ]
- * add german translation
-
- [ Yun Liu ]
- * Fix GUI crash in Ubuntu8.10 64bit system
-
- -- Yun Liu <yun.liu@savoirfairelinux.com> Thu, 08 Jan 2009 13:08:51 -0500
-
-sflphone (0.9.2-2ubuntu3) SYSTEM; urgency=low
-
- [ Emmanuel Milou ]
- * The main thread synchronizes the ringtone thread
- * disable custom ringtone for the ALSA layer
- * Fix the Makefile.am in man directory, add a SEE ALSO section
-
- [ Yun Liu ]
- * Fix daemon crash caused by the previous patch ( for bug ticket #129)
-
- -- Yun Liu <yun.liu@savoirfairelinux.com> Tue, 06 Jan 2009 16:18:38 -0500
-
-sflphone (0.9.2-2ubuntu2) SYSTEM; urgency=low
-
- * Fix bug ticket #129
-
- -- Yun Liu <yun.liu@savoirfairelinux.com> Wed, 5 Jan 2009 15:54:53 -0500
-
-sflphone (0.9.2-2ubuntu1) SYSTEM; urgency=low
-
- * Migrate from eXosip library to pjsip
- * Add multiple SIP accounts support
- * Fix ringtones problems
- * Add a pulseaudio support
- * Improve audio quality with ALSA
- * Add chinese translation
- * Improve spanish translation
- * Migrate to a maintained C++ DBus bindings
- * Clean and improve the build system
- * Add build-dependency on Perl because we need pod2man to generate manpages
-
- -- Yun Liu <yun.liu@savoirfairelinux.com> Wed, 26 Nov 2008 09:47:53 -0500
-
-sflphone (0.9.1) unstable; urgency=low
- * Add a search tool in the history
- * Migrate some gtk_entry_new to sexy_icon_entry_new
- * Bug fix (Ticket #78): The voicemail password isn't displayed anymore in
- the history tab
- * Add the SIP registration expire value in the user file.
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Thu, 22 May 2008 11:14:25 -0500
-
-sflphone (0.9.0) unstable; urgency=low
- * Add history features
- * Call date
- * Call duration
- * Mouse events in the history tab
- * Smooth switch from the history tab to the calls tab
- * Remove most of GTK-Critical warnings
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Wed, 13 May 2008 16:58:25 -0500
-
-sflphone (0.9-2008-06-06) unstable; urgency=low
- * Audio bug correction: capture stopped after a few minutes of conversation
- with USB Plantronics sound card
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Tue, 06 May 2008 16:58:25 -0500
-
-sflphone (0.9-2008-05-06) unstable; urgency=low
- * Bug correction: account creation with the assistant
- * GTK+ warnings removal
- * libnotify warnings removal
- * Remove aliasing on the SFLphone logo
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Mon, 05 May 2008 16:58:25 -0500
-
-sflphone (0.9) unstable; urgency=low
- * Clean dependencies ( removal of libboost )
- * Several GTK improvement and updates
- -account window
- -configuration window
- * Migrate from GtkCheckMenuItem to GtkImageMenuItem
- * ALSA standard I/O transfers: MMAP instead of R/W
- * Fix speex audio quality
- * IAX2 protocol
- -Fix hold/unhold situation
- -Add on hold music
- * SIP protocol
- -Ringtone on incoming call
- -Fix transfer situation
- * Add desktop notification ( libnotify )
- * Improve the system tray icon behaviour
- * Improve registration error handling
- * Register/unregister from the account window takes effect without starting back SFLphone
- * Compilation warnings removal
- * Call history
- * Add an account configuration wizard
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Wed, 30 Apr 2008 16:58:25 -0500
-
-sflphone (0.8.2) unstable; urgency=low
- * Internationalization of the GTK GUI
- * English / French
- * STUN support
- * Slight modifications of the graphical interface ( tooltips, dialpad, ...)
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Fri, 21 Mar 2008 11:37:53 -0500
diff --git a/tools/build-system/launchpad/sflphone-daemon/debian/compat b/tools/build-system/launchpad/sflphone-daemon/debian/compat
deleted file mode 100644
index 7ed6ff82de6bcc2a78243fc9c54d3ef5ac14da69..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/sflphone-daemon/debian/compat
+++ /dev/null
@@ -1 +0,0 @@
-5
diff --git a/tools/build-system/launchpad/sflphone-daemon/debian/control b/tools/build-system/launchpad/sflphone-daemon/debian/control
deleted file mode 100644
index 8275b9dc6791d7a41350093e010b69a4ad666f92..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/sflphone-daemon/debian/control
+++ /dev/null
@@ -1,29 +0,0 @@
-Source: sflphone-daemon
-Maintainer: SavoirFaireLinux Inc <julien.bonjean@savoirfairelinux.com>
-Section: gnome
-Priority: optional
-Build-Depends: debhelper (>= 7.0.50), libgcc1, autoconf, automake, libpulse-dev, libsamplerate0-dev, libccrtp-dev, libgsm1-dev, libspeex-dev, libtool, libdbus-1-dev, libasound2-dev, libopus-dev, libspeexdsp-dev, libexpat1-dev, libzrtpcpp-dev, libssl-dev, libgnutls-dev, libpcre3-dev, libyaml-cpp-dev, libboost-dev, libdbus-c++-dev, libsndfile1-dev, libpjproject-dev, libsrtp-dev, libjack-dev
-Standards-Version: 3.7.3
-
-Package: sflphone-daemon
-Priority: optional
-Architecture: any
-Depends: ${shlibs:Depends}, ${misc:Depends}
-Replaces: sflphone, sflphone-common
-Provides: sflphone-common
-Conflicts: sflphone-common-video, sflphone-daemon-video, sflphone-data
-Homepage: http://www.sflphone.org
-Description: SIP and IAX2 compatible softphone - Core
- SFLphone is meant to be a robust enterprise-class desktop phone.
- SFLphone is released under the GNU General Public License.
- SFLphone is being developed by the global community, and maintained by
- Savoir-faire Linux, a Montreal, Quebec, Canada-based Linux consulting company.
-
-Package: sflphone-daemon-dbg
-Architecture: any
-Section: debug
-Priority: extra
-Depends:
- sflphone-daemon (= ${binary:Version}),
- ${misc:Depends}
-Description: debugging symbols for sflphone-daemon
diff --git a/tools/build-system/launchpad/sflphone-daemon/debian/copyright b/tools/build-system/launchpad/sflphone-daemon/debian/copyright
deleted file mode 100644
index 7b3bdc5eebd9324f7616980b85dd03f0354d32a5..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/sflphone-daemon/debian/copyright
+++ /dev/null
@@ -1,28 +0,0 @@
-This package was debianized by Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> on
-Fri, 3 Apr 2009 09:47:53 -0500.
-
-It was downloaded from the git repository of SFLphone: git://sflphone.org/git/sflphone.git
-
-Upstream Author: Emmanuel Milou <emmanuel.milou@savoirfairelinux.com>
-
-Copyright:
-
-Savoir-faire Linux Inc.
-
-License:
-
-This software is copyright (c) 2004-2016 Savoir-faire Linux Inc.
-
-You are free to distribute this software under the terms of
-the GNU General Public License version 3.
-On Debian systems, the complete text of the GNU General Public
-License can be found in the file `/usr/share/common-licenses/GPL'.
-
-This program is distributed in the hope that it will be useful,
-but WITHOUT ANY WARRANTY; without even the implied warranty of
-MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
-GNU General Public License for more details.
-
-You should have received a copy of the GNU General Public License
-along with this program; if not, write to the Free Software
-Foundation, Inc., 51 Franklyn St, Fifth Floor, Boston, MA 02110-1301, USA.
diff --git a/tools/build-system/launchpad/sflphone-daemon/debian/cron.d b/tools/build-system/launchpad/sflphone-daemon/debian/cron.d
deleted file mode 100644
index d11e61177739b56bce3aac6de6483b48e797a258..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/sflphone-daemon/debian/cron.d
+++ /dev/null
@@ -1,4 +0,0 @@
-#
-# Regular cron jobs for the sflphone package
-#
-0 4 * * * root sflphone_maintenance
diff --git a/tools/build-system/launchpad/sflphone-daemon/debian/dirs b/tools/build-system/launchpad/sflphone-daemon/debian/dirs
deleted file mode 100644
index 93e7926139602286c03695199324cd3d2bcdcb39..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/sflphone-daemon/debian/dirs
+++ /dev/null
@@ -1,9 +0,0 @@
-usr/bin
-usr/lib
-usr/lib/sflphone
-usr/share/applications
-usr/share/dbus-1/services
-usr/share/sflphone/ringtones
-usr/share/locale
-usr/share/doc
-usr/share/man
diff --git a/tools/build-system/launchpad/sflphone-daemon/debian/docs b/tools/build-system/launchpad/sflphone-daemon/debian/docs
deleted file mode 100644
index f1dd08af0258ad3bbadc728c41f5d3cc680a7ef4..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/sflphone-daemon/debian/docs
+++ /dev/null
@@ -1,6 +0,0 @@
-NEWS
-README
-TODO
-ChangeLog
-AUTHORS
-
diff --git a/tools/build-system/launchpad/sflphone-daemon/debian/manpages b/tools/build-system/launchpad/sflphone-daemon/debian/manpages
deleted file mode 100644
index 0b7e5f1c26cb4203d1b4d6dcfd0c761be11e9a09..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/sflphone-daemon/debian/manpages
+++ /dev/null
@@ -1 +0,0 @@
-debian/sflphone-daemon/usr/share/man/man1/sflphoned.1
diff --git a/tools/build-system/launchpad/sflphone-daemon/debian/postinst b/tools/build-system/launchpad/sflphone-daemon/debian/postinst
deleted file mode 100644
index 7cbd04ec6550e04cc8233ee3d74022f4cc57609f..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/sflphone-daemon/debian/postinst
+++ /dev/null
@@ -1,56 +0,0 @@
-#!/bin/bash
-# postinst script for sflphone-daemon
-#
-# see: dh_installdeb(1)
-
-# Script to copy and move, if exists, configuration file sflphonedrc and history in the XDG directory
-# Freedesktop specifications: http://standards.freedesktop.org/basedir-spec/latest/
-
-set -e
-
-INST_CONFIG="$HOME/.sflphone/sflphonedrc";
-INST_DATA="$HOME/.sflphone/history";
-INST_CACHE="$HOME/.sflphone/sfl.pid";
-
-NEW_INST_CONFIG=
-NEW_INST_DATA=
-NEW_INST_CACHE=
-
-# Set the XDG CONFIG directory to the default one or to the path set in the environment variable
-if [ -z $XDG_CONFIG_HOME ]; then
- NEW_INST_CONFIG=$HOME"/.config/sflphone/"; # This is the standard path
-else
- NEW_INST_CONFIG=$XDG_CONFIG_HOME;
-fi;
-
-# Set the XDG DATA directory to the default one or to the path set in the environment variable
-if [ -z $XDG_DATA_HOME ]; then
- NEW_INST_DATA=$HOME"/.local/share/sflphone/"; # This is the standard path
-else
- NEW_INST_DATA=$XDG_DATA_HOME;
-fi;
-
-# Move the configuration file
-if [ -f $INST_CONFIG ] ; then
- echo "Moving the configuration file into $NEW_INST_CONFIG directory";
- if [ ! -d $NEW_INST_CONFIG ]; then
- mkdir $NEW_INST_CONFIG;
- fi
- mv $INST_CONFIG $NEW_INST_CONFIG;
-fi
-
-# Move the history
-if [ -f $INST_DATA ] ; then
- echo "Moving the history file into $NEW_INST_DATA directory";
- if [ ! -d $NEW_INST_DATA ]; then
- mkdir $NEW_INST_DATA;
- fi
- mv $INST_DATA $NEW_INST_DATA;
-fi
-
-# Remove the directory
-# rmdir $HOME"/.sflphone";
-
-echo "You may remove the $HOME/.sflphone, the application won't use it anymore, but the XDG directories instead. Thank you.";
-
-exit 0
diff --git a/tools/build-system/launchpad/sflphone-daemon/debian/postrm b/tools/build-system/launchpad/sflphone-daemon/debian/postrm
deleted file mode 100644
index 70be710bd108ddadbfa790f9ac8795984cc139f8..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/sflphone-daemon/debian/postrm
+++ /dev/null
@@ -1,36 +0,0 @@
-#!/bin/sh
-# postrm script for sflphone
-#
-# see: dh_installdeb(1)
-
-set -e
-
-# summary of how this script can be called:
-# * <postrm> `remove'
-# * <postrm> `purge'
-# * <old-postrm> `upgrade' <new-version>
-# * <new-postrm> `failed-upgrade' <old-version>
-# * <new-postrm> `abort-install'
-# * <new-postrm> `abort-install' <old-version>
-# * <new-postrm> `abort-upgrade' <old-version>
-# * <disappearer's-postrm> `disappear' <overwriter>
-# <overwriter-version>
-# for details, see http://www.debian.org/doc/debian-policy/ or
-# the debian-policy package
-
-if [ "$1" = "purge" ]
-then
-
- # remove the user config file
- rm -f $HOME/.sflphone/sflphonedrc
-
-fi
-
-# dh_installdeb will replace this with shell code automatically
-# generated by other debhelper scripts.
-
-#DEBHELPER#
-
-exit 0
-
-
diff --git a/tools/build-system/launchpad/sflphone-daemon/debian/preinst b/tools/build-system/launchpad/sflphone-daemon/debian/preinst
deleted file mode 100644
index 6d04e97b452606720e35f07523b60c32ad7f9d6b..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/sflphone-daemon/debian/preinst
+++ /dev/null
@@ -1,16 +0,0 @@
-#!/bin/sh
-# postrm script for sflphone
-#
-# see: dh_installdeb(1)
-
-set -e
-
-package=sflphone
-
-case "$1" in
- install|upgrade)
- # Clear the old dbus-c++ and iax2 if presents
- ;;
-esac
-
-exit 0
diff --git a/tools/build-system/launchpad/sflphone-daemon/debian/rules b/tools/build-system/launchpad/sflphone-daemon/debian/rules
deleted file mode 100755
index ff8ac6dcc2ed3fb5b996939cf18e2254de274d1b..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/sflphone-daemon/debian/rules
+++ /dev/null
@@ -1,93 +0,0 @@
-#!/usr/bin/make -f
-# -*- makefile -*-
-# Sample debian/rules that uses debhelper.
-# This file was originally written by Joey Hess and Craig Small.
-# As a special exception, when this file is copied by dh-make into a
-# dh-make output file, you may use that output file without restriction.
-# This special exception was added by Craig Small in version 0.37 of dh-make.
-
-# Uncomment this to turn on verbose mode.
-#export DH_VERBOSE=1
-export DH_OPTIONS
-
-package=sflphone-daemon
-
-CXX = g++-4.0
-CFLAGS = -Wall -g
-DEB_INSTALL_MANPAGES_sflphone_common = sflphoned.1
-
-configure: configure-stamp
-configure-stamp:
- dh_testdir
- # Add here commands to configure the package.
- # build iax and opendht with contrib since they are not packaged
- cd contrib && mkdir -p native && cd native && ../bootstrap && make .iax && make .dht && cd ../..
- ./autogen.sh
- ./configure --prefix=/usr --disable-video
- touch configure-stamp
-
-#Architecture
-build: build-arch
-
-build-arch: build-arch-stamp
-build-arch-stamp: configure-stamp
-
- # Add here commands to compile the arch part of the package.
- $(MAKE)
- touch $@
-
-clean:
- dh_testdir
- dh_testroot
- rm -f build-arch-stamp configure-stamp
- # Add here commands to clean up after the build process.
- [ ! -f Makefile ] || $(MAKE) distclean
-
-ifneq "$(wildcard /usr/share/misc/config.sub)" ""
- cp -f /usr/share/misc/config.sub config.sub
-endif
-ifneq "$(wildcard /usr/share/misc/config.guess)" ""
- cp -f /usr/share/misc/config.guess config.guess
-endif
- dh_clean
-
-install: install-arch
-
-install-arch:
- dh_testdir
- dh_testroot
- dh_clean -k -s
- dh_installdirs -s
- # Add here commands to install the arch part of the package into
- # debian/tmp.
- $(MAKE) DESTDIR=$(CURDIR)/debian/$(package) install
- rm -rf $(CURDIR)/debian/$(package)/usr/include
- dh_install -s
-
-binary-common:
- dh_testdir
- dh_testroot
- dh_installchangelogs ChangeLog
- dh_installdocs
- dh_installexamples
- dh_installman
- dh_link
- dh_strip --dbg-package=sflphone-daemon-dbg
- dh_compress
- dh_fixperms
- dh_makeshlibs
- dh_installdeb
- dh_shlibdeps
- dh_gencontrol
- dh_md5sums
- dh_builddeb
-
-# Build architecture dependant packages using the common target.
-binary-arch: build-arch install-arch
- $(MAKE) -f debian/rules DH_OPTIONS=-s binary-common
-
-override_dh_strip:
- dh_strip --dbg-package=sflphone-daemon-dbg
-
-binary: binary-arch
-.PHONY: build clean binary-arch binary install install-arch configure override_dh_strip
diff --git a/tools/build-system/launchpad/sflphone-gnome-video/debian/changelog b/tools/build-system/launchpad/sflphone-gnome-video/debian/changelog
deleted file mode 100644
index cdaabfde1543d49cdc4465601244a7c987566471..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/sflphone-gnome-video/debian/changelog
+++ /dev/null
@@ -1,3138 +0,0 @@
-sflphone-gnome-video (1.1.0-rc20120607~ppa1~SYSTEM) SYSTEM; urgency=low
-
- ** SNAPSHOT 1.1.0-rc20120607~ppa1~SYSTEM **
-
- * gnome: cleanup logging
- * * #12061: gnome: fix video codec list display
- * * #12050: gnome: fixed regression in active codec handling
- * #11732: Do not align menu with other interface component
- * * #11818: gnome: stop daemon on SIGTERM, SIGINT or SIGHUP
- * #10304: updatePlaybackScale dbus method uses int 32 bit for size and
- position (allow for 24 days long recording playback)
- * #10304: Add time lable for seekslider
- * * #10725: gnome: seekslider: fixed compiler warning due to missing
- header
- * * #11732: gnome:calltree: fix warning when searchbar is not created
- * * #11252: daemon: removed deprecated zrtp code
- * #10725: Consolidate test to determine if there is a recording,
- remove log on NULL pointer
- * #10725: Remove logging from stop callback in seekslider
- * #10725: No need of playback related callbacks in uimanager
- * #10725: Remove logging in seekslider when no selected call
- * #11732: Align Icons with dialpad in gnome client
- * #10304: Make sure that the playback won't be updated after reset
- * #10304: Prevent playback widget reset from stoping dtmf
- * #10304: Move Playback widget at the bottom of the main window
- * #10304: Fix logic to determine outgoing/incoming calls for recording
- icon
- * * #11685: gnome: fixed memory leaks in config menus
- * #10304: Fix reset seekslider that prevent scale to be updated at
- first read
-
- -- Julien Bonjean <julien.bonjean@savoirfairelinux.com> Thu, 07 Jun 2012 16:04:59 -0400
-
-sflphone-client-gnome (1.0.0-rc20110930~ppa1~SYSTEM) SYSTEM; urgency=low
-
- ** SNAPSHOT 1.0.0-rc20110930~ppa1~SYSTEM **
-
- * update kde .gitignore
- * Fix bug in volume widget
- * More polishing for release
- * Bump version to 1.0.0
- * [#7023] Add the ability to load an abstract contact backend in the
- library to resolve more data, polish code
- * [#7021] More cleanup for release
- * Cleanup
- * [#7021] Refactor KDE client dbus handling, add a missing call in
- daemon and port the DataEngine to the new API
- * Remove some annoying debug
- * merge language scripts
- * remove obsolete 'VERSION' files
- * update install instructions
- * Add missing translations to gnome
- * language update
- * Revert "Don't reference count DBus clients, exit core immediately
- when one of them request it"
- * Don't reference count DBus clients, exit core immediately when one
- of them request it
- * [7021] Add contact abstraction support
- * [#7121] Polishing library (over). Indentation, spacing and naming
- are now consistent
- * codecs: link to libccrtp, don't use logger
- * Fix a daemon bug
- * [#7038] Fix adding contact
- * * #7037 : stop audio stream after all calls have been hanged up
- * [#7025] Add full support for bookmark
- * SFLPhone KDE do not destroy history anymore
- * Fix config skeleton
- * Close the daemon once and for all, no more automatic respawning
- * Fix "unregistered account" bug (I hope so)
- * Close SFLPhone at the right place, it still respawn, I don't know
- why
- * Remove dead code
- * Fix regressions introduced in the last commit
- * Dead code elimination 1/3
- * Fix bug, add "add contact" option, fix warning
- * * #7019: Fix IAX codec negociation
- * Remove or comment unnecessary/unhelpful debug output
- * Fix "same as local" account setting, fix IP2IP LED color
- * Add support for some more advanced config options and add missing
- config dialog icons
- * Fix crash with noise suppressor
- * Alternative can now be selected from the call view context menu
- * Add drag and drop support, initial context menu and fix 3 bugs in
- the account dialog
- * Add basic history drag and drop support
- * Complete contact support is back
- * * #6991 : fix IAX problems
- * Fix IAX accounts being disabled by default
- * Revert "deb: forge -g flags for pjsip"
- * * #5884: Disable debug code in pjsip
- * echo suppressor : more assertions
- * Don't let the daemon think crypto is enabled when it's not
- * Simplify ToneList
- * Some progress on contact support
- * Remove unused getRegistrationCount()
- * remove annoying debug
- * revert SIP bit of e27e5c39bad27bae28f574eb2cba7717e8956229
- * Simplify CallManager::placeCallFirstAccount
- * Fix crash on hold
- * * #6905 : SIP refactor
- * gnome client: be sure key exchange is set correctly
- * Move code into createSipTransport
- * Fix account registration on start
- * ManagerImpl::registerAccounts(): simplify
- * * #5884: don't mess with pjsip threads in echo suppressor
- * * #6905 : simplify udp/stun/tls pjsip transport creation
- * Restore and improve support for Call history
- * fix launchpad build
- * SIPVoIPLink: simplify / refactor
- * Fix libwidget linking
- * SIP: simplify
- * IM : simplify
- * gnome: remove some debug
- * AudioRtpFactory::stop() cannot fail
- * * #6905: simplify SIP code
- * pjlib: fix build without SSLv2, fix warnings
- * Port history to the new syntax
- * Test a dock widget based implementation for contact and history
- * Disable SSLv2 support from pjsip and sflphone
- * deb: forge -g flags for pjsip
- * Fix deb packaging to get debug symbols
- * remove debug
- * pjproject: update to last stable release (1.10)
- * Require gtk >= 2.20 and glib >= 2.24
- * tlsadvanceddialog: simplify
- * * #6902 : fix errors spotted by -DGSEAL_ENABLE
- * Update daemon dbus XML and port KDE config backend from dbus to
- local
- * Remove unused but set variables
- * * #6929 : fix IM widget, cleanup
- * Unconditionally enable debug symbols
- * Should fix many KDE issues
- * * #6886 : hitting backspace on empty number have no side effects
- * * #6905 : fix AudioCodecFactory access in optimized builds (-O > 0)
- * Remove unsupported and broken jaunty/karmic packages
- * * #6902 : avoid using some gtk deprecated functions
- * Update dbus introspection files
- * * #6904: removed unused contactmanager
- * * #6903 : use correct dbus-cxx package name
- * * #6902: don't use individual gtk headers
- * Fix a segfault when config is not present
- * Merge latest (0.9.13) KDE code. This version is not yet ready for
- git master, but better than the previous one
- * addressbook : simplify
- * * #5659 : sflphone-plugins doesn't depend on libedataserverui
- * * #5659 : addressbook doesn't use libedataserverui
- * gnome client doesn't depend on evolution
- * * #5695: addressbook: simplify
- * * #5695: addressbook : remove AddrBookHandle from plugin
- * * #5695 : addressbook : remove unused stuff in the client
- * * #5695 : addressbook : remove unused stuff, use static mutex
- * gnome client doesn't use evolution
- * gnome: use proper API to set GTK_CAN_FOCUS
- * * #6897: removed unused focus state vars/callbacks
- * gnome: fix calls to sflphone_fill_codec_list_per_account
- * * #6623: gnome: don't leak in mainwindow
- * gnome: mainwindow whitespace cleanup
- * gnome: actions.c parameter doesn't have to be a double pointer
- * * #6895: fix memleaks, cleanup in accountconfigdialog
- * * #6893: fixes segfault in client on clean history
- * * #6894: fix leaks, cleanup in sflnotify
- * daemon: fixed prints in main
- * * #6892: simplify, fix leaks in dialpad
- * * #6887: audiopreference creates audio layer
- * * #6660: use const char * const, not std::string for globally
- visible constants
- * * #6852: Preferences now solely responsible for audiolayer creation.
- * * #6860: refactor uimanager, also fixes #6865
- * * #6853: hangup as soon as all digits have been deleted
- * * #6852: alsa: retry if device is busy
- * * #6852: audiolayer creation depends only on preference.audioApi
- * * #6850: gnome: fix build for gtk < 2.22.0
- * cleanup in iax
- * alsa: typo
- * pulse: if we can't peek in audio input, we can't drop samples
- * * #6849: show error window if codecs are missing, instead of dying
- * EchoCancel: unused, remove
- * * #6629 : use number of samples as arguments for audio filters
- * * #6629 : remove unused Algorithm interface
- * * #6629 : use helper to call alsa functions and display error msgs
- * Remove unused type
- * * #6841: fix some error handling
- * * #6629: simplify AlsaLayer::alsa_set_params()
- * Get gdk key definition from header
- * * #6828: Replace raw key codes by gdk defines
- * remove some debug, enhance some other
- * mainbuffer: simplify
- * * #6561 : fix phantom call after transfer
- * Conference Participant set : simplify
- * SIPCall: remove unused functions, make invite session public
- * * #6229 : remove malloc/free from pulse audio loop
- * * #6629 : simplify pulse callbacks
- * * #6629
- * Simplify widgets
- * * #6629 : keep the correct audio module when frequency changes
- * * #6751: fixed erroneous debug msgs
- * callable_obj.h: removed unneeded pthread header
- * alsalayer: cleanup
- * * #6629: Always restart audio driver when changing parameters (ALSA
- only)
- * gnome GUI: don't block in DBus signal errorAlert()
- * * #6629 : simplify AudioLayer creation
- * * #6629 : remove unused and unconfigurable frameSize from audiolayer
- * * #6629 : remove unused error message from audio layer
- * Fix logic error when switching audio API
- * Remove unused AudioProcessing class
- * AudioRtpRecordHandler::initNoiseSuppress() : use noiseSuppress
- directly
- * * #6629 : use DC blocker directly in audio layers
- * * #6629 : clean AudioLayer
- * * #6629 : don't store mainbuffer inside audiolayer
- * * #6629 : correct AudioLayer::notifyincomingCall()
- * * #6554: cleanup, refactoring in sipvoiplink
- * * #6554: cleanup in iaxvoiplink
- * * #6554: throw exception in getSIPCall if pointer is NULL
- * * #6554: make some methods of sipvoiplink static
- * * #6655: cleanup in managerimpl
- * * #6554: refactoring, fix memleaks in sipvoiplink
- * * #6478: remove throw specs, cleanup in voiplink
- * * #6629 : remove unused AudioDevice
- * * #6655: removed more dependencies from managerimpl
- * * #6744: simplified numbercleaner
- * conference : remove one prototype
- * * #6743: fix ip2ip
- * Don't give glib warnings if icons are not found
- * gnome: fixed includes
- * Codec.h: removed unused function
- * * #6742 : clean dbus & icons
- * * #6699: refactor/cleanup accounts
- * icons: cleanup
- * timer : use second precision, not millisecond
- * calltree_update_clock : use correct type, returns something
- * * #6737: fixed typo in dbus call
- * * #6737: removed tests for removed API
- * * #6737: dbus: fixed bug from merge
- * * #6737: cleanup in accountlist
- * * #6737: cleanup in dbus
- * * #6740 : fix history double free
- * * #6740 : remove time updating thread from calls
- * * #6737 : use c99 for client
- * * #6738 : make history loading faster
- * sipvoiplink : don't crash on transfers
- * fixed typo
- * Remove unused file
- * Don't build networkmanager.cpp at all if NM is disabled
- * _debug* -> _debug
- * * #6554 : simplify sipvoiplink
- * hudson: added -x to git clean command
- * added git clean to hudson script
- * audiocodecfactory: cleanup
- * * #6718: refactored setTlsSettings into SIPAccount
- * * #6718: removed more unused methods
- * * #6718: refactored confmanager code into sipaccount
- * remove unused functions
- * * #6718: confmanager: removed more unused methods
- * AudioCodecFactory : cleanup
- * #6697 : Turn callableElement struct into union
- * * #6718: confmanager: removed more unused methods
- * * #6718: confmanager: removed more unused methods
- * * #6718: removed unused dbus methods, refactoring
- * * #6699: accounts: cleanup/refactoring
- * * #6699: refactoring, cleanup in accounts
- * * #6699: more account cleanup
- * remove unused autoconf variable
- * * #6714: fixed hudson script
- * make distclean in hudson
- * added || exit 1 to run_tests.sh call
- * * #6714: fixed make distcheck for sflphone-plugins
- * * #6714: fixed make distcheck for gnome client
- * * #6714: fixed make distcheck for daemon
- * git: #6698 split the main .gitignore file
- * gnome: gpointer is already a pointer
- * gnome: calltab_init: use calloc instead of malloc
- * * #6699: more account cleanup
- * * #6699: cleanup account
- * * #6554 : more *voiplink cleanup
- * * #6558 : more sipvoiplink simplification
- * * #6558: saner loadSIPLocalIP prototype
- * gnome: #6623 clean calllists
- * * #6692: more audiolayer cleanup
- * * #6692: cleanup/refactoring in audiolayers
- * * #6692: more forward declarations, AudioThread->AlsaThread
- * * #6692: audiolayer cleanup
- * * #6692: alsalayer cleanup
- * * #6558 : remove account creator
- * * #6558 : clean sipvoiplink
- * * #6554 : cleanup sipvoiplink
- * audiortp: cleanup
- * * #6657 : fix launchpad builds for good
- * * #6675 : send RTP dtmf events only once
- * * #6655: more cleanup
- * AudioRtpSession::updateSessionMedia() : simplify
- * * #6655: more cleanup in managerimpl
- * * #6655: removed more code, cleanup
- * * #6655: more cleanup, fixed infinite loop
- * * #6655: removed more unused files
- * * #6655: removed unused mutex
- * * #6655 removed more unused code
- * * #6655: removed unused methods
- * * #6655: cleanup in main
- * * #6663: fixed segfault when off hold from transfer
- * * #6658: user's active codec selection is respected
- * * #6660: static global string should be static const char* const
- class member
- * * #6659: use g_strcmp0, not strcmp for vals that may be null
- * callable_obj: fix double free
- * calltree_display_call_info() : simplify
- * * #6657: Fix launchpad builds
- * Logger::log() : simplify
- * AudioRtpSession : privatize members
- * * #6655: more constness, cleaned up/simplified methods
- * * #6654: call DBus::_init_threading so that dbus-c++ to make it
- threadaware
- * set default credentials on account creation
- * AudioCodecFactory::scanCodecDirectory() : simplify and correct
- * * #6623: fixed typos
- * * #6623: fixed more leaks
- * * #6623: fixed more leaks
- * * #6623: fixed more leaks, don't print codec name if null
- * * #6623: more leaks fixed in client
- * * #6623: fix more leaks, fixed some warnings
- * * #6623: fixed leak in history
- * updated gitignore
- * initialize dbus dispatcher correctly
- * Fix tests, hudson doesn't have a dbus daemon running
- * remove unused code
- * removeCall() : simplify , fix leak
- * stopRtpThread() : simplify
- * *CurrentCall : simplify
- * Fix memleak
- * fix serialization of audio api (pulse / alsa)
- * account map : simplify
- * remove call from callmap before terminating it, avoid use after free
- * * #6630 : don't make DBusManager a singleton
- * call: return confID by value
- * add back history code deleted by error
- * history : reverse logic
- * simplify history serialization and remove some debug
- * remove annoying debug
- * * #6464 : replace cerr with _error
- * * #6464: replace cout with logger macros
- * replace printf() with logger macros
- * update .gitignore
- * remove unused function
- * update eclipse projects
- * uimanager_new() : simplify
- * rename directories
- * celt: simplify a bit
- * Fix CELT configure.ac test
- * * #6612 : template speex codecs
- * * #6623: refactored conference obj
- * * #6623: refactored callable object, removed leaks
- * * #6623: more cleanup, fix leaks, make global vars static and rename
- them
- * * #6623: calltree: fixed memleaks, simplified code.
- * audiolayer: init pointer members
- * manager: catch exception on invalid hangup
- * * #6623: don't leak on calls to create_new_call
- * * #6611 : clarify codecs prototypes
- * ringtones : .au and .ul files are both ulaw
- * * #6611 : make sure samplerate converters are called correctly
- * ManagerImpl::switchAudioManager() : simplify
- * * #6623: fixed more leaks
- * * #6623: fixed more leaks
- * * #6623: fixed more leaks
- * * #6623: fixed leak, line-endings in imwidget
- * * #6627: zero-initialize pointers if they're going to be deleted
- * * #6628: don't leak calls on exceptions
- * Revert "audiortp: call join after calling stop on RtpThread"
- * sflphone-client: more constness
- * audiortp: call join after calling stop on RtpThread
- * * #6625: return 0 on successful completion
- * * #6624: fix segfault on servercallfailure
- * * #6621: Fixed double free, unlock mutex in ManagerImpl::terminate
- * * #6220: remove audio stream when peer hangs up
- * * #6596: AudioSymmetricSession shouldn't self-delete
- * resampler: grow internal buffers dynamically
- * merge up and down sampling => resampling
- * Leave test directory unchanged when running make check
- * audio algorithms : remove unused prototype
- * ringtone: detect codec from file extension
- * *AudioFile : simplify
- * * #6596: create local SDP on the stack, not the heap
- * * #6596: don't call Ost::Thread::terminate from dtor
- * audiofile: cleanup (samplerate -> unsigned)
- * remove unused func
- * samplerateconverter: cleanup
- * RingBuffer::Put() : remove unused return value
- * MainBuffer::putData() : remove unused return argument
- * audiolayer::putMain() : remove unused func
- * AudioLayer::putUrgent() : remove unused return value
- * * #6618: delete any remaining ringbuffers in destructor
- * RingBuffer::availForPut() : remove
- * * #6617: return from main rather than calling exit
- * MainBuffer::availForPut(): remove
- * RingBuffer: simplify
- * alsa : remove write only variable
- * fix memcpy declaration
- * bcopy(src, dst) -> memcpy(dst, src)
- * RingBuffer::Get() : remove constant volume argument
- * return a copy of the call ID, not just a reference.
- * MainBuffer::getDataById() : remove volume argument (always 100)
- * MainBuffer::getData() : remove constant volume argument
- * RingBuffer::Put() : remove constant volume argument
- * MainBuffer::putData() : remove constant (=100) volume argument
- * audiolayer: remove constant _defaultvolume
- * AudioRtpRecordHandler / AudioRtpSession : simplify
- * mainbuffer: fix test
- * iaxvoiplink : simplify
- * sip registration callback: fix a dbus crash
- * MainBuffer: simplify
- * AudioRtpFactory: return cached type of rtp session. The rtp session
- can have disappeared if the call was put on hold
- * AudioRtpFactory: remove unused setters
- * Fix launchpad builds
- * * #6611 : remove unused bandwidth codec information
- * * #6611: AudioCodec: remove useless/unused setters
- * make sure buffer string is initialized correctly
- * * #6596: declare certain destructors virtual
- * audiolayer : cleanup
- * Simplify doc build rules
- * * #6270: don't build dbus-api doc with make, should require make all
- * configure.ac: cleanup
- * Remove copy of dbus-c++ from libs/
- * * #6596: stop clock thread when peer hangs up
- * removed unused Fmtp.h
- * * #6595: more logical initialization order
- * * #6600 : fix account creation
- * * #6601 : fix configure.ac tests
- * remove unused variable
- * Don't mix stack and heap based allocations
- * Fix copyright (2009, 2008, 2009 -> 2008, 2009)
- * Fix warnings found by clang
- * * #6595: fix initialization order for AudioRTP
- * * #6592: removed typedef std::string CallID
- * * #6586: implement local g_slist_free_full for older glib versions
- * * #6579: fix memory leaks in client (there's a lot left)
- * ShortcutPreferences::setShortcuts() : simplify
- * Fix merge
- * * #6548: remove call to non thread-safe strerror()
- * AudioRtpFactory: each instance is associated to exactly one SipCall
- * create_audiocodecs_configuration() : make static
- * * #6269 : refactor AudioRtpSession
- * Fix AudioSymmetricRtpSession.h inclusion guard (cherry picked from
- commit c3081dce1cc1370d6d3558a4c4ef5cfac0d21caf)
- * * #6269: Rename AudioRtpSession to AudioSymmetricRtpSession
- * * #6574: Don't exit when connection to pulseaudio server fails
- * accountconfigdialog.h : remove some stuff from header
- * * #6560: fix configuration test
- * Fix warning in test
- * * #6560: don't hide password entry in security tab
- * * #6560: set initial password for SIP accounts
- * * #6506: remove useless pointer indirection
- * * 6560: password is now specific to IAX accounts
- * * #6560 : actually use, store, restore, transmit SIP credentials
- * * #6560: YamlEmitter: serialize sequences
- * YamlEmitterException: typo
- * ManagerImpl::computeMd5HashFromCredential() : simplify, fix memleak
- * * #6561: invite_session_state_changed_cb() : simplify
- * * #6561: More useful debug in VoIPLink::removeCall
- * * #6561 : fix ghost call reappearing in GUI after transfer
- * while -> for (make the code smaller)
- * * #6558 : Account::loadConfig() : move IAX code to IAXAccount
- * IAXVoIPLink::getAccountPtr : simplify
- * * #6554 : access the SIPVoIPLink directly, not per account
- * SIPVoIPLink is instanciated only once and is not associated to a
- single account
- * yamlnode: use const references when possible (still some left to do)
- * Account::_accountID: constify
- * VoIPLink: simplify, remove unused method
- * hudson test : no need to call run_tests.sh anymore
- * Remove AccountID type and AccountNULL define
- * Make check runs the test (no need to call run_tests.sh manually
- anymore)
- * gnome GUI: Fix tests
- * Revert "Move registration information from SIPAccount to
- SIPVoIPLink"
- * * #6392: pluginmanagertest: fix warnings reported by valgrind
- * * #6547 : remove unused exceptions
- * * #6547: CallManagerException: use runtime exceptions
- * * #6547: InstantMessageException: use runtime exceptions
- * * #6547: do not throw exceptions if some settings are not present in
- config file
- * * #6547: YamlParserException: use runtime exceptions
- * * #6547: VoipLinkException: use runtime exceptions
- * * #6547: YamlEmitterException: use runtime exceptions
- * * #6547: DTMFException: use runtime exceptions
- * * #6547: AudioFile: use runtime exceptions
- * * 6547: AudioZRtpSession: remove impossible error case
- * * #6547 : AudioRtpSession: remove impossible error case
- * * #6547: AudioZrtp: use runtime exceptions
- * * #6408 : send authenticationUsername to GUI
- * * #6408 : store/restore authenticationUsername from config file
- * SIPAccount: simplify
- * Move registration information from SIPAccount to SIPVoIPLink
- * SIPAccount::getAccountDetails : simplify
- * * #6540: yaml parser: simplify
- * sdp.cpp : fix a warning
- * * #6540: yaml parser : remove std::string typedefs
- * * #6540: Simplify yaml unserialization
- * * #6540 : add a Conf::ScalarNode constructor for booleans
- * setAccountDetails(): simplify
- * * #6408: store authentication username in daemon
- * * #6408: Be able to set the authentication username in the GUI
- * * #6507 : do not crash if the program is not sflphoned
- * Fix tests
- * macroify SIPAccount::unserialize()
- * Move all .cpp files from sflphoned target to libsflphone.la, except
- main.c
- * main() : simplify, return positive error codes
- * * #6507 : find codecs dir in build directory
- * * #6392: Sdp: move clean functions to destructor
- * AlsaLayer::adjustVolume() : simplify
- * alsalayer : reduce indentation
- * malloc/free -> new/delete
- * malloc/free -> new[]/delete[]
- * malloc/free -> new/delete
- * AudioSrtpSession: simplify base64 encoding
- * * #6392: Initialize std::string from pj_str_t correctly
- * * #6392: AudioRtpSession: Initialize remote port
- * Audio settings : Initialize _echoCancelTailLength and
- _echoCancelDelay(0)
- * Initialize variable
- * YamlParserException : fix use of stack variable after it has been
- deallocated
- * * #6392: fix memory leak in history
- * * #6392 AudioCodec : fix memory leak
- * * #6392 : fix memory leak in sip account
- * * #6408: clean up sipaccount (cosmetics mostly)
- * sipaccount.cpp serialize() : reduce number of lines
- * * #6392: invalid memory access
- * * #6392 : fix invalid memory access
- * * #6479: merged useful code from MimeParameters into Codec interface
- * * #6462: fixed hangup on IP2IP call
- * added run_daemon.sh script
- * test: remove unused variable
- * Remove functions only used by a failing test (cherry picked from
- commit fcf718cb75de7f1882dc61c07bb8d300dfa10f85)
- * * #6360 : make client tests build (cherry picked from commit
- 028b2835f040e51ab8ab979b32732b07b8798fce)
- * * #6360 : fix warnings in check_global test (cherry picked from
- commit 9e2bd6a7496dd64f6f48595e385760019aab1193)
- * * 6360: updated API calls in tests, but they're not building yet
- (cherry picked from commit 548f6f0f919b43772a3e9c667e5e292791281795)
- * Fixed include in tests (cherry picked from commit
- aeadc7525c1e31f936670ac8b02f0bcf387c38a8)
- * Remove unused variables and functions
- * IAX: fix warnings (cherry picked from commit
- fd7a113a11cac2cd9a7c36929e88ad28195c4c35)
- * Remove unused DEBUG define which interferes with logger.h (cherry
- picked from commit b2f72b91d0f43cb1dd94d138882a8caa9c841c24)
- * * #6392: no need to check for account NULLity since it is
- dereferenced above
- * * #6392: fix a memory leak, replace by stack allocation
- * * #6392: remove a variable assignement which confuses cppcheck
- * process_conference_participant_from_serialized() : remove unused
- function
- * * #6392: s/free/g_free/
- * * #6392: fix a memory leak in abookfactory_load_module()
- * * #6392: remove generate_call_id() used only once
- * * #6392: fix memory leak (opendir() without closedir())
- * * #6392: AudioRecorder(): ensures mbuffer is set
- * Remove SFLPHONED_VERSION from global.h, use autoconf PACKAGE_VERSION
- * #6298: Cleanup
- * #6331: Fix deleting ringtone file after call have been answered
- * * #6330: merged user_cfg into headers
- * #6298: Fix conference recording file update at conference end
- * #6298: Fix record file name serialization for conference
- * * #6295: cleanup of codec hierarchy
- * #6298: Fix gtk warnings
- * * #6300: added script to run tests
- * #6109: Add recording playback for conference
- * * #6300: tests do not require an installed sflphone
- * * #6295: re-removed clone methods
- * #6109: Fix gtk_critical warnings for incoming calls
- * #6109: Fix GTK_CRITICAL warning
- * #6109: Fix icons when history is not activated
- * #6109: Fix warnings
- * #6109: Implement stop recorded file playback signal
- * Revert "* #6295: removed unused clone method"
- * * #6295: removed unused clone method
- * * #6296: removed non existant file from Makefile.am
- * #6109: Stop fileplayback for outgoing call
- * #6109: Implement stop recording playback button
- * Fix binding names errors in dbus introspection file
- * #6109: Implement playback recorded file callback in client
- * #6109: Store recorded file path on client side
- * #6109: Add dbus methods for call recording playback
- * * #6290: remove unused classes from utilspp
- * * #6288: cleanup sdp
- * * #6288: fix exception usage
- * * #6288: simplify SdpException
- * * #6288: cleanup in sdp.cpp/h
- * #6109: Only display playback button if record file is set and valid
- * * 6290: updated configure.ac to remove functor Makefile
- * * #6290, #6289: removed unused classes from utilspp, fixed make
- check
- * #6109: Add button for history playback of recorded file
- * * #6289: removed unused observer class
- * * #6282: forward declare sdpMedia in sdp.h
- * * #6281: renamed setCallAudioLocal->setCallMediaLocal
- * #6183: Handle conference with more tahn two calls
- * #6183: Fix history icons when calling back a conference from history
- * #6183: Fix icons inconsistencies in history for conference hang up
- * #6183: Fix toolbar actions when selecting a conference in history
- * #6183: Fix conference serialization
- * #6268: Serialize only calls
- * * #6269: removed useless type testing
- * ignore some files in test/
- * * #6268: Remove dead class AudioSymmetricRtpSession
- * #6251: Do not had history calls in calllist when loading history
- file
- * #6251: Fix insertion in history map in before saving history file in
- daemon
- * #6251: Fix history unit tests
- * #6251: Order the list before serailization, get rid of the hashtable
- in history
- * #6251: Implement history serialization using a list wether than a
- map
- * * #6253: remove external audioport from header, make all members
- private
- * * #6253: don't store external local audio port (used for NAT) in
- Call
- * #6251: Add start_time timestamp in history serialization
- * #6251: Fix call insertion in conference items
- * #6233: Fix serialized account list terminated with a ";" character
- * #6238: Fix draggable history calls into current calls
- * #6233: Fix toolbar updates
- * #6233: Fix history
- * * #6235: remove pyc files from git tree
- * #6233: Handle cases when one or manuy calls are unreachable in
- createConfFomrParticipantList
- * #6233: Handle wrong numbers in createConferenceFromParticipantList
- * #6231: Fix drag-n-drop issue
- * * #6173 : move sippxml in tools
- * #6231: Fix merging issue
- * #6183: Implement conference unserialize
- * * #6212: remove extraneous flags from globals.mak
- * #6183: Unserialize conference data in conference
- * #6183: Add account information in request for conference call from
- history
- * #5755: Add -ldl to liker in sflphone-client-gnome
- * #5755: Fix fedora 15 compilation issue
- * #6183: Serialize conference participant phone number and account
- * #6183: Add conference timestamp in serialization
- * * #6186: don't include global.h, just logger.h
- * #6183: Fix saving history to file
- * #6183: Fix removing call from calllist
- * * #6184: remove pointers to Manager from AudioRtpSessions
- * #6183: Calling calltree_add_call explicitely for history
- * #6183: Ability to store conference inside history tab queue
- * * 6181: remove unused API from sipcall
- * #6171: Implment nreCallCreated callback
- * #6167: Fix participant list NULL ending
- * #6149: First draft of conference creation from history
- * #6149: Fix multiple call/conf selection callbacks ...
- * #6129: Fix place_call function called twice for pressing enter
- action
- * #6129: Fix double click action for history
- * #6149: Add dbus call for creating conference from history
- * #6129: Fix placing call from history and addressbook (still need to
- fix icon)
- * * #6148: removed unused AudioRtpFactory constructor
- * * #6145: remove unused isAudioStarted
- * * #6145: remove unused isAudioStarted
- * #6129: Add conference into history, fix call/conference selection
- * * #6143: don't use getType outside of serialization methods
- * * #6132: forward declarations instead of includes
- * * #6132: add constness, remove redundant "inline" keywords
- * #6129: Add timestamp to conference object to order history entries
- * * #6128: remove unused forward declarations from header
- * * #6127: make noncopyable class actually noncopyable
- * * #6125: don't include AudioRtpFactory in sipcall.h
- * #6123: Fix alsa ringback audio file
- * #6123: Fix raw audio file loading problem
- * #6109: Fix daemon plugin manager unit test
- * #6109: Fix history manager unit tests
- * #6109: Recording filename in daemon and client for history items +
- serialization
- * #6109: Refactor AudioFile to play recorded call
- * * #6104: AudioCodec moved to sfl namespace
- * * #6099: remove active flags from codec classes
- * #6095: Add notification-daemon as a runtime dependencies for rpm
- packages
- * #6095: Fix fedora 15 compilation in MineParameters.h
- * #6095: Declare static variable explicitely for client
- * #6095: Add logs to build OSC build machine
- * * #6098: global variables should have file-scope to avoid name
- conflicts
- * #6095: Fix compilation error for Fedora 15
- * #6095: Update SFLphone version to 0.9.14
- * #6095: Add specification file in opensusse build service for
- sflphone-plugins
- * #6073: Fix sflphone-plugins build on launchpad
- * #6093: Rename CodecDescriptor for AudioCodecFactory
- * * #6089: fix warnings in make check
- * * #6086: renamed codecs methods to audio_codecs
- * * #6085: renamed codec related dbus calls to audio_codec
- * #6065: Remove g_print from client, use DEBUG instead
- * #6065: Add actions name for addressbook
- * * #6085: renamed codecs* widgets/functions audiocodecs*
- * #6065: Fix Addressbook runtime warnings
- * #6065: Replace Codecs tab for Audio in account preference dialog
- * #6065: Fix "transfert" typo
- * #6065: Fix addressbook action runtime warning in uimanager
- * * #6082: fixes make check by adding libcrypto libs to test
- dependencies
- * #6073: Rename plugin/addressbook folders for addressbook/evolution
- in sflphone-plugins
- * #6074: Removed AC_SUBST from configure.ac when using
- PKG_CHECK_MODULE
- * #6073: Fix sflphone-plugins package build
- * #6073: Fix sflphone-common build
- * #6065: Fix runtime gtk warning when initializing searchbar without
- addressbook
- * #6063: Fix mozilla-tellify gitignore
- * #6063: Remove stream copy file using ifdef macro
- * * #6012: fix make dist for sflphone-common
- * #6063: Update .gitignore file
- * #6058: Fix base64 encoding related warnings
- * #6056: Fix SdpException handling
- * #6055: Fix unknown pargma warning for gcc <= 4.5
- * * #5949: test gcc version before disabling unused-but-set warning
- * #6054: Fix addressbook plugin compilation warning
- * #6048: Fix uimanager static initialization
- * #6046: Fix addressbook factory static initialization of member
- addrbook
- * #5979: Fix implicit function declaration warning
- * #6042: Fixed discarding qualifier warnings in client
- * #6041: Fix instant messaging unhandled case warning
- * #5994: Implement set current addressbook name and search type in
- addressbook plugin
- * #5994: add rules for launchpad packaging of addressbook plugin
- * #5994: Fix addressbook plugin configuration loading
- * #6027: Fix addressbook enabled test from configuration
- * #6027: No need of gnomedoc related macros in addressbook plugin
- * #6027: Add NEWS file required for build
- * #6027: Add addressbook plugin autogen.sh script
- * #6027: Remove plugins from client
- * #6027: Add sflphone-plugins folder at project's root level
- * #5994: Move addressbook folder from contacts to plugin folder
- * * #6011: removed unused Makefiles
- * * #6010: remove unused headers
- * * #5952: fix "string constant to char*" warnings
- * * #6009 fixed warnings
- * * #6003: finished cleanup of account classes
- * * #6003, #6004: cleanup of account classes, defaultAccount no longer
- global
- * * #6000: fix memory leak of args object
- * * #5998: removed using namespace std from networkmanager
- * * #5998: removed "using namespace std" from ZrtpSessionCallback
- * * #5998: removed using namespacestd from AudioZrtpSession.h
- * * #5998: remove "using namespace std" from auriorecord.h and
- MimeParameters.h
- * * #5998: remove using namespace std in main
- * * #5998: removed "using namespace std" from logger
- * * #5949: test gcc version before disabling unused-but-set warning
- * #5994: Installation of addressbook plugin
- * #5979: Implement codec full addressbook search from plugin
- * #5979: Implement addressbook factory and plugin
- * * #5981: unused webwidget removed
- * #5966: Account config synchronization fix (for stun)
- * #5954: Handle media name exception
- * #5954: Fix audio codec name display in client
- * #5954: Clean up getSessionMedia methods
- * * #5957: getRecordingSmplRate returns a value
- * #5954: Clean up getCurrentCodec methods
- * * #5950: remove "converting to non-pointer type 'int' from NULL"
- warnings
- * #5915: Full gain control version
- * * #5949: remove more unused variable warnings
- * * #5949: remove unused/unused-but-set variable warnings
- * * #5949: show_preferences_dialog returns a success value
- * * #5946: cleanup of include directives, undefined function
- * * #5515: comment out SSLv2 calls in pjsip
- * #5915: Implement different slope for attack tme and release time for
- gain control
- * #5915: use only one input signal for gain control (removed output
- buffer)
- * #5921: Fix no audio after holding a conference
- * #5916: Add gaincontrol files
- * #5916: Implement FFMPEG/CCRTP video streaming prototype
- * #5903: Fix call transfer during a conference
- * #5915: implement rms detector, first order averager, limiter for
- gain control
- * #5914: Fix call transfer when no notification request is required
- * #5899: Fix conference right-click segfault
- * #5884: temporary fix segfault in pjsip memory pool
- * #5883: Fix compilation issues on maverick and lucid
- * #5755: Fix fedora 15 compilation without patching ccrtp
- * [#5855] Make echo canceller optional
- * #5855: Fix echo suppression activation/deactivation
- * #5855: Implement pjsip echo canceller
- * #5814: Speex initialization function uses samples, not bytes
- * #5814: Test using more unbalanced signals
- * #5814: Fix buffer size for long echo length or long echo delay
- * #5814: Adjust level for echo cancellation at runtime
- * #5814: Process noise reduction before echo cancelling
- * #5814: Implement speex post echo canceller processing
- * #5814: Dump echo cancel file to disk
- * #5814: Add parameters for echo cancel
- * #5809: Add configuration parameters
- * #5809: Implement speex echo canceller in audio rtp session
- * #5814: Code cleanup
- * #5814: Fix conf creation with several incomming ringing calls
- * #5814: Fix conf creation segfault when dragging a call on hold on a
- ringing call
- * #5809: Added unit test for echo cancellation and implemented
- "process" virtual method
- * #5709: Add always recording option in configuration
- * #5709: Add always recording option in audio conference panel
- * #5709: Add core functionnality for always recording (missing config
- options)
- * #5769: Fix conference participant handling (detach/attach) and hold
- actions
- * #5747: Fix recording icons and state for conference when adding new
- participant
- * #5769: Code cleanup
- * #5769: Fix hangup unsent calls
- * #5769: Fix remove/add additional participant to conference
- * 5769: Several fixes concerning confererence handling
- * #5769: Fix compilation error
- * [#5769] Fix audio streams binding in main buffer
- * #5769: Removed access to audio mixer from audio layer
- * #5765: Fix audio crash for illformated wavefiles
- * #5765: Add maximum iteration for finding fmt and data "chunck"
- * #5589: Fix compilation of libnotify under
- * #5757: Fix abort signal when receiving INFO
- * #5747: Add usersDetached.svg
- * #5747: Handle offhold action for recording conference
- * #5747: Fix off hold action for conferences
- * #5747: Implement update conference in record action in calltree
- * #5747: Add new icons for recording conferences
- * #5747: Add recording state for conferences
- * [#5738] Remove getAudioDriver call from manager (replace by
- _audiodriver var)
- * [#5738] Refactor mutex protecting audiolayer
- * [#5737] Fix HD conference recording
- * [#5730] Fix start audio session after changing sampling rate
- * [#5714] Fix enter keyboard event for addressbbok and history
- * [5695] Fix addressbook combo box update when no addressbook selected
- * [#5695] Fix addressbook initialization and search bar update
- * [#5695] Add mutex for books_data in addressbook to protect async
- calls
- * [#5695] Get back addressbook open from uri
- * [#5695] Fix absolute addressbook URI for local addressbooks
- * [#5695] Implement libebook 3.0 interface
- * [#5571] Better logic for hangup (for case where call have not been
- sent yet)
- * [#5571] Update error handling in voip links
- * [#5571] Fix compile time warnings
- * [#5696] Fix installation dependencies for Natty
- * [#5669] Add mention that sflphone.org is for testing only
- * [#5693] Add natty in teh dput.conf file
- * [#5690] Remove not useful logs
- * [#5670] Use dynamic payload type for rtp dtmf
- * [#5668] Clean up sflphone configuration logging
- * [#5668] Fix hook checkbox configuration update
- * [#5666] Fix unit tests
- * [#5666] Manage event subscription
- * [#5666] Emit bye request when subscription is terminated
- * [#5666] Bye request should be sent after event subscription
- notification is done on transfer
- * [#5666] Make reinvite method static (to be called in pjsip
- callbacks)
- * [#5666] Hangup Call in manager for AccountNULL and IP2IP
- * [#5589] Use PKG_CHECK_MODULE for every client's dependencies
- * [#5623] Enlarge initial size of pjsip memory pool for calls (16k)
- * [#5564] Fix audio recording resampling for g722
- * [#5571] Move attribute handling for onhold/offhold actions in SDP
- session
- * [#5571] Codec negotiation refactored and unittested
- * [#5571] Implement tests
- * [#5571] Implement pjsip negociator
- * [#5571] Fix unit tests
- * [#5571] Add Fmtp.h to repository
- * [#5571] Integrate mime types and codec factory
- * [#5571] Handle exception when SDP negotiation fails
- * [#5570] Add sflphoned-sample.yml in repository
- * [#5564]: Implement stereo to mono mixing for rigntone
- * [#5342] Update audio stream initialization
- * [#5514] Restore test ni historytest suite
- * [#5514] Fix
- * [#5514] Disable test_create_history_path
- * [#5514] use pulseaudio in sample config file
- * [#5514] Fix test: load history from file
- * [#5514] Do not use X
- * [#5513] Make unit tests compile successfully
- * [#3947] Enable unit tests in Jenkins
- * [#5454] Fix build system to handle new version number
- * [#5454] Update languages from launchpad
- * [#5454] Add --without-celt in OpenSuse build service
- * [#5454] Change version number
- * [#5331] Added first SDP session tests
- * [#5273] Update nightly build version tags to conform dpkg rules
- * [#5211] Refactor send register method for iaxvoiplink and
- sipvoiplink
- * [#3950] Remove call being transfered from calltree
- * [#5211] Use appropriate memory pool for transport selector
- * [#5211] Fix strict aliasing rules warning in pjsip
- * [#5211] Bring back pjsip shutting down sleep to 1000 ms
- * [#5211] Fix registration callback segfault when closing the
- application
- * [#5211] Use the dialog memory pool for Route header in INVITE
- request
- * [#5211] Add temporary memory pool for findLocalAddressFromUri and
- findLocalPortFromUri
- * [#5211] Use individual memory pool for dtmfs
- * [#5211] SipVoipLink refactoring
- * [#3950] Attended transfer for conference calls
- * [#5284] Fix DNS resolution for Route with specified port number
- * [#5284] Some code cleanup
- * [#3947] Fix typo in hudson script
- * [#5284] Added sip route to REGISTER, INVITE, BYE request, plus DNS
- resolution
- * [#5266] Use RTP dtmf as default
- * [#5284] Added pjsip_process_route_set after setting routes in regc
- structure
- * [#5286] Fix parsing error due to long configuration file (removed
- max event)
- * [#5286] Fix false test in configuration emmiter
- * [#5286] Code cleanup
- * [#5286] Updated exception handling in configuration system
- * [#4969] Fix put SRTP call on hold
- * [#3950] Add debug messages
- * [#3950] Ability to perform an attended transfer
- * [#5276] Fix initialization problem in g722
- * [#3950] Add replace header in SIPVoIPLink::transferWithReplaces
- method
- * [#3950] Implemented attended method in SIPVoIPLink
- * [#3950] Cleanup transaction request received callback
- * [#3950] Implement dummy attended transfer in gnome-client
- * [#5249] Fix audio samplerate update algorithm for g722
- * [#5249] Fix uninitialized variable used in conditional jumps
- * [#5249] Fix conditional jump error in audiolayer (uninitialized
- value)
- * [#5267] Use autoconf 2.65 as a requirement (instead of 2.67)
- * [#5267] Restore manual pjsip configuration and compilation
- * [#5267] Autodetect celt version (0.9.1, 0.7.1)
- * [#5267] Fix deprecated macros in gnome client configure.ac
- * [#5267] Update configuration for libcelt-dev
- * [#5267] Fix build autoconf and automake
- * [#5227] Deactivate automatic call to astyle after compilation
- * [#5242] Hangup every calls before leaving
- * [#5237] Will now nightly-build for natty, Karmic deprecated
- * [#5229] Use inner class for rtp thread instead of inheritance
- * [#5211] Move mainbuffer unbind call in rtp final method
- * [#5211] Initialize sip call memory pool using 16 kb
- * [#5211] Use call memory pool in session reinvite
- * [#5211] Add debug messages
- * [#5211] Use and internal pool for calls
- * [#5211] Reduce pjsip memory pool usage for stateless error messages
- * [#5211] Refactor call deletion
- * [#5212]
- * [#5208] Refactor codec management for accounts
- * [#5168] Remove printf from codec's encode & decode method
- * [#5168] Fix celt compilation on launchpad
- * [#5168] Fix sflphoned compilation warnings in audiocodec.h
- * [#[#5168] Must keep the g722 specific RTP rate to avoid incoming
- packet timeout
- * [#5168] Fix static/dynamic payload rtp session update
- * [#5168] Throw SIPVoipLink Error if codec not instantiated in new
- outgoing call
- * [#5168] Fix dynamic/static codec payload type ambiguity
- * [#5169] Fix doubled IP2IP profile when no config file
- * [#4867] Add gtkinfobar in configuration panel
- * [#4867] Disable input/output/ringtone selection when using default
- alsa plugin
- * [#4952] Patches for possible buffer overflows
- * [$4885] Fix schemas problem
- * [#4885] sflphone-client-gnome.schemas not present during build
- * [#4885] Add gconf shemas directories in opensuse build system
- * [#4885] Add file/folder ownership for opensuse-factory build system
- * [#4906] Fix opensuse-factory build
- * [#4885] Update name dependency for libedataserver
- * [#4885] Fix non-void function without return in dbus-c++
- * [#4895] Update language translation
- * [#4896] Update session timestamp when updating media
- * [#4896] Reapply RTP hack for G722 payload type
- * [#4896] Update recording sampling rate when updating codec
- * [#4897] Save codecs in config for each configuration changes
- * [#4895] Do not save config when sflphone quit
- * [#4885] Update date for copyright
- * [#4885] Deactivate siptest that require more than one sipp instance
- * [#4879] Remove inmcoming call notification from IAX
- * [#4885] Some cleanup
- * [#4874] Add setCancel immediate/deffered for ost::Thread
- * [#4879] Fix incoming call notification
- * [#4878] Set keyboard focus on searchbar when selecting addressbook
- * [#4874] Fixed compilation warning
- * [#4874] Fixed compilation warning in sipvoiplink
- * [#4874] Fix compile time warning in RTP record handler
- * [#4874] Fix conditional jump in SDP
- * [#4874] Fix conditional jump based on uninitialized value
- * [#4874] Store call id within rtp thread context
- * [#4874] Fixed conditional jump based on uninitialised value in
- conference
- * [#4871] Fix default account fetching
- * [#4870] Delete RTP session when Refusing an incoming call
- * Restore IP to IP call
- * [#4857] Fix audio codec negotiation problem
- * [#3947] Adjust ressources allocated to compilation
- * [#3947] Disable unit tests in Hudson
- * [#4305] Free mutex only when really quiting SFLphone
- * [#4859] Update copyright to 2011 in every source file
- * [#3218] Character '.' stripped by the caller engine
- * [#4854] Fix typos, desktop entry
- * [#4847] Apply RTP modification to ZRTP session
- * [#4852] Update Karmic and Lucid dependencies
- * [#4852] Add Libedataserver and libedataserverui as gnome client
- dependencies
- * [#4852] Add authentication mechanism for EDS
- * [#4851] Fix segfault when closing pulseaudio layer too rapidly
- * [#4808] Some otehr cleanup
- * [#4808] Made some cleanup
- * [#4808] Added mutex in rtp session for codecs and noise process
- * [#4847] Update audio processing when updating RTP media
- * [#4842] Add support for linking with gold/ld --no-add-needed
- * [#4808] Make update g722 related static/dynamic payload logic
- * [#4827] Upper limit on the number of contacts to import from EDS is
- hard-coded to 500
- * [#4808] Fix put call on/off hold
- * [#4808] Implement early RTP start for incoming calls
- * [#4808] Audio stream is no longer start within RTP session.
- * [#4808] Removed coupling between audio layer and and RTP session
- * [#4702] Start audio rtp session as soon as it is created
- * [#4702] Init timestamp to 0
- * #4702: Send RTP packets immediately, no need of outgoing queue
- * [#4784] Update dbus-c++ version from gitorious
- * [#4702] Update RTP timeouts
- * [#4702] Lengthen RTP timeouts
- * [PATCH] Fixed compatibility with old libtool versions.
- * [PATCH] Accept older libebook (Maemo 5 has 1.4.2)
- * [PATCH] Fixed double-free error in preferences dialog
- * [PATCH] Fixed building of sflphone-common on Maemo5
- * [PATCH] Improved Gnome client initialization error handling. 1. It
- no longer segfaults when sflphoned isn't available. 2. User is
- provided with GUI error dialog.
- * [PATCH] Improved autogen.sh scripts 1. They do not require bash
- anymore 2. Added workaround for Debian bug #565663 3. Replaced
- manual autotools invocations with single autoreconf call 4. Non-zero
- return status on failure
- * Revert "[#4468] libtool <= 2.2 doesn't have LT_INIT macro so
- AC_PROG_LIBTOOL should be used instead."
- * Revert "[#4468] Libebook 1.4 is sufficient"
- * Revert "[#4468] Apply big path on dbus communication system"
- * [#4468] Apply big path on dbus communication system
- * [#4468] Libebook 1.4 is sufficient
- * [#4468] libtool <= 2.2 doesn't have LT_INIT macro so AC_PROG_LIBTOOL
- should be used instead.
- * [#4639] Fix determining default addressbook if this property is not
- set in gconf
- * [#4639] Fix memory leaks in Addressbook
- * [#4637] Fix opening default addressbook at sflphone init
- * [#4622] Free yaml events while parsing configuration file
- * [#4623] Fix conditional jumps based on uninitialized variable
- * [#4622] Fix leaks in yaml serialization engine
- * [#4616] Fix addressbook warnings
- * [#4514] Adjust RTP timestamp
- * #4527: Rename Karmic libyaml and Celt package in debian control file
- * #4495: Rework addressbook opening loop
- * [#4524] Increment RTP count when sending data
- * [#4524] DO NOT start RTP session twice
- * [#4367] Use PKG_CHECK_MODULE for celt
- * [#4367] Fedora package celt as celt (not libcelt)
- * [#4367] Astyling
- * [#4367] Update .po files
- * [#4367] Fix segfault in gensin
- * [#4354] Make celt a direct dependency on launchpad opensuse build
- service
- * [#4367] Make celt a required package, option --without-celt valid
- * [#4367] Fix zrtp timestamping error
- * [#4367] Fix audio zrtp timing
- * [#4367] Dispatch ZRTP packets
- * [#4367] Fix segfault when unloading account map
- * [#4367] Fix zrtp session
- * [#4367] Implement on packet receive
- * [#4367] use symetric audio rtp session, not dual
- * [#4367] Reduce packet receive/sent timeout
- * [#4367] Reduce RTP timeouts
- * [#4367] Move speaker data receive
- * [#4367] Move speaker data receive
- * [#4367] Move receive speaker data method
- * [#4367] Remove debug in rtp session
- * [#4367] Fix g722 codec clock rate
- * [#4367] Fix noise suppression initialization
- * [#4367] Fix segfault in RTP mic fadein method
- * [#4367] Refactor mic data encoding in rtp session
- * [#4367] Implement RTP main loop
- * [#4367] Fix compilation problem
- * [#4367] Fix AudioRtpclass using TRTPSessionBase
- * [#4367] Fix AudioRtpSession putDtmfEvent shadowing
- * [#4367] Fix AudioRtpSession putDtmfEvent shadowing
- * [#4367] Refactor RTP session (phase 2)
- * [#4367] Refactor RTP session (phase 1)
- * [#4367] Remove Redeclaration of SymetricAudioRtpSession in
- rtpfactory
- * [#4265] Add continue statement in for loop for invalid addressbook
- * [#4261] Makes addressbook initialization more robust
- * [#4257] Add maverick in build system
- * [#4233] Add sdp related unit tests
- * [#4233] Add condition and signal in two incoming call test
- * [#4243] Fix segfault in AudioSrtpSession
- * [#4243] Fix memory leak in AudioSrtpSession
- * [#4243] Make audio srtp optional in for incoming call
- * [#4243] Add boolean variable to make sure remote crypto context
- initialized only once
- * [#4243] Add documentation to AudioSrtpSession
- * [#4243] Use 80 bits authentication tags by default
- * [#4243] Init audio srtp remote crypto context in
- call_on_media_update
- * [#4243] Move SDP negotiastion in mod_on_rx_request
- * [#4243] Implement initLocalCryptoInfo to be called at different
- momment
- * [#4243] Init init local crypto context in when initializing audiortp
- * [#4243] Change key length according to sdes negociation
- * [#4243] Associate callid to accountid for incoming calls
- * [#4242] Fix no SDES keys in IP2IP calls
- * [#4242] Fix no SDES keys in IP2IP calls
- * [#4233] Test for call on/off hold
- * [#4233] Add two incoming call test
- * [#4233]
- * [#4233] Add 2 outgoing simultaneous call unit tests
-
- -- Julien Bonjean <julien.bonjean@savoirfairelinux.com> Fri, 30 Sep 2011 13:44:57 -0400
-
-sflphone-client-gnome (0.9.7~rc1~ppa1~SYSTEM) SYSTEM; urgency=low
-
- ** 0.9.7~rc1~ppa1~SYSTEM **
-
- * [#2462] Set explicitly the transport on incoming call too
- * [#2462] fix typo
- * [#2462] Use different address for SDP and call IP
- * [#2462] Use published address in SIP-SDP
- * [#2181] Fixed changelog files
- * [#2181] Updated spec file
- * [#2402] Fix pointer to int conversion warning (atoi)
- * [#2402] Remove daemon warnings, make indent
- * [#2459] Make sure the stream is opened when the call is answered
- * [#2402] Add conference related picture in documentation
- * [#2443] Not much ...
- * [#2399] Fix dialing display problem
- * [#2450] Fix incoming call already in conference crash
- * [#2399] Display peer name on the first line and peer number on the
- second
- * [#2450] Handle 403 FORBIDDEN when refused
- * [#2447] Bind offHold/onHold actions to button in gtk client
- * [#2447] Bind hangup action to button for conference
- * [#2447] Add conference action in gtk client's ToolBar
- * [#2381] Disable the password hashing in config file
- * [#2402] Cleanup
- * [#2366] Set callback to null when deleting Pulseaudio streams
- * [#1313] Fix main buffer unit test
- * [#1313] Fix audio layer unit test
- * [#2315] Hide pw in security tab, display when editing, sync with
- basic tab
- * [#1313] UnitTest change AudioRtpSession for AudioSymetricRtpSession
- instance
- * [#2402] Code cleanup
- * [#2444] Add debug to catch occasional crash when loading client's
- config
- * [#2444] Add debug info to catch occasional crash when loading config
- dialog
- * [#2402] Restore Call menu translations
- * [#2403] Use the published address if checked in GUI
- * [#2442] Add protection test in sdp
- * [#1841] Reapply pjsip patch concerning DNS SRV resolution
- * [#2384] Tags incoming call as direct SIP call, if applicable
- * [#2402] Change the monkey face
- * [#2315] Enable user to display password in clear text
- * [#2434] Force optimization level at 2
- * [#2284] Fix dbus_get_all_ip_interface compilation warnings
- * [#2431] Popup main window on incoming if applicable
- * [$2402] Fix simple warnings
- * [#2402] Fix implicit variable init order in LibraryManagerException
- * [#2402] Fixing implicit variable initialization warnings in
- AudioRtpSession
- * [#2402] Revert atoi change, fixing codec list doubled entries
- * [#2402] Fix gpointer to gint conversion
- * [#2402] Fix pointer casting to integer different size warning in
- codec list
- * [#2402] Fix warning discarting qualifiers from pointer target
- * [#2402] Fix gtk tree view assignement from incompatible type warning
- * [#1669] Fix audio recording folder utf-8 non compatibility issue
- * [#2414] Clean up debugs
- * [#2414] Use transport set in iptoip Account and update it frm
- preference
- * [#2348] Use macro N_() to mark ui.xml strings as translatable
- * [#2414] Rename getSipAddress/setSipAddress functions
- * [#2407] Fix volume controls display
- * [#2407] Fixes dialpad
- * [#2383] Set ip to ip config when clicking apply button
- * [#2404] Update call-to script - Maxime Chambreuil
- * [#2405] Client handles unknown call in current state as well
- * [#2383] Add DBUS signal to send IPtoIP local address and port as
- string
- * [#2383] Add Ip to IP config change apply call back
- * Clonflict
- * [#2402] Code cleanup
- * [#2383] Do the same for IPtoIP (init localn ip with first in the
- list)
- * [#2383] Use first interface in the list if local addresss is not
- defined
- * [#2403] Clean up unuseful addresses/ports
- * [#2403] Use the IP profile SIP port as global SIP port
- * [#2383] Fix dbus_get_all_ip_interface warnings
- * [#2383] Take into account sameAsLocal when loading published address
- * [#2383] Tsake into account sameAsLocal option when saving published
- address
- * [#2383] Update local ip address in ip to ip config
- * [#2383] Save ip 2 ip local port in config
- * [#2406] Update toolbar at startup
- * [#2284] Remove redefinition warnings + speex warnings
- * [#2383] Fix security table in account config
- * [#2383] Save ip 2 ip network interface parameters in config
- * [#2403] Restore sip transport selector
- * [#2383] Fix filling the Localt IP Address on account creation
- * [#2383] Fix Gtk-Critical when checking STUN
- * [#2383] Fix reopening account configuration display issue
- * [#2383] Load IPtoIP local address and port in preference iptoiptab
- * [#2383] Add LocalAddress and Localport in Preference IpToIp tab
- * [#2403] Use the address and port associated to the account as often
- as possible
- * [#1753] Removed pjsip generated files
- * [#1753] Removed remaining milenage lib references
- * [#2383] Add _publishedSameasLocal variable in sipaccount
- * [#2383] Add PUBLISHED_SAMEAS_LOCAL variable in config
- * [#2383] Fix stun set active or not when opening config
- * [#2181] Added RPM 64bits dbus patch
- * [#2402] Code indentation
- * [#2313] Force $(HOME).cache directory creation at startup
- * [#2383] Separate network interface and published address in account
- config
- * [#2400] Change dbus service installation path to libdir
- * [#2382] Move TLS related published address options in security tab
- * [#2382] Indent accountconfigdialog.c
- * [#2181] Install libdbus-c++ in $pkglib instead of $lib
- * [#1753] Remove ILBC code and disable it by default in the configure
- * [#1753] Remove milenage directory
- * [#2382] Fix switching interaface instabilities
- * [#2396] Save local ip in account creation wizard
- * [#2284] Remove warning on hold
- * [#2387] Fixes history searching and filtering
- * [#1215] Add samplerate display in the GUI
- * [#1663] Voicemail icon reflects voice messages
- * [#2395] Fix account registration ( specifically with callcentric)
- * [#2386] Strip "sip:" on incoming call, fixing history call back
- * [#2181] Updated spec files
- * [#1215] Display codec name in calltree instead of status bar
- * [#2390] Move back nbCalls and stopStream higher in refuseCall
- * [#2392] Fix ringtone during call in IAX
- * [#2391] Stop audio streams when there is 0 calls only
- * [#2391] Add debug when call state is not valid
- * [#2390] Clear returns in IAXvoipLink::sendAudioFromMic() method
- * [#2380] Fixing IncomingCallNotification not regular
- * [#2339] Query conference at client startup
- * [#2339] Working conference querying at startup
- * [#2339] Add conference in call tree
- * [#2339] Primitives to query conferences at client startup
- * [#2320] Add account selection in history
- * [#2355] Temporary solution: do not delete pointer when removing
- account
- * [#2380] Change algorithm in AudioRtp to trigger an
- IncomingCallNotification
- * [#2274] Comment sdebug in MainBuffer flush method
- * [#2274] Add flushMain() in ManagerImpl::addStream
- * [#2274] Add getBufferID() method in ring buffer
- * [#2274] Fix warning, comment debug in ringbuffer's flush method
- * [#2274] Use AudioLayer flushMain() and flushUrgent() in ALSA
- * [#2274] Clean up unused variable warning
- * [#2274] Protect minbudffer pointer on flushing
- * [#2274] Fix playATone method which writing empty buffer in urgent
- ringbuffer
- * [#2274] Use audio layer flushUrgent and flushMain in createStreams
- * [#2274] Use flush audio calls from audiolayer
- * [#2274] Flush when peer answered call
- * [#2375] Flush main buffer in iax when answering a call
- * [#2274] Parse displayname using c++ string method
- * [#2375] Flush main buffer when off holding calls
- * [#2375] Flush main buffer mon RTP startup
- * [#2376] Use now Pulseaudio module-cork-music-on-phone
- * Updated OSC packaging
-
- -- Julien Bonjean <julien.bonjean@savoirfairelinux.com> Fri, 20 Nov 2009 13:59:02 -0500
-
-sflphone-client-gnome (0.9.7~beta~ppa1~SYSTEM) SYSTEM; urgency=low
-
- ** 0.9.7~beta~ppa1~SYSTEM **
-
- * [#1933] Cleanup debug
- * [#1933] Clean up debug
- * Fix mic
- * [#1933] Set the IAx format earlier
- * [#1933] Move IAX sendAudioFromMic outside if (call) statement
- * [#1933] Fix startstream when offhold in iax and add debug concerning
- codec neg.
- * [#2371] sflphone_notify_voice_mail: minor gettext message formatting
- cleanup
- * [#2371] select_account_cb: properly gettextize status message
- * [#2371] show_account_list_config_dialog: properly gettextize status
- message
- * INSTALL: Minor tidyup of core install guide
- * Add /sflphone-client-gnome/src/icons/Makefile to .gitignore
- * [#2181] Updated OpenSUSE files (tmp)
- * [#1933] Add debug for codec negociation for iax
- * [#1933] Get rid of getMicAvail and getMicData in audiolayer (not
- used anymore)
- * [#1933] Add "audio codec not determined" error in IAX
- * [#1933] Test flush data
- * [#1933] Do not need to start audio stream in iax anymore
- * [#1933] Protecting pointer
- * [#2284] Remove more compilation/execution warnings
- * [#2284] Cleanup debug in client, use DEBUG instead of g_print
- * [#2284] Clean up uimanager
- * [#2370] Remove warnings
- * [#2366] Clean up other debug
- * [#2366] Clean up debug
- * [#2366] Call pa_xfree explicitely in writeToSpeaker
- * [#2284] Remove address book warnings
- * [#2365] Fixes bad cast
- * [#2352] Fix continuous ringing when peer hangup and call not yet
- answered
- * [#2181] Added version support
- * [#2181] Fixed some minor issues
- * [#2360] Moved MainBuffer from AudioLayer to ManagerImpl
- * [#2352] Makes getMainBuffer() everywhere
- * [#2352] Use 50 sec latency on pulseaudio stream creation
- * [#2352] Add alsa debug
- * [#2359] Update repository documentation
- * [#2354] Move pulseaudio disconnectAudioStream after stopping main
- loop
- * [#2352] Adjust nb byte copied in pulseaudio according to
- writeableSize
- * [#2352] Specify pulseaudio tlength parameters using pa_usec_to_bytes
- * [#2322] Convert italian translation to UTF-8
- * [#2357] Fixes window size
- * [#2357] Display only actionnable tool item
- * [#2333] Update streams parameters
- * [#2347] Use GNOME user settings for Menu and Toolbar appareance
- * [#2349] Load/Save properly audio params
- * [#2322] Update translations from Launchpad
- * [#2181] Added Francois Marier script
- * [#2350] Remove non-valid test
- * [#2181] Updated launchpad packaging
- * [#2333] Fix Pulseaudio Capture
- * [#2333] Use pulseaudio ADJUST_LATENCY flag and ALSA RT-SCHEDULING
- * [#2333] Pulseaudio Interpolate timing
- * [#2333] Change (again) Pulseaudio settings to fit logiteck usb hdw
- requirement
- * [#2333] Adjust pulseaudio fragment size to 4096 (max sflphone's
- frames per buffer)
- * [#2284] Remove recurrent compilation warning (g++ linker problem)
- * [#2333] Safer Audiostream parameters
- * [#2333] Fix alsa playback to reduce underrun
- * [#2333] Better audiostream parameters
- * [#2181] Updated version management
- * [#2333] Exclusive test in playback loop
- * [#2181] Updated build system
- * [#2333] Less underrun with these value
- * [#2333] Update playback audiostream parameters
- * [#2333] Lengthen the audio buffer reduce number of underrun in
- pulseaudio
- * [#2333] Add ALSA recovery functions for underrun (begin)
- * [#2333] Add pa_stream_trigger in pulse audio underrun callabck
- * [#2048] Reduce prebuffering in pulseaudio (which affect incomming
- calls' plbck)
- * [#2316] Do not display any icons to the right on the history tab
- * [#2333] Comment pa_stream_trigger in pulseaudio underrun
- * [#2333] Modify pulseaudio streams parameters
- * [#2318] Fix transfer tool button double signal
- * [#2181] Updated
- * [#2333] Fix ALSA ringtone
- * [#2333] Flush all main buffer before starting audio
- * [#2333] Open/Close Alsa thread between calls while there is no audio
- * [#2333] Add debug message and test condition on starting playback
- and capture
- * [#2181] Fixed gnome client makefile
- * [#2181] Updated
- * [#2308] Remove getTelephoneTone debug
- * [#2308] Change plughw for default in ALSA
- * [#2308] Oups, forgot to change function name in audiolayertest.cpp
- * [#2308] Cleanup in pulseaudio code (debug, function name)
- * [#2308] Fix pulseaudio stream closing assertion failure
- * [#2308] Moved pulseaudio mainloop locking from AudioStream
- disconnect stream
- * [2308] Fix latency at the beginning of a call, when playing DTMF and
- wehn starting tone
- * [#2181] Updated karmic
- * [#2317] [#2319] Fix address book toggle button contextual behaviour
- * [#2308] Stop stream when refusing a call
- * [#2308] Stop pulseaudio stream when peer hungup
- * [#2308] Fix tone and ringtone
- * [#2312] Display the STUN entry widget when opening the tab
- * [#2308] Implement two different callbacks for capture/playback in
- pulseaudio
- * [#2309] Open/close pulseaudio connections in startStream/stopStream
- * [2308] Leave pulseaudio stream running, do not cork/uncork them
- anymore
- * [#2295] Set gtk file chooser to None if nothing is set in
- configuration
- * [#1976] Add codec and conference documentation
- * [#2209] Fix recording in regard of resamling
- * [#2297] Update .gitignore
- * [#2297] Update translation files
- * [#2297] Add reference to our coding standards
- * [#2297] Remove old docbook code
- * [#2296] Reinit tls account settings after modification
- * [#2253] Add DcBlocker class to remove capture's dc offset
- * [#2034] Fixes for TLS transport to initialize
- * [#2284] Add silent build rule + client clean warnings
- * [#2274] Fix unserialize history items in cilent at startup
- * [#2274] Complete display name parsing and displaying
- * [#2274] Parse the Display Name in sip INVITE message
- * [#2050] Fix capture volume control in ALSA
- * [#1970] Volume controls disable when using pulseaudio
- * [#1970] Disable volume controls when using pulseaudio
- * [#2277] Fix direct ip2ip ZRTP enabling/disabling in ip2ip
- preferences
- * [#2181] Added launchpad debian files
- * [#2181] Added spec files for OSC
- * [#2274] Set display name for "Contact" sip header as the hostname
- * [#2181] Fixed daemon issues
- * [#2181] Fixed gnome client issues
- * [#1976] Remove warnings - need to fix the transfer
- * [#2006] Add init is_rec variable in ManagerImpl
- * [#2006] Update codec display on call selection
- * [#2006] Restore double click actions in history and contact calltree
- (GTK)
- * [#2176] use XDG_CACHE_HOME when initializing sfl.zid file
- * [#1976] Fix calltree switching from history
- * [#2209] (Re)Fix cache for zid
- * [#2209] Clean up debug messages
- * [#2209] Clean debug messages
- * [#2209] Fix trasnfering a call during a conference
- * [#2209] Speex decode must return the number of bytes
- * [#2209] Change frameSize speex 32kHz
- * [#2209] Fix speex codec framesize
- * [#2209] Reinit converterSamplingRate in RTP sessions
- * [#2209] Change speex ultra wide band framesize
- * [#1747] Add pixmap data
- * [#2252] Fix Receiving a server error 488 crashes the callee
- * [#2209] Fix iax low rate packate sending
- * [#2209] Clean up debug messages
- * [#2209] Add resampling changes for IAX
- * [#2209] Clean up resampling code
- * [#2209] Fix latency introduced by pulseaudio
- * [#2209] Fix initialization of mainbuffer's internal sampling rate
- * [#2176] Fix upsampling buffer size in audiolayer
- * [#2209] Add dynamic converter sampling rate in audiortp sessions
- * [#1747] Fixes runtime warnings
- * [#1747] Remove from repo
- * [#1747] register our icons to be used as stock icons
- * [#2209] Fix number of byte in alsa's write to speaker
- * [#2209] Fix putting non-resampled data in RTP's mainbuffer
- * [#2209] Add alsa resampler
- * [#2209] Add a samplerate converter in PulseLayer
- * [#2209] Add mainbuffer's internal sampling rate and flushall method
- * [#2176] Add mainbuffer stateInfo debug method
- * [#2209] Resampling is optimal using SRC_LINEAR not SRC_FASTEST
- * [#2176] Remove debug recordings
- * [#2176] Fix Holding a conference participant on new calls
- * [#2224] Add confID in callable object
- * [#2176] Fix putting onhold a call participating to a conference when
- pressing new call
- * [#2176] Reset auidio buffers when adding streams (rtp, audiolayer)
- * [#1976] Use xml to describe toolbars - Add a naviguation toolbar
- * [#2176] Remove conference default_id in joinParticipant
- * [#2176] Display error message in alsa's snd_pcm_avail_update call
- * [#2176] Alsa mic avail data debug
- * [#2176] Add some debug message for mic loss problem
- * [#2176] Flush mic ring buffer when offholding a call
- * [#2176] Reset ringbuffers' readpointer when adding main participant
- * [#2176] Fix getAvailData algorithm
- * [#2176] Reset ringbuffer's readpointer when adding a new participant
- to a conference
- * [#1744] Regex object renamed to Pattern. Previous attempt at
- providing
- * [#2176] Fix detach main participant problem when adding new one
- * [#1976] Use right domain to translate
- * [#1976] Add xml menu description
- * [#2176] Store a list of confernece participant in client
- * [#2176] Fix add participant, joinparticipant methods
- * [#2181] Do not install dbus-c++ headers + add return value
- * [#2176] Fix minor call handling instabilities
- * [#2174] Fix incoming IP call contact address
- * [#2211] Add test to protect NULL pointer
- * [#1163] Add Advanced account configuration section
- * [#2176] Add some usefull comments and debugging info
- * [#2176] Add conditions to display security icons in conference
- * [#2176] Fix detaching one participant while keeping communication to
- others
- * [#2176] Reenable userActive.svg in call tree
- * [#2176] Make user active blue (not red)
- * [#2176] Fix user active picture
- * [#2176] Fix "hidden" merge conflict in sipvoiplink
- * [#2176] Remove iax audio stream on peer hungup
- * [#2174] Multiple UDP transports functional (TESTED with 2 accounts
- and 3 calls)
- * [#2176] Fix fix audio stream binding in iax
- * [#2174] Create a default UDP transport + use tp selector for dialogs
- also
- * [#2176] Register iax audio stream in mainbuffer
- * [#2176] Fix getAudioCodecName in IAXvoipLink
- * [#2176] Fix iax account init
- * [#2176] Handle multiple account using the same sip transport
- * [#2165] Add .png files
- * [#2176] Small fixes concerning dtmf
- * [#2176] Fix make uninstall in codecs
- * [#2174] remove stund makefile generation
- * [#2176] Add conference lock
- * [#2174] Add transport selector for multiple accounts
- * [#2176] Change userActive picture from red to blue
- * [#2176] Fix security pixbuff in calltree
- * [#2176] Replace sfl.zid in .cache/sflphone instead of .sflphone
- * [#2176] Fix add call description
- * [#2176] Remove detach button from toolbar
- * [#2176] Fix calltree call description state and state code in
- conferences
- * [#2176] Fix pulse audio double free
- * [#2176] Fix conference selection
- * [#2174] Clean up - remove stun settings in client network
- configuration panel
- * [#2174] Remove voviva stun code
- * [#2174] Rsolve STUN with pjsip - DO NOT WORK
- * [#2165] Add user svg
- * [#2165] Debugging sip call failed
- * [#929] Link against uuid if installed
- * Oops
- * Fixed bugs related to libsexy (with GTK < 2.16)
- * [#929] Remove uuid-dev dependency in the core
- * [#2165] Debugging no negociated codecs at communicatio start
- * [#2165] Fix calltree bug (gtktreestore instead of gtkliststore)
- * [#2165] Fix several merge problems
- * Updated opensuse packaging script
- * [#1163] Add missing figures
- * [#1163] Update INSTALL file
- * [#2165] Fix IAX
- * [#2165] Add recordabe interface
- * [#2165] Finish recording refactoring for call (not for conference)
- * [#2165] Enable speaker recording for two different calls
- simultanously
- * [#2165] Implement call recording using the Recordable interface
- * [#2165] Add get and set to AudioLayer's audio recorder
- * [#2165] Add class recordable from which inherit call and conference
- * [#2006] Fix G722 and Speex 8khz codec conferencing
- * [#2006] add recording of audio buffers
- * [#1163] Add general settings section
- * [#1163] Fixes makefile error
- * [#2006] Fix some minor issues
- * [#2006] Drag a conference call on another conference call
- (difference conferences)
- * [#2006] Fix dragging a conference on itself
- * [#1744] Integrating some of the needed regular expression patterns
- in order
- * COmplete call features
- * [#1744] Added support for named subgroup in the Regex object. Also,
- new
- * [#1744] Adds thread safety features, compile() and setPattern()
- methods to the Regex class.
- * [#1744] Fix inconsistency in the finditer method from the last
- commit.
- * [#1744] Added regex pattern object built on top of libpcre. To be
- used
- * [#1744] Initial commit towards implementing RFC4568. Unimplemented
- in the
- * [#2157] Hide "security" and "advanced" tabs for IAX under account
- * [#1163] Add call features section
- * [#2006] Add joinConference capabilities
- * [#2006] Add dbus joinConference signal
- * [#2006] Drag a conference call onto a conference to add it
- * [#1163] Add addressbook section
- * [#2006] Drag a conference call onto a single call to create a
- conference
- * [#2006] Expand rows automatically
- * [#2006] Add minimal multiple conference handling
- * [#2006] Add atached/detached conference icons
- * [#2006] Add function processRemainingParticipant
- * [#2006] Deep refactoring, fix hangup bug
- * [#1163] Update documentation - Accounts part
- * [#1976] Integrate user doc to gnome client build system
- * [#2122] Remove double inclusion in dbus-c++/src/Makefile.am
- * Remove pjproject version number
- * [#2006] Fix peerHungup
- * [#1976] Make Yelp accessible from the GNOME client (need to install
- the sflphone.xml first)
- * [#2006] Fix multiconferencing hangup
- * [#2006] Fix hangup calls in a conference
- * [#2150] Make IAx2 reappear
- * [#2006] Fix detach participant on multiple call
- * [#2006] Can remove rining call from a conference
- * [#2006] Reinit confID when removing a participant
- * [#2006] Remove get isCurrentCAll in hangup/peerhungup (SipVoipLink)
- * [#2006] Fix refuse call
- * [#2006] Fix answerring incoming call
- * [#2006] Refactor conference's participant list
- * [#2101] Re-integrate test compilation in main build system
- * [#2101] Make the test directory compile
- * [#2136] Restore history functionality
- * [#2006] Fix binding main participant to himself
- * [#2006] Fix add current/incoming/onHold participant to an existing
- conference
- * [#2006] Fix add incoming calls to an already created conference
- * [#2006] Fix remove stream
- * [#2006] Fix detachParticipant/removeParticipant switchCall ids
- * [#2006] Fix adding a call in conference having state "CURRENT"
- * [#2006] Remove/add main participant from conferences
- * [#2006] Hold/unHold conference
- * [#2006] Detach a partcipant from drag n drop
- * [#2006] Hangup a conference
- * [#2006] Add hold/unhold conference dbus messages
- * [#2034] gtk-ui fix under the "basic" tab.
- * [#2006] Fix dragging calls on conference calls
- * [#2006] Fix detach participant from a conference
- * [#2034] Added default message is status bar under the account config
- dialog
- * [#2112] Fix a crashed caused when a non-md5 password was sent to
- pjsip.
- * [#2006] Detach participant by ID
- * [#2006] Fix addParticipant method in managerImpl to handle
- incoming/answered calls
- * [#2006] Add addParticipant method in managerimpl and related dbus
- messages
- * [#2111] Added the ability to configure zrtp on sip.sflphone.org from
- * [#2106] Fixed problem in the account assistant under gtk-ui. Also,
- assistant.c
- * [#2006] Fix dragging a conference call on another conference call
- (same conference)
- * [#1904] Small UI fix. Assistant was moved from "Call" to "Edit"
- menu.
- * [#1904] Fix a wrong label under gtk-ui.
- * [#2034] Renaming and source code splitting.
- * [#2034] Status bar added to account window to better reflect the
- registration
- * [#2006] Make calltree_remove_call recursive (for GtkTreeStore)
- * [#1110] Small gtk-UI fix in the account window (alignment).
- * [#2006] Fix remove conference, display children which are still
- active
- * [#2006] Recursive function call in calltree_update_call
- * [#2006] Add multilayered capabilities to calltree (GtkTreeStore)
- * [#2006] Implement remove conference in calltree
- * [#2034] Now useless as Direct Ip calls settings moved under
- Preferences.
- * [#2034] Edit/add buttons were set insensitive all the time under
- gtk-ui.
- * [#1887] Information about the state of the current SIP call is
- displayed
- * [#2006] Add call tree remove callback
- * [#2006] Fix create_conference function
- * [#2006] Update conference_added_cb to add new conference to the list
- * [#812] Added new tab under GTK-ui Preferences. Moving Direct Ip
- Calls from
- * [#2121] Disable temporarily test compilation
- * [#2006] Fix conferencelist to handle conference_obj_t instead of
- gchar
- * [#2006] Add conference_obj structure
- * [#2121] Update version
- * [#2006] Fix conference selection
- * [#2101] Use the new source tree to fetch the right object files
- * [#2006] Add conference in calltree
- * [#2006] Add Dbus signal conference added/removed/changed
- * [#2006] Add getConferenceDetails call on dbus
- * [#1904] Registration expire now appears as a spin box under gtk-ui.
- * [#812] Fixing a segmentation fault caused by a non-existing account
- ID
- * [#2006] Add getConfList method over dbus
- * [#2006] Add a conferencelist data structure in client-gnome
- * [#812] Defaults value are now sent if a non-existing account is
- requested
- * [#2006] Add sflphone action sflphone_join_participant
- * [#2006] Fix buffer read pointer problem deletion
- * [pjsip] Attempt at fixing via header incompatibility with
- Freeswitch.
- * [#1797] forget something
- * [#2006] Add call new state conferencing in deamon
- * [#2006] Remove addParticipant method for conference, use
- joinParticipant only
- * [#1163] Update INSTALL documentation
- * [#812] Msec/sec values were not taken into account.
- * [#1797] Make pjproject-1.4 compile
- * [#2006] Add Detach participant method
- * [#2006] Dragndrop fully functional with INCOMING and HOLD call
- * [#1797] Add pjproject-1.4
- * [#1797] Remove pjproject-1.0.3
- * [#2006] Get call state in conference related function
- * [#2006] Add joinParticipant (conference) method in ManagerImpl
- * [#2006] Add joinConference DBUS message
- * [#2006] Store the previously selected call_id on dragndrop
- * [#2006] Fix GValue pointer unref in selection callback
- * [#2006] Store dragged call_id
- * [#2006] Update drag_data_received_cb callback to manipulate CallIDs
- * [#2006] Add dragndrop signals
- * [#2006] Set calltree reordable
- * [#812] Adds the ability to create a TLS listener in case the user
- requests
- * [#812] Adds the ability to configure local/published address from
- * [#1883] Move switchCall in onHoldCall function
- * [#812] Deals with the published address/port problem when
- integrating TLS.
- * [#1883] Switch call id in managerimpl when peerHungUp
- * [#1883] Switch call id before hangup
- * [#1883] Add usefull and permanent debug info for conference
- cretion/deletion
- * [#812] Fix various segmentation faults related to Direct IP kind of
- calls.
- * [#1883] Fix deletion of std::map elements using iterators
- * [#2014] Add libzrtpcpp build dependency
- * [#1883] Still some for loop test ambiguity (while loop instead)
- * [#1883] Fix for loop initial test ambiguity (use while loop instead)
- * [#1883] We must discard data in urgent ring buffer if data is get in
- mainbuf
- * [#1883] Fix availForGet same id for ringbuffer and readpointer
- * [#812] Match "sips" as a Direct IP Call when the user enter a sip
- uri
- * [#812] Fix segmentation fault related to SIP URI creation.
- * [#812] Towards integrating multiple tls listeners at the same time.
- This
- * [#1883] Add debug messages in conference and fix mainbufferTest
- * [#812] gkt-ui fix. Private key must be fed as a filename and not as-
- is.
- * [#812] TLS integration within sipvoiplink and pjsip. Also,
- configure.ac
- * [#1883] Fix Alsa/Pulse mallocation
- * [#1883] Fix data corruption in AudioRtp's micData buffer
- * [#812] Full dbus integration for all the tls related options under
- gtk-ui.
- * [#1883] Fix memory leaks in audiortp session
- * [#1883] Fix mem leaks in audio rtp
- * [#812] Fix setAccountDetails where TLS_ENABLE was set to the value
- * [#812] Small gtk-ui fix.
- * [#811][#812] Small gtk-ui fix.
- * [#812] Introduced a mechanism for configuration files that makes
- possible
- * [#812] New dbus bindings added. Also, configuration compliance was
- enforced
- * [#1881] Remove default buffer from MainBuffer (update unit-tests)
- * [#1881] Add ring buffer read pointer tests
- * [#1883] Fix issues in ringbuffer reader pointers
- * [#2034] Implementing a new configuration dialogue for TLS transport
- settings
- * [#1883] Add some usefull debug and safety checks
- * [#2028] Notify the client with libnotify when the zrtp negotiation
- failed.
- * [#811] Harmless no to throw an exception, an makes the application
- less
- * [#2028] A minidialog is showed to the user under sflphone-client-
- gnome
- * Removed useless file.
- * Ignoring Makefile in src/widget
- * [#2027] Fix segmentation fault when showMessage callback is called
- after
- * [#2026] keyExchange was set to ZRTP instead of "1"
- * [#2024] Fix the wrong summary at the end of the assistant.
- * [#1883] Fix mnagerimpl conference map insertion
- * [#1883] Add Mutexes in MainBuffer
- * [#811] Gtk ui was not presenting the right information about zrtp
- for
- * [#2023] security icons were not installed in sflphone-client-gnome.
- * [#2021] Fix a mistake in the readme from sflphone-common that gives
- wrong
- * [#811] The current SRTP mode was not properly displayed for the
- IP2IP
- * [#1743] Re-implementation of the "automatically remove error dialogs
- [...]"
- * [#2017] [#2019] Fix the inability to dial a number and place a
- registered
- * [#811] Final re-integration of ZRTP support in the main branch from
- 0.9.6
- * [#1883] Fix map insertion methods
- * [#811] Combo box now is now set to the active key exchange method
- * [#811] ZRTP options now configurable back again from the Gtk UI.
- IP2IP
- * Updated hostname for git clone
- * [#1883] Add minimal functionalities to create a conference
- * [#811] re-integration of all the methods and signals on dbus.
- ManagerImpl
- * [#811] Got out of a precarious position were nothing would compile.
- * [#1976] Build documentation squeleton with docbook
- * [#1883] Add sflphone-client "addParticipant" button for conference
- * [#1994] Better organize the source directory structure. New
- subdirectories
- * [#1883] Add a simple Conference class
- * [#1882] Use static audio buffer in Pulse and ALSA layer (instead of
- malloc)
- * [#811] First commit toward re-integration and refactoring of ZRTP
- * [#1882] Flush RTP ring buffer before entering mainloop
- * [#1882] Fixed MainBuffer::UnBinCallID() in case there is no
- ringbuffer
- * [#1882] Test (and fixe) high level conference and mixing
- functionalities
- * [#1772] Apply patch to compile on fedora (sent by Marcin
- Zajączkowski <mszpak@wp.pl>)
- * [#1882] Update Bind, unBind call_id in MainBuffer
- * [#1959] This adds the ability to store password as an MD5 Hash in
- the
- * [#1538] Fixes rules compilation
- * [#1930][#1931] Fixed a mistake (again) related to index and
- credential count
- * [#1753] Remove ILBC from pjproject - Hacks in pjsip
- * [#1930][#1931] Credential was not selected properly using realm
- * [#1882] Finilize multiple reading pointer in RingBuffer
- * [#1538] Remove configure from autogen.sh to respect debian upstream
- authors policy
- * [#1773] Remove generated files from repo
- * [#1791] Use XDG_CACHE_HOME to save pid file
- * [#1791] Fixes path to save history
- * [#1791] Fix debian installation scripts
- * [#1930][#1931] Settings are now taken into account in the server.
- * [#1882] Add ringbuffer default ring buffer pointer in methods
- involving mStart
- * [#1882] Add default ringbuffer pointer
- * [#1882] Add RingBuffer multiple read pointer basic functionnalities
- * [#1882] Fix MainBuffer flushData unit test
- * [#1930][#1931] Ability to save and retreive the configuration from
- * [#1882] Added Multiple CallID mapping to MainBuffer
- * [#1791] Not much
- * [#1791] If XDG env variables are not null but empty, use default
- ones
- * [#1791] Make XDG_CONFIG_HOME writable
- * [#1930][#1931] Partial commit. Not working yet. Cannot delete
- account
- * [#1881] Fixed alsa capture latency problem
- * [#1881] Fixed Alsa capture temporarily
- * [#1930] [#1931] Partial unbroken commit providing the ability to
- * [#1881] MainBuffer implemented in AudioLayer/AudioRTP
- * [#1881] Add discard and flush unit-tests
- * [#1881] Add discard and flush functionnalites to MainRingBuffer
- * [#1881] Add availForGet in MainBuffer
- * [#1881] Add availForPut function to MainBuffer
- * [#1880] Remove AudioRTP* pointer from SipVoIP (reapered while
- merging master)
- * [#1881] Add a map between call id and coresponding ring buffer
- * [#1855] Refresh pot file and upload on Launchpad
- * [#1881] MainBuffe now robust to false ids on getData and putData
- * [#1881] Fix big big big memory leak
- * [#1881] Add getData and putData to mainBuffer
- * [#1881] Unit-test basic ring buffer functionnaities
- * [#1881] Add class MainBuffer and basic buffer creation unit-tests
- * [#1880] Fix call transfer (step2) issues
- * [#1880] Moved AudioRtp* pointer from SIPVoIPLink to SIPCall class
- * [#1791] Add postinst script to keep user data when migrating
- config/history file
- * [#1797] Make pjsip compile
- * [#1777] Code indentation
- * [#1791] Use XDG_DATA_HOME and XDG_CONFIG_HOME for sflphonedrc and
- history + unit tests
- * [#1746] Useless space does not appear anymore when volume sliders
- and
- * [#1643] GtkCheckMenuItem is used instead of icons for elements in
- the
- * [#1110] [#1668] STUN parameters are now located in the preferences,
- under
-
- -- Julien Bonjean <julien.bonjean@savoirfairelinux.com> Fri, 06 Nov 2009 11:20:01 -0500
-
-sflphone-client-gnome (0.9.6-SYSTEM) SYSTEM; urgency=low
-
- ** 0.9.6 **
-
- * Documentation on echo test
- * [redmine_down] codec names not displayed in total
- * [redmine_down] crash when hanging up a dialing call because tries to
- add it to history whereas no starttime
- * [#1927] alternate every time screen changed to call history
- * [#1886] clean code
- * [#1886] debug messages when loading history removed
- * [redmine_down] sflphone-kde icons
- * [#1855] Update language files
- * [#1502] Update version number
- * [redmine_down] setHistory at close
- * [#redmine_down] Handle PJ_DECLINE_SC as failure
- * [#1923] Fix segmentation fault when adding a new account
- * [#1923] Check on iterator before setting the config
- * [#1904] Added mnemonic to tabs in sflphone-client-gnome.
- * [#1905] The daemon was not sending the currentSelectedCodec signal
- on dbus when answering a call.
- * [#1922] Default values set to all account details
- * [#1886] Spinbox reg expire enables apply, and address book is not
- visible when disabled
- * [#1905] Bug fix for segmentation fault caused by an empty string,
- * [#1910] Warnings in test directory
- * [#1919] Error fixed
- * [#1855] Update russian translation - Hussein Abdallah
- * [#1910] Remove files
- * [#1919] fixed
- * [#1777] Code indentation
- * [#1918] fixed
- * [#1917] fixed
- * [#1910] Remove warnings compilation in src
- * [#1886] removed AccountListModel in configskeleton
- * [#1914]
- * [#1911] check previous and new port
- * [#1910] Remove compilation warnings in src/dbus and src/history
- * [#1910] Remove compilation warnings in src/audio
- * [1855] Update german translation - Sven Werlen
- * [#1909] removed
- * [#1906] Done
- * [#1904] The registration expire value is now configurable from the
- * Cleaned up debug messages.
- * [#1886] separated initCallItem in two functions
- * [#1886] reversed error in commit
- * [#1886] clean debug
- * [#1886] changed Name of classes and files
- * [#1886] clean
- * [#1870] In call_state_cb (dbus.c:126), _time_stop was overridden by
- the actual time.
- * [#1884] Added some new gpg flags to prevent tty warnings
- * [#1886] Clean audio config dialog
- * [#1886] No more compile warnings. + 1 comm
- * [#1872] Check if the user input is smaller than PJ_MAX_HOSTNAME.
- * [#1886]
- * [#1785] Fixed build when no new commit
- * [#1852] If chosen by the user, the hostname can now be solved and
- used
- * [#1871] * and # inverted back
- * [#1869] Conditional compilation that checks if
- * [#1309] removed test in main
- * [#1425] Put actions in SFLPhone window class instead of ui view,
- made a separate toolbar for screens.
-
- -- SFLphone Automatic Build System <team@sflphone.org> Mon, 27 Jul 2009 09:53:19 -0400
-
-sflphone-client-gnome (0.9.6~rc2-SYSTEM) SYSTEM; urgency=low
-
- ** 0.9.6~rc2 **
-
- * [#1755] Remove generated file
- * [#1753] restore ilbc ...
- * [#1866] Methods getSipPort and setSipPort now have an effect on the
- * [#1753] make pjsip compile without ilbc. Use ./autogen.sh --disable-
- ilbc-codec
- * [#1855] Fix error in russian translation
- * [#1805] Remove the old flawed signal mechanism which was failing in
- * [#1855] Refresh translation
- * Spanish translation finished + po README files updated + echo's in
- copy-in-clients
- * [#1850] Yun made the chinese HK-CN translation
- * [#1848] Fix transfer interface bug
- * [#1862] At install, kde client installs only french translation file
- * [#1841] A new fallback mechanism was added to the internal resolver
- in PJSIP.
- * Started AccountList model/view
- * [#1855] Remove po subdir in Makefile.am
- * [#1855] Fix typo error in sflphone-client-gnome
- * [#1855] Do not generate Makefile in sflphone-common/po
- * [#1855] Copy translation files into both clients dirs
- * [#1855] Remove po dir from sflphone-common
- * Comments added
- * [#1860] mailbox->voicemail...
- * make scripts executable
- * [#1855] French translation
- * [#1855] Chinese zh_HK partially filled...
- * [#1859] An unnamed pipe monitored by poll() was added. When we want
- to
- * [#1855] Sven completed the first part of the german translation
- * [#1855] Cantonese manually filled for already translated, almost
- equal strings
- * [#1855] Merge russian translation
- * [#1855] Spanish manually filled for already translated, almost equal
- strings
- * [#1855] Update german translation in ./lang/de
- * [#1858] This problem was fixed by removing a useless line in
- * [#1855] merged existing translations in lang/ sflphone.po's
- * [#1842] [#1843] An attempt at improving the expected behaviour that
- can't
- * [#1855] added po folder in gnome client and scripts for copying from
- common lang folder to clients
- * [#1853] Edit before call does nothing on call history
- * Put most language entries possible in common. From 300 to 250
- entries. Stays underscores problem. Scripts for copy in clients.
- * commit to merge master
- * [#1825] Changed "Bad authentification" to "Authentication Failed".
- * common po files
- * [#1753] Remove ILBC from pjproject
-
- -- SFLphone Automatic Build System <team@sflphone.org> Fri, 17 Jul 2009 19:12:58 -0400
-
-sflphone-client-gnome (0.9.6~rc1-SYSTEM) SYSTEM; urgency=low
-
- ** 0.9.6~rc1 **
-
- * Update some version number
- * [#1792] Creates .sflphone directory with permission 600. Also,
- "chmod 600" after
- * [#1810] GUI is now notified that the call failed. Also, a segfault
- was
- * [#1816] Address book search disabled when disabled address book and
- enabled it back plus button stays triggered
- * codeclistmodel + asynchronous loading of address book +
- enable/disable address book
- * [#1810] Now checking SDP answer after 200 OK. Still need to
- implement full
- * [#1794] Can't use the interface during a call
- * Updated translation files
- * Russian translation integrated
- * Codec list model/view started.
- * [#1807] Add configure.ac in pjproject-1.0.3
- * [#1787] closeRtpSession added in some places where it should have
- been
- * Use Item class for contacts and accounts
- * Comments + clean code
- * [#1794] Improved debug messages
- * [#1805] Replaced the old and unreliable mecanism that was was
- waiting for
- * [#1794] Can't use the interface during a call
- * [#1787] For those cases where no registered SIP account is
- configured
- * [#1797] Make pjsip compile
- * [#1787] Minor changes. Removed useless commented line. Changed order
- of
- * [#1777] Code indentation
- * [#1797] Update package generation with new pjsip version
- * [#1798] Does not hang up when the call is building up
- * [#1797] Update .gitignore with new pjsip version
- * [#1797] Remove generated files from repo
- * [#1797] Main build system now uses pjproject-1.0.3
- * [#1797] Add pjproject-1.0.3
- * [#1797] Remove pjproject-1.0.2
- * [#1796] Computing time optimization (samplerate conversion)
- * [#1787] _audiortp->start() moved away from offhold(),
- SIPCallAnswered()
- * [#1312] Added new states for calls initialized by other clients
- * [#1795] Crashes when adding a new account, checking it and applying
- * [#1782] Missing icons
- * [#1793] KDE client compilation problem
- * Fake ringtone files can no longer be set.
- * indentation
- * [#1312] Able to fetch to differentiate incoming/ringing call state
- * [#1784] Use DESTDIR variable in po Makefile - fix language file
- installation
- * [#1785] Fixed typo
- * [#1785] Fixed changelog update
- * [#1759] ./autogen.sh --prefix=/usr --with-debug to use optimization
- level 0
- * [#1773] Changed snapshot naming convention
- * [#1773] Removed gpg agent use, added repository cache cleaning
- * [#1759] Use optimization level 0 for repository, 2 for packages
- * [#1777] Code indentation/formatting
- * Translated new features in french
- * [#1785] Added missing changelog entry
- * [#1781] Window title is SFLPhone
- * [#1777] Add code indentation/formatting in the buil system
- * [#1774] Can't set voicemail number in KDE account creation wizard
- * [#1775] Can't modify account information for account created with
- the wizard
- * [#1771] Add a "Default" button in context menu to disable chosen
- prior account
- * [#1705]
- * [#1224] Remove generated file from the repo
- * [#1224] Remove generated file from the repo
- * [#1762] distclean target should remove kconfig generated files
- (settings.h, settings.cpp). Rename them?
- * [#1761] clear history button should really clear history
- * Dialpad works.
- * Implemented Dialpad widget instead of building it in main view.
- * Removed last occurence of the old config dialog, that made the build
- crash.
- * [#1755] Do not consider G722 as a dynamic payload elsewhere than in
- RTP layer
- * [#1753] Remove ilbc Makefile generation
- * [#1756] Implement a kde configuration dialog with kconfig xt and
- kconfigdialog class
- * [#1755] fix audiocodec folder parsing problem
- * [#1450] Reinit timestamp comparison in RTP, create session in
- newOutgoingCall
- * [#1753] Remove milenage third party code from pjsip
- * New Config Dialog integrated in GUI.(without codecs)
- * [#1753] Remove ILBC codec
- * kconfig started, tr2i18n -> i18n, icons folder, accountList changed
- * [#1705] Fixed Audio RTP thread creation/start
- * [#1714] Fix codec negociation result handling
- * [#1678] Fix audiortp payload setting
- * [#1678] Put bac putData method in rtp
- * [#1669] gtk_file_chooser_get_filename() support UTF-8 by default
- * [#1735] Add conditions to sdp update call if call declined
- * [#1737] substr of recordings destination folder to remove "file://"
- should be done in client rather than in daemon
- * [#1731] Enlarge audio stream buffer size
- * [#1714] Missing true
- * [#1317] Fixed Mandriva timeout
- * [#1317] Changed tag convention
- * [#1317] Cleaned git-dch
-
- -- SFLphone Automatic Build System <team@sflphone.org> Fri, 10 Jul 2009 15:50:26 -0400
-
-sflphone-client-gnome (0.9.6~beta-SYSTEM) SYSTEM; urgency=low
-
- ** 0.9.6~beta **
-
- * spec files for mandriva and opensuse updated with buildrequires
- libqt4-dev >=4.3
- * [#1700] Cannot build on ubuntu 8.10 and a few other distribs
- * [#1502] Update version number where applicable
- * [#1642] Update client icons
- * [#1450] Clean up useless debug and comments in sipvoiplink and
- audiortp
- * [#1450] Remove Semaphore object in AudioRtp thread deletion
- * [#1450] Audio RTP init now synchronized with Sip/SDP
- * [#1693] kde client crashes when changing codecs order/activation
- * [#1450] Deep refactoring of audiortp
- * [#1450] setRtpSessionRemoteIp
- * [#1689] getCallList at start
- * [#1224] Change path in package files
- * [#1450] Audio RTP initialized only once, payload and remote ip set
- at runtime
- * [#1450] Add setRtpSessionMedia and setRtpSessionRemoteIp address
- * [#1642] Make GNOME GUI fresher and younger ;)
- * [#1686] Status bar displaying used account
- * added sflphone-kde icon so that it compiles
- * [#1659] Ending a call causes the daemon to crash
- * corrected introspection XMLs, po files...
- * [#1211] g722 media descriptor in codecDescriptor
- * [#1310] Install sflphoned in $(prefix)/lib/sflphone
- * [#1502] Do not install test binaries and dbus utilitaries
- * [#1224] hack for pjsip build system!
- * [#1224] Remove pjsip binaries from repo
- * [#1224] Upgrade to pjsip 1.0.2
- * [#1658] About SFLphone (bugs)
- * [#1658] About SFLphone
- * [#1660] Displaying all dialed numbers in a call
- * Tested status bar.
- * [#790] Optimize pulse audio streams parameters
- * [#1678] Some usefull debug messages for mutex/semaphore deadlock
- problem
- * [#1669] Add/remove some usefull/unusefull debug
- * [#1665] Fix latency related to pulse audio stream openning/closing
- * [#1457] Make the menus and panels accessible in french
- * [#1457] Improve broken keyboard accessibility in menus and conf
- panels
- * [#961] Instanciate only once the searchbar icons
- * [#961] Restore transfer fonction
- * [#961] Filter on the history type OK
- * [#961] Fix compilation problems on hardy/intrepid
- * [#1157] Commit missing files
- * [#790] Reduce number of start/stop streams call on pulse audio
- * [#1639] kde client crashes when no account registered
- * [#1620] Fix the searchbar
- * [#1620] Get back caltree as it was during gtkcritical area
- * [#1620] Add history filter reinit function
- * [#1335] Add a missing label in address book preferences
- * [#1561] Update russian translation - Hussein Abdallah
- * [#1605] Fix edit menu french translation
- * [#961] Enable to search in the history according to the call type
- * [#1449] Searchbar does not work anymore
- * [#961] Add popup menu on the entry primary icon for history
- * [#1317] Fixed KDE client package dependency
- * [#936] speex 32 khz integration completed
- * [#936] Use 320 frame size
- * [#936] Test using a frame size at 320 smpls
- * [#1214] Enable / Disable history
- * [#1607] Fix compilation problem for ubuntu 8.10 (libsexy)
- * [#1313] Implement processDataEncode processDataDecode in audiortp
- * [#1613] codec list order can't be set
- * Better handling of localisation + added languages + corrected
- warnings + begginning of new config dialog with kconfig + 14px
- account leds
- * [#1214] Save and load history according to the limit timestamp +
- unit tests
- * [1609] Fix call number copy/paste feature
- * [1607] Restore clear action icon in searchbar
- * [#936] Try to decode using 1280 samples
- * [#936] Add some debug
- * [#936] Add .cpp file
- * [#936] Oops Forgot speex 32 khz
- * [#1214] Add configuration panel for history + D-Bus calls
- * [#1313] Test rtp thread function, frame size, nbbytes, resampling
- * [#790] Flush audio data before closing audio streams
- * [#1214] History displays local time
- * [#1214] Skip empty field on display
- * [#1214] Associate an account to an history entry
- * [#1342] Get addressbook options sensitive/non-sensitive
- * [#1211] Clean up and comments
- * [#1211] Get back to 20 ms framesize
- * [#1211] Use sendImmediate instead of putData in RTP
- * [#1211] Fix nb byte available in RTP
- * [#1211] Clear condition on maxNbSamples in RTP
- * [#1211] Fix max byte available in RTP session
- * [#1211] G722: Use 160 samples per frame instead of 320
- * [#1211] Test using a dynamic payload
- * [#1211] Test using a dynamic payload type
- * [#1211] Rename size variable (nb_samples, nb_bytes)
- * [#1211] Test g722 ip-to-ip sending twice the data lenth
- * [#1211] Test g722 ip-to-ip
- * [#1214] Do not select an history item by default at startup
- * [#1214] Remove some compilation warnings
- * [#1214] Handle empty field - remove g_print
- * [#1214] Add each history item only once
- * [#1214] Handle call timestamps properlier
- * [#1214] Do not need timestamp files anymore
- * [#1214] Use the saved date for history entry
- * Clean up
- * [#1214] Client doesn't crash if the D-Bus call fails
- * [#1214] Client is able to save its history - still some glitches
- * [#1211] Forgot 16000 for g722
- * [#1211] G722 initialization
- * [#1214] Save name/number, successfully load the history if no fields
- are empty
- * [#1499] Fixed destination directory bug
- * [#1214] Restore all the functionalities; peer name/number way more
- easy to handle !!
- * [#1214] Add callable_object instead of call_t, refactoring
- * [#1211] Test with polycom soundstation 16000
- * [#1211] Remove C like inline function in g722 codec
- * [#1342] Finalize gnome client preference window formating
- * [#1214] Retrieve the history when the gnome client startsup
- * [#1306] Implement localization for KDE client
- * [#1593] enable accounts apply button when account checked/unchecked
- * [#1214] Implement the dbus calls on server side
- * [#1214] Add serialized/unserialized functions to pass data on DBUS
- * [#1342] Formating gnome client configuration windows
- * [#1214] Save sucessfully a map of history items
- * [#1499] Removed multiple jobs compilation for KDE client (2)
- * [#1214] Load history from file into memory, add unit tests
- * [#1534] Throws a length_error exception in case URL exceeds
- std::string max_size
- * [#1499] Removed multiple jobs compilation for KDE client
- * [#1565] make account leds smaller
- * [1430] Fix dbus debug
- * [#1562] crashes when trying to change item of a call of state "OVER"
- * [#1116] Fix compilation bug
- * [#1317] Added mandriva and opensuse-11 64 bits
- * [#1108] Add messges in main window concerning transfer success
- failure
- * [#1116] Fix compilation problems
- * [#1211] g722 Makefile
- * [#1108] Client side transferFailed/trasferSucceded signals handling
- * [#1211] G722 mostly completed,
- * [#1555] make bigger toolbar (24x24)
- * [#1551] remove default mailbox number in wizard and disable mailbox
- button when first account doesn't have mailbox number
- * [#1342] Re-add sflphone manpages
- * [#1116] Fix compilation on non-jaunty distros
- * [#1317] Fixed opensuse startup sleep
- * [#1108] Add a signal in the client to notify successful or failed
- transfer
- * [#1108] Dbus signals concerning call transfer success/failure
- * [#1317] Added opensuse to automatic build system
- * [#1223] Fix manpages bug
- * [#1060] german translation glitch
- * Clean up some gnome client warnings
- * [#1547] replace ugly account leds by beautiful icons
- * [#1548] add close button that hides windowand just hide on clicking
- the cross
- * [#1549] put introspec XMLs in the client's source
- * [#1312] Implement getCallList D-BUS method
- * [#1116] Clear text in history and contacts
- * [#1499] KDE integration
- * [#1469] Modify header linkers in dbus-c++'s Makefile.am's
- * [#1469] Remove examples folder from dbus-c++
- * [#1214] History integration in build system; unit test squeleton
- * [#1317] Cleaning
- * [#1469] Remove configure stuff in dbus-c++
- * [#1469] Add unofficial mainline dbus-c++
- * [#1469] Remove dbus-c++ from freedesktop
- * [#1430] Bring account changed signal/callback back to normal
- * [#1060] Update german translation - Sven Werlen
- * [#1430] Add marshaller one string define
- * [#1430] Send account change signal broadcast using account id
- * [#1430] Remove condition on setRegistrationState, cause stun to
- crash
- * [#1317] Centralized version handling
- * [#1317] Fixed version number on sfl-git-dch
- * [#1317] Refactoring for new distributions
- * [#1215] Fix account order at startup if latency
- * [#1088] Restore sip dns srv
- * [#1214] Add squeleton for history manager
- * [#1430] Add accout id to accout changed method
- * [#1430] No connectionStatusNotification (account changed) if no
- changes
- * [#1538] Add COPYING file
- * [#1430] Add audio rtp thread tests
- * [#1317] Changed version detection
- * [#1538] Document license in libs/stund
- * [#1317] Added version files
- * [#1538] Apply François patches - debian packages
- * [#1317] Updated spec files
- * add files
- * [#1538] Apply François patches - debian packages
- * [#1535] Change program file structure (directory src...)
- * [#1317] Updated build system scripts
- * [#1317] Cleaning
- * [#1317] Copied introspect files to gnome client
- * [#1317] Added opensuse to build-system : first-shot
- * [#1317] Remove spec files from configure
- * [#1317] Added missing prefix
- * removed debug for daemon account fix
- * [#1430] Add a connection reference which most likely belong to
- libdbus
- * [#1430] Use shared connection instead of private
- * make daemon find the account, added userMatch
- * Clean code, add comments...
- * [#1317] Fixed packaging rules
- * [#1317] Updated autogen
- * Updated autogen.sh for pjsip
- * [#1526] Set accounts order
- * [#1317] Fixed pjsip lib dirs
- * [#1317] Updated debian packaging for new pjsip configuration script
- * [#1317] Switch to autogenerated guess and sub files
- * [#1317] Updated pjsip inclusion in build system
- * [#1317] Replaced pjsip guess and sub files
- * [#1317] Fixed compilation issues on opensuse 11
- * [#1505] account list seem to crash the application when clicking
- Apply very fast...
- * [#1456] Add a flag to be replaced in the control files
- * [#1456] Added version dependancy handling
- * put account alias in AccountWidgetItem rather than in the item with
- " " before.
- * [#1034] The KDE client should start sflphoned if it is not started
- * [#1500] Handle options for notifications and display on incoming
- call.
- * [#1443] Client should not crash when receive an unexpected
- stateChanged signal
- * [#1403] Do not stop the notification anymore
- * [#1456] Added version dependancy handling
- * [#1426] Daemon crashes when get alsa plugin
- * [#1422] Improved error messages
- * commit for merge
- * [#1424] Change logo in tray icon and put a different one when
- incoming call
- * [#1425] first part done, window title...
- * [#1413] add manpages creating and installing in build system
- * [#1417] The client should start the account creation wizard if
- started for the first time (if config file doesn't exist)
- * [#1421] Make volume bars horizontal when dialpad is hidden.
- * Changed main window title and fixed a mistake in sflphone_const.h
- * [#1412] make debian package building work
- * changelog changed.
- * Changed addAccount method in gnome client.
- * Debian and man folders added.
- * [#1388] Change project name from sflphone_kde to sflphone-client-kde
- * Better handle of kabc check.
- * [#1351] Automatic generation of dbus interfaces in makefile
- generated by cmake
- * [#1307] Implement "edit before call" in history and address book.
- * [#1344] change action_call label in call history from "call" to
- "call back".
- * [#1308] Implement Hook feature in kde client
- * Improved build system.
- * #1219 : Add address book configuration page
- * Better handling of registration to the daemon.
- * #1039 : Add tray icon in kde.
- * Issue no 1216 : Double click on item in history or address book
- causes call.
- * display peer name in call list and call history when called from
- address book.
- * Address book functionnal with photo displayed.
- * Help menu kde available but actions disappeared. All fonctions in
- view.
- * Address book functionnal but ugly and making its own sort in the
- complete address book.
- * Account choice on right click, clean out includes, page address
- book, fixed bugs...
- * Wizard, double click, context menu...
- * Removed sflphone_kde.kdevelop.filelist
- * Added account creation wizard and translated interface in english.
- * Transfer functionnal but ugly.
- * transfer not functionnal
- * Bug fixed : unholding (UNHOLD_CURRENT, UNHOLD_RECORD)
- * Commit functional for push. With install.sh
- * Before merge.
- * Problem with enable accounts. Account display increased.
- * Functional with codec order working , playDTMF.
- * Commit functional.
- * sflphone_kde/build added in .gitignore.
- * complete commit for checkout previous.
- * Commit before checkout previous version to check the display
- bug(little font everywhere...)
- * Functionnal client. Rest : history icons, config icons and
- functionalities
- * commit before merge asavard for isRecording.
- * Call and Automate fusion done and seems to work.
- * Commiting before putting Automate class in Call class.
- * Functionnal main window without recording, history, voicemail, kio
- widgets.
- * client kde avec kdevelop.
- * Config Dialog almost finished.
- * Base of QT client
-
- -- SFLphone Automatic Build System <team@sflphone.org> Tue, 23 Jun 2009 11:13:42 -0400
-
-sflphone-client-gnome (0.9.5-SYSTEM) SYSTEM; urgency=low
-
- ** 0.9.5 release **
-
- * [#1060] FIx bug in chinese translation
- * [#1313] git add rtpTest.cpp rtpTest.h
- * [#1313] Add init/close rtp tests
- * [#1313] Basic instanciation of the rtp layer
- * [#1449] Gtk-Critical concerning history filters and new calls
- * [#1400] Make the match with the hostname instead of username
- * [#1324] Change status bar label for "Using %s (%s)"
- * [#1403] Icon size: 60x60 px
- * [#1403] Do not remove notification, improve icon quality
- * [#1403] Add smaller icon for gnome notifications
- * [#1403] Prevent crash when hangup && no notification
- * [#1403] Remove all actions on notifications; code refactoring
- * [#1451] Use stun.sflphone.org as default STUN server
- * [#1060] New po files - need to be translated
- * [#1060] Update french translation - Rebuild template file
- * [#1456] Add a flag to be replaced in the control files
- * [#1454] Make cppunit optional; remove from build deps in control
- files
- * [#1401] Add libexpat1-dev dependency in control files
- * [#1448] Take off these ugly debug messages
- * [#1448] fixed getTelephoneTone and getTelephoneFile() called
- repeatedly
- * [#1406] add liblog4c-dev in build-depends
- * [#1409] Restore .desktop icon
-
- -- SFLphone Automatic Build System <team@sflphone.org> Mon, 25 May 2009 11:34:48 -0400
-
-sflphone-client-gnome (0.9.5-SYSTEM~rc2) SYSTEM; urgency=low
-
- ** 0.9.5 rc2 **
-
- * [#1422] Improved error message
- * [#1402] Fix pjsip build
- * [#1404] Clear GTK-Critical Bug at client startup
- * [#1422] Added automatic VM shutdown when building on more than one
- VM
- * [#1422] Fixed some issues with new changelog generation script
- * [#1422] Moved distribution update to specific file
- * [#1422] Dropped git-dch, replace by home made implementation
- * [#1402] Fix pjsip build
- * [#1404] Clear GTK-Critical Bug at client startup
- * Changes for name based dbus connection
- * Clean changelogs
- * [#1343] Gnome: Implement a callback system to handle focus on
- different widgets
- * Debus Session
- * Refactoring Python code, PEP8
- * [#1430] Get back dbus_g_proxy_new_for_name
- * [#1430] Get back DBUS_BUS_SESSION type
- * [#1430] Dbus fixed owner message binding
- * Second test with DBUS owner
- * [#1404] Gnome -> Preferences -> Hooks
- * [#1404] Gnome -> Preferences -> Recordings
- * [#1404] Call History
- * [#1404] Gnome -> Preferences -> Address Book
- * [#1404] IF the first notification option disable the second
- notification
- * Dbus with fixed owner does not automatically start the deamon
- * Add codec debug tests in pysflphone
- * [#1407] Some print info
- * [#1407] Add a scenario to pick_up action
- * Test client dbus connection to a fixed owner
- * Add python dbus test suite
- * [#1161] Modified version handling in build system
- * [#1314] Test pulse audio and audio streams connect and disconnect
- * [#1402] Add info message after configure
- * [#1402] Build the daemon with the local pjsip library (vs the
- installed one)
- * [#1009] Fix Codec Sampling Rate set to zeros
- * [#1314] Add mutex to pulse layer audio streams
- * [#1314] Refactoring pulseaudio stream to test connect disconnect
- * [#1314] Refactoring of pulselayer to test conect/disconnect
- * Add debug messages in debus calls concerning account
- * [#1314] Add some return values to audio init functions
- * [#1406] add liblog4c-dev in build-depends
- * [#1409] Restore .desktop icon
- * Bug #1405: Fix strings as requested.
- * Bug #1404: Fix strings in preferences panel.
-
- -- SFLphone Automatic Build System <team@sflphone.org> Tue, 19 May 2009 12:08:18 -0400
-
-sflphone-client-gnome (0.9.5-0ubuntu1~rc1) SYSTEM; urgency=low
-
- [ SFLphone Project ]
- * [#1262] Updated changelogs for version 0.9.5-0ubuntu1 Snapshot 2009-
- 05-05
-
- [ Emmanuel Milou ]
- * Add some python CLI client code; not really functional
- * [#1108] Fix peerHungup method for IP to IP call
-
- [ Alexandre Savard ]
- * [#1108] Correct setting of SIP contact for direct IP call
- * [#1108] SIP user agent handles incoming REFER
-
- [ Emmanuel Milou ]
- * Remove website from repository
- * Update translation
-
- [ Alexandre Savard ]
- * Sflphone icon's tooltip changed for "configured" instead of
- "registered"
-
- [ Emmanuel Milou ]
- * Update translation
-
- [ Sflphone Project ]
-
- -- Sflphone Project <sflphone@mtl.savoirfairelinux.net> Tue, 05 May 2009 19:16:13 -0400
-
-sflphone-client-gnome (0.9.5-0ubuntu1~beta) SYSTEM; urgency=low
-
- [ Julien Bonjean ]
- * Updated Eclipse stuff
- * Improved addressbook config window
- * Added sflphone Eclipse stuff
- * Implemented addressbook list server side
- * Moved dbus stuff in dbus directory
- * Updated addressbook configuration
-
- [ Emmanuel Milou ]
- * Remove unuseful installation scripts. Use apt-get build-dep sflphone
- instead
- * fix bug #1090
-
- [ Alexandre Savard ]
- * defining speex 16khz
-
- [ Emmanuel Milou ]
- * Remove unuseful file from build system
- * Start dns srv resolver
-
- [ Alexandre Savard ]
- * Basic ogg/vorbis initialization
-
- [ Emmanuel Milou ]
- * Handle incoming IP-to-IP invite correctly
-
- [ Alexandre Savard ]
- * speex wideband 16000
-
- [ Emmanuel Milou ]
- * Better handling of incoming IP to IP call
- * DNS SRV resolution functional
- * Implement IAX2 incoming URL
- * Allow user to make IP call without any accounts configured
- * Add a contextual menu to edit a number from the contacts tab
- * Add comments, tooltip and new button to the contextual menu
- * add delete event, migrate to GTK 2.16 for sexy icons
- * Resolve ticket #1118
- * Update suse spec file
- * Add phone number cleanup functions, unit tests and panel
- configuration
- * Add pertinent test that fails
- * fix dependencies for suse package
- * Add contextual edit menu in history - #1120
-
- [ Alexandre Savard ]
- * Temporary comit: make speex wideband (16 khz)
- * Temporary: shared object for speex narrow band
- * Temporary: speex narrowband and wideband coexist
-
- [ Julien Bonjean ]
- * Fixed bug when no book selected
- * Fixed addressbook related compilation warnings
- * Fixed GTK client remaining compilation warnings
- * Fixed segfault when book removed since last sflphone run
- * Fixed bug when book is unreachable (ldap error)
-
- [ Alexandre Savard ]
- * Fix codec list in audio config window
- * Active/inactive speex codec by payload
-
- [ Julien Bonjean ]
- * Updated gitignore
- * Added some comments
-
- [ Emmanuel Milou ]
- * Add callto: handler script for browsers and al.
- * Integrate test compilation in the daemon build-system
-
- [ Julien Bonjean ]
- * Fixed g_object_unref warning for pixbuf
- * Cleaned too verbose output
- * Fixed toolbar update warning
- * Added support for asynchornous books open (first shot)
-
- [ Emmanuel Milou ]
- * Add a DBus call to fetch the call details from a call ID - Ticket
- #928
-
- [ Julien Bonjean ]
- * Improved async open books
- * Fixed bug #1139
-
- [ Emmanuel Milou ]
- * Add a way to save account order
- * commit missing files
-
- [ Julien Bonjean ]
- * Introduced log4c (ticket #1162)
-
- [ Emmanuel Milou ]
- * Load/save account order functionnal - ticket #813
-
- [ Alexandre Savard ]
- * Add CELT codec (#1143)
- * Make celt frame size 256 (*1143)
-
- [ Julien Bonjean ]
- * Switched everything to log4c (ticket #1162)
- * Updated eclipse settings
-
- [ Emmanuel Milou ]
- * Restore adding account - ticket #1172
- * Add liblog4c dependecy - ticket #1179
-
- [ Alexandre Savard ]
- * Double maxAvailByte for frame size in rtp (#1143)
-
- [ Emmanuel Milou ]
- * Add User-Agent SIP header - Ticket #1173
-
- [ Julien Bonjean ]
- * Fixed autoresize issue (#708)
-
- [ Emmanuel Milou ]
- * Remove libcppuint dependency for the debian packages
- * Look for libsexy only if gtk version < 2.16 - Ticket #1116
- * Remove libsexy dependency for jaunty. ticket #1116
-
- [ Julien Bonjean ]
- * Introduced unit tests (#1146)
- * Updated gitignore
- * Fixed Makefile (#1146)
-
- [ Emmanuel Milou ]
- * [TICKET #1112] Add a test on the voice buffer to send through iax
- packets
- * Remove doublon in dependencies
- * Remove warnings from the client test framework
- * Update version number to 0.9.5~beta
- * Update build-package script
- * Add check dependency in build-deps control file field
- * Create debian files for the new sflphone-client-gnome
- * [TICKET #1212] Add Replaces field in control files
- * [TICKET #1212] Fix manpages installation path
- * [TICKET #1212] Add maintainer scripts to create alternatives
- * [#1212] Update the manpages generation - edit preinst maintainer
- script
- * [#1212] Fix reference error in manpage
- * [#1212] Add missing files on the client side
- * [#1212] Fix debian docs files - no TODO file
- * [1212] Fix manpage creation problem
- * [#1220] Generate client-side glue files and marshaller at
- compilation time
- * [#1220] Generate server-side glue files at compilation time
- * [#1212] Change binary name to sflphone-client-gnome
- * [#1212] Update .gitignore to fit the new working tree
- * [#1220] Explicitly generate glue files before building the library
- * [#1220] Compile dbus directory before audio
- * [#1212] Create sflphone-common at the root of the repository
- * [#1212] Re-add pjproject
- * [#1212] Remove Makefile from repo
- * [#1220] Fix Makefile.am
- * [#1212] New working directory functional
- * [#1212] Update .gitignore
- * [#1212] Hack to make pjsip compile..
- * [#1220] Use non-installed binary for dbusxx-xml2cpp
- * [#1212] Add descriptive files, remove unuseful scripts from tools/
-
- [ Alexandre Savard ]
- * Restore speex codecs
- * add frame size for celt (#1143)
- * add framesize to codec, independant from audiolayer (#1143)
- * use codec frame size in rtp (#1143)
- * compute fixed_codec_framesize (#1143)
- * do not resample if not required (#1143)
- * add condition on resampling for decoder (#1143)
- * add a condition on bytesAvail == 0 from mic data
- * no maximum in rtp decode (#1143)
- * compute maximum for decoding (#1143)
-
- [ Emmanuel Milou ]
- * [#1146] Implement unitary tests on the client-side
-
- [ Alexandre Savard ]
- * use float instead of int to compute max nb of sample (#1143)
- * add nbSampleMax for unresampled data (#1143)
- * make thread sleep during 5 ms insead of 20 (#1143)
- * use unix usleep (#1143)
- * 50 usecond thread!!!!! (#1143)
- * try with the smallest compression (#1143)
- * use timer set at framesize (#1143)
-
- [ Emmanuel Milou ]
- * [#1161] Restore changelog version
-
- [ Alexandre Savard ]
- * Remove celt stuff
-
- [ Emmanuel Milou ]
- * [#1161] Update changelog
- * [#1220] Add Conflicts: sflphone in debian control files
- * [#1179] Add liblog4c3 runtime dependency
- * [#1212] FIx typo error in dependency list for itnrepid
- * [#1212] FIx .desktop file to point on the right exec
- * [#1212] Modify changelog replacing tag
-
- [ Sflphone Project ]
- * "[#1262] Updated changelogs for version 0.9.5-0ubuntu1~beta"
-
- [ Emmanuel Milou ]
- * [#1212] restore changelogs
-
- [ Sflphone Project ]
- * [#1262] Updated changelogs for version 0.9.5-0ubuntu1 Snapshot 2009-
- 04-27
-
- [ Emmanuel Milou ]
- * [#1212] restore changelogs
-
- [ Sflphone Project ]
- * [#1262] Updated changelogs for version 0.9.5-0ubuntu1~beta
-
- [ Emmanuel Milou ]
- * [#1212] restore changelogs
-
- [ Sflphone Project ]
-
- -- Sflphone Project <sflphone@mtl.savoirfairelinux.net> Mon, 27 Apr 2009 17:00:03 -0400
-
-sflphone-client-gnome (0.9.4-0ubuntu2) SYSTEM; urgency=low
-
- [ Alexandre Savard ]
- * Restore speex and GSM detection
-
- [ Emmanuel Milou ]
- * Fix bug #1090
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Wed, 8 Apr 2009 11:29:15 -0500
-
-sflphone (0.9.4-0ubuntu1) SYSTEM; urgency=low
-
- [ Emmanuel Milou ]
- * Integrate DBus-c++ and libiax2 in the main build system
- * Clean up in the working repository
- * Reorder hooks configuration panel
- * Protect case when no codecs are active
- * Fix some return values
- * Add unitary tests for the hook manager (premisces)
-
- [Yun Liu]
- * Update chinese translation
-
- [Sven Werlen]
- * Update german translation
-
- [Hussein Abdallah]
- * Update russian translation
-
- [Maxime Chambreuil]
- * Update spanish translation
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Fri, 3 Apr 2009 18:29:15 -0500
-
-
-sflphone (0.9.4-rc1) SYSTEM; urgency=low
-
- [ Emmanuel Milou ]
- * Fix bug while trying to hold/unhold several simultaneous call
- * Improve address book build system
- * Implement SIP url popup on incoming call
- * Improve GTK+ panel configuration
- [ Julien Bonjean ]
- * GTK+ client refactoring
- * GTK+ clean up
- * Address book improvment
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Fri, 27 Mar 2009 18:29:15 -0500
-
-sflphone (0.9.4-0beta1) SYSTEM; urgency=low
-
- [ Alexandre Savard ]
- * Display codec used during conversation on the GUI
- * Enable/disable STUN parameters at runtime
- * Refactor search bar use
- [ Emmanuel Milou ]
- * Build system fixes
- * Implement SIP re-invite
- * Implement IP to IP call
- [ Julien Bonjean ]
- * Integrate GNOME address book based on evolution data server
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Fri, 20 Mar 2009 18:29:15 -0500
-
-
-sflphone (0.9.3-0ubuntu3) SYSTEM; urgency=low
-
- [ Alexandre Savard ]
- * Both playback and record streams in PA_STREAM_CORKED (pulseaudio)
- * Use PLUGHW device for ALSA capture
- * Functional IAX and SIP recording for voicemail
- * Use the less CPU-consuming interpolator algorithm for resampling
- * Display in GTK GUI the codec used in conversation
- * GTK GUI use ASCII instread of utf-8
- * Add record menus in GTK GUI
- * Put on hold when dialing a new number
- * AccountID's are saved in the history
-
- [ Emmanuel Milou ]
- * Integrate DBUS C++, libiax2 in the git repository
- * Update website
- * Use libspeexdsp only if available on the system
- * Updated .gitignore file
-
- [Cyrille Béraud]
- * Account assistant manager improvment
- * Add an email request when creating a new account to receive voicemails
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Sat, 14 Feb 2009 13:29:15 -0500
-
-sflphone (0.9.3-0ubuntu2) SYSTEM; urgency=low
-
- [ Emmanuel Milou ]
- * Add compilation note in README
- * Use default ALSA plugin for capture
- * Fix the ALSA capture problem one more time
- * Clean up debug messages in dbus.c
- * Add libspeexdsp dependency
- * Remove implicit declaration compilation warnings
- * Fix links in the website, add release note
- * Change capture for the website front page
- * Add alsa devel dependency in build-depends control file field
- * Clean up, indentation, try to handle latency problems in iax/pulseaudio
- * Remove pjsip generated files from the repo
- * Use the previous declared curAlias function in accountwindow
- * Fix bug in history call duration when the call fails
- * Remove runtime warning in the GTK+ client
- * Add librsvg2-common dependency to load SVG under KDE
- * Refresh .gitignore
- * Update locales files + french translation
- * Add configuration panel for future noise reduction
- * Add configuration panel for audio record module
- * Daemon less verbose; accounts don't try to access STUn options anymore
- * Fix typo in configwindow
- * Add content in the official website
- * use a GTK_STOCK icon for the record button
- * Complete description text in the assistant manager
- * Add libtool flags in client configure.ac
- * Remove unuseful dependency (snd)
- * Fix SIP transfer problems
- * Remove previous version of PJSIP from the repo
- * Upgrade PJSIP to version 1.0.1
- * Add the new website source in the repository
- * Use libspeexdsp for silence detection only if available
-
- [ Loïc Faure-Lacroix ]
- * Ajout du logo gpl3
- * Ajout des images
- * Ajout de la section screenshot pour le site
- * Ajout du favicon dans le header
- * Modification des cartes
-
- [ Alexandre Savard ]
- * Clean up <speex/libspeexdsp>
- * Small cleanup
- * Save Wave fixed
- * Fix new call button when recording
- * libspeexdsp added
- * Recording: default home folder at startup
- * Minor changes to config window
- * IAX recording fixed
- * Set / get recording path, still need some GTK for client
- * AudioRecord file name format
- * Now recording in HOME folder
-
- [ Cyrille Béraud ]
- * Fix bug in reqaccount.c
-
- [ Maxime Chambreuil ]
- * Update spanish translation
-
- [Yun Liu ]
- * Update chinese translation
-
- [ Hussein Abdallah ]
- * Update russian translation
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Sat, 14 Feb 2009 13:29:15 -0500
-
-sflphone (0.9.3-0ubuntu1) SYSTEM; urgency=low
-
- * Remove debug
- * Join thread before leaving
- * Fix implicit declaration in reqaccount
- * Add REST code to build the request to server
- * Fix GValue initialization warnings
- * Update version number, fix implicit declaration, fix GTK markup
- warnings
- * Apply patch to create custom SIP account from our own server
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Fri, 06 Feb 2009 19:17:32 -0500
-
-sflphone (0.9.2-2ubuntu9) SYSTEM; urgency=low
-
- [ Alexandre Savard ]
- * Speex audio codec preprocessing initialization
- * peer hung up segmentation fault solved
- * Stop recording when transfering
- * Terminate only one call
- * Add isRecording() function
- * Fix call_icon GTK client
- * Fix SIPCallClose() function, recorded file now close properly
- * Function terminateSIPCall added in sipvoiplink and managerimpl
- * Fix thread destructor
- * setRecordingOption function implement in audiorecord
- * Record now implemented in Call class
- * Record interface complete (on hold erase previous recording)
- * Added recButton in client
- * Added: record button related icons
- * Record button added
- * Overload AudioRecord::recData to get mic and speaker data mixed
- * Recording now in audiortp::run() method
- * Audio recording working in AudioRTP: receiveSessionForSpeaker
- * Open/close a wave file when pulse audio stream start/stop
-
- [ Emmanuel Milou ]
- * Fix path for GTK+ icons; clean up
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Thu, 05 Feb 2009 18:27:53 -0500
-
-sflphone (0.9.2-2ubuntu8) SYSTEM; urgency=low
-
- [ Emmanuel Milou ]
- * Update changelogs
- * Fix bug in merge and in Makefile.am
- * Terminate only one call
- * Disable PJsip shutdown when changing STUN parameters
- * Function terminateSIPCall added in sipvoiplink and managerimpl
- * Add a timer to the alsa thread to not jam the CPU load
- * Fix bug in sipvoiplink.cpp
- * Clean shutdown of pulseaudio on quiting
- * Fix DTMF at first start with Pulseaudio
- * Remove zeroconf from the build system
- * Add a library manager + exception handling
- * Clean up in the working directory
- * Better handling of capture XRUNs
- * Restore mic adjust volume on ALSA layer
- * Protect device ALSA operation if not opened
- * Fix the switching layer bug
- * Use dynamic_cast<> to use audiolayer-specific methods
- * Open the audio devices only once at startup
- * Refactoring of the ALSA part
- * Functional plug-in manager
- * Use a C++ thread to handle tones and DTMF in ALSA
- * Restore IAXVoIPLink, restore Mutex
- * Make the plugins registering against the plugin manager
- * Migrate to 1->N relationship between voiplink and accounts
- * API plugin for registration
- * Use C++ thread in SIP, move everything in sipvoiplink
- * Complete singleton pattern for the plugin manager
- * Add -Wno-return-type compilation flag to remove warnings; Update
- version number in configure.ac
- * Add the dynamic loading for the plugin framework; integate unittest
-
- [ Yun Liu ]
- * Update rpm spec file
- * modify build package script and spec file for suse
-
- [ Alexandre Savard ]
- * Add audiorecorder plugin and testaudiorecorder
- * Add audio Recording class, edit global.h
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Wed, 04 Feb 2009 14:00:30 -0500
-
-sflphone (0.9.2-2ubuntu7) SYSTEM; urgency=low
-
- [ Emmanuel Milou ]
- * Update changelog to 0.9.2-6
- * Fix some dbus-glib implementation details on the client side
- * Init history after dbus initialization
- * Add error checking in useragent; Clean sipvoiplink
- * Prevent crash when trying to call an empty number
- * Set the volume of the playback stream to PA_VOLUME_NORM at startup
- * Fix GTK+ generic value double initialization
- * Fix jaunty control file dependency problems
- * Fix jaunty control file dependency problems
-
- [ Yun Liu ]
- * Fix bug ticket # 137
- * Tolerant to gsm library of OpenSuse 11
-
- [ Sven Werlen ]
- * Update german translation
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Fri, 23 Jan 2009 17:48:13 -0500
-
-sflphone (0.9.2-2ubuntu6) SYSTEM; urgency=low
-
- [ Emmanuel Milou ]
- * Migrate STUN configuration to the main config window
- * Update french translation
- * Other tiny memory leaks
- * Fix memory leak in sampleconverter.cpp
- * Generate packages from the release branch
- * update the build package script
- * modify the control files with architecture=any
- * Remove valgring uninitialized value
- * IAX and SIP use the same global variables to set account
- configuration ; fix broken code
-
- [ Maxime Chambreuil ]
- * Update spanish translation
-
- [ Hussein Abdallah ]
- * Update russian translation
-
- [ Yun Liu ]
- * Update translation files
- * Fix the bug when user uncheck the account which fails in the
- previous registration
- * Add stun error status
- * Fix bug ticket #143
- * Script for auto-install dependencies
- * Fix bug ticket #140
- * Fix bug ticket 141
- * Fix the reregister process when user change the details of an
- account
-
- -- Emmanuel Milou <manu@sulfur.inside.savoirfairelinux.net> Fri, 16 Jan 2009 18:19:05 -0500
-
-sflphone (0.9.2-2ubuntu5) SYSTEM; urgency=low
-
- * Fix memory leak in the pulseaudio callback
- * Update debian package generation script
- * Warnings removal in GTK+ client
- * Clean adjust volume method in alsalayer
- * Plug the sflphone playback volume control to the pulseaudio volume
- manager
- * Display the date in history according to the current locale
- * Generate the changelog according to the git commit messages
- * Complete header in chinese translation file
- * Use the right gpg key to sign the packages
- * add debian jaunty jackalope support
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Wed, 14 Jan 2009 21:17:20 -0500
-
-sflphone (0.9.2-2ubuntu4) SYSTEM; urgency=low
-
- [ Emmanuel Milou ]
- * add german translation
-
- [ Yun Liu ]
- * Fix GUI crash in Ubuntu8.10 64bit system
-
- -- Yun Liu <yun.liu@savoirfairelinux.com> Thu, 08 Jan 2009 13:08:51 -0500
-
-sflphone (0.9.2-2ubuntu3) SYSTEM; urgency=low
-
- [ Emmanuel Milou ]
- * The main thread synchronizes the ringtone thread
- * disable custom ringtone for the ALSA layer
- * Fix the Makefile.am in man directory, add a SEE ALSO section
-
- [ Yun Liu ]
- * Fix daemon crash caused by the previous patch ( for bug ticket #129)
-
- -- Yun Liu <yun.liu@savoirfairelinux.com> Tue, 06 Jan 2009 16:18:38 -0500
-
-sflphone (0.9.2-2ubuntu2) SYSTEM; urgency=low
-
- * Fix bug ticket #129
-
- -- Yun Liu <yun.liu@savoirfairelinux.com> Wed, 5 Jan 2009 15:54:53 -0500
-
-sflphone (0.9.2-2ubuntu1) SYSTEM; urgency=low
-
- * Migrate from eXosip library to pjsip
- * Add multiple SIP accounts support
- * Fix ringtones problems
- * Add a pulseaudio support
- * Improve audio quality with ALSA
- * Add chinese translation
- * Improve spanish translation
- * Migrate to a maintained C++ DBus bindings
- * Clean and improve the build system
- * Add build-dependency on Perl because we need pod2man to generate manpages
-
- -- Yun Liu <yun.liu@savoirfairelinux.com> Wed, 26 Nov 2008 09:47:53 -0500
-
-sflphone (0.9.1) unstable; urgency=low
- * Add a search tool in the history
- * Migrate some gtk_entry_new to sexy_icon_entry_new
- * Bug fix (Ticket #78): The voicemail password isn't displayed anymore in
- the history tab
- * Add the SIP registration expire value in the user file.
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Thu, 22 May 2008 11:14:25 -0500
-
-sflphone (0.9.0) unstable; urgency=low
- * Add history features
- * Call date
- * Call duration
- * Mouse events in the history tab
- * Smooth switch from the history tab to the calls tab
- * Remove most of GTK-Critical warnings
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Wed, 13 May 2008 16:58:25 -0500
-
-sflphone (0.9-2008-06-06) unstable; urgency=low
- * Audio bug correction: capture stopped after a few minutes of conversation
- with USB Plantronics sound card
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Tue, 06 May 2008 16:58:25 -0500
-
-sflphone (0.9-2008-05-06) unstable; urgency=low
- * Bug correction: account creation with the assistant
- * GTK+ warnings removal
- * libnotify warnings removal
- * Remove aliasing on the SFLphone logo
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Mon, 05 May 2008 16:58:25 -0500
-
-sflphone (0.9) unstable; urgency=low
- * Clean dependencies ( removal of libboost )
- * Several GTK improvement and updates
- -account window
- -configuration window
- * Migrate from GtkCheckMenuItem to GtkImageMenuItem
- * ALSA standard I/O transfers: MMAP instead of R/W
- * Fix speex audio quality
- * IAX2 protocol
- -Fix hold/unhold situation
- -Add on hold music
- * SIP protocol
- -Ringtone on incoming call
- -Fix transfer situation
- * Add desktop notification ( libnotify )
- * Improve the system tray icon behaviour
- * Improve registration error handling
- * Register/unregister from the account window takes effect without starting back SFLphone
- * Compilation warnings removal
- * Call history
- * Add an account configuration wizard
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Wed, 30 Apr 2008 16:58:25 -0500
-
-sflphone (0.8.2) unstable; urgency=low
- * Internationalization of the GTK GUI
- * English / French
- * STUN support
- * Slight modifications of the graphical interface ( tooltips, dialpad, ...)
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Fri, 21 Mar 2008 11:37:53 -0500
diff --git a/tools/build-system/launchpad/sflphone-gnome-video/debian/compat b/tools/build-system/launchpad/sflphone-gnome-video/debian/compat
deleted file mode 100644
index 7ed6ff82de6bcc2a78243fc9c54d3ef5ac14da69..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/sflphone-gnome-video/debian/compat
+++ /dev/null
@@ -1 +0,0 @@
-5
diff --git a/tools/build-system/launchpad/sflphone-gnome-video/debian/control b/tools/build-system/launchpad/sflphone-gnome-video/debian/control
deleted file mode 100644
index 9699203f535bac9bf53f9945d7e51ae934662a32..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/sflphone-gnome-video/debian/control
+++ /dev/null
@@ -1,21 +0,0 @@
-Source: sflphone-gnome-video
-Maintainer: SavoirFaireLinux Inc <julien.bonjean@savoirfairelinux.com>
-Section: gnome
-Priority: optional
-Build-Depends: debhelper, libgcc1, intltool, autopoint, autoconf, automake, libtool, libgtk-3-dev, libdbus-glib-1-dev, libnotify4-dev | libnotify-dev (>= 0.7), check, gnome-doc-utils, rarian-compat, librsvg2-common, gnome-common, yelp-tools, libclutter-1.0-dev, libclutter-gtk-1.0-dev
-Standards-Version: 3.7.3
-
-Package: sflphone-gnome-video
-Priority: optional
-Architecture: any
-Depends: sflphone-daemon-video (=${source:Version}), ${shlibs:Depends}, ${misc:Depends}
-Provides: sflphone-client-gnome-video
-Replaces: sflphone-client-gnome-video, sflphone
-Conflicts: sflphone-client-gnome, sflphone-gnome, sflphone-data
-Homepage: http://www.sflphone.org
-Description: GNOME client for SFLphone, with video support
- Provide a GNOME client for SFLphone.
- SFLphone is meant to be a robust enterprise-class desktop phone.
- SFLphone is released under the GNU General Public License.
- SFLphone is being developed by the global community, and maintained by
- Savoir-faire Linux, a Montreal, Quebec, Canada-based Linux consulting company.
diff --git a/tools/build-system/launchpad/sflphone-gnome-video/debian/copyright b/tools/build-system/launchpad/sflphone-gnome-video/debian/copyright
deleted file mode 100644
index 7b3bdc5eebd9324f7616980b85dd03f0354d32a5..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/sflphone-gnome-video/debian/copyright
+++ /dev/null
@@ -1,28 +0,0 @@
-This package was debianized by Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> on
-Fri, 3 Apr 2009 09:47:53 -0500.
-
-It was downloaded from the git repository of SFLphone: git://sflphone.org/git/sflphone.git
-
-Upstream Author: Emmanuel Milou <emmanuel.milou@savoirfairelinux.com>
-
-Copyright:
-
-Savoir-faire Linux Inc.
-
-License:
-
-This software is copyright (c) 2004-2016 Savoir-faire Linux Inc.
-
-You are free to distribute this software under the terms of
-the GNU General Public License version 3.
-On Debian systems, the complete text of the GNU General Public
-License can be found in the file `/usr/share/common-licenses/GPL'.
-
-This program is distributed in the hope that it will be useful,
-but WITHOUT ANY WARRANTY; without even the implied warranty of
-MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
-GNU General Public License for more details.
-
-You should have received a copy of the GNU General Public License
-along with this program; if not, write to the Free Software
-Foundation, Inc., 51 Franklyn St, Fifth Floor, Boston, MA 02110-1301, USA.
diff --git a/tools/build-system/launchpad/sflphone-gnome-video/debian/cron.d b/tools/build-system/launchpad/sflphone-gnome-video/debian/cron.d
deleted file mode 100644
index d11e61177739b56bce3aac6de6483b48e797a258..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/sflphone-gnome-video/debian/cron.d
+++ /dev/null
@@ -1,4 +0,0 @@
-#
-# Regular cron jobs for the sflphone package
-#
-0 4 * * * root sflphone_maintenance
diff --git a/tools/build-system/launchpad/sflphone-gnome-video/debian/dirs b/tools/build-system/launchpad/sflphone-gnome-video/debian/dirs
deleted file mode 100644
index e2dc98dcb24907fb5a7ceb0f0651276b702d6030..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/sflphone-gnome-video/debian/dirs
+++ /dev/null
@@ -1,7 +0,0 @@
-usr/bin
-usr/share/applications
-usr/share/pixmaps
-usr/share/sflphone
-usr/share/locale
-usr/share/doc
-usr/share/man
diff --git a/tools/build-system/launchpad/sflphone-gnome-video/debian/docs b/tools/build-system/launchpad/sflphone-gnome-video/debian/docs
deleted file mode 100644
index f757754f072a3dad7d52809b4736c54994481fcb..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/sflphone-gnome-video/debian/docs
+++ /dev/null
@@ -1,4 +0,0 @@
-NEWS
-README
-ChangeLog
-AUTHORS
diff --git a/tools/build-system/launchpad/sflphone-gnome-video/debian/manpages b/tools/build-system/launchpad/sflphone-gnome-video/debian/manpages
deleted file mode 100644
index 91e6d06b26c6f68a8ced9a9b28e117b41526b1b0..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/sflphone-gnome-video/debian/manpages
+++ /dev/null
@@ -1,2 +0,0 @@
-debian/sflphone-gnome-video/usr/share/man/man1/sflphone-client-gnome.1
-debian/sflphone-gnome-video/usr/share/man/man1/sflphone.1
diff --git a/tools/build-system/launchpad/sflphone-gnome-video/debian/postinst b/tools/build-system/launchpad/sflphone-gnome-video/debian/postinst
deleted file mode 100644
index ebee7fa2bb049bf0e6f826e28569d05cb51d451a..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/sflphone-gnome-video/debian/postinst
+++ /dev/null
@@ -1,7 +0,0 @@
-#!/bin/sh -e
-
-update-alternatives --install /usr/bin/sflphone sflphone /usr/bin/sflphone-client-gnome 100 \
- --slave /usr/share/man/man1/sflphone.1.gz sflphone.1.gz \
- /usr/share/man/man1/sflphone-client-gnome.1.gz
-
-exit 0
diff --git a/tools/build-system/launchpad/sflphone-gnome-video/debian/postrm b/tools/build-system/launchpad/sflphone-gnome-video/debian/postrm
deleted file mode 100644
index e6107444fa259e7d87e9d0d0fcfff80ca8ff144d..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/sflphone-gnome-video/debian/postrm
+++ /dev/null
@@ -1,34 +0,0 @@
-#!/bin/sh
-# postrm script for sflphone
-#
-# see: dh_installdeb(1)
-
-set -e
-
-# summary of how this script can be called:
-# * <postrm> `remove'
-# * <postrm> `purge'
-# * <old-postrm> `upgrade' <new-version>
-# * <new-postrm> `failed-upgrade' <old-version>
-# * <new-postrm> `abort-install'
-# * <new-postrm> `abort-install' <old-version>
-# * <new-postrm> `abort-upgrade' <old-version>
-# * <disappearer's-postrm> `disappear' <overwriter>
-# <overwriter-version>
-# for details, see http://www.debian.org/doc/debian-policy/ or
-# the debian-policy package
-
-if [ "$1" = "purge" ]
-then
-
- # remove the user config file
- rm -f $HOME/.sflphone/sflphonedrc
-
-fi
-
-# dh_installdeb will replace this with shell code automatically
-# generated by other debhelper scripts.
-
-#DEBHELPER#
-
-exit 0
diff --git a/tools/build-system/launchpad/sflphone-gnome-video/debian/preinst b/tools/build-system/launchpad/sflphone-gnome-video/debian/preinst
deleted file mode 100644
index d5e20248d97258ec7a5c80e4b5077ad19a77bcde..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/sflphone-gnome-video/debian/preinst
+++ /dev/null
@@ -1,19 +0,0 @@
-#!/bin/sh
-# postrm script for sflphone
-#
-# see: dh_installdeb(1)
-
-set -e
-
-package=sflphone-gnome
-
-case "$1" in
- install|upgrade)
- ## Clean up the previous manpage
- if [ -f /usr/share/man/man1/sflphone-gtk.1 ]; then
- rm /usr/share/man/man1/sflphone-gtk.1
- fi
- ;;
-esac
-
-exit 0
diff --git a/tools/build-system/launchpad/sflphone-gnome-video/debian/prerm b/tools/build-system/launchpad/sflphone-gnome-video/debian/prerm
deleted file mode 100644
index 5e9021706875bb08a56c8c54f35cef96a7ca6055..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/sflphone-gnome-video/debian/prerm
+++ /dev/null
@@ -1,7 +0,0 @@
-#!/bin/sh -e
-
-
-if [ "$1" = "remove" ]; then
- # Remove alternatives symlink set in postinst
- update-alternatives --remove sflphone /usr/bin/sflphone
-fi
diff --git a/tools/build-system/launchpad/sflphone-gnome-video/debian/rules b/tools/build-system/launchpad/sflphone-gnome-video/debian/rules
deleted file mode 100755
index bdb37e4f600badb411a6ae4745f38c25ea415d6e..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/sflphone-gnome-video/debian/rules
+++ /dev/null
@@ -1,117 +0,0 @@
-#!/usr/bin/make -f
-# -*- makefile -*-
-# Sample debian/rules that uses debhelper.
-# This file was originally written by Joey Hess and Craig Small.
-# As a special exception, when this file is copied by dh-make into a
-# dh-make output file, you may use that output file without restriction.
-# This special exception was added by Craig Small in version 0.37 of dh-make.
-
-# Uncomment this to turn on verbose mode.
-#export DH_VERBOSE=1
-export DH_OPTIONS
-
-package=sflphone-gnome-video
-
-CXX = g++-4.0
-CFLAGS = -Wall -g
-DEB_INSTALL_MANPAGES_sflphone_gnome_video = sflphone.1 sflphone-client-gnome.1
-
-configure: configure-stamp
-configure-stamp:
- dh_testdir
- # Add here commands to configure the package.
- NOCONFIGURE=1 ./autogen.sh
- ./configure --prefix=/usr --enable-video
- touch configure-stamp
-
-
-#Architecture
-build: build-arch build-indep
-
-build-arch: build-arch-stamp
-build-arch-stamp: configure-stamp
-
- # Add here commands to compile the arch part of the package.
- $(MAKE)
- touch $@
-
-build-indep: build-indep-stamp
-build-indep-stamp: configure-stamp
-
- # Add here commands to compile the indep part of the package.
- #$(MAKE) doc
- touch $@
-clean:
- dh_testdir
- dh_testroot
- rm -f build-arch-stamp build-indep-stamp configure-stamp
- # Add here commands to clean up after the build process.
- [ ! -f Makefile ] || $(MAKE) distclean
-
-ifneq "$(wildcard /usr/share/misc/config.sub)" ""
- cp -f /usr/share/misc/config.sub config.sub
-endif
-ifneq "$(wildcard /usr/share/misc/config.guess)" ""
- cp -f /usr/share/misc/config.guess config.guess
-endif
- dh_clean
-
-install: install-indep install-arch
-install-indep:
- dh_testdir
- dh_testroot
- dh_clean -k -i
- dh_installdirs -i
- # Add here commands to install the package into debian/sflphone.
-
-install-arch:
- dh_testdir
- dh_testroot
- dh_clean -k -s
- dh_installdirs -s
- # Add here commands to install the arch part of the package into
- # debian/tmp.
- $(MAKE) DESTDIR=$(CURDIR)/debian/$(package) install
- dh_install -s
-# Must not depend on anything. This is to be called by
-# binary-arch/binary-indep
-# in another 'make' thread.
-
-binary-common:
- dh_testdir
- dh_testroot
- dh_installchangelogs ChangeLog
- dh_installdocs
- dh_installexamples
-# dh_installmenu
-# dh_installdebconf
-# dh_installlogrotate
-# dh_installemacsen
-# dh_installpam
-# dh_installmime
-# dh_python
-# dh_installinit
-# dh_installcron
-# dh_installinfo
-# dh_installman
- dh_link
- dh_strip
- dh_compress
- dh_fixperms
-# dh_perl
- dh_makeshlibs
- dh_installdeb
- dh_shlibdeps
- dh_gencontrol
- dh_md5sums
- dh_builddeb
-# Build architecture independant packages using the common target.
-binary-indep: build-indep install-indep
- $(MAKE) -f debian/rules DH_OPTIONS=-i binary-common
-
-# Build architecture dependant packages using the common target.
-binary-arch: build-arch install-arch
- $(MAKE) -f debian/rules DH_OPTIONS=-s binary-common
-
-binary: binary-arch binary-indep
-.PHONY: build clean binary-indep binary-arch binary install install-indep install-arch configure
diff --git a/tools/build-system/launchpad/sflphone-gnome-video/debian/substvars b/tools/build-system/launchpad/sflphone-gnome-video/debian/substvars
deleted file mode 100644
index 566a162f0d3708c2c131a6eff863df6727922259..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/sflphone-gnome-video/debian/substvars
+++ /dev/null
@@ -1 +0,0 @@
-plop=0.9.6
diff --git a/tools/build-system/launchpad/sflphone-gnome/debian/changelog b/tools/build-system/launchpad/sflphone-gnome/debian/changelog
deleted file mode 100644
index c26c39793864226c1696e01b20dec881a45ef400..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/sflphone-gnome/debian/changelog
+++ /dev/null
@@ -1,3138 +0,0 @@
-sflphone-gnome (1.1.0-rc20120607~ppa1~SYSTEM) SYSTEM; urgency=low
-
- ** SNAPSHOT 1.1.0-rc20120607~ppa1~SYSTEM **
-
- * gnome: cleanup logging
- * * #12061: gnome: fix video codec list display
- * * #12050: gnome: fixed regression in active codec handling
- * #11732: Do not align menu with other interface component
- * * #11818: gnome: stop daemon on SIGTERM, SIGINT or SIGHUP
- * #10304: updatePlaybackScale dbus method uses int 32 bit for size and
- position (allow for 24 days long recording playback)
- * #10304: Add time lable for seekslider
- * * #10725: gnome: seekslider: fixed compiler warning due to missing
- header
- * * #11732: gnome:calltree: fix warning when searchbar is not created
- * * #11252: daemon: removed deprecated zrtp code
- * #10725: Consolidate test to determine if there is a recording,
- remove log on NULL pointer
- * #10725: Remove logging from stop callback in seekslider
- * #10725: No need of playback related callbacks in uimanager
- * #10725: Remove logging in seekslider when no selected call
- * #11732: Align Icons with dialpad in gnome client
- * #10304: Make sure that the playback won't be updated after reset
- * #10304: Prevent playback widget reset from stoping dtmf
- * #10304: Move Playback widget at the bottom of the main window
- * #10304: Fix logic to determine outgoing/incoming calls for recording
- icon
- * * #11685: gnome: fixed memory leaks in config menus
- * #10304: Fix reset seekslider that prevent scale to be updated at
- first read
-
- -- Julien Bonjean <julien.bonjean@savoirfairelinux.com> Thu, 07 Jun 2012 16:04:59 -0400
-
-sflphone-gnome (1.0.0-rc20110930~ppa1~SYSTEM) SYSTEM; urgency=low
-
- ** SNAPSHOT 1.0.0-rc20110930~ppa1~SYSTEM **
-
- * update kde .gitignore
- * Fix bug in volume widget
- * More polishing for release
- * Bump version to 1.0.0
- * [#7023] Add the ability to load an abstract contact backend in the
- library to resolve more data, polish code
- * [#7021] More cleanup for release
- * Cleanup
- * [#7021] Refactor KDE client dbus handling, add a missing call in
- daemon and port the DataEngine to the new API
- * Remove some annoying debug
- * merge language scripts
- * remove obsolete 'VERSION' files
- * update install instructions
- * Add missing translations to gnome
- * language update
- * Revert "Don't reference count DBus clients, exit core immediately
- when one of them request it"
- * Don't reference count DBus clients, exit core immediately when one
- of them request it
- * [7021] Add contact abstraction support
- * [#7121] Polishing library (over). Indentation, spacing and naming
- are now consistent
- * codecs: link to libccrtp, don't use logger
- * Fix a daemon bug
- * [#7038] Fix adding contact
- * * #7037 : stop audio stream after all calls have been hanged up
- * [#7025] Add full support for bookmark
- * SFLPhone KDE do not destroy history anymore
- * Fix config skeleton
- * Close the daemon once and for all, no more automatic respawning
- * Fix "unregistered account" bug (I hope so)
- * Close SFLPhone at the right place, it still respawn, I don't know
- why
- * Remove dead code
- * Fix regressions introduced in the last commit
- * Dead code elimination 1/3
- * Fix bug, add "add contact" option, fix warning
- * * #7019: Fix IAX codec negociation
- * Remove or comment unnecessary/unhelpful debug output
- * Fix "same as local" account setting, fix IP2IP LED color
- * Add support for some more advanced config options and add missing
- config dialog icons
- * Fix crash with noise suppressor
- * Alternative can now be selected from the call view context menu
- * Add drag and drop support, initial context menu and fix 3 bugs in
- the account dialog
- * Add basic history drag and drop support
- * Complete contact support is back
- * * #6991 : fix IAX problems
- * Fix IAX accounts being disabled by default
- * Revert "deb: forge -g flags for pjsip"
- * * #5884: Disable debug code in pjsip
- * echo suppressor : more assertions
- * Don't let the daemon think crypto is enabled when it's not
- * Simplify ToneList
- * Some progress on contact support
- * Remove unused getRegistrationCount()
- * remove annoying debug
- * revert SIP bit of e27e5c39bad27bae28f574eb2cba7717e8956229
- * Simplify CallManager::placeCallFirstAccount
- * Fix crash on hold
- * * #6905 : SIP refactor
- * gnome client: be sure key exchange is set correctly
- * Move code into createSipTransport
- * Fix account registration on start
- * ManagerImpl::registerAccounts(): simplify
- * * #5884: don't mess with pjsip threads in echo suppressor
- * * #6905 : simplify udp/stun/tls pjsip transport creation
- * Restore and improve support for Call history
- * fix launchpad build
- * SIPVoIPLink: simplify / refactor
- * Fix libwidget linking
- * SIP: simplify
- * IM : simplify
- * gnome: remove some debug
- * AudioRtpFactory::stop() cannot fail
- * * #6905: simplify SIP code
- * pjlib: fix build without SSLv2, fix warnings
- * Port history to the new syntax
- * Test a dock widget based implementation for contact and history
- * Disable SSLv2 support from pjsip and sflphone
- * deb: forge -g flags for pjsip
- * Fix deb packaging to get debug symbols
- * remove debug
- * pjproject: update to last stable release (1.10)
- * Require gtk >= 2.20 and glib >= 2.24
- * tlsadvanceddialog: simplify
- * * #6902 : fix errors spotted by -DGSEAL_ENABLE
- * Update daemon dbus XML and port KDE config backend from dbus to
- local
- * Remove unused but set variables
- * * #6929 : fix IM widget, cleanup
- * Unconditionally enable debug symbols
- * Should fix many KDE issues
- * * #6886 : hitting backspace on empty number have no side effects
- * * #6905 : fix AudioCodecFactory access in optimized builds (-O > 0)
- * Remove unsupported and broken jaunty/karmic packages
- * * #6902 : avoid using some gtk deprecated functions
- * Update dbus introspection files
- * * #6904: removed unused contactmanager
- * * #6903 : use correct dbus-cxx package name
- * * #6902: don't use individual gtk headers
- * Fix a segfault when config is not present
- * Merge latest (0.9.13) KDE code. This version is not yet ready for
- git master, but better than the previous one
- * addressbook : simplify
- * * #5659 : sflphone-plugins doesn't depend on libedataserverui
- * * #5659 : addressbook doesn't use libedataserverui
- * gnome client doesn't depend on evolution
- * * #5695: addressbook: simplify
- * * #5695: addressbook : remove AddrBookHandle from plugin
- * * #5695 : addressbook : remove unused stuff in the client
- * * #5695 : addressbook : remove unused stuff, use static mutex
- * gnome client doesn't use evolution
- * gnome: use proper API to set GTK_CAN_FOCUS
- * * #6897: removed unused focus state vars/callbacks
- * gnome: fix calls to sflphone_fill_codec_list_per_account
- * * #6623: gnome: don't leak in mainwindow
- * gnome: mainwindow whitespace cleanup
- * gnome: actions.c parameter doesn't have to be a double pointer
- * * #6895: fix memleaks, cleanup in accountconfigdialog
- * * #6893: fixes segfault in client on clean history
- * * #6894: fix leaks, cleanup in sflnotify
- * daemon: fixed prints in main
- * * #6892: simplify, fix leaks in dialpad
- * * #6887: audiopreference creates audio layer
- * * #6660: use const char * const, not std::string for globally
- visible constants
- * * #6852: Preferences now solely responsible for audiolayer creation.
- * * #6860: refactor uimanager, also fixes #6865
- * * #6853: hangup as soon as all digits have been deleted
- * * #6852: alsa: retry if device is busy
- * * #6852: audiolayer creation depends only on preference.audioApi
- * * #6850: gnome: fix build for gtk < 2.22.0
- * cleanup in iax
- * alsa: typo
- * pulse: if we can't peek in audio input, we can't drop samples
- * * #6849: show error window if codecs are missing, instead of dying
- * EchoCancel: unused, remove
- * * #6629 : use number of samples as arguments for audio filters
- * * #6629 : remove unused Algorithm interface
- * * #6629 : use helper to call alsa functions and display error msgs
- * Remove unused type
- * * #6841: fix some error handling
- * * #6629: simplify AlsaLayer::alsa_set_params()
- * Get gdk key definition from header
- * * #6828: Replace raw key codes by gdk defines
- * remove some debug, enhance some other
- * mainbuffer: simplify
- * * #6561 : fix phantom call after transfer
- * Conference Participant set : simplify
- * SIPCall: remove unused functions, make invite session public
- * * #6229 : remove malloc/free from pulse audio loop
- * * #6629 : simplify pulse callbacks
- * * #6629
- * Simplify widgets
- * * #6629 : keep the correct audio module when frequency changes
- * * #6751: fixed erroneous debug msgs
- * callable_obj.h: removed unneeded pthread header
- * alsalayer: cleanup
- * * #6629: Always restart audio driver when changing parameters (ALSA
- only)
- * gnome GUI: don't block in DBus signal errorAlert()
- * * #6629 : simplify AudioLayer creation
- * * #6629 : remove unused and unconfigurable frameSize from audiolayer
- * * #6629 : remove unused error message from audio layer
- * Fix logic error when switching audio API
- * Remove unused AudioProcessing class
- * AudioRtpRecordHandler::initNoiseSuppress() : use noiseSuppress
- directly
- * * #6629 : use DC blocker directly in audio layers
- * * #6629 : clean AudioLayer
- * * #6629 : don't store mainbuffer inside audiolayer
- * * #6629 : correct AudioLayer::notifyincomingCall()
- * * #6554: cleanup, refactoring in sipvoiplink
- * * #6554: cleanup in iaxvoiplink
- * * #6554: throw exception in getSIPCall if pointer is NULL
- * * #6554: make some methods of sipvoiplink static
- * * #6655: cleanup in managerimpl
- * * #6554: refactoring, fix memleaks in sipvoiplink
- * * #6478: remove throw specs, cleanup in voiplink
- * * #6629 : remove unused AudioDevice
- * * #6655: removed more dependencies from managerimpl
- * * #6744: simplified numbercleaner
- * conference : remove one prototype
- * * #6743: fix ip2ip
- * Don't give glib warnings if icons are not found
- * gnome: fixed includes
- * Codec.h: removed unused function
- * * #6742 : clean dbus & icons
- * * #6699: refactor/cleanup accounts
- * icons: cleanup
- * timer : use second precision, not millisecond
- * calltree_update_clock : use correct type, returns something
- * * #6737: fixed typo in dbus call
- * * #6737: removed tests for removed API
- * * #6737: dbus: fixed bug from merge
- * * #6737: cleanup in accountlist
- * * #6737: cleanup in dbus
- * * #6740 : fix history double free
- * * #6740 : remove time updating thread from calls
- * * #6737 : use c99 for client
- * * #6738 : make history loading faster
- * sipvoiplink : don't crash on transfers
- * fixed typo
- * Remove unused file
- * Don't build networkmanager.cpp at all if NM is disabled
- * _debug* -> _debug
- * * #6554 : simplify sipvoiplink
- * hudson: added -x to git clean command
- * added git clean to hudson script
- * audiocodecfactory: cleanup
- * * #6718: refactored setTlsSettings into SIPAccount
- * * #6718: removed more unused methods
- * * #6718: refactored confmanager code into sipaccount
- * remove unused functions
- * * #6718: confmanager: removed more unused methods
- * AudioCodecFactory : cleanup
- * #6697 : Turn callableElement struct into union
- * * #6718: confmanager: removed more unused methods
- * * #6718: confmanager: removed more unused methods
- * * #6718: removed unused dbus methods, refactoring
- * * #6699: accounts: cleanup/refactoring
- * * #6699: refactoring, cleanup in accounts
- * * #6699: more account cleanup
- * remove unused autoconf variable
- * * #6714: fixed hudson script
- * make distclean in hudson
- * added || exit 1 to run_tests.sh call
- * * #6714: fixed make distcheck for sflphone-plugins
- * * #6714: fixed make distcheck for gnome client
- * * #6714: fixed make distcheck for daemon
- * git: #6698 split the main .gitignore file
- * gnome: gpointer is already a pointer
- * gnome: calltab_init: use calloc instead of malloc
- * * #6699: more account cleanup
- * * #6699: cleanup account
- * * #6554 : more *voiplink cleanup
- * * #6558 : more sipvoiplink simplification
- * * #6558: saner loadSIPLocalIP prototype
- * gnome: #6623 clean calllists
- * * #6692: more audiolayer cleanup
- * * #6692: cleanup/refactoring in audiolayers
- * * #6692: more forward declarations, AudioThread->AlsaThread
- * * #6692: audiolayer cleanup
- * * #6692: alsalayer cleanup
- * * #6558 : remove account creator
- * * #6558 : clean sipvoiplink
- * * #6554 : cleanup sipvoiplink
- * audiortp: cleanup
- * * #6657 : fix launchpad builds for good
- * * #6675 : send RTP dtmf events only once
- * * #6655: more cleanup
- * AudioRtpSession::updateSessionMedia() : simplify
- * * #6655: more cleanup in managerimpl
- * * #6655: removed more code, cleanup
- * * #6655: more cleanup, fixed infinite loop
- * * #6655: removed more unused files
- * * #6655: removed unused mutex
- * * #6655 removed more unused code
- * * #6655: removed unused methods
- * * #6655: cleanup in main
- * * #6663: fixed segfault when off hold from transfer
- * * #6658: user's active codec selection is respected
- * * #6660: static global string should be static const char* const
- class member
- * * #6659: use g_strcmp0, not strcmp for vals that may be null
- * callable_obj: fix double free
- * calltree_display_call_info() : simplify
- * * #6657: Fix launchpad builds
- * Logger::log() : simplify
- * AudioRtpSession : privatize members
- * * #6655: more constness, cleaned up/simplified methods
- * * #6654: call DBus::_init_threading so that dbus-c++ to make it
- threadaware
- * set default credentials on account creation
- * AudioCodecFactory::scanCodecDirectory() : simplify and correct
- * * #6623: fixed typos
- * * #6623: fixed more leaks
- * * #6623: fixed more leaks
- * * #6623: fixed more leaks, don't print codec name if null
- * * #6623: more leaks fixed in client
- * * #6623: fix more leaks, fixed some warnings
- * * #6623: fixed leak in history
- * updated gitignore
- * initialize dbus dispatcher correctly
- * Fix tests, hudson doesn't have a dbus daemon running
- * remove unused code
- * removeCall() : simplify , fix leak
- * stopRtpThread() : simplify
- * *CurrentCall : simplify
- * Fix memleak
- * fix serialization of audio api (pulse / alsa)
- * account map : simplify
- * remove call from callmap before terminating it, avoid use after free
- * * #6630 : don't make DBusManager a singleton
- * call: return confID by value
- * add back history code deleted by error
- * history : reverse logic
- * simplify history serialization and remove some debug
- * remove annoying debug
- * * #6464 : replace cerr with _error
- * * #6464: replace cout with logger macros
- * replace printf() with logger macros
- * update .gitignore
- * remove unused function
- * update eclipse projects
- * uimanager_new() : simplify
- * rename directories
- * celt: simplify a bit
- * Fix CELT configure.ac test
- * * #6612 : template speex codecs
- * * #6623: refactored conference obj
- * * #6623: refactored callable object, removed leaks
- * * #6623: more cleanup, fix leaks, make global vars static and rename
- them
- * * #6623: calltree: fixed memleaks, simplified code.
- * audiolayer: init pointer members
- * manager: catch exception on invalid hangup
- * * #6623: don't leak on calls to create_new_call
- * * #6611 : clarify codecs prototypes
- * ringtones : .au and .ul files are both ulaw
- * * #6611 : make sure samplerate converters are called correctly
- * ManagerImpl::switchAudioManager() : simplify
- * * #6623: fixed more leaks
- * * #6623: fixed more leaks
- * * #6623: fixed more leaks
- * * #6623: fixed leak, line-endings in imwidget
- * * #6627: zero-initialize pointers if they're going to be deleted
- * * #6628: don't leak calls on exceptions
- * Revert "audiortp: call join after calling stop on RtpThread"
- * sflphone-client: more constness
- * audiortp: call join after calling stop on RtpThread
- * * #6625: return 0 on successful completion
- * * #6624: fix segfault on servercallfailure
- * * #6621: Fixed double free, unlock mutex in ManagerImpl::terminate
- * * #6220: remove audio stream when peer hangs up
- * * #6596: AudioSymmetricSession shouldn't self-delete
- * resampler: grow internal buffers dynamically
- * merge up and down sampling => resampling
- * Leave test directory unchanged when running make check
- * audio algorithms : remove unused prototype
- * ringtone: detect codec from file extension
- * *AudioFile : simplify
- * * #6596: create local SDP on the stack, not the heap
- * * #6596: don't call Ost::Thread::terminate from dtor
- * audiofile: cleanup (samplerate -> unsigned)
- * remove unused func
- * samplerateconverter: cleanup
- * RingBuffer::Put() : remove unused return value
- * MainBuffer::putData() : remove unused return argument
- * audiolayer::putMain() : remove unused func
- * AudioLayer::putUrgent() : remove unused return value
- * * #6618: delete any remaining ringbuffers in destructor
- * RingBuffer::availForPut() : remove
- * * #6617: return from main rather than calling exit
- * MainBuffer::availForPut(): remove
- * RingBuffer: simplify
- * alsa : remove write only variable
- * fix memcpy declaration
- * bcopy(src, dst) -> memcpy(dst, src)
- * RingBuffer::Get() : remove constant volume argument
- * return a copy of the call ID, not just a reference.
- * MainBuffer::getDataById() : remove volume argument (always 100)
- * MainBuffer::getData() : remove constant volume argument
- * RingBuffer::Put() : remove constant volume argument
- * MainBuffer::putData() : remove constant (=100) volume argument
- * audiolayer: remove constant _defaultvolume
- * AudioRtpRecordHandler / AudioRtpSession : simplify
- * mainbuffer: fix test
- * iaxvoiplink : simplify
- * sip registration callback: fix a dbus crash
- * MainBuffer: simplify
- * AudioRtpFactory: return cached type of rtp session. The rtp session
- can have disappeared if the call was put on hold
- * AudioRtpFactory: remove unused setters
- * Fix launchpad builds
- * * #6611 : remove unused bandwidth codec information
- * * #6611: AudioCodec: remove useless/unused setters
- * make sure buffer string is initialized correctly
- * * #6596: declare certain destructors virtual
- * audiolayer : cleanup
- * Simplify doc build rules
- * * #6270: don't build dbus-api doc with make, should require make all
- * configure.ac: cleanup
- * Remove copy of dbus-c++ from libs/
- * * #6596: stop clock thread when peer hangs up
- * removed unused Fmtp.h
- * * #6595: more logical initialization order
- * * #6600 : fix account creation
- * * #6601 : fix configure.ac tests
- * remove unused variable
- * Don't mix stack and heap based allocations
- * Fix copyright (2009, 2008, 2009 -> 2008, 2009)
- * Fix warnings found by clang
- * * #6595: fix initialization order for AudioRTP
- * * #6592: removed typedef std::string CallID
- * * #6586: implement local g_slist_free_full for older glib versions
- * * #6579: fix memory leaks in client (there's a lot left)
- * ShortcutPreferences::setShortcuts() : simplify
- * Fix merge
- * * #6548: remove call to non thread-safe strerror()
- * AudioRtpFactory: each instance is associated to exactly one SipCall
- * create_audiocodecs_configuration() : make static
- * * #6269 : refactor AudioRtpSession
- * Fix AudioSymmetricRtpSession.h inclusion guard (cherry picked from
- commit c3081dce1cc1370d6d3558a4c4ef5cfac0d21caf)
- * * #6269: Rename AudioRtpSession to AudioSymmetricRtpSession
- * * #6574: Don't exit when connection to pulseaudio server fails
- * accountconfigdialog.h : remove some stuff from header
- * * #6560: fix configuration test
- * Fix warning in test
- * * #6560: don't hide password entry in security tab
- * * #6560: set initial password for SIP accounts
- * * #6506: remove useless pointer indirection
- * * 6560: password is now specific to IAX accounts
- * * #6560 : actually use, store, restore, transmit SIP credentials
- * * #6560: YamlEmitter: serialize sequences
- * YamlEmitterException: typo
- * ManagerImpl::computeMd5HashFromCredential() : simplify, fix memleak
- * * #6561: invite_session_state_changed_cb() : simplify
- * * #6561: More useful debug in VoIPLink::removeCall
- * * #6561 : fix ghost call reappearing in GUI after transfer
- * while -> for (make the code smaller)
- * * #6558 : Account::loadConfig() : move IAX code to IAXAccount
- * IAXVoIPLink::getAccountPtr : simplify
- * * #6554 : access the SIPVoIPLink directly, not per account
- * SIPVoIPLink is instanciated only once and is not associated to a
- single account
- * yamlnode: use const references when possible (still some left to do)
- * Account::_accountID: constify
- * VoIPLink: simplify, remove unused method
- * hudson test : no need to call run_tests.sh anymore
- * Remove AccountID type and AccountNULL define
- * Make check runs the test (no need to call run_tests.sh manually
- anymore)
- * gnome GUI: Fix tests
- * Revert "Move registration information from SIPAccount to
- SIPVoIPLink"
- * * #6392: pluginmanagertest: fix warnings reported by valgrind
- * * #6547 : remove unused exceptions
- * * #6547: CallManagerException: use runtime exceptions
- * * #6547: InstantMessageException: use runtime exceptions
- * * #6547: do not throw exceptions if some settings are not present in
- config file
- * * #6547: YamlParserException: use runtime exceptions
- * * #6547: VoipLinkException: use runtime exceptions
- * * #6547: YamlEmitterException: use runtime exceptions
- * * #6547: DTMFException: use runtime exceptions
- * * #6547: AudioFile: use runtime exceptions
- * * 6547: AudioZRtpSession: remove impossible error case
- * * #6547 : AudioRtpSession: remove impossible error case
- * * #6547: AudioZrtp: use runtime exceptions
- * * #6408 : send authenticationUsername to GUI
- * * #6408 : store/restore authenticationUsername from config file
- * SIPAccount: simplify
- * Move registration information from SIPAccount to SIPVoIPLink
- * SIPAccount::getAccountDetails : simplify
- * * #6540: yaml parser: simplify
- * sdp.cpp : fix a warning
- * * #6540: yaml parser : remove std::string typedefs
- * * #6540: Simplify yaml unserialization
- * * #6540 : add a Conf::ScalarNode constructor for booleans
- * setAccountDetails(): simplify
- * * #6408: store authentication username in daemon
- * * #6408: Be able to set the authentication username in the GUI
- * * #6507 : do not crash if the program is not sflphoned
- * Fix tests
- * macroify SIPAccount::unserialize()
- * Move all .cpp files from sflphoned target to libsflphone.la, except
- main.c
- * main() : simplify, return positive error codes
- * * #6507 : find codecs dir in build directory
- * * #6392: Sdp: move clean functions to destructor
- * AlsaLayer::adjustVolume() : simplify
- * alsalayer : reduce indentation
- * malloc/free -> new/delete
- * malloc/free -> new[]/delete[]
- * malloc/free -> new/delete
- * AudioSrtpSession: simplify base64 encoding
- * * #6392: Initialize std::string from pj_str_t correctly
- * * #6392: AudioRtpSession: Initialize remote port
- * Audio settings : Initialize _echoCancelTailLength and
- _echoCancelDelay(0)
- * Initialize variable
- * YamlParserException : fix use of stack variable after it has been
- deallocated
- * * #6392: fix memory leak in history
- * * #6392 AudioCodec : fix memory leak
- * * #6392 : fix memory leak in sip account
- * * #6408: clean up sipaccount (cosmetics mostly)
- * sipaccount.cpp serialize() : reduce number of lines
- * * #6392: invalid memory access
- * * #6392 : fix invalid memory access
- * * #6479: merged useful code from MimeParameters into Codec interface
- * * #6462: fixed hangup on IP2IP call
- * added run_daemon.sh script
- * test: remove unused variable
- * Remove functions only used by a failing test (cherry picked from
- commit fcf718cb75de7f1882dc61c07bb8d300dfa10f85)
- * * #6360 : make client tests build (cherry picked from commit
- 028b2835f040e51ab8ab979b32732b07b8798fce)
- * * #6360 : fix warnings in check_global test (cherry picked from
- commit 9e2bd6a7496dd64f6f48595e385760019aab1193)
- * * 6360: updated API calls in tests, but they're not building yet
- (cherry picked from commit 548f6f0f919b43772a3e9c667e5e292791281795)
- * Fixed include in tests (cherry picked from commit
- aeadc7525c1e31f936670ac8b02f0bcf387c38a8)
- * Remove unused variables and functions
- * IAX: fix warnings (cherry picked from commit
- fd7a113a11cac2cd9a7c36929e88ad28195c4c35)
- * Remove unused DEBUG define which interferes with logger.h (cherry
- picked from commit b2f72b91d0f43cb1dd94d138882a8caa9c841c24)
- * * #6392: no need to check for account NULLity since it is
- dereferenced above
- * * #6392: fix a memory leak, replace by stack allocation
- * * #6392: remove a variable assignement which confuses cppcheck
- * process_conference_participant_from_serialized() : remove unused
- function
- * * #6392: s/free/g_free/
- * * #6392: fix a memory leak in abookfactory_load_module()
- * * #6392: remove generate_call_id() used only once
- * * #6392: fix memory leak (opendir() without closedir())
- * * #6392: AudioRecorder(): ensures mbuffer is set
- * Remove SFLPHONED_VERSION from global.h, use autoconf PACKAGE_VERSION
- * #6298: Cleanup
- * #6331: Fix deleting ringtone file after call have been answered
- * * #6330: merged user_cfg into headers
- * #6298: Fix conference recording file update at conference end
- * #6298: Fix record file name serialization for conference
- * * #6295: cleanup of codec hierarchy
- * #6298: Fix gtk warnings
- * * #6300: added script to run tests
- * #6109: Add recording playback for conference
- * * #6300: tests do not require an installed sflphone
- * * #6295: re-removed clone methods
- * #6109: Fix gtk_critical warnings for incoming calls
- * #6109: Fix GTK_CRITICAL warning
- * #6109: Fix icons when history is not activated
- * #6109: Fix warnings
- * #6109: Implement stop recorded file playback signal
- * Revert "* #6295: removed unused clone method"
- * * #6295: removed unused clone method
- * * #6296: removed non existant file from Makefile.am
- * #6109: Stop fileplayback for outgoing call
- * #6109: Implement stop recording playback button
- * Fix binding names errors in dbus introspection file
- * #6109: Implement playback recorded file callback in client
- * #6109: Store recorded file path on client side
- * #6109: Add dbus methods for call recording playback
- * * #6290: remove unused classes from utilspp
- * * #6288: cleanup sdp
- * * #6288: fix exception usage
- * * #6288: simplify SdpException
- * * #6288: cleanup in sdp.cpp/h
- * #6109: Only display playback button if record file is set and valid
- * * 6290: updated configure.ac to remove functor Makefile
- * * #6290, #6289: removed unused classes from utilspp, fixed make
- check
- * #6109: Add button for history playback of recorded file
- * * #6289: removed unused observer class
- * * #6282: forward declare sdpMedia in sdp.h
- * * #6281: renamed setCallAudioLocal->setCallMediaLocal
- * #6183: Handle conference with more tahn two calls
- * #6183: Fix history icons when calling back a conference from history
- * #6183: Fix icons inconsistencies in history for conference hang up
- * #6183: Fix toolbar actions when selecting a conference in history
- * #6183: Fix conference serialization
- * #6268: Serialize only calls
- * * #6269: removed useless type testing
- * ignore some files in test/
- * * #6268: Remove dead class AudioSymmetricRtpSession
- * #6251: Do not had history calls in calllist when loading history
- file
- * #6251: Fix insertion in history map in before saving history file in
- daemon
- * #6251: Fix history unit tests
- * #6251: Order the list before serailization, get rid of the hashtable
- in history
- * #6251: Implement history serialization using a list wether than a
- map
- * * #6253: remove external audioport from header, make all members
- private
- * * #6253: don't store external local audio port (used for NAT) in
- Call
- * #6251: Add start_time timestamp in history serialization
- * #6251: Fix call insertion in conference items
- * #6233: Fix serialized account list terminated with a ";" character
- * #6238: Fix draggable history calls into current calls
- * #6233: Fix toolbar updates
- * #6233: Fix history
- * * #6235: remove pyc files from git tree
- * #6233: Handle cases when one or manuy calls are unreachable in
- createConfFomrParticipantList
- * #6233: Handle wrong numbers in createConferenceFromParticipantList
- * #6231: Fix drag-n-drop issue
- * * #6173 : move sippxml in tools
- * #6231: Fix merging issue
- * #6183: Implement conference unserialize
- * * #6212: remove extraneous flags from globals.mak
- * #6183: Unserialize conference data in conference
- * #6183: Add account information in request for conference call from
- history
- * #5755: Add -ldl to liker in sflphone-gnome
- * #5755: Fix fedora 15 compilation issue
- * #6183: Serialize conference participant phone number and account
- * #6183: Add conference timestamp in serialization
- * * #6186: don't include global.h, just logger.h
- * #6183: Fix saving history to file
- * #6183: Fix removing call from calllist
- * * #6184: remove pointers to Manager from AudioRtpSessions
- * #6183: Calling calltree_add_call explicitely for history
- * #6183: Ability to store conference inside history tab queue
- * * 6181: remove unused API from sipcall
- * #6171: Implment nreCallCreated callback
- * #6167: Fix participant list NULL ending
- * #6149: First draft of conference creation from history
- * #6149: Fix multiple call/conf selection callbacks ...
- * #6129: Fix place_call function called twice for pressing enter
- action
- * #6129: Fix double click action for history
- * #6149: Add dbus call for creating conference from history
- * #6129: Fix placing call from history and addressbook (still need to
- fix icon)
- * * #6148: removed unused AudioRtpFactory constructor
- * * #6145: remove unused isAudioStarted
- * * #6145: remove unused isAudioStarted
- * #6129: Add conference into history, fix call/conference selection
- * * #6143: don't use getType outside of serialization methods
- * * #6132: forward declarations instead of includes
- * * #6132: add constness, remove redundant "inline" keywords
- * #6129: Add timestamp to conference object to order history entries
- * * #6128: remove unused forward declarations from header
- * * #6127: make noncopyable class actually noncopyable
- * * #6125: don't include AudioRtpFactory in sipcall.h
- * #6123: Fix alsa ringback audio file
- * #6123: Fix raw audio file loading problem
- * #6109: Fix daemon plugin manager unit test
- * #6109: Fix history manager unit tests
- * #6109: Recording filename in daemon and client for history items +
- serialization
- * #6109: Refactor AudioFile to play recorded call
- * * #6104: AudioCodec moved to sfl namespace
- * * #6099: remove active flags from codec classes
- * #6095: Add notification-daemon as a runtime dependencies for rpm
- packages
- * #6095: Fix fedora 15 compilation in MineParameters.h
- * #6095: Declare static variable explicitely for client
- * #6095: Add logs to build OSC build machine
- * * #6098: global variables should have file-scope to avoid name
- conflicts
- * #6095: Fix compilation error for Fedora 15
- * #6095: Update SFLphone version to 0.9.14
- * #6095: Add specification file in opensusse build service for
- sflphone-plugins
- * #6073: Fix sflphone-plugins build on launchpad
- * #6093: Rename CodecDescriptor for AudioCodecFactory
- * * #6089: fix warnings in make check
- * * #6086: renamed codecs methods to audio_codecs
- * * #6085: renamed codec related dbus calls to audio_codec
- * #6065: Remove g_print from client, use DEBUG instead
- * #6065: Add actions name for addressbook
- * * #6085: renamed codecs* widgets/functions audiocodecs*
- * #6065: Fix Addressbook runtime warnings
- * #6065: Replace Codecs tab for Audio in account preference dialog
- * #6065: Fix "transfert" typo
- * #6065: Fix addressbook action runtime warning in uimanager
- * * #6082: fixes make check by adding libcrypto libs to test
- dependencies
- * #6073: Rename plugin/addressbook folders for addressbook/evolution
- in sflphone-plugins
- * #6074: Removed AC_SUBST from configure.ac when using
- PKG_CHECK_MODULE
- * #6073: Fix sflphone-plugins package build
- * #6073: Fix sflphone-common build
- * #6065: Fix runtime gtk warning when initializing searchbar without
- addressbook
- * #6063: Fix mozilla-tellify gitignore
- * #6063: Remove stream copy file using ifdef macro
- * * #6012: fix make dist for sflphone-common
- * #6063: Update .gitignore file
- * #6058: Fix base64 encoding related warnings
- * #6056: Fix SdpException handling
- * #6055: Fix unknown pargma warning for gcc <= 4.5
- * * #5949: test gcc version before disabling unused-but-set warning
- * #6054: Fix addressbook plugin compilation warning
- * #6048: Fix uimanager static initialization
- * #6046: Fix addressbook factory static initialization of member
- addrbook
- * #5979: Fix implicit function declaration warning
- * #6042: Fixed discarding qualifier warnings in client
- * #6041: Fix instant messaging unhandled case warning
- * #5994: Implement set current addressbook name and search type in
- addressbook plugin
- * #5994: add rules for launchpad packaging of addressbook plugin
- * #5994: Fix addressbook plugin configuration loading
- * #6027: Fix addressbook enabled test from configuration
- * #6027: No need of gnomedoc related macros in addressbook plugin
- * #6027: Add NEWS file required for build
- * #6027: Add addressbook plugin autogen.sh script
- * #6027: Remove plugins from client
- * #6027: Add sflphone-plugins folder at project's root level
- * #5994: Move addressbook folder from contacts to plugin folder
- * * #6011: removed unused Makefiles
- * * #6010: remove unused headers
- * * #5952: fix "string constant to char*" warnings
- * * #6009 fixed warnings
- * * #6003: finished cleanup of account classes
- * * #6003, #6004: cleanup of account classes, defaultAccount no longer
- global
- * * #6000: fix memory leak of args object
- * * #5998: removed using namespace std from networkmanager
- * * #5998: removed "using namespace std" from ZrtpSessionCallback
- * * #5998: removed using namespacestd from AudioZrtpSession.h
- * * #5998: remove "using namespace std" from auriorecord.h and
- MimeParameters.h
- * * #5998: remove using namespace std in main
- * * #5998: removed "using namespace std" from logger
- * * #5949: test gcc version before disabling unused-but-set warning
- * #5994: Installation of addressbook plugin
- * #5979: Implement codec full addressbook search from plugin
- * #5979: Implement addressbook factory and plugin
- * * #5981: unused webwidget removed
- * #5966: Account config synchronization fix (for stun)
- * #5954: Handle media name exception
- * #5954: Fix audio codec name display in client
- * #5954: Clean up getSessionMedia methods
- * * #5957: getRecordingSmplRate returns a value
- * #5954: Clean up getCurrentCodec methods
- * * #5950: remove "converting to non-pointer type 'int' from NULL"
- warnings
- * #5915: Full gain control version
- * * #5949: remove more unused variable warnings
- * * #5949: remove unused/unused-but-set variable warnings
- * * #5949: show_preferences_dialog returns a success value
- * * #5946: cleanup of include directives, undefined function
- * * #5515: comment out SSLv2 calls in pjsip
- * #5915: Implement different slope for attack tme and release time for
- gain control
- * #5915: use only one input signal for gain control (removed output
- buffer)
- * #5921: Fix no audio after holding a conference
- * #5916: Add gaincontrol files
- * #5916: Implement FFMPEG/CCRTP video streaming prototype
- * #5903: Fix call transfer during a conference
- * #5915: implement rms detector, first order averager, limiter for
- gain control
- * #5914: Fix call transfer when no notification request is required
- * #5899: Fix conference right-click segfault
- * #5884: temporary fix segfault in pjsip memory pool
- * #5883: Fix compilation issues on maverick and lucid
- * #5755: Fix fedora 15 compilation without patching ccrtp
- * [#5855] Make echo canceller optional
- * #5855: Fix echo suppression activation/deactivation
- * #5855: Implement pjsip echo canceller
- * #5814: Speex initialization function uses samples, not bytes
- * #5814: Test using more unbalanced signals
- * #5814: Fix buffer size for long echo length or long echo delay
- * #5814: Adjust level for echo cancellation at runtime
- * #5814: Process noise reduction before echo cancelling
- * #5814: Implement speex post echo canceller processing
- * #5814: Dump echo cancel file to disk
- * #5814: Add parameters for echo cancel
- * #5809: Add configuration parameters
- * #5809: Implement speex echo canceller in audio rtp session
- * #5814: Code cleanup
- * #5814: Fix conf creation with several incomming ringing calls
- * #5814: Fix conf creation segfault when dragging a call on hold on a
- ringing call
- * #5809: Added unit test for echo cancellation and implemented
- "process" virtual method
- * #5709: Add always recording option in configuration
- * #5709: Add always recording option in audio conference panel
- * #5709: Add core functionnality for always recording (missing config
- options)
- * #5769: Fix conference participant handling (detach/attach) and hold
- actions
- * #5747: Fix recording icons and state for conference when adding new
- participant
- * #5769: Code cleanup
- * #5769: Fix hangup unsent calls
- * #5769: Fix remove/add additional participant to conference
- * 5769: Several fixes concerning confererence handling
- * #5769: Fix compilation error
- * [#5769] Fix audio streams binding in main buffer
- * #5769: Removed access to audio mixer from audio layer
- * #5765: Fix audio crash for illformated wavefiles
- * #5765: Add maximum iteration for finding fmt and data "chunck"
- * #5589: Fix compilation of libnotify under
- * #5757: Fix abort signal when receiving INFO
- * #5747: Add usersDetached.svg
- * #5747: Handle offhold action for recording conference
- * #5747: Fix off hold action for conferences
- * #5747: Implement update conference in record action in calltree
- * #5747: Add new icons for recording conferences
- * #5747: Add recording state for conferences
- * [#5738] Remove getAudioDriver call from manager (replace by
- _audiodriver var)
- * [#5738] Refactor mutex protecting audiolayer
- * [#5737] Fix HD conference recording
- * [#5730] Fix start audio session after changing sampling rate
- * [#5714] Fix enter keyboard event for addressbbok and history
- * [5695] Fix addressbook combo box update when no addressbook selected
- * [#5695] Fix addressbook initialization and search bar update
- * [#5695] Add mutex for books_data in addressbook to protect async
- calls
- * [#5695] Get back addressbook open from uri
- * [#5695] Fix absolute addressbook URI for local addressbooks
- * [#5695] Implement libebook 3.0 interface
- * [#5571] Better logic for hangup (for case where call have not been
- sent yet)
- * [#5571] Update error handling in voip links
- * [#5571] Fix compile time warnings
- * [#5696] Fix installation dependencies for Natty
- * [#5669] Add mention that sflphone.org is for testing only
- * [#5693] Add natty in teh dput.conf file
- * [#5690] Remove not useful logs
- * [#5670] Use dynamic payload type for rtp dtmf
- * [#5668] Clean up sflphone configuration logging
- * [#5668] Fix hook checkbox configuration update
- * [#5666] Fix unit tests
- * [#5666] Manage event subscription
- * [#5666] Emit bye request when subscription is terminated
- * [#5666] Bye request should be sent after event subscription
- notification is done on transfer
- * [#5666] Make reinvite method static (to be called in pjsip
- callbacks)
- * [#5666] Hangup Call in manager for AccountNULL and IP2IP
- * [#5589] Use PKG_CHECK_MODULE for every client's dependencies
- * [#5623] Enlarge initial size of pjsip memory pool for calls (16k)
- * [#5564] Fix audio recording resampling for g722
- * [#5571] Move attribute handling for onhold/offhold actions in SDP
- session
- * [#5571] Codec negotiation refactored and unittested
- * [#5571] Implement tests
- * [#5571] Implement pjsip negociator
- * [#5571] Fix unit tests
- * [#5571] Add Fmtp.h to repository
- * [#5571] Integrate mime types and codec factory
- * [#5571] Handle exception when SDP negotiation fails
- * [#5570] Add sflphoned-sample.yml in repository
- * [#5564]: Implement stereo to mono mixing for rigntone
- * [#5342] Update audio stream initialization
- * [#5514] Restore test ni historytest suite
- * [#5514] Fix
- * [#5514] Disable test_create_history_path
- * [#5514] use pulseaudio in sample config file
- * [#5514] Fix test: load history from file
- * [#5514] Do not use X
- * [#5513] Make unit tests compile successfully
- * [#3947] Enable unit tests in Jenkins
- * [#5454] Fix build system to handle new version number
- * [#5454] Update languages from launchpad
- * [#5454] Add --without-celt in OpenSuse build service
- * [#5454] Change version number
- * [#5331] Added first SDP session tests
- * [#5273] Update nightly build version tags to conform dpkg rules
- * [#5211] Refactor send register method for iaxvoiplink and
- sipvoiplink
- * [#3950] Remove call being transfered from calltree
- * [#5211] Use appropriate memory pool for transport selector
- * [#5211] Fix strict aliasing rules warning in pjsip
- * [#5211] Bring back pjsip shutting down sleep to 1000 ms
- * [#5211] Fix registration callback segfault when closing the
- application
- * [#5211] Use the dialog memory pool for Route header in INVITE
- request
- * [#5211] Add temporary memory pool for findLocalAddressFromUri and
- findLocalPortFromUri
- * [#5211] Use individual memory pool for dtmfs
- * [#5211] SipVoipLink refactoring
- * [#3950] Attended transfer for conference calls
- * [#5284] Fix DNS resolution for Route with specified port number
- * [#5284] Some code cleanup
- * [#3947] Fix typo in hudson script
- * [#5284] Added sip route to REGISTER, INVITE, BYE request, plus DNS
- resolution
- * [#5266] Use RTP dtmf as default
- * [#5284] Added pjsip_process_route_set after setting routes in regc
- structure
- * [#5286] Fix parsing error due to long configuration file (removed
- max event)
- * [#5286] Fix false test in configuration emmiter
- * [#5286] Code cleanup
- * [#5286] Updated exception handling in configuration system
- * [#4969] Fix put SRTP call on hold
- * [#3950] Add debug messages
- * [#3950] Ability to perform an attended transfer
- * [#5276] Fix initialization problem in g722
- * [#3950] Add replace header in SIPVoIPLink::transferWithReplaces
- method
- * [#3950] Implemented attended method in SIPVoIPLink
- * [#3950] Cleanup transaction request received callback
- * [#3950] Implement dummy attended transfer in gnome-client
- * [#5249] Fix audio samplerate update algorithm for g722
- * [#5249] Fix uninitialized variable used in conditional jumps
- * [#5249] Fix conditional jump error in audiolayer (uninitialized
- value)
- * [#5267] Use autoconf 2.65 as a requirement (instead of 2.67)
- * [#5267] Restore manual pjsip configuration and compilation
- * [#5267] Autodetect celt version (0.9.1, 0.7.1)
- * [#5267] Fix deprecated macros in gnome client configure.ac
- * [#5267] Update configuration for libcelt-dev
- * [#5267] Fix build autoconf and automake
- * [#5227] Deactivate automatic call to astyle after compilation
- * [#5242] Hangup every calls before leaving
- * [#5237] Will now nightly-build for natty, Karmic deprecated
- * [#5229] Use inner class for rtp thread instead of inheritance
- * [#5211] Move mainbuffer unbind call in rtp final method
- * [#5211] Initialize sip call memory pool using 16 kb
- * [#5211] Use call memory pool in session reinvite
- * [#5211] Add debug messages
- * [#5211] Use and internal pool for calls
- * [#5211] Reduce pjsip memory pool usage for stateless error messages
- * [#5211] Refactor call deletion
- * [#5212]
- * [#5208] Refactor codec management for accounts
- * [#5168] Remove printf from codec's encode & decode method
- * [#5168] Fix celt compilation on launchpad
- * [#5168] Fix sflphoned compilation warnings in audiocodec.h
- * [#[#5168] Must keep the g722 specific RTP rate to avoid incoming
- packet timeout
- * [#5168] Fix static/dynamic payload rtp session update
- * [#5168] Throw SIPVoipLink Error if codec not instantiated in new
- outgoing call
- * [#5168] Fix dynamic/static codec payload type ambiguity
- * [#5169] Fix doubled IP2IP profile when no config file
- * [#4867] Add gtkinfobar in configuration panel
- * [#4867] Disable input/output/ringtone selection when using default
- alsa plugin
- * [#4952] Patches for possible buffer overflows
- * [$4885] Fix schemas problem
- * [#4885] sflphone-gnome.schemas not present during build
- * [#4885] Add gconf shemas directories in opensuse build system
- * [#4885] Add file/folder ownership for opensuse-factory build system
- * [#4906] Fix opensuse-factory build
- * [#4885] Update name dependency for libedataserver
- * [#4885] Fix non-void function without return in dbus-c++
- * [#4895] Update language translation
- * [#4896] Update session timestamp when updating media
- * [#4896] Reapply RTP hack for G722 payload type
- * [#4896] Update recording sampling rate when updating codec
- * [#4897] Save codecs in config for each configuration changes
- * [#4895] Do not save config when sflphone quit
- * [#4885] Update date for copyright
- * [#4885] Deactivate siptest that require more than one sipp instance
- * [#4879] Remove inmcoming call notification from IAX
- * [#4885] Some cleanup
- * [#4874] Add setCancel immediate/deffered for ost::Thread
- * [#4879] Fix incoming call notification
- * [#4878] Set keyboard focus on searchbar when selecting addressbook
- * [#4874] Fixed compilation warning
- * [#4874] Fixed compilation warning in sipvoiplink
- * [#4874] Fix compile time warning in RTP record handler
- * [#4874] Fix conditional jump in SDP
- * [#4874] Fix conditional jump based on uninitialized value
- * [#4874] Store call id within rtp thread context
- * [#4874] Fixed conditional jump based on uninitialised value in
- conference
- * [#4871] Fix default account fetching
- * [#4870] Delete RTP session when Refusing an incoming call
- * Restore IP to IP call
- * [#4857] Fix audio codec negotiation problem
- * [#3947] Adjust ressources allocated to compilation
- * [#3947] Disable unit tests in Hudson
- * [#4305] Free mutex only when really quiting SFLphone
- * [#4859] Update copyright to 2011 in every source file
- * [#3218] Character '.' stripped by the caller engine
- * [#4854] Fix typos, desktop entry
- * [#4847] Apply RTP modification to ZRTP session
- * [#4852] Update Karmic and Lucid dependencies
- * [#4852] Add Libedataserver and libedataserverui as gnome client
- dependencies
- * [#4852] Add authentication mechanism for EDS
- * [#4851] Fix segfault when closing pulseaudio layer too rapidly
- * [#4808] Some otehr cleanup
- * [#4808] Made some cleanup
- * [#4808] Added mutex in rtp session for codecs and noise process
- * [#4847] Update audio processing when updating RTP media
- * [#4842] Add support for linking with gold/ld --no-add-needed
- * [#4808] Make update g722 related static/dynamic payload logic
- * [#4827] Upper limit on the number of contacts to import from EDS is
- hard-coded to 500
- * [#4808] Fix put call on/off hold
- * [#4808] Implement early RTP start for incoming calls
- * [#4808] Audio stream is no longer start within RTP session.
- * [#4808] Removed coupling between audio layer and and RTP session
- * [#4702] Start audio rtp session as soon as it is created
- * [#4702] Init timestamp to 0
- * #4702: Send RTP packets immediately, no need of outgoing queue
- * [#4784] Update dbus-c++ version from gitorious
- * [#4702] Update RTP timeouts
- * [#4702] Lengthen RTP timeouts
- * [PATCH] Fixed compatibility with old libtool versions.
- * [PATCH] Accept older libebook (Maemo 5 has 1.4.2)
- * [PATCH] Fixed double-free error in preferences dialog
- * [PATCH] Fixed building of sflphone-common on Maemo5
- * [PATCH] Improved Gnome client initialization error handling. 1. It
- no longer segfaults when sflphoned isn't available. 2. User is
- provided with GUI error dialog.
- * [PATCH] Improved autogen.sh scripts 1. They do not require bash
- anymore 2. Added workaround for Debian bug #565663 3. Replaced
- manual autotools invocations with single autoreconf call 4. Non-zero
- return status on failure
- * Revert "[#4468] libtool <= 2.2 doesn't have LT_INIT macro so
- AC_PROG_LIBTOOL should be used instead."
- * Revert "[#4468] Libebook 1.4 is sufficient"
- * Revert "[#4468] Apply big path on dbus communication system"
- * [#4468] Apply big path on dbus communication system
- * [#4468] Libebook 1.4 is sufficient
- * [#4468] libtool <= 2.2 doesn't have LT_INIT macro so AC_PROG_LIBTOOL
- should be used instead.
- * [#4639] Fix determining default addressbook if this property is not
- set in gconf
- * [#4639] Fix memory leaks in Addressbook
- * [#4637] Fix opening default addressbook at sflphone init
- * [#4622] Free yaml events while parsing configuration file
- * [#4623] Fix conditional jumps based on uninitialized variable
- * [#4622] Fix leaks in yaml serialization engine
- * [#4616] Fix addressbook warnings
- * [#4514] Adjust RTP timestamp
- * #4527: Rename Karmic libyaml and Celt package in debian control file
- * #4495: Rework addressbook opening loop
- * [#4524] Increment RTP count when sending data
- * [#4524] DO NOT start RTP session twice
- * [#4367] Use PKG_CHECK_MODULE for celt
- * [#4367] Fedora package celt as celt (not libcelt)
- * [#4367] Astyling
- * [#4367] Update .po files
- * [#4367] Fix segfault in gensin
- * [#4354] Make celt a direct dependency on launchpad opensuse build
- service
- * [#4367] Make celt a required package, option --without-celt valid
- * [#4367] Fix zrtp timestamping error
- * [#4367] Fix audio zrtp timing
- * [#4367] Dispatch ZRTP packets
- * [#4367] Fix segfault when unloading account map
- * [#4367] Fix zrtp session
- * [#4367] Implement on packet receive
- * [#4367] use symetric audio rtp session, not dual
- * [#4367] Reduce packet receive/sent timeout
- * [#4367] Reduce RTP timeouts
- * [#4367] Move speaker data receive
- * [#4367] Move speaker data receive
- * [#4367] Move receive speaker data method
- * [#4367] Remove debug in rtp session
- * [#4367] Fix g722 codec clock rate
- * [#4367] Fix noise suppression initialization
- * [#4367] Fix segfault in RTP mic fadein method
- * [#4367] Refactor mic data encoding in rtp session
- * [#4367] Implement RTP main loop
- * [#4367] Fix compilation problem
- * [#4367] Fix AudioRtpclass using TRTPSessionBase
- * [#4367] Fix AudioRtpSession putDtmfEvent shadowing
- * [#4367] Fix AudioRtpSession putDtmfEvent shadowing
- * [#4367] Refactor RTP session (phase 2)
- * [#4367] Refactor RTP session (phase 1)
- * [#4367] Remove Redeclaration of SymetricAudioRtpSession in
- rtpfactory
- * [#4265] Add continue statement in for loop for invalid addressbook
- * [#4261] Makes addressbook initialization more robust
- * [#4257] Add maverick in build system
- * [#4233] Add sdp related unit tests
- * [#4233] Add condition and signal in two incoming call test
- * [#4243] Fix segfault in AudioSrtpSession
- * [#4243] Fix memory leak in AudioSrtpSession
- * [#4243] Make audio srtp optional in for incoming call
- * [#4243] Add boolean variable to make sure remote crypto context
- initialized only once
- * [#4243] Add documentation to AudioSrtpSession
- * [#4243] Use 80 bits authentication tags by default
- * [#4243] Init audio srtp remote crypto context in
- call_on_media_update
- * [#4243] Move SDP negotiastion in mod_on_rx_request
- * [#4243] Implement initLocalCryptoInfo to be called at different
- momment
- * [#4243] Init init local crypto context in when initializing audiortp
- * [#4243] Change key length according to sdes negociation
- * [#4243] Associate callid to accountid for incoming calls
- * [#4242] Fix no SDES keys in IP2IP calls
- * [#4242] Fix no SDES keys in IP2IP calls
- * [#4233] Test for call on/off hold
- * [#4233] Add two incoming call test
- * [#4233]
- * [#4233] Add 2 outgoing simultaneous call unit tests
-
- -- Julien Bonjean <julien.bonjean@savoirfairelinux.com> Fri, 30 Sep 2011 13:44:57 -0400
-
-sflphone-gnome (0.9.7~rc1~ppa1~SYSTEM) SYSTEM; urgency=low
-
- ** 0.9.7~rc1~ppa1~SYSTEM **
-
- * [#2462] Set explicitly the transport on incoming call too
- * [#2462] fix typo
- * [#2462] Use different address for SDP and call IP
- * [#2462] Use published address in SIP-SDP
- * [#2181] Fixed changelog files
- * [#2181] Updated spec file
- * [#2402] Fix pointer to int conversion warning (atoi)
- * [#2402] Remove daemon warnings, make indent
- * [#2459] Make sure the stream is opened when the call is answered
- * [#2402] Add conference related picture in documentation
- * [#2443] Not much ...
- * [#2399] Fix dialing display problem
- * [#2450] Fix incoming call already in conference crash
- * [#2399] Display peer name on the first line and peer number on the
- second
- * [#2450] Handle 403 FORBIDDEN when refused
- * [#2447] Bind offHold/onHold actions to button in gtk client
- * [#2447] Bind hangup action to button for conference
- * [#2447] Add conference action in gtk client's ToolBar
- * [#2381] Disable the password hashing in config file
- * [#2402] Cleanup
- * [#2366] Set callback to null when deleting Pulseaudio streams
- * [#1313] Fix main buffer unit test
- * [#1313] Fix audio layer unit test
- * [#2315] Hide pw in security tab, display when editing, sync with
- basic tab
- * [#1313] UnitTest change AudioRtpSession for AudioSymetricRtpSession
- instance
- * [#2402] Code cleanup
- * [#2444] Add debug to catch occasional crash when loading client's
- config
- * [#2444] Add debug info to catch occasional crash when loading config
- dialog
- * [#2402] Restore Call menu translations
- * [#2403] Use the published address if checked in GUI
- * [#2442] Add protection test in sdp
- * [#1841] Reapply pjsip patch concerning DNS SRV resolution
- * [#2384] Tags incoming call as direct SIP call, if applicable
- * [#2402] Change the monkey face
- * [#2315] Enable user to display password in clear text
- * [#2434] Force optimization level at 2
- * [#2284] Fix dbus_get_all_ip_interface compilation warnings
- * [#2431] Popup main window on incoming if applicable
- * [$2402] Fix simple warnings
- * [#2402] Fix implicit variable init order in LibraryManagerException
- * [#2402] Fixing implicit variable initialization warnings in
- AudioRtpSession
- * [#2402] Revert atoi change, fixing codec list doubled entries
- * [#2402] Fix gpointer to gint conversion
- * [#2402] Fix pointer casting to integer different size warning in
- codec list
- * [#2402] Fix warning discarting qualifiers from pointer target
- * [#2402] Fix gtk tree view assignement from incompatible type warning
- * [#1669] Fix audio recording folder utf-8 non compatibility issue
- * [#2414] Clean up debugs
- * [#2414] Use transport set in iptoip Account and update it frm
- preference
- * [#2348] Use macro N_() to mark ui.xml strings as translatable
- * [#2414] Rename getSipAddress/setSipAddress functions
- * [#2407] Fix volume controls display
- * [#2407] Fixes dialpad
- * [#2383] Set ip to ip config when clicking apply button
- * [#2404] Update call-to script - Maxime Chambreuil
- * [#2405] Client handles unknown call in current state as well
- * [#2383] Add DBUS signal to send IPtoIP local address and port as
- string
- * [#2383] Add Ip to IP config change apply call back
- * Clonflict
- * [#2402] Code cleanup
- * [#2383] Do the same for IPtoIP (init localn ip with first in the
- list)
- * [#2383] Use first interface in the list if local addresss is not
- defined
- * [#2403] Clean up unuseful addresses/ports
- * [#2403] Use the IP profile SIP port as global SIP port
- * [#2383] Fix dbus_get_all_ip_interface warnings
- * [#2383] Take into account sameAsLocal when loading published address
- * [#2383] Tsake into account sameAsLocal option when saving published
- address
- * [#2383] Update local ip address in ip to ip config
- * [#2383] Save ip 2 ip local port in config
- * [#2406] Update toolbar at startup
- * [#2284] Remove redefinition warnings + speex warnings
- * [#2383] Fix security table in account config
- * [#2383] Save ip 2 ip network interface parameters in config
- * [#2403] Restore sip transport selector
- * [#2383] Fix filling the Localt IP Address on account creation
- * [#2383] Fix Gtk-Critical when checking STUN
- * [#2383] Fix reopening account configuration display issue
- * [#2383] Load IPtoIP local address and port in preference iptoiptab
- * [#2383] Add LocalAddress and Localport in Preference IpToIp tab
- * [#2403] Use the address and port associated to the account as often
- as possible
- * [#1753] Removed pjsip generated files
- * [#1753] Removed remaining milenage lib references
- * [#2383] Add _publishedSameasLocal variable in sipaccount
- * [#2383] Add PUBLISHED_SAMEAS_LOCAL variable in config
- * [#2383] Fix stun set active or not when opening config
- * [#2181] Added RPM 64bits dbus patch
- * [#2402] Code indentation
- * [#2313] Force $(HOME).cache directory creation at startup
- * [#2383] Separate network interface and published address in account
- config
- * [#2400] Change dbus service installation path to libdir
- * [#2382] Move TLS related published address options in security tab
- * [#2382] Indent accountconfigdialog.c
- * [#2181] Install libdbus-c++ in $pkglib instead of $lib
- * [#1753] Remove ILBC code and disable it by default in the configure
- * [#1753] Remove milenage directory
- * [#2382] Fix switching interaface instabilities
- * [#2396] Save local ip in account creation wizard
- * [#2284] Remove warning on hold
- * [#2387] Fixes history searching and filtering
- * [#1215] Add samplerate display in the GUI
- * [#1663] Voicemail icon reflects voice messages
- * [#2395] Fix account registration ( specifically with callcentric)
- * [#2386] Strip "sip:" on incoming call, fixing history call back
- * [#2181] Updated spec files
- * [#1215] Display codec name in calltree instead of status bar
- * [#2390] Move back nbCalls and stopStream higher in refuseCall
- * [#2392] Fix ringtone during call in IAX
- * [#2391] Stop audio streams when there is 0 calls only
- * [#2391] Add debug when call state is not valid
- * [#2390] Clear returns in IAXvoipLink::sendAudioFromMic() method
- * [#2380] Fixing IncomingCallNotification not regular
- * [#2339] Query conference at client startup
- * [#2339] Working conference querying at startup
- * [#2339] Add conference in call tree
- * [#2339] Primitives to query conferences at client startup
- * [#2320] Add account selection in history
- * [#2355] Temporary solution: do not delete pointer when removing
- account
- * [#2380] Change algorithm in AudioRtp to trigger an
- IncomingCallNotification
- * [#2274] Comment sdebug in MainBuffer flush method
- * [#2274] Add flushMain() in ManagerImpl::addStream
- * [#2274] Add getBufferID() method in ring buffer
- * [#2274] Fix warning, comment debug in ringbuffer's flush method
- * [#2274] Use AudioLayer flushMain() and flushUrgent() in ALSA
- * [#2274] Clean up unused variable warning
- * [#2274] Protect minbudffer pointer on flushing
- * [#2274] Fix playATone method which writing empty buffer in urgent
- ringbuffer
- * [#2274] Use audio layer flushUrgent and flushMain in createStreams
- * [#2274] Use flush audio calls from audiolayer
- * [#2274] Flush when peer answered call
- * [#2375] Flush main buffer in iax when answering a call
- * [#2274] Parse displayname using c++ string method
- * [#2375] Flush main buffer when off holding calls
- * [#2375] Flush main buffer mon RTP startup
- * [#2376] Use now Pulseaudio module-cork-music-on-phone
- * Updated OSC packaging
-
- -- Julien Bonjean <julien.bonjean@savoirfairelinux.com> Fri, 20 Nov 2009 13:59:02 -0500
-
-sflphone-gnome (0.9.7~beta~ppa1~SYSTEM) SYSTEM; urgency=low
-
- ** 0.9.7~beta~ppa1~SYSTEM **
-
- * [#1933] Cleanup debug
- * [#1933] Clean up debug
- * Fix mic
- * [#1933] Set the IAx format earlier
- * [#1933] Move IAX sendAudioFromMic outside if (call) statement
- * [#1933] Fix startstream when offhold in iax and add debug concerning
- codec neg.
- * [#2371] sflphone_notify_voice_mail: minor gettext message formatting
- cleanup
- * [#2371] select_account_cb: properly gettextize status message
- * [#2371] show_account_list_config_dialog: properly gettextize status
- message
- * INSTALL: Minor tidyup of core install guide
- * Add /sflphone-gnome/src/icons/Makefile to .gitignore
- * [#2181] Updated OpenSUSE files (tmp)
- * [#1933] Add debug for codec negociation for iax
- * [#1933] Get rid of getMicAvail and getMicData in audiolayer (not
- used anymore)
- * [#1933] Add "audio codec not determined" error in IAX
- * [#1933] Test flush data
- * [#1933] Do not need to start audio stream in iax anymore
- * [#1933] Protecting pointer
- * [#2284] Remove more compilation/execution warnings
- * [#2284] Cleanup debug in client, use DEBUG instead of g_print
- * [#2284] Clean up uimanager
- * [#2370] Remove warnings
- * [#2366] Clean up other debug
- * [#2366] Clean up debug
- * [#2366] Call pa_xfree explicitely in writeToSpeaker
- * [#2284] Remove address book warnings
- * [#2365] Fixes bad cast
- * [#2352] Fix continuous ringing when peer hangup and call not yet
- answered
- * [#2181] Added version support
- * [#2181] Fixed some minor issues
- * [#2360] Moved MainBuffer from AudioLayer to ManagerImpl
- * [#2352] Makes getMainBuffer() everywhere
- * [#2352] Use 50 sec latency on pulseaudio stream creation
- * [#2352] Add alsa debug
- * [#2359] Update repository documentation
- * [#2354] Move pulseaudio disconnectAudioStream after stopping main
- loop
- * [#2352] Adjust nb byte copied in pulseaudio according to
- writeableSize
- * [#2352] Specify pulseaudio tlength parameters using pa_usec_to_bytes
- * [#2322] Convert italian translation to UTF-8
- * [#2357] Fixes window size
- * [#2357] Display only actionnable tool item
- * [#2333] Update streams parameters
- * [#2347] Use GNOME user settings for Menu and Toolbar appareance
- * [#2349] Load/Save properly audio params
- * [#2322] Update translations from Launchpad
- * [#2181] Added Francois Marier script
- * [#2350] Remove non-valid test
- * [#2181] Updated launchpad packaging
- * [#2333] Fix Pulseaudio Capture
- * [#2333] Use pulseaudio ADJUST_LATENCY flag and ALSA RT-SCHEDULING
- * [#2333] Pulseaudio Interpolate timing
- * [#2333] Change (again) Pulseaudio settings to fit logiteck usb hdw
- requirement
- * [#2333] Adjust pulseaudio fragment size to 4096 (max sflphone's
- frames per buffer)
- * [#2284] Remove recurrent compilation warning (g++ linker problem)
- * [#2333] Safer Audiostream parameters
- * [#2333] Fix alsa playback to reduce underrun
- * [#2333] Better audiostream parameters
- * [#2181] Updated version management
- * [#2333] Exclusive test in playback loop
- * [#2181] Updated build system
- * [#2333] Less underrun with these value
- * [#2333] Update playback audiostream parameters
- * [#2333] Lengthen the audio buffer reduce number of underrun in
- pulseaudio
- * [#2333] Add ALSA recovery functions for underrun (begin)
- * [#2333] Add pa_stream_trigger in pulse audio underrun callabck
- * [#2048] Reduce prebuffering in pulseaudio (which affect incomming
- calls' plbck)
- * [#2316] Do not display any icons to the right on the history tab
- * [#2333] Comment pa_stream_trigger in pulseaudio underrun
- * [#2333] Modify pulseaudio streams parameters
- * [#2318] Fix transfer tool button double signal
- * [#2181] Updated
- * [#2333] Fix ALSA ringtone
- * [#2333] Flush all main buffer before starting audio
- * [#2333] Open/Close Alsa thread between calls while there is no audio
- * [#2333] Add debug message and test condition on starting playback
- and capture
- * [#2181] Fixed gnome client makefile
- * [#2181] Updated
- * [#2308] Remove getTelephoneTone debug
- * [#2308] Change plughw for default in ALSA
- * [#2308] Oups, forgot to change function name in audiolayertest.cpp
- * [#2308] Cleanup in pulseaudio code (debug, function name)
- * [#2308] Fix pulseaudio stream closing assertion failure
- * [#2308] Moved pulseaudio mainloop locking from AudioStream
- disconnect stream
- * [2308] Fix latency at the beginning of a call, when playing DTMF and
- wehn starting tone
- * [#2181] Updated karmic
- * [#2317] [#2319] Fix address book toggle button contextual behaviour
- * [#2308] Stop stream when refusing a call
- * [#2308] Stop pulseaudio stream when peer hungup
- * [#2308] Fix tone and ringtone
- * [#2312] Display the STUN entry widget when opening the tab
- * [#2308] Implement two different callbacks for capture/playback in
- pulseaudio
- * [#2309] Open/close pulseaudio connections in startStream/stopStream
- * [2308] Leave pulseaudio stream running, do not cork/uncork them
- anymore
- * [#2295] Set gtk file chooser to None if nothing is set in
- configuration
- * [#1976] Add codec and conference documentation
- * [#2209] Fix recording in regard of resamling
- * [#2297] Update .gitignore
- * [#2297] Update translation files
- * [#2297] Add reference to our coding standards
- * [#2297] Remove old docbook code
- * [#2296] Reinit tls account settings after modification
- * [#2253] Add DcBlocker class to remove capture's dc offset
- * [#2034] Fixes for TLS transport to initialize
- * [#2284] Add silent build rule + client clean warnings
- * [#2274] Fix unserialize history items in cilent at startup
- * [#2274] Complete display name parsing and displaying
- * [#2274] Parse the Display Name in sip INVITE message
- * [#2050] Fix capture volume control in ALSA
- * [#1970] Volume controls disable when using pulseaudio
- * [#1970] Disable volume controls when using pulseaudio
- * [#2277] Fix direct ip2ip ZRTP enabling/disabling in ip2ip
- preferences
- * [#2181] Added launchpad debian files
- * [#2181] Added spec files for OSC
- * [#2274] Set display name for "Contact" sip header as the hostname
- * [#2181] Fixed daemon issues
- * [#2181] Fixed gnome client issues
- * [#1976] Remove warnings - need to fix the transfer
- * [#2006] Add init is_rec variable in ManagerImpl
- * [#2006] Update codec display on call selection
- * [#2006] Restore double click actions in history and contact calltree
- (GTK)
- * [#2176] use XDG_CACHE_HOME when initializing sfl.zid file
- * [#1976] Fix calltree switching from history
- * [#2209] (Re)Fix cache for zid
- * [#2209] Clean up debug messages
- * [#2209] Clean debug messages
- * [#2209] Fix trasnfering a call during a conference
- * [#2209] Speex decode must return the number of bytes
- * [#2209] Change frameSize speex 32kHz
- * [#2209] Fix speex codec framesize
- * [#2209] Reinit converterSamplingRate in RTP sessions
- * [#2209] Change speex ultra wide band framesize
- * [#1747] Add pixmap data
- * [#2252] Fix Receiving a server error 488 crashes the callee
- * [#2209] Fix iax low rate packate sending
- * [#2209] Clean up debug messages
- * [#2209] Add resampling changes for IAX
- * [#2209] Clean up resampling code
- * [#2209] Fix latency introduced by pulseaudio
- * [#2209] Fix initialization of mainbuffer's internal sampling rate
- * [#2176] Fix upsampling buffer size in audiolayer
- * [#2209] Add dynamic converter sampling rate in audiortp sessions
- * [#1747] Fixes runtime warnings
- * [#1747] Remove from repo
- * [#1747] register our icons to be used as stock icons
- * [#2209] Fix number of byte in alsa's write to speaker
- * [#2209] Fix putting non-resampled data in RTP's mainbuffer
- * [#2209] Add alsa resampler
- * [#2209] Add a samplerate converter in PulseLayer
- * [#2209] Add mainbuffer's internal sampling rate and flushall method
- * [#2176] Add mainbuffer stateInfo debug method
- * [#2209] Resampling is optimal using SRC_LINEAR not SRC_FASTEST
- * [#2176] Remove debug recordings
- * [#2176] Fix Holding a conference participant on new calls
- * [#2224] Add confID in callable object
- * [#2176] Fix putting onhold a call participating to a conference when
- pressing new call
- * [#2176] Reset auidio buffers when adding streams (rtp, audiolayer)
- * [#1976] Use xml to describe toolbars - Add a naviguation toolbar
- * [#2176] Remove conference default_id in joinParticipant
- * [#2176] Display error message in alsa's snd_pcm_avail_update call
- * [#2176] Alsa mic avail data debug
- * [#2176] Add some debug message for mic loss problem
- * [#2176] Flush mic ring buffer when offholding a call
- * [#2176] Reset ringbuffers' readpointer when adding main participant
- * [#2176] Fix getAvailData algorithm
- * [#2176] Reset ringbuffer's readpointer when adding a new participant
- to a conference
- * [#1744] Regex object renamed to Pattern. Previous attempt at
- providing
- * [#2176] Fix detach main participant problem when adding new one
- * [#1976] Use right domain to translate
- * [#1976] Add xml menu description
- * [#2176] Store a list of confernece participant in client
- * [#2176] Fix add participant, joinparticipant methods
- * [#2181] Do not install dbus-c++ headers + add return value
- * [#2176] Fix minor call handling instabilities
- * [#2174] Fix incoming IP call contact address
- * [#2211] Add test to protect NULL pointer
- * [#1163] Add Advanced account configuration section
- * [#2176] Add some usefull comments and debugging info
- * [#2176] Add conditions to display security icons in conference
- * [#2176] Fix detaching one participant while keeping communication to
- others
- * [#2176] Reenable userActive.svg in call tree
- * [#2176] Make user active blue (not red)
- * [#2176] Fix user active picture
- * [#2176] Fix "hidden" merge conflict in sipvoiplink
- * [#2176] Remove iax audio stream on peer hungup
- * [#2174] Multiple UDP transports functional (TESTED with 2 accounts
- and 3 calls)
- * [#2176] Fix fix audio stream binding in iax
- * [#2174] Create a default UDP transport + use tp selector for dialogs
- also
- * [#2176] Register iax audio stream in mainbuffer
- * [#2176] Fix getAudioCodecName in IAXvoipLink
- * [#2176] Fix iax account init
- * [#2176] Handle multiple account using the same sip transport
- * [#2165] Add .png files
- * [#2176] Small fixes concerning dtmf
- * [#2176] Fix make uninstall in codecs
- * [#2174] remove stund makefile generation
- * [#2176] Add conference lock
- * [#2174] Add transport selector for multiple accounts
- * [#2176] Change userActive picture from red to blue
- * [#2176] Fix security pixbuff in calltree
- * [#2176] Replace sfl.zid in .cache/sflphone instead of .sflphone
- * [#2176] Fix add call description
- * [#2176] Remove detach button from toolbar
- * [#2176] Fix calltree call description state and state code in
- conferences
- * [#2176] Fix pulse audio double free
- * [#2176] Fix conference selection
- * [#2174] Clean up - remove stun settings in client network
- configuration panel
- * [#2174] Remove voviva stun code
- * [#2174] Rsolve STUN with pjsip - DO NOT WORK
- * [#2165] Add user svg
- * [#2165] Debugging sip call failed
- * [#929] Link against uuid if installed
- * Oops
- * Fixed bugs related to libsexy (with GTK < 2.16)
- * [#929] Remove uuid-dev dependency in the core
- * [#2165] Debugging no negociated codecs at communicatio start
- * [#2165] Fix calltree bug (gtktreestore instead of gtkliststore)
- * [#2165] Fix several merge problems
- * Updated opensuse packaging script
- * [#1163] Add missing figures
- * [#1163] Update INSTALL file
- * [#2165] Fix IAX
- * [#2165] Add recordabe interface
- * [#2165] Finish recording refactoring for call (not for conference)
- * [#2165] Enable speaker recording for two different calls
- simultanously
- * [#2165] Implement call recording using the Recordable interface
- * [#2165] Add get and set to AudioLayer's audio recorder
- * [#2165] Add class recordable from which inherit call and conference
- * [#2006] Fix G722 and Speex 8khz codec conferencing
- * [#2006] add recording of audio buffers
- * [#1163] Add general settings section
- * [#1163] Fixes makefile error
- * [#2006] Fix some minor issues
- * [#2006] Drag a conference call on another conference call
- (difference conferences)
- * [#2006] Fix dragging a conference on itself
- * [#1744] Integrating some of the needed regular expression patterns
- in order
- * COmplete call features
- * [#1744] Added support for named subgroup in the Regex object. Also,
- new
- * [#1744] Adds thread safety features, compile() and setPattern()
- methods to the Regex class.
- * [#1744] Fix inconsistency in the finditer method from the last
- commit.
- * [#1744] Added regex pattern object built on top of libpcre. To be
- used
- * [#1744] Initial commit towards implementing RFC4568. Unimplemented
- in the
- * [#2157] Hide "security" and "advanced" tabs for IAX under account
- * [#1163] Add call features section
- * [#2006] Add joinConference capabilities
- * [#2006] Add dbus joinConference signal
- * [#2006] Drag a conference call onto a conference to add it
- * [#1163] Add addressbook section
- * [#2006] Drag a conference call onto a single call to create a
- conference
- * [#2006] Expand rows automatically
- * [#2006] Add minimal multiple conference handling
- * [#2006] Add atached/detached conference icons
- * [#2006] Add function processRemainingParticipant
- * [#2006] Deep refactoring, fix hangup bug
- * [#1163] Update documentation - Accounts part
- * [#1976] Integrate user doc to gnome client build system
- * [#2122] Remove double inclusion in dbus-c++/src/Makefile.am
- * Remove pjproject version number
- * [#2006] Fix peerHungup
- * [#1976] Make Yelp accessible from the GNOME client (need to install
- the sflphone.xml first)
- * [#2006] Fix multiconferencing hangup
- * [#2006] Fix hangup calls in a conference
- * [#2150] Make IAx2 reappear
- * [#2006] Fix detach participant on multiple call
- * [#2006] Can remove rining call from a conference
- * [#2006] Reinit confID when removing a participant
- * [#2006] Remove get isCurrentCAll in hangup/peerhungup (SipVoipLink)
- * [#2006] Fix refuse call
- * [#2006] Fix answerring incoming call
- * [#2006] Refactor conference's participant list
- * [#2101] Re-integrate test compilation in main build system
- * [#2101] Make the test directory compile
- * [#2136] Restore history functionality
- * [#2006] Fix binding main participant to himself
- * [#2006] Fix add current/incoming/onHold participant to an existing
- conference
- * [#2006] Fix add incoming calls to an already created conference
- * [#2006] Fix remove stream
- * [#2006] Fix detachParticipant/removeParticipant switchCall ids
- * [#2006] Fix adding a call in conference having state "CURRENT"
- * [#2006] Remove/add main participant from conferences
- * [#2006] Hold/unHold conference
- * [#2006] Detach a partcipant from drag n drop
- * [#2006] Hangup a conference
- * [#2006] Add hold/unhold conference dbus messages
- * [#2034] gtk-ui fix under the "basic" tab.
- * [#2006] Fix dragging calls on conference calls
- * [#2006] Fix detach participant from a conference
- * [#2034] Added default message is status bar under the account config
- dialog
- * [#2112] Fix a crashed caused when a non-md5 password was sent to
- pjsip.
- * [#2006] Detach participant by ID
- * [#2006] Fix addParticipant method in managerImpl to handle
- incoming/answered calls
- * [#2006] Add addParticipant method in managerimpl and related dbus
- messages
- * [#2111] Added the ability to configure zrtp on sip.sflphone.org from
- * [#2106] Fixed problem in the account assistant under gtk-ui. Also,
- assistant.c
- * [#2006] Fix dragging a conference call on another conference call
- (same conference)
- * [#1904] Small UI fix. Assistant was moved from "Call" to "Edit"
- menu.
- * [#1904] Fix a wrong label under gtk-ui.
- * [#2034] Renaming and source code splitting.
- * [#2034] Status bar added to account window to better reflect the
- registration
- * [#2006] Make calltree_remove_call recursive (for GtkTreeStore)
- * [#1110] Small gtk-UI fix in the account window (alignment).
- * [#2006] Fix remove conference, display children which are still
- active
- * [#2006] Recursive function call in calltree_update_call
- * [#2006] Add multilayered capabilities to calltree (GtkTreeStore)
- * [#2006] Implement remove conference in calltree
- * [#2034] Now useless as Direct Ip calls settings moved under
- Preferences.
- * [#2034] Edit/add buttons were set insensitive all the time under
- gtk-ui.
- * [#1887] Information about the state of the current SIP call is
- displayed
- * [#2006] Add call tree remove callback
- * [#2006] Fix create_conference function
- * [#2006] Update conference_added_cb to add new conference to the list
- * [#812] Added new tab under GTK-ui Preferences. Moving Direct Ip
- Calls from
- * [#2121] Disable temporarily test compilation
- * [#2006] Fix conferencelist to handle conference_obj_t instead of
- gchar
- * [#2006] Add conference_obj structure
- * [#2121] Update version
- * [#2006] Fix conference selection
- * [#2101] Use the new source tree to fetch the right object files
- * [#2006] Add conference in calltree
- * [#2006] Add Dbus signal conference added/removed/changed
- * [#2006] Add getConferenceDetails call on dbus
- * [#1904] Registration expire now appears as a spin box under gtk-ui.
- * [#812] Fixing a segmentation fault caused by a non-existing account
- ID
- * [#2006] Add getConfList method over dbus
- * [#2006] Add a conferencelist data structure in client-gnome
- * [#812] Defaults value are now sent if a non-existing account is
- requested
- * [#2006] Add sflphone action sflphone_join_participant
- * [#2006] Fix buffer read pointer problem deletion
- * [pjsip] Attempt at fixing via header incompatibility with
- Freeswitch.
- * [#1797] forget something
- * [#2006] Add call new state conferencing in deamon
- * [#2006] Remove addParticipant method for conference, use
- joinParticipant only
- * [#1163] Update INSTALL documentation
- * [#812] Msec/sec values were not taken into account.
- * [#1797] Make pjproject-1.4 compile
- * [#2006] Add Detach participant method
- * [#2006] Dragndrop fully functional with INCOMING and HOLD call
- * [#1797] Add pjproject-1.4
- * [#1797] Remove pjproject-1.0.3
- * [#2006] Get call state in conference related function
- * [#2006] Add joinParticipant (conference) method in ManagerImpl
- * [#2006] Add joinConference DBUS message
- * [#2006] Store the previously selected call_id on dragndrop
- * [#2006] Fix GValue pointer unref in selection callback
- * [#2006] Store dragged call_id
- * [#2006] Update drag_data_received_cb callback to manipulate CallIDs
- * [#2006] Add dragndrop signals
- * [#2006] Set calltree reordable
- * [#812] Adds the ability to create a TLS listener in case the user
- requests
- * [#812] Adds the ability to configure local/published address from
- * [#1883] Move switchCall in onHoldCall function
- * [#812] Deals with the published address/port problem when
- integrating TLS.
- * [#1883] Switch call id in managerimpl when peerHungUp
- * [#1883] Switch call id before hangup
- * [#1883] Add usefull and permanent debug info for conference
- cretion/deletion
- * [#812] Fix various segmentation faults related to Direct IP kind of
- calls.
- * [#1883] Fix deletion of std::map elements using iterators
- * [#2014] Add libzrtpcpp build dependency
- * [#1883] Still some for loop test ambiguity (while loop instead)
- * [#1883] Fix for loop initial test ambiguity (use while loop instead)
- * [#1883] We must discard data in urgent ring buffer if data is get in
- mainbuf
- * [#1883] Fix availForGet same id for ringbuffer and readpointer
- * [#812] Match "sips" as a Direct IP Call when the user enter a sip
- uri
- * [#812] Fix segmentation fault related to SIP URI creation.
- * [#812] Towards integrating multiple tls listeners at the same time.
- This
- * [#1883] Add debug messages in conference and fix mainbufferTest
- * [#812] gkt-ui fix. Private key must be fed as a filename and not as-
- is.
- * [#812] TLS integration within sipvoiplink and pjsip. Also,
- configure.ac
- * [#1883] Fix Alsa/Pulse mallocation
- * [#1883] Fix data corruption in AudioRtp's micData buffer
- * [#812] Full dbus integration for all the tls related options under
- gtk-ui.
- * [#1883] Fix memory leaks in audiortp session
- * [#1883] Fix mem leaks in audio rtp
- * [#812] Fix setAccountDetails where TLS_ENABLE was set to the value
- * [#812] Small gtk-ui fix.
- * [#811][#812] Small gtk-ui fix.
- * [#812] Introduced a mechanism for configuration files that makes
- possible
- * [#812] New dbus bindings added. Also, configuration compliance was
- enforced
- * [#1881] Remove default buffer from MainBuffer (update unit-tests)
- * [#1881] Add ring buffer read pointer tests
- * [#1883] Fix issues in ringbuffer reader pointers
- * [#2034] Implementing a new configuration dialogue for TLS transport
- settings
- * [#1883] Add some usefull debug and safety checks
- * [#2028] Notify the client with libnotify when the zrtp negotiation
- failed.
- * [#811] Harmless no to throw an exception, an makes the application
- less
- * [#2028] A minidialog is showed to the user under sflphone-client-
- gnome
- * Removed useless file.
- * Ignoring Makefile in src/widget
- * [#2027] Fix segmentation fault when showMessage callback is called
- after
- * [#2026] keyExchange was set to ZRTP instead of "1"
- * [#2024] Fix the wrong summary at the end of the assistant.
- * [#1883] Fix mnagerimpl conference map insertion
- * [#1883] Add Mutexes in MainBuffer
- * [#811] Gtk ui was not presenting the right information about zrtp
- for
- * [#2023] security icons were not installed in sflphone-gnome.
- * [#2021] Fix a mistake in the readme from sflphone-common that gives
- wrong
- * [#811] The current SRTP mode was not properly displayed for the
- IP2IP
- * [#1743] Re-implementation of the "automatically remove error dialogs
- [...]"
- * [#2017] [#2019] Fix the inability to dial a number and place a
- registered
- * [#811] Final re-integration of ZRTP support in the main branch from
- 0.9.6
- * [#1883] Fix map insertion methods
- * [#811] Combo box now is now set to the active key exchange method
- * [#811] ZRTP options now configurable back again from the Gtk UI.
- IP2IP
- * Updated hostname for git clone
- * [#1883] Add minimal functionalities to create a conference
- * [#811] re-integration of all the methods and signals on dbus.
- ManagerImpl
- * [#811] Got out of a precarious position were nothing would compile.
- * [#1976] Build documentation squeleton with docbook
- * [#1883] Add sflphone-client "addParticipant" button for conference
- * [#1994] Better organize the source directory structure. New
- subdirectories
- * [#1883] Add a simple Conference class
- * [#1882] Use static audio buffer in Pulse and ALSA layer (instead of
- malloc)
- * [#811] First commit toward re-integration and refactoring of ZRTP
- * [#1882] Flush RTP ring buffer before entering mainloop
- * [#1882] Fixed MainBuffer::UnBinCallID() in case there is no
- ringbuffer
- * [#1882] Test (and fixe) high level conference and mixing
- functionalities
- * [#1772] Apply patch to compile on fedora (sent by Marcin
- Zajączkowski <mszpak@wp.pl>)
- * [#1882] Update Bind, unBind call_id in MainBuffer
- * [#1959] This adds the ability to store password as an MD5 Hash in
- the
- * [#1538] Fixes rules compilation
- * [#1930][#1931] Fixed a mistake (again) related to index and
- credential count
- * [#1753] Remove ILBC from pjproject - Hacks in pjsip
- * [#1930][#1931] Credential was not selected properly using realm
- * [#1882] Finilize multiple reading pointer in RingBuffer
- * [#1538] Remove configure from autogen.sh to respect debian upstream
- authors policy
- * [#1773] Remove generated files from repo
- * [#1791] Use XDG_CACHE_HOME to save pid file
- * [#1791] Fixes path to save history
- * [#1791] Fix debian installation scripts
- * [#1930][#1931] Settings are now taken into account in the server.
- * [#1882] Add ringbuffer default ring buffer pointer in methods
- involving mStart
- * [#1882] Add default ringbuffer pointer
- * [#1882] Add RingBuffer multiple read pointer basic functionnalities
- * [#1882] Fix MainBuffer flushData unit test
- * [#1930][#1931] Ability to save and retreive the configuration from
- * [#1882] Added Multiple CallID mapping to MainBuffer
- * [#1791] Not much
- * [#1791] If XDG env variables are not null but empty, use default
- ones
- * [#1791] Make XDG_CONFIG_HOME writable
- * [#1930][#1931] Partial commit. Not working yet. Cannot delete
- account
- * [#1881] Fixed alsa capture latency problem
- * [#1881] Fixed Alsa capture temporarily
- * [#1930] [#1931] Partial unbroken commit providing the ability to
- * [#1881] MainBuffer implemented in AudioLayer/AudioRTP
- * [#1881] Add discard and flush unit-tests
- * [#1881] Add discard and flush functionnalites to MainRingBuffer
- * [#1881] Add availForGet in MainBuffer
- * [#1881] Add availForPut function to MainBuffer
- * [#1880] Remove AudioRTP* pointer from SipVoIP (reapered while
- merging master)
- * [#1881] Add a map between call id and coresponding ring buffer
- * [#1855] Refresh pot file and upload on Launchpad
- * [#1881] MainBuffe now robust to false ids on getData and putData
- * [#1881] Fix big big big memory leak
- * [#1881] Add getData and putData to mainBuffer
- * [#1881] Unit-test basic ring buffer functionnaities
- * [#1881] Add class MainBuffer and basic buffer creation unit-tests
- * [#1880] Fix call transfer (step2) issues
- * [#1880] Moved AudioRtp* pointer from SIPVoIPLink to SIPCall class
- * [#1791] Add postinst script to keep user data when migrating
- config/history file
- * [#1797] Make pjsip compile
- * [#1777] Code indentation
- * [#1791] Use XDG_DATA_HOME and XDG_CONFIG_HOME for sflphonedrc and
- history + unit tests
- * [#1746] Useless space does not appear anymore when volume sliders
- and
- * [#1643] GtkCheckMenuItem is used instead of icons for elements in
- the
- * [#1110] [#1668] STUN parameters are now located in the preferences,
- under
-
- -- Julien Bonjean <julien.bonjean@savoirfairelinux.com> Fri, 06 Nov 2009 11:20:01 -0500
-
-sflphone-gnome (0.9.6-SYSTEM) SYSTEM; urgency=low
-
- ** 0.9.6 **
-
- * Documentation on echo test
- * [redmine_down] codec names not displayed in total
- * [redmine_down] crash when hanging up a dialing call because tries to
- add it to history whereas no starttime
- * [#1927] alternate every time screen changed to call history
- * [#1886] clean code
- * [#1886] debug messages when loading history removed
- * [redmine_down] sflphone-kde icons
- * [#1855] Update language files
- * [#1502] Update version number
- * [redmine_down] setHistory at close
- * [#redmine_down] Handle PJ_DECLINE_SC as failure
- * [#1923] Fix segmentation fault when adding a new account
- * [#1923] Check on iterator before setting the config
- * [#1904] Added mnemonic to tabs in sflphone-gnome.
- * [#1905] The daemon was not sending the currentSelectedCodec signal
- on dbus when answering a call.
- * [#1922] Default values set to all account details
- * [#1886] Spinbox reg expire enables apply, and address book is not
- visible when disabled
- * [#1905] Bug fix for segmentation fault caused by an empty string,
- * [#1910] Warnings in test directory
- * [#1919] Error fixed
- * [#1855] Update russian translation - Hussein Abdallah
- * [#1910] Remove files
- * [#1919] fixed
- * [#1777] Code indentation
- * [#1918] fixed
- * [#1917] fixed
- * [#1910] Remove warnings compilation in src
- * [#1886] removed AccountListModel in configskeleton
- * [#1914]
- * [#1911] check previous and new port
- * [#1910] Remove compilation warnings in src/dbus and src/history
- * [#1910] Remove compilation warnings in src/audio
- * [1855] Update german translation - Sven Werlen
- * [#1909] removed
- * [#1906] Done
- * [#1904] The registration expire value is now configurable from the
- * Cleaned up debug messages.
- * [#1886] separated initCallItem in two functions
- * [#1886] reversed error in commit
- * [#1886] clean debug
- * [#1886] changed Name of classes and files
- * [#1886] clean
- * [#1870] In call_state_cb (dbus.c:126), _time_stop was overridden by
- the actual time.
- * [#1884] Added some new gpg flags to prevent tty warnings
- * [#1886] Clean audio config dialog
- * [#1886] No more compile warnings. + 1 comm
- * [#1872] Check if the user input is smaller than PJ_MAX_HOSTNAME.
- * [#1886]
- * [#1785] Fixed build when no new commit
- * [#1852] If chosen by the user, the hostname can now be solved and
- used
- * [#1871] * and # inverted back
- * [#1869] Conditional compilation that checks if
- * [#1309] removed test in main
- * [#1425] Put actions in SFLPhone window class instead of ui view,
- made a separate toolbar for screens.
-
- -- SFLphone Automatic Build System <team@sflphone.org> Mon, 27 Jul 2009 09:53:19 -0400
-
-sflphone-gnome (0.9.6~rc2-SYSTEM) SYSTEM; urgency=low
-
- ** 0.9.6~rc2 **
-
- * [#1755] Remove generated file
- * [#1753] restore ilbc ...
- * [#1866] Methods getSipPort and setSipPort now have an effect on the
- * [#1753] make pjsip compile without ilbc. Use ./autogen.sh --disable-
- ilbc-codec
- * [#1855] Fix error in russian translation
- * [#1805] Remove the old flawed signal mechanism which was failing in
- * [#1855] Refresh translation
- * Spanish translation finished + po README files updated + echo's in
- copy-in-clients
- * [#1850] Yun made the chinese HK-CN translation
- * [#1848] Fix transfer interface bug
- * [#1862] At install, kde client installs only french translation file
- * [#1841] A new fallback mechanism was added to the internal resolver
- in PJSIP.
- * Started AccountList model/view
- * [#1855] Remove po subdir in Makefile.am
- * [#1855] Fix typo error in sflphone-gnome
- * [#1855] Do not generate Makefile in sflphone-common/po
- * [#1855] Copy translation files into both clients dirs
- * [#1855] Remove po dir from sflphone-common
- * Comments added
- * [#1860] mailbox->voicemail...
- * make scripts executable
- * [#1855] French translation
- * [#1855] Chinese zh_HK partially filled...
- * [#1859] An unnamed pipe monitored by poll() was added. When we want
- to
- * [#1855] Sven completed the first part of the german translation
- * [#1855] Cantonese manually filled for already translated, almost
- equal strings
- * [#1855] Merge russian translation
- * [#1855] Spanish manually filled for already translated, almost equal
- strings
- * [#1855] Update german translation in ./lang/de
- * [#1858] This problem was fixed by removing a useless line in
- * [#1855] merged existing translations in lang/ sflphone.po's
- * [#1842] [#1843] An attempt at improving the expected behaviour that
- can't
- * [#1855] added po folder in gnome client and scripts for copying from
- common lang folder to clients
- * [#1853] Edit before call does nothing on call history
- * Put most language entries possible in common. From 300 to 250
- entries. Stays underscores problem. Scripts for copy in clients.
- * commit to merge master
- * [#1825] Changed "Bad authentification" to "Authentication Failed".
- * common po files
- * [#1753] Remove ILBC from pjproject
-
- -- SFLphone Automatic Build System <team@sflphone.org> Fri, 17 Jul 2009 19:12:58 -0400
-
-sflphone-gnome (0.9.6~rc1-SYSTEM) SYSTEM; urgency=low
-
- ** 0.9.6~rc1 **
-
- * Update some version number
- * [#1792] Creates .sflphone directory with permission 600. Also,
- "chmod 600" after
- * [#1810] GUI is now notified that the call failed. Also, a segfault
- was
- * [#1816] Address book search disabled when disabled address book and
- enabled it back plus button stays triggered
- * codeclistmodel + asynchronous loading of address book +
- enable/disable address book
- * [#1810] Now checking SDP answer after 200 OK. Still need to
- implement full
- * [#1794] Can't use the interface during a call
- * Updated translation files
- * Russian translation integrated
- * Codec list model/view started.
- * [#1807] Add configure.ac in pjproject-1.0.3
- * [#1787] closeRtpSession added in some places where it should have
- been
- * Use Item class for contacts and accounts
- * Comments + clean code
- * [#1794] Improved debug messages
- * [#1805] Replaced the old and unreliable mecanism that was was
- waiting for
- * [#1794] Can't use the interface during a call
- * [#1787] For those cases where no registered SIP account is
- configured
- * [#1797] Make pjsip compile
- * [#1787] Minor changes. Removed useless commented line. Changed order
- of
- * [#1777] Code indentation
- * [#1797] Update package generation with new pjsip version
- * [#1798] Does not hang up when the call is building up
- * [#1797] Update .gitignore with new pjsip version
- * [#1797] Remove generated files from repo
- * [#1797] Main build system now uses pjproject-1.0.3
- * [#1797] Add pjproject-1.0.3
- * [#1797] Remove pjproject-1.0.2
- * [#1796] Computing time optimization (samplerate conversion)
- * [#1787] _audiortp->start() moved away from offhold(),
- SIPCallAnswered()
- * [#1312] Added new states for calls initialized by other clients
- * [#1795] Crashes when adding a new account, checking it and applying
- * [#1782] Missing icons
- * [#1793] KDE client compilation problem
- * Fake ringtone files can no longer be set.
- * indentation
- * [#1312] Able to fetch to differentiate incoming/ringing call state
- * [#1784] Use DESTDIR variable in po Makefile - fix language file
- installation
- * [#1785] Fixed typo
- * [#1785] Fixed changelog update
- * [#1759] ./autogen.sh --prefix=/usr --with-debug to use optimization
- level 0
- * [#1773] Changed snapshot naming convention
- * [#1773] Removed gpg agent use, added repository cache cleaning
- * [#1759] Use optimization level 0 for repository, 2 for packages
- * [#1777] Code indentation/formatting
- * Translated new features in french
- * [#1785] Added missing changelog entry
- * [#1781] Window title is SFLPhone
- * [#1777] Add code indentation/formatting in the buil system
- * [#1774] Can't set voicemail number in KDE account creation wizard
- * [#1775] Can't modify account information for account created with
- the wizard
- * [#1771] Add a "Default" button in context menu to disable chosen
- prior account
- * [#1705]
- * [#1224] Remove generated file from the repo
- * [#1224] Remove generated file from the repo
- * [#1762] distclean target should remove kconfig generated files
- (settings.h, settings.cpp). Rename them?
- * [#1761] clear history button should really clear history
- * Dialpad works.
- * Implemented Dialpad widget instead of building it in main view.
- * Removed last occurence of the old config dialog, that made the build
- crash.
- * [#1755] Do not consider G722 as a dynamic payload elsewhere than in
- RTP layer
- * [#1753] Remove ilbc Makefile generation
- * [#1756] Implement a kde configuration dialog with kconfig xt and
- kconfigdialog class
- * [#1755] fix audiocodec folder parsing problem
- * [#1450] Reinit timestamp comparison in RTP, create session in
- newOutgoingCall
- * [#1753] Remove milenage third party code from pjsip
- * New Config Dialog integrated in GUI.(without codecs)
- * [#1753] Remove ILBC codec
- * kconfig started, tr2i18n -> i18n, icons folder, accountList changed
- * [#1705] Fixed Audio RTP thread creation/start
- * [#1714] Fix codec negociation result handling
- * [#1678] Fix audiortp payload setting
- * [#1678] Put bac putData method in rtp
- * [#1669] gtk_file_chooser_get_filename() support UTF-8 by default
- * [#1735] Add conditions to sdp update call if call declined
- * [#1737] substr of recordings destination folder to remove "file://"
- should be done in client rather than in daemon
- * [#1731] Enlarge audio stream buffer size
- * [#1714] Missing true
- * [#1317] Fixed Mandriva timeout
- * [#1317] Changed tag convention
- * [#1317] Cleaned git-dch
-
- -- SFLphone Automatic Build System <team@sflphone.org> Fri, 10 Jul 2009 15:50:26 -0400
-
-sflphone-gnome (0.9.6~beta-SYSTEM) SYSTEM; urgency=low
-
- ** 0.9.6~beta **
-
- * spec files for mandriva and opensuse updated with buildrequires
- libqt4-dev >=4.3
- * [#1700] Cannot build on ubuntu 8.10 and a few other distribs
- * [#1502] Update version number where applicable
- * [#1642] Update client icons
- * [#1450] Clean up useless debug and comments in sipvoiplink and
- audiortp
- * [#1450] Remove Semaphore object in AudioRtp thread deletion
- * [#1450] Audio RTP init now synchronized with Sip/SDP
- * [#1693] kde client crashes when changing codecs order/activation
- * [#1450] Deep refactoring of audiortp
- * [#1450] setRtpSessionRemoteIp
- * [#1689] getCallList at start
- * [#1224] Change path in package files
- * [#1450] Audio RTP initialized only once, payload and remote ip set
- at runtime
- * [#1450] Add setRtpSessionMedia and setRtpSessionRemoteIp address
- * [#1642] Make GNOME GUI fresher and younger ;)
- * [#1686] Status bar displaying used account
- * added sflphone-kde icon so that it compiles
- * [#1659] Ending a call causes the daemon to crash
- * corrected introspection XMLs, po files...
- * [#1211] g722 media descriptor in codecDescriptor
- * [#1310] Install sflphoned in $(prefix)/lib/sflphone
- * [#1502] Do not install test binaries and dbus utilitaries
- * [#1224] hack for pjsip build system!
- * [#1224] Remove pjsip binaries from repo
- * [#1224] Upgrade to pjsip 1.0.2
- * [#1658] About SFLphone (bugs)
- * [#1658] About SFLphone
- * [#1660] Displaying all dialed numbers in a call
- * Tested status bar.
- * [#790] Optimize pulse audio streams parameters
- * [#1678] Some usefull debug messages for mutex/semaphore deadlock
- problem
- * [#1669] Add/remove some usefull/unusefull debug
- * [#1665] Fix latency related to pulse audio stream openning/closing
- * [#1457] Make the menus and panels accessible in french
- * [#1457] Improve broken keyboard accessibility in menus and conf
- panels
- * [#961] Instanciate only once the searchbar icons
- * [#961] Restore transfer fonction
- * [#961] Filter on the history type OK
- * [#961] Fix compilation problems on hardy/intrepid
- * [#1157] Commit missing files
- * [#790] Reduce number of start/stop streams call on pulse audio
- * [#1639] kde client crashes when no account registered
- * [#1620] Fix the searchbar
- * [#1620] Get back caltree as it was during gtkcritical area
- * [#1620] Add history filter reinit function
- * [#1335] Add a missing label in address book preferences
- * [#1561] Update russian translation - Hussein Abdallah
- * [#1605] Fix edit menu french translation
- * [#961] Enable to search in the history according to the call type
- * [#1449] Searchbar does not work anymore
- * [#961] Add popup menu on the entry primary icon for history
- * [#1317] Fixed KDE client package dependency
- * [#936] speex 32 khz integration completed
- * [#936] Use 320 frame size
- * [#936] Test using a frame size at 320 smpls
- * [#1214] Enable / Disable history
- * [#1607] Fix compilation problem for ubuntu 8.10 (libsexy)
- * [#1313] Implement processDataEncode processDataDecode in audiortp
- * [#1613] codec list order can't be set
- * Better handling of localisation + added languages + corrected
- warnings + begginning of new config dialog with kconfig + 14px
- account leds
- * [#1214] Save and load history according to the limit timestamp +
- unit tests
- * [1609] Fix call number copy/paste feature
- * [1607] Restore clear action icon in searchbar
- * [#936] Try to decode using 1280 samples
- * [#936] Add some debug
- * [#936] Add .cpp file
- * [#936] Oops Forgot speex 32 khz
- * [#1214] Add configuration panel for history + D-Bus calls
- * [#1313] Test rtp thread function, frame size, nbbytes, resampling
- * [#790] Flush audio data before closing audio streams
- * [#1214] History displays local time
- * [#1214] Skip empty field on display
- * [#1214] Associate an account to an history entry
- * [#1342] Get addressbook options sensitive/non-sensitive
- * [#1211] Clean up and comments
- * [#1211] Get back to 20 ms framesize
- * [#1211] Use sendImmediate instead of putData in RTP
- * [#1211] Fix nb byte available in RTP
- * [#1211] Clear condition on maxNbSamples in RTP
- * [#1211] Fix max byte available in RTP session
- * [#1211] G722: Use 160 samples per frame instead of 320
- * [#1211] Test using a dynamic payload
- * [#1211] Test using a dynamic payload type
- * [#1211] Rename size variable (nb_samples, nb_bytes)
- * [#1211] Test g722 ip-to-ip sending twice the data lenth
- * [#1211] Test g722 ip-to-ip
- * [#1214] Do not select an history item by default at startup
- * [#1214] Remove some compilation warnings
- * [#1214] Handle empty field - remove g_print
- * [#1214] Add each history item only once
- * [#1214] Handle call timestamps properlier
- * [#1214] Do not need timestamp files anymore
- * [#1214] Use the saved date for history entry
- * Clean up
- * [#1214] Client doesn't crash if the D-Bus call fails
- * [#1214] Client is able to save its history - still some glitches
- * [#1211] Forgot 16000 for g722
- * [#1211] G722 initialization
- * [#1214] Save name/number, successfully load the history if no fields
- are empty
- * [#1499] Fixed destination directory bug
- * [#1214] Restore all the functionalities; peer name/number way more
- easy to handle !!
- * [#1214] Add callable_object instead of call_t, refactoring
- * [#1211] Test with polycom soundstation 16000
- * [#1211] Remove C like inline function in g722 codec
- * [#1342] Finalize gnome client preference window formating
- * [#1214] Retrieve the history when the gnome client startsup
- * [#1306] Implement localization for KDE client
- * [#1593] enable accounts apply button when account checked/unchecked
- * [#1214] Implement the dbus calls on server side
- * [#1214] Add serialized/unserialized functions to pass data on DBUS
- * [#1342] Formating gnome client configuration windows
- * [#1214] Save sucessfully a map of history items
- * [#1499] Removed multiple jobs compilation for KDE client (2)
- * [#1214] Load history from file into memory, add unit tests
- * [#1534] Throws a length_error exception in case URL exceeds
- std::string max_size
- * [#1499] Removed multiple jobs compilation for KDE client
- * [#1565] make account leds smaller
- * [1430] Fix dbus debug
- * [#1562] crashes when trying to change item of a call of state "OVER"
- * [#1116] Fix compilation bug
- * [#1317] Added mandriva and opensuse-11 64 bits
- * [#1108] Add messges in main window concerning transfer success
- failure
- * [#1116] Fix compilation problems
- * [#1211] g722 Makefile
- * [#1108] Client side transferFailed/trasferSucceded signals handling
- * [#1211] G722 mostly completed,
- * [#1555] make bigger toolbar (24x24)
- * [#1551] remove default mailbox number in wizard and disable mailbox
- button when first account doesn't have mailbox number
- * [#1342] Re-add sflphone manpages
- * [#1116] Fix compilation on non-jaunty distros
- * [#1317] Fixed opensuse startup sleep
- * [#1108] Add a signal in the client to notify successful or failed
- transfer
- * [#1108] Dbus signals concerning call transfer success/failure
- * [#1317] Added opensuse to automatic build system
- * [#1223] Fix manpages bug
- * [#1060] german translation glitch
- * Clean up some gnome client warnings
- * [#1547] replace ugly account leds by beautiful icons
- * [#1548] add close button that hides windowand just hide on clicking
- the cross
- * [#1549] put introspec XMLs in the client's source
- * [#1312] Implement getCallList D-BUS method
- * [#1116] Clear text in history and contacts
- * [#1499] KDE integration
- * [#1469] Modify header linkers in dbus-c++'s Makefile.am's
- * [#1469] Remove examples folder from dbus-c++
- * [#1214] History integration in build system; unit test squeleton
- * [#1317] Cleaning
- * [#1469] Remove configure stuff in dbus-c++
- * [#1469] Add unofficial mainline dbus-c++
- * [#1469] Remove dbus-c++ from freedesktop
- * [#1430] Bring account changed signal/callback back to normal
- * [#1060] Update german translation - Sven Werlen
- * [#1430] Add marshaller one string define
- * [#1430] Send account change signal broadcast using account id
- * [#1430] Remove condition on setRegistrationState, cause stun to
- crash
- * [#1317] Centralized version handling
- * [#1317] Fixed version number on sfl-git-dch
- * [#1317] Refactoring for new distributions
- * [#1215] Fix account order at startup if latency
- * [#1088] Restore sip dns srv
- * [#1214] Add squeleton for history manager
- * [#1430] Add accout id to accout changed method
- * [#1430] No connectionStatusNotification (account changed) if no
- changes
- * [#1538] Add COPYING file
- * [#1430] Add audio rtp thread tests
- * [#1317] Changed version detection
- * [#1538] Document license in libs/stund
- * [#1317] Added version files
- * [#1538] Apply François patches - debian packages
- * [#1317] Updated spec files
- * add files
- * [#1538] Apply François patches - debian packages
- * [#1535] Change program file structure (directory src...)
- * [#1317] Updated build system scripts
- * [#1317] Cleaning
- * [#1317] Copied introspect files to gnome client
- * [#1317] Added opensuse to build-system : first-shot
- * [#1317] Remove spec files from configure
- * [#1317] Added missing prefix
- * removed debug for daemon account fix
- * [#1430] Add a connection reference which most likely belong to
- libdbus
- * [#1430] Use shared connection instead of private
- * make daemon find the account, added userMatch
- * Clean code, add comments...
- * [#1317] Fixed packaging rules
- * [#1317] Updated autogen
- * Updated autogen.sh for pjsip
- * [#1526] Set accounts order
- * [#1317] Fixed pjsip lib dirs
- * [#1317] Updated debian packaging for new pjsip configuration script
- * [#1317] Switch to autogenerated guess and sub files
- * [#1317] Updated pjsip inclusion in build system
- * [#1317] Replaced pjsip guess and sub files
- * [#1317] Fixed compilation issues on opensuse 11
- * [#1505] account list seem to crash the application when clicking
- Apply very fast...
- * [#1456] Add a flag to be replaced in the control files
- * [#1456] Added version dependancy handling
- * put account alias in AccountWidgetItem rather than in the item with
- " " before.
- * [#1034] The KDE client should start sflphoned if it is not started
- * [#1500] Handle options for notifications and display on incoming
- call.
- * [#1443] Client should not crash when receive an unexpected
- stateChanged signal
- * [#1403] Do not stop the notification anymore
- * [#1456] Added version dependancy handling
- * [#1426] Daemon crashes when get alsa plugin
- * [#1422] Improved error messages
- * commit for merge
- * [#1424] Change logo in tray icon and put a different one when
- incoming call
- * [#1425] first part done, window title...
- * [#1413] add manpages creating and installing in build system
- * [#1417] The client should start the account creation wizard if
- started for the first time (if config file doesn't exist)
- * [#1421] Make volume bars horizontal when dialpad is hidden.
- * Changed main window title and fixed a mistake in sflphone_const.h
- * [#1412] make debian package building work
- * changelog changed.
- * Changed addAccount method in gnome client.
- * Debian and man folders added.
- * [#1388] Change project name from sflphone_kde to sflphone-client-kde
- * Better handle of kabc check.
- * [#1351] Automatic generation of dbus interfaces in makefile
- generated by cmake
- * [#1307] Implement "edit before call" in history and address book.
- * [#1344] change action_call label in call history from "call" to
- "call back".
- * [#1308] Implement Hook feature in kde client
- * Improved build system.
- * #1219 : Add address book configuration page
- * Better handling of registration to the daemon.
- * #1039 : Add tray icon in kde.
- * Issue no 1216 : Double click on item in history or address book
- causes call.
- * display peer name in call list and call history when called from
- address book.
- * Address book functionnal with photo displayed.
- * Help menu kde available but actions disappeared. All fonctions in
- view.
- * Address book functionnal but ugly and making its own sort in the
- complete address book.
- * Account choice on right click, clean out includes, page address
- book, fixed bugs...
- * Wizard, double click, context menu...
- * Removed sflphone_kde.kdevelop.filelist
- * Added account creation wizard and translated interface in english.
- * Transfer functionnal but ugly.
- * transfer not functionnal
- * Bug fixed : unholding (UNHOLD_CURRENT, UNHOLD_RECORD)
- * Commit functional for push. With install.sh
- * Before merge.
- * Problem with enable accounts. Account display increased.
- * Functional with codec order working , playDTMF.
- * Commit functional.
- * sflphone_kde/build added in .gitignore.
- * complete commit for checkout previous.
- * Commit before checkout previous version to check the display
- bug(little font everywhere...)
- * Functionnal client. Rest : history icons, config icons and
- functionalities
- * commit before merge asavard for isRecording.
- * Call and Automate fusion done and seems to work.
- * Commiting before putting Automate class in Call class.
- * Functionnal main window without recording, history, voicemail, kio
- widgets.
- * client kde avec kdevelop.
- * Config Dialog almost finished.
- * Base of QT client
-
- -- SFLphone Automatic Build System <team@sflphone.org> Tue, 23 Jun 2009 11:13:42 -0400
-
-sflphone-gnome (0.9.5-SYSTEM) SYSTEM; urgency=low
-
- ** 0.9.5 release **
-
- * [#1060] FIx bug in chinese translation
- * [#1313] git add rtpTest.cpp rtpTest.h
- * [#1313] Add init/close rtp tests
- * [#1313] Basic instanciation of the rtp layer
- * [#1449] Gtk-Critical concerning history filters and new calls
- * [#1400] Make the match with the hostname instead of username
- * [#1324] Change status bar label for "Using %s (%s)"
- * [#1403] Icon size: 60x60 px
- * [#1403] Do not remove notification, improve icon quality
- * [#1403] Add smaller icon for gnome notifications
- * [#1403] Prevent crash when hangup && no notification
- * [#1403] Remove all actions on notifications; code refactoring
- * [#1451] Use stun.sflphone.org as default STUN server
- * [#1060] New po files - need to be translated
- * [#1060] Update french translation - Rebuild template file
- * [#1456] Add a flag to be replaced in the control files
- * [#1454] Make cppunit optional; remove from build deps in control
- files
- * [#1401] Add libexpat1-dev dependency in control files
- * [#1448] Take off these ugly debug messages
- * [#1448] fixed getTelephoneTone and getTelephoneFile() called
- repeatedly
- * [#1406] add liblog4c-dev in build-depends
- * [#1409] Restore .desktop icon
-
- -- SFLphone Automatic Build System <team@sflphone.org> Mon, 25 May 2009 11:34:48 -0400
-
-sflphone-gnome (0.9.5-SYSTEM~rc2) SYSTEM; urgency=low
-
- ** 0.9.5 rc2 **
-
- * [#1422] Improved error message
- * [#1402] Fix pjsip build
- * [#1404] Clear GTK-Critical Bug at client startup
- * [#1422] Added automatic VM shutdown when building on more than one
- VM
- * [#1422] Fixed some issues with new changelog generation script
- * [#1422] Moved distribution update to specific file
- * [#1422] Dropped git-dch, replace by home made implementation
- * [#1402] Fix pjsip build
- * [#1404] Clear GTK-Critical Bug at client startup
- * Changes for name based dbus connection
- * Clean changelogs
- * [#1343] Gnome: Implement a callback system to handle focus on
- different widgets
- * Debus Session
- * Refactoring Python code, PEP8
- * [#1430] Get back dbus_g_proxy_new_for_name
- * [#1430] Get back DBUS_BUS_SESSION type
- * [#1430] Dbus fixed owner message binding
- * Second test with DBUS owner
- * [#1404] Gnome -> Preferences -> Hooks
- * [#1404] Gnome -> Preferences -> Recordings
- * [#1404] Call History
- * [#1404] Gnome -> Preferences -> Address Book
- * [#1404] IF the first notification option disable the second
- notification
- * Dbus with fixed owner does not automatically start the deamon
- * Add codec debug tests in pysflphone
- * [#1407] Some print info
- * [#1407] Add a scenario to pick_up action
- * Test client dbus connection to a fixed owner
- * Add python dbus test suite
- * [#1161] Modified version handling in build system
- * [#1314] Test pulse audio and audio streams connect and disconnect
- * [#1402] Add info message after configure
- * [#1402] Build the daemon with the local pjsip library (vs the
- installed one)
- * [#1009] Fix Codec Sampling Rate set to zeros
- * [#1314] Add mutex to pulse layer audio streams
- * [#1314] Refactoring pulseaudio stream to test connect disconnect
- * [#1314] Refactoring of pulselayer to test conect/disconnect
- * Add debug messages in debus calls concerning account
- * [#1314] Add some return values to audio init functions
- * [#1406] add liblog4c-dev in build-depends
- * [#1409] Restore .desktop icon
- * Bug #1405: Fix strings as requested.
- * Bug #1404: Fix strings in preferences panel.
-
- -- SFLphone Automatic Build System <team@sflphone.org> Tue, 19 May 2009 12:08:18 -0400
-
-sflphone-gnome (0.9.5-0ubuntu1~rc1) SYSTEM; urgency=low
-
- [ SFLphone Project ]
- * [#1262] Updated changelogs for version 0.9.5-0ubuntu1 Snapshot 2009-
- 05-05
-
- [ Emmanuel Milou ]
- * Add some python CLI client code; not really functional
- * [#1108] Fix peerHungup method for IP to IP call
-
- [ Alexandre Savard ]
- * [#1108] Correct setting of SIP contact for direct IP call
- * [#1108] SIP user agent handles incoming REFER
-
- [ Emmanuel Milou ]
- * Remove website from repository
- * Update translation
-
- [ Alexandre Savard ]
- * Sflphone icon's tooltip changed for "configured" instead of
- "registered"
-
- [ Emmanuel Milou ]
- * Update translation
-
- [ Sflphone Project ]
-
- -- Sflphone Project <sflphone@mtl.savoirfairelinux.net> Tue, 05 May 2009 19:16:13 -0400
-
-sflphone-gnome (0.9.5-0ubuntu1~beta) SYSTEM; urgency=low
-
- [ Julien Bonjean ]
- * Updated Eclipse stuff
- * Improved addressbook config window
- * Added sflphone Eclipse stuff
- * Implemented addressbook list server side
- * Moved dbus stuff in dbus directory
- * Updated addressbook configuration
-
- [ Emmanuel Milou ]
- * Remove unuseful installation scripts. Use apt-get build-dep sflphone
- instead
- * fix bug #1090
-
- [ Alexandre Savard ]
- * defining speex 16khz
-
- [ Emmanuel Milou ]
- * Remove unuseful file from build system
- * Start dns srv resolver
-
- [ Alexandre Savard ]
- * Basic ogg/vorbis initialization
-
- [ Emmanuel Milou ]
- * Handle incoming IP-to-IP invite correctly
-
- [ Alexandre Savard ]
- * speex wideband 16000
-
- [ Emmanuel Milou ]
- * Better handling of incoming IP to IP call
- * DNS SRV resolution functional
- * Implement IAX2 incoming URL
- * Allow user to make IP call without any accounts configured
- * Add a contextual menu to edit a number from the contacts tab
- * Add comments, tooltip and new button to the contextual menu
- * add delete event, migrate to GTK 2.16 for sexy icons
- * Resolve ticket #1118
- * Update suse spec file
- * Add phone number cleanup functions, unit tests and panel
- configuration
- * Add pertinent test that fails
- * fix dependencies for suse package
- * Add contextual edit menu in history - #1120
-
- [ Alexandre Savard ]
- * Temporary comit: make speex wideband (16 khz)
- * Temporary: shared object for speex narrow band
- * Temporary: speex narrowband and wideband coexist
-
- [ Julien Bonjean ]
- * Fixed bug when no book selected
- * Fixed addressbook related compilation warnings
- * Fixed GTK client remaining compilation warnings
- * Fixed segfault when book removed since last sflphone run
- * Fixed bug when book is unreachable (ldap error)
-
- [ Alexandre Savard ]
- * Fix codec list in audio config window
- * Active/inactive speex codec by payload
-
- [ Julien Bonjean ]
- * Updated gitignore
- * Added some comments
-
- [ Emmanuel Milou ]
- * Add callto: handler script for browsers and al.
- * Integrate test compilation in the daemon build-system
-
- [ Julien Bonjean ]
- * Fixed g_object_unref warning for pixbuf
- * Cleaned too verbose output
- * Fixed toolbar update warning
- * Added support for asynchornous books open (first shot)
-
- [ Emmanuel Milou ]
- * Add a DBus call to fetch the call details from a call ID - Ticket
- #928
-
- [ Julien Bonjean ]
- * Improved async open books
- * Fixed bug #1139
-
- [ Emmanuel Milou ]
- * Add a way to save account order
- * commit missing files
-
- [ Julien Bonjean ]
- * Introduced log4c (ticket #1162)
-
- [ Emmanuel Milou ]
- * Load/save account order functionnal - ticket #813
-
- [ Alexandre Savard ]
- * Add CELT codec (#1143)
- * Make celt frame size 256 (*1143)
-
- [ Julien Bonjean ]
- * Switched everything to log4c (ticket #1162)
- * Updated eclipse settings
-
- [ Emmanuel Milou ]
- * Restore adding account - ticket #1172
- * Add liblog4c dependecy - ticket #1179
-
- [ Alexandre Savard ]
- * Double maxAvailByte for frame size in rtp (#1143)
-
- [ Emmanuel Milou ]
- * Add User-Agent SIP header - Ticket #1173
-
- [ Julien Bonjean ]
- * Fixed autoresize issue (#708)
-
- [ Emmanuel Milou ]
- * Remove libcppuint dependency for the debian packages
- * Look for libsexy only if gtk version < 2.16 - Ticket #1116
- * Remove libsexy dependency for jaunty. ticket #1116
-
- [ Julien Bonjean ]
- * Introduced unit tests (#1146)
- * Updated gitignore
- * Fixed Makefile (#1146)
-
- [ Emmanuel Milou ]
- * [TICKET #1112] Add a test on the voice buffer to send through iax
- packets
- * Remove doublon in dependencies
- * Remove warnings from the client test framework
- * Update version number to 0.9.5~beta
- * Update build-package script
- * Add check dependency in build-deps control file field
- * Create debian files for the new sflphone-gnome
- * [TICKET #1212] Add Replaces field in control files
- * [TICKET #1212] Fix manpages installation path
- * [TICKET #1212] Add maintainer scripts to create alternatives
- * [#1212] Update the manpages generation - edit preinst maintainer
- script
- * [#1212] Fix reference error in manpage
- * [#1212] Add missing files on the client side
- * [#1212] Fix debian docs files - no TODO file
- * [1212] Fix manpage creation problem
- * [#1220] Generate client-side glue files and marshaller at
- compilation time
- * [#1220] Generate server-side glue files at compilation time
- * [#1212] Change binary name to sflphone-gnome
- * [#1212] Update .gitignore to fit the new working tree
- * [#1220] Explicitly generate glue files before building the library
- * [#1220] Compile dbus directory before audio
- * [#1212] Create sflphone-common at the root of the repository
- * [#1212] Re-add pjproject
- * [#1212] Remove Makefile from repo
- * [#1220] Fix Makefile.am
- * [#1212] New working directory functional
- * [#1212] Update .gitignore
- * [#1212] Hack to make pjsip compile..
- * [#1220] Use non-installed binary for dbusxx-xml2cpp
- * [#1212] Add descriptive files, remove unuseful scripts from tools/
-
- [ Alexandre Savard ]
- * Restore speex codecs
- * add frame size for celt (#1143)
- * add framesize to codec, independant from audiolayer (#1143)
- * use codec frame size in rtp (#1143)
- * compute fixed_codec_framesize (#1143)
- * do not resample if not required (#1143)
- * add condition on resampling for decoder (#1143)
- * add a condition on bytesAvail == 0 from mic data
- * no maximum in rtp decode (#1143)
- * compute maximum for decoding (#1143)
-
- [ Emmanuel Milou ]
- * [#1146] Implement unitary tests on the client-side
-
- [ Alexandre Savard ]
- * use float instead of int to compute max nb of sample (#1143)
- * add nbSampleMax for unresampled data (#1143)
- * make thread sleep during 5 ms insead of 20 (#1143)
- * use unix usleep (#1143)
- * 50 usecond thread!!!!! (#1143)
- * try with the smallest compression (#1143)
- * use timer set at framesize (#1143)
-
- [ Emmanuel Milou ]
- * [#1161] Restore changelog version
-
- [ Alexandre Savard ]
- * Remove celt stuff
-
- [ Emmanuel Milou ]
- * [#1161] Update changelog
- * [#1220] Add Conflicts: sflphone in debian control files
- * [#1179] Add liblog4c3 runtime dependency
- * [#1212] FIx typo error in dependency list for itnrepid
- * [#1212] FIx .desktop file to point on the right exec
- * [#1212] Modify changelog replacing tag
-
- [ Sflphone Project ]
- * "[#1262] Updated changelogs for version 0.9.5-0ubuntu1~beta"
-
- [ Emmanuel Milou ]
- * [#1212] restore changelogs
-
- [ Sflphone Project ]
- * [#1262] Updated changelogs for version 0.9.5-0ubuntu1 Snapshot 2009-
- 04-27
-
- [ Emmanuel Milou ]
- * [#1212] restore changelogs
-
- [ Sflphone Project ]
- * [#1262] Updated changelogs for version 0.9.5-0ubuntu1~beta
-
- [ Emmanuel Milou ]
- * [#1212] restore changelogs
-
- [ Sflphone Project ]
-
- -- Sflphone Project <sflphone@mtl.savoirfairelinux.net> Mon, 27 Apr 2009 17:00:03 -0400
-
-sflphone-gnome (0.9.4-0ubuntu2) SYSTEM; urgency=low
-
- [ Alexandre Savard ]
- * Restore speex and GSM detection
-
- [ Emmanuel Milou ]
- * Fix bug #1090
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Wed, 8 Apr 2009 11:29:15 -0500
-
-sflphone (0.9.4-0ubuntu1) SYSTEM; urgency=low
-
- [ Emmanuel Milou ]
- * Integrate DBus-c++ and libiax2 in the main build system
- * Clean up in the working repository
- * Reorder hooks configuration panel
- * Protect case when no codecs are active
- * Fix some return values
- * Add unitary tests for the hook manager (premisces)
-
- [Yun Liu]
- * Update chinese translation
-
- [Sven Werlen]
- * Update german translation
-
- [Hussein Abdallah]
- * Update russian translation
-
- [Maxime Chambreuil]
- * Update spanish translation
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Fri, 3 Apr 2009 18:29:15 -0500
-
-
-sflphone (0.9.4-rc1) SYSTEM; urgency=low
-
- [ Emmanuel Milou ]
- * Fix bug while trying to hold/unhold several simultaneous call
- * Improve address book build system
- * Implement SIP url popup on incoming call
- * Improve GTK+ panel configuration
- [ Julien Bonjean ]
- * GTK+ client refactoring
- * GTK+ clean up
- * Address book improvment
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Fri, 27 Mar 2009 18:29:15 -0500
-
-sflphone (0.9.4-0beta1) SYSTEM; urgency=low
-
- [ Alexandre Savard ]
- * Display codec used during conversation on the GUI
- * Enable/disable STUN parameters at runtime
- * Refactor search bar use
- [ Emmanuel Milou ]
- * Build system fixes
- * Implement SIP re-invite
- * Implement IP to IP call
- [ Julien Bonjean ]
- * Integrate GNOME address book based on evolution data server
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Fri, 20 Mar 2009 18:29:15 -0500
-
-
-sflphone (0.9.3-0ubuntu3) SYSTEM; urgency=low
-
- [ Alexandre Savard ]
- * Both playback and record streams in PA_STREAM_CORKED (pulseaudio)
- * Use PLUGHW device for ALSA capture
- * Functional IAX and SIP recording for voicemail
- * Use the less CPU-consuming interpolator algorithm for resampling
- * Display in GTK GUI the codec used in conversation
- * GTK GUI use ASCII instread of utf-8
- * Add record menus in GTK GUI
- * Put on hold when dialing a new number
- * AccountID's are saved in the history
-
- [ Emmanuel Milou ]
- * Integrate DBUS C++, libiax2 in the git repository
- * Update website
- * Use libspeexdsp only if available on the system
- * Updated .gitignore file
-
- [Cyrille Béraud]
- * Account assistant manager improvment
- * Add an email request when creating a new account to receive voicemails
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Sat, 14 Feb 2009 13:29:15 -0500
-
-sflphone (0.9.3-0ubuntu2) SYSTEM; urgency=low
-
- [ Emmanuel Milou ]
- * Add compilation note in README
- * Use default ALSA plugin for capture
- * Fix the ALSA capture problem one more time
- * Clean up debug messages in dbus.c
- * Add libspeexdsp dependency
- * Remove implicit declaration compilation warnings
- * Fix links in the website, add release note
- * Change capture for the website front page
- * Add alsa devel dependency in build-depends control file field
- * Clean up, indentation, try to handle latency problems in iax/pulseaudio
- * Remove pjsip generated files from the repo
- * Use the previous declared curAlias function in accountwindow
- * Fix bug in history call duration when the call fails
- * Remove runtime warning in the GTK+ client
- * Add librsvg2-common dependency to load SVG under KDE
- * Refresh .gitignore
- * Update locales files + french translation
- * Add configuration panel for future noise reduction
- * Add configuration panel for audio record module
- * Daemon less verbose; accounts don't try to access STUn options anymore
- * Fix typo in configwindow
- * Add content in the official website
- * use a GTK_STOCK icon for the record button
- * Complete description text in the assistant manager
- * Add libtool flags in client configure.ac
- * Remove unuseful dependency (snd)
- * Fix SIP transfer problems
- * Remove previous version of PJSIP from the repo
- * Upgrade PJSIP to version 1.0.1
- * Add the new website source in the repository
- * Use libspeexdsp for silence detection only if available
-
- [ Loïc Faure-Lacroix ]
- * Ajout du logo gpl3
- * Ajout des images
- * Ajout de la section screenshot pour le site
- * Ajout du favicon dans le header
- * Modification des cartes
-
- [ Alexandre Savard ]
- * Clean up <speex/libspeexdsp>
- * Small cleanup
- * Save Wave fixed
- * Fix new call button when recording
- * libspeexdsp added
- * Recording: default home folder at startup
- * Minor changes to config window
- * IAX recording fixed
- * Set / get recording path, still need some GTK for client
- * AudioRecord file name format
- * Now recording in HOME folder
-
- [ Cyrille Béraud ]
- * Fix bug in reqaccount.c
-
- [ Maxime Chambreuil ]
- * Update spanish translation
-
- [Yun Liu ]
- * Update chinese translation
-
- [ Hussein Abdallah ]
- * Update russian translation
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Sat, 14 Feb 2009 13:29:15 -0500
-
-sflphone (0.9.3-0ubuntu1) SYSTEM; urgency=low
-
- * Remove debug
- * Join thread before leaving
- * Fix implicit declaration in reqaccount
- * Add REST code to build the request to server
- * Fix GValue initialization warnings
- * Update version number, fix implicit declaration, fix GTK markup
- warnings
- * Apply patch to create custom SIP account from our own server
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Fri, 06 Feb 2009 19:17:32 -0500
-
-sflphone (0.9.2-2ubuntu9) SYSTEM; urgency=low
-
- [ Alexandre Savard ]
- * Speex audio codec preprocessing initialization
- * peer hung up segmentation fault solved
- * Stop recording when transfering
- * Terminate only one call
- * Add isRecording() function
- * Fix call_icon GTK client
- * Fix SIPCallClose() function, recorded file now close properly
- * Function terminateSIPCall added in sipvoiplink and managerimpl
- * Fix thread destructor
- * setRecordingOption function implement in audiorecord
- * Record now implemented in Call class
- * Record interface complete (on hold erase previous recording)
- * Added recButton in client
- * Added: record button related icons
- * Record button added
- * Overload AudioRecord::recData to get mic and speaker data mixed
- * Recording now in audiortp::run() method
- * Audio recording working in AudioRTP: receiveSessionForSpeaker
- * Open/close a wave file when pulse audio stream start/stop
-
- [ Emmanuel Milou ]
- * Fix path for GTK+ icons; clean up
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Thu, 05 Feb 2009 18:27:53 -0500
-
-sflphone (0.9.2-2ubuntu8) SYSTEM; urgency=low
-
- [ Emmanuel Milou ]
- * Update changelogs
- * Fix bug in merge and in Makefile.am
- * Terminate only one call
- * Disable PJsip shutdown when changing STUN parameters
- * Function terminateSIPCall added in sipvoiplink and managerimpl
- * Add a timer to the alsa thread to not jam the CPU load
- * Fix bug in sipvoiplink.cpp
- * Clean shutdown of pulseaudio on quiting
- * Fix DTMF at first start with Pulseaudio
- * Remove zeroconf from the build system
- * Add a library manager + exception handling
- * Clean up in the working directory
- * Better handling of capture XRUNs
- * Restore mic adjust volume on ALSA layer
- * Protect device ALSA operation if not opened
- * Fix the switching layer bug
- * Use dynamic_cast<> to use audiolayer-specific methods
- * Open the audio devices only once at startup
- * Refactoring of the ALSA part
- * Functional plug-in manager
- * Use a C++ thread to handle tones and DTMF in ALSA
- * Restore IAXVoIPLink, restore Mutex
- * Make the plugins registering against the plugin manager
- * Migrate to 1->N relationship between voiplink and accounts
- * API plugin for registration
- * Use C++ thread in SIP, move everything in sipvoiplink
- * Complete singleton pattern for the plugin manager
- * Add -Wno-return-type compilation flag to remove warnings; Update
- version number in configure.ac
- * Add the dynamic loading for the plugin framework; integate unittest
-
- [ Yun Liu ]
- * Update rpm spec file
- * modify build package script and spec file for suse
-
- [ Alexandre Savard ]
- * Add audiorecorder plugin and testaudiorecorder
- * Add audio Recording class, edit global.h
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Wed, 04 Feb 2009 14:00:30 -0500
-
-sflphone (0.9.2-2ubuntu7) SYSTEM; urgency=low
-
- [ Emmanuel Milou ]
- * Update changelog to 0.9.2-6
- * Fix some dbus-glib implementation details on the client side
- * Init history after dbus initialization
- * Add error checking in useragent; Clean sipvoiplink
- * Prevent crash when trying to call an empty number
- * Set the volume of the playback stream to PA_VOLUME_NORM at startup
- * Fix GTK+ generic value double initialization
- * Fix jaunty control file dependency problems
- * Fix jaunty control file dependency problems
-
- [ Yun Liu ]
- * Fix bug ticket # 137
- * Tolerant to gsm library of OpenSuse 11
-
- [ Sven Werlen ]
- * Update german translation
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Fri, 23 Jan 2009 17:48:13 -0500
-
-sflphone (0.9.2-2ubuntu6) SYSTEM; urgency=low
-
- [ Emmanuel Milou ]
- * Migrate STUN configuration to the main config window
- * Update french translation
- * Other tiny memory leaks
- * Fix memory leak in sampleconverter.cpp
- * Generate packages from the release branch
- * update the build package script
- * modify the control files with architecture=any
- * Remove valgring uninitialized value
- * IAX and SIP use the same global variables to set account
- configuration ; fix broken code
-
- [ Maxime Chambreuil ]
- * Update spanish translation
-
- [ Hussein Abdallah ]
- * Update russian translation
-
- [ Yun Liu ]
- * Update translation files
- * Fix the bug when user uncheck the account which fails in the
- previous registration
- * Add stun error status
- * Fix bug ticket #143
- * Script for auto-install dependencies
- * Fix bug ticket #140
- * Fix bug ticket 141
- * Fix the reregister process when user change the details of an
- account
-
- -- Emmanuel Milou <manu@sulfur.inside.savoirfairelinux.net> Fri, 16 Jan 2009 18:19:05 -0500
-
-sflphone (0.9.2-2ubuntu5) SYSTEM; urgency=low
-
- * Fix memory leak in the pulseaudio callback
- * Update debian package generation script
- * Warnings removal in GTK+ client
- * Clean adjust volume method in alsalayer
- * Plug the sflphone playback volume control to the pulseaudio volume
- manager
- * Display the date in history according to the current locale
- * Generate the changelog according to the git commit messages
- * Complete header in chinese translation file
- * Use the right gpg key to sign the packages
- * add debian jaunty jackalope support
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Wed, 14 Jan 2009 21:17:20 -0500
-
-sflphone (0.9.2-2ubuntu4) SYSTEM; urgency=low
-
- [ Emmanuel Milou ]
- * add german translation
-
- [ Yun Liu ]
- * Fix GUI crash in Ubuntu8.10 64bit system
-
- -- Yun Liu <yun.liu@savoirfairelinux.com> Thu, 08 Jan 2009 13:08:51 -0500
-
-sflphone (0.9.2-2ubuntu3) SYSTEM; urgency=low
-
- [ Emmanuel Milou ]
- * The main thread synchronizes the ringtone thread
- * disable custom ringtone for the ALSA layer
- * Fix the Makefile.am in man directory, add a SEE ALSO section
-
- [ Yun Liu ]
- * Fix daemon crash caused by the previous patch ( for bug ticket #129)
-
- -- Yun Liu <yun.liu@savoirfairelinux.com> Tue, 06 Jan 2009 16:18:38 -0500
-
-sflphone (0.9.2-2ubuntu2) SYSTEM; urgency=low
-
- * Fix bug ticket #129
-
- -- Yun Liu <yun.liu@savoirfairelinux.com> Wed, 5 Jan 2009 15:54:53 -0500
-
-sflphone (0.9.2-2ubuntu1) SYSTEM; urgency=low
-
- * Migrate from eXosip library to pjsip
- * Add multiple SIP accounts support
- * Fix ringtones problems
- * Add a pulseaudio support
- * Improve audio quality with ALSA
- * Add chinese translation
- * Improve spanish translation
- * Migrate to a maintained C++ DBus bindings
- * Clean and improve the build system
- * Add build-dependency on Perl because we need pod2man to generate manpages
-
- -- Yun Liu <yun.liu@savoirfairelinux.com> Wed, 26 Nov 2008 09:47:53 -0500
-
-sflphone (0.9.1) unstable; urgency=low
- * Add a search tool in the history
- * Migrate some gtk_entry_new to sexy_icon_entry_new
- * Bug fix (Ticket #78): The voicemail password isn't displayed anymore in
- the history tab
- * Add the SIP registration expire value in the user file.
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Thu, 22 May 2008 11:14:25 -0500
-
-sflphone (0.9.0) unstable; urgency=low
- * Add history features
- * Call date
- * Call duration
- * Mouse events in the history tab
- * Smooth switch from the history tab to the calls tab
- * Remove most of GTK-Critical warnings
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Wed, 13 May 2008 16:58:25 -0500
-
-sflphone (0.9-2008-06-06) unstable; urgency=low
- * Audio bug correction: capture stopped after a few minutes of conversation
- with USB Plantronics sound card
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Tue, 06 May 2008 16:58:25 -0500
-
-sflphone (0.9-2008-05-06) unstable; urgency=low
- * Bug correction: account creation with the assistant
- * GTK+ warnings removal
- * libnotify warnings removal
- * Remove aliasing on the SFLphone logo
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Mon, 05 May 2008 16:58:25 -0500
-
-sflphone (0.9) unstable; urgency=low
- * Clean dependencies ( removal of libboost )
- * Several GTK improvement and updates
- -account window
- -configuration window
- * Migrate from GtkCheckMenuItem to GtkImageMenuItem
- * ALSA standard I/O transfers: MMAP instead of R/W
- * Fix speex audio quality
- * IAX2 protocol
- -Fix hold/unhold situation
- -Add on hold music
- * SIP protocol
- -Ringtone on incoming call
- -Fix transfer situation
- * Add desktop notification ( libnotify )
- * Improve the system tray icon behaviour
- * Improve registration error handling
- * Register/unregister from the account window takes effect without starting back SFLphone
- * Compilation warnings removal
- * Call history
- * Add an account configuration wizard
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Wed, 30 Apr 2008 16:58:25 -0500
-
-sflphone (0.8.2) unstable; urgency=low
- * Internationalization of the GTK GUI
- * English / French
- * STUN support
- * Slight modifications of the graphical interface ( tooltips, dialpad, ...)
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Fri, 21 Mar 2008 11:37:53 -0500
diff --git a/tools/build-system/launchpad/sflphone-gnome/debian/compat b/tools/build-system/launchpad/sflphone-gnome/debian/compat
deleted file mode 100644
index 7ed6ff82de6bcc2a78243fc9c54d3ef5ac14da69..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/sflphone-gnome/debian/compat
+++ /dev/null
@@ -1 +0,0 @@
-5
diff --git a/tools/build-system/launchpad/sflphone-gnome/debian/control b/tools/build-system/launchpad/sflphone-gnome/debian/control
deleted file mode 100644
index 656bc4f249d2d54d65c9f830a0483229cd0cdcc3..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/sflphone-gnome/debian/control
+++ /dev/null
@@ -1,21 +0,0 @@
-Source: sflphone-gnome
-Maintainer: SavoirFaireLinux Inc <julien.bonjean@savoirfairelinux.com>
-Section: gnome
-Priority: optional
-Build-Depends: debhelper, libgcc1, intltool, autopoint, autoconf, automake, libtool, libgtk-3-dev, libdbus-glib-1-dev, libnotify4-dev | libnotify-dev (>= 0.7), check, gnome-doc-utils, rarian-compat, librsvg2-common, yelp-tools, gnome-common
-Standards-Version: 3.7.3
-
-Package: sflphone-gnome
-Priority: optional
-Architecture: any
-Depends: sflphone-daemon (=${source:Version}), ${shlibs:Depends}, ${misc:Depends}
-Provides: sflphone-client-gnome
-Replaces: sflphone-client-gnome, sflphone
-Conflicts: sflphone-client-gnome-video, sflphone-gnome-video, sflphone-data
-Homepage: http://www.sflphone.org
-Description: GNOME client for SFLphone
- Provide a GNOME client for SFLphone.
- SFLphone is meant to be a robust enterprise-class desktop phone.
- SFLphone is released under the GNU General Public License.
- SFLphone is being developed by the global community, and maintained by
- Savoir-faire Linux, a Montreal, Quebec, Canada-based Linux consulting company.
diff --git a/tools/build-system/launchpad/sflphone-gnome/debian/copyright b/tools/build-system/launchpad/sflphone-gnome/debian/copyright
deleted file mode 100644
index 7b3bdc5eebd9324f7616980b85dd03f0354d32a5..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/sflphone-gnome/debian/copyright
+++ /dev/null
@@ -1,28 +0,0 @@
-This package was debianized by Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> on
-Fri, 3 Apr 2009 09:47:53 -0500.
-
-It was downloaded from the git repository of SFLphone: git://sflphone.org/git/sflphone.git
-
-Upstream Author: Emmanuel Milou <emmanuel.milou@savoirfairelinux.com>
-
-Copyright:
-
-Savoir-faire Linux Inc.
-
-License:
-
-This software is copyright (c) 2004-2016 Savoir-faire Linux Inc.
-
-You are free to distribute this software under the terms of
-the GNU General Public License version 3.
-On Debian systems, the complete text of the GNU General Public
-License can be found in the file `/usr/share/common-licenses/GPL'.
-
-This program is distributed in the hope that it will be useful,
-but WITHOUT ANY WARRANTY; without even the implied warranty of
-MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
-GNU General Public License for more details.
-
-You should have received a copy of the GNU General Public License
-along with this program; if not, write to the Free Software
-Foundation, Inc., 51 Franklyn St, Fifth Floor, Boston, MA 02110-1301, USA.
diff --git a/tools/build-system/launchpad/sflphone-gnome/debian/cron.d b/tools/build-system/launchpad/sflphone-gnome/debian/cron.d
deleted file mode 100644
index d11e61177739b56bce3aac6de6483b48e797a258..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/sflphone-gnome/debian/cron.d
+++ /dev/null
@@ -1,4 +0,0 @@
-#
-# Regular cron jobs for the sflphone package
-#
-0 4 * * * root sflphone_maintenance
diff --git a/tools/build-system/launchpad/sflphone-gnome/debian/dirs b/tools/build-system/launchpad/sflphone-gnome/debian/dirs
deleted file mode 100644
index e2dc98dcb24907fb5a7ceb0f0651276b702d6030..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/sflphone-gnome/debian/dirs
+++ /dev/null
@@ -1,7 +0,0 @@
-usr/bin
-usr/share/applications
-usr/share/pixmaps
-usr/share/sflphone
-usr/share/locale
-usr/share/doc
-usr/share/man
diff --git a/tools/build-system/launchpad/sflphone-gnome/debian/docs b/tools/build-system/launchpad/sflphone-gnome/debian/docs
deleted file mode 100644
index 9830da213fdb4baf4d68538e8c8e490248e209e1..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/sflphone-gnome/debian/docs
+++ /dev/null
@@ -1,5 +0,0 @@
-NEWS
-README
-ChangeLog
-AUTHORS
-
diff --git a/tools/build-system/launchpad/sflphone-gnome/debian/manpages b/tools/build-system/launchpad/sflphone-gnome/debian/manpages
deleted file mode 100644
index 27631d29ed1d9d3949db19e5d8c6340413ea90f8..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/sflphone-gnome/debian/manpages
+++ /dev/null
@@ -1,2 +0,0 @@
-debian/sflphone-gnome/usr/share/man/man1/sflphone-client-gnome.1
-debian/sflphone-gnome/usr/share/man/man1/sflphone.1
diff --git a/tools/build-system/launchpad/sflphone-gnome/debian/postinst b/tools/build-system/launchpad/sflphone-gnome/debian/postinst
deleted file mode 100644
index ebee7fa2bb049bf0e6f826e28569d05cb51d451a..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/sflphone-gnome/debian/postinst
+++ /dev/null
@@ -1,7 +0,0 @@
-#!/bin/sh -e
-
-update-alternatives --install /usr/bin/sflphone sflphone /usr/bin/sflphone-client-gnome 100 \
- --slave /usr/share/man/man1/sflphone.1.gz sflphone.1.gz \
- /usr/share/man/man1/sflphone-client-gnome.1.gz
-
-exit 0
diff --git a/tools/build-system/launchpad/sflphone-gnome/debian/postrm b/tools/build-system/launchpad/sflphone-gnome/debian/postrm
deleted file mode 100644
index 70be710bd108ddadbfa790f9ac8795984cc139f8..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/sflphone-gnome/debian/postrm
+++ /dev/null
@@ -1,36 +0,0 @@
-#!/bin/sh
-# postrm script for sflphone
-#
-# see: dh_installdeb(1)
-
-set -e
-
-# summary of how this script can be called:
-# * <postrm> `remove'
-# * <postrm> `purge'
-# * <old-postrm> `upgrade' <new-version>
-# * <new-postrm> `failed-upgrade' <old-version>
-# * <new-postrm> `abort-install'
-# * <new-postrm> `abort-install' <old-version>
-# * <new-postrm> `abort-upgrade' <old-version>
-# * <disappearer's-postrm> `disappear' <overwriter>
-# <overwriter-version>
-# for details, see http://www.debian.org/doc/debian-policy/ or
-# the debian-policy package
-
-if [ "$1" = "purge" ]
-then
-
- # remove the user config file
- rm -f $HOME/.sflphone/sflphonedrc
-
-fi
-
-# dh_installdeb will replace this with shell code automatically
-# generated by other debhelper scripts.
-
-#DEBHELPER#
-
-exit 0
-
-
diff --git a/tools/build-system/launchpad/sflphone-gnome/debian/preinst b/tools/build-system/launchpad/sflphone-gnome/debian/preinst
deleted file mode 100644
index d5e20248d97258ec7a5c80e4b5077ad19a77bcde..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/sflphone-gnome/debian/preinst
+++ /dev/null
@@ -1,19 +0,0 @@
-#!/bin/sh
-# postrm script for sflphone
-#
-# see: dh_installdeb(1)
-
-set -e
-
-package=sflphone-gnome
-
-case "$1" in
- install|upgrade)
- ## Clean up the previous manpage
- if [ -f /usr/share/man/man1/sflphone-gtk.1 ]; then
- rm /usr/share/man/man1/sflphone-gtk.1
- fi
- ;;
-esac
-
-exit 0
diff --git a/tools/build-system/launchpad/sflphone-gnome/debian/prerm b/tools/build-system/launchpad/sflphone-gnome/debian/prerm
deleted file mode 100644
index 5e9021706875bb08a56c8c54f35cef96a7ca6055..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/sflphone-gnome/debian/prerm
+++ /dev/null
@@ -1,7 +0,0 @@
-#!/bin/sh -e
-
-
-if [ "$1" = "remove" ]; then
- # Remove alternatives symlink set in postinst
- update-alternatives --remove sflphone /usr/bin/sflphone
-fi
diff --git a/tools/build-system/launchpad/sflphone-gnome/debian/rules b/tools/build-system/launchpad/sflphone-gnome/debian/rules
deleted file mode 100755
index 3a3522cccd87a27106f8b6ebd4028269ac431cd0..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/sflphone-gnome/debian/rules
+++ /dev/null
@@ -1,117 +0,0 @@
-#!/usr/bin/make -f
-# -*- makefile -*-
-# Sample debian/rules that uses debhelper.
-# This file was originally written by Joey Hess and Craig Small.
-# As a special exception, when this file is copied by dh-make into a
-# dh-make output file, you may use that output file without restriction.
-# This special exception was added by Craig Small in version 0.37 of dh-make.
-
-# Uncomment this to turn on verbose mode.
-#export DH_VERBOSE=1
-export DH_OPTIONS
-
-package=sflphone-gnome
-
-CXX = g++-4.0
-CFLAGS = -Wall -g
-DEB_INSTALL_MANPAGES_sflphone_gnome = sflphone.1 sflphone-client-gnome.1
-
-configure: configure-stamp
-configure-stamp:
- dh_testdir
- # Add here commands to configure the package.
- NOCONFIGURE=1 ./autogen.sh
- ./configure --prefix=/usr --disable-video
- touch configure-stamp
-
-
-#Architecture
-build: build-arch build-indep
-
-build-arch: build-arch-stamp
-build-arch-stamp: configure-stamp
-
- # Add here commands to compile the arch part of the package.
- $(MAKE)
- touch $@
-
-build-indep: build-indep-stamp
-build-indep-stamp: configure-stamp
-
- # Add here commands to compile the indep part of the package.
- #$(MAKE) doc
- touch $@
-clean:
- dh_testdir
- dh_testroot
- rm -f build-arch-stamp build-indep-stamp configure-stamp
- # Add here commands to clean up after the build process.
- [ ! -f Makefile ] || $(MAKE) distclean
-
-ifneq "$(wildcard /usr/share/misc/config.sub)" ""
- cp -f /usr/share/misc/config.sub config.sub
-endif
-ifneq "$(wildcard /usr/share/misc/config.guess)" ""
- cp -f /usr/share/misc/config.guess config.guess
-endif
- dh_clean
-
-install: install-indep install-arch
-install-indep:
- dh_testdir
- dh_testroot
- dh_clean -k -i
- dh_installdirs -i
- # Add here commands to install the package into debian/sflphone.
-
-install-arch:
- dh_testdir
- dh_testroot
- dh_clean -k -s
- dh_installdirs -s
- # Add here commands to install the arch part of the package into
- # debian/tmp.
- $(MAKE) DESTDIR=$(CURDIR)/debian/$(package) install
- dh_install -s
-# Must not depend on anything. This is to be called by
-# binary-arch/binary-indep
-# in another 'make' thread.
-
-binary-common:
- dh_testdir
- dh_testroot
- dh_installchangelogs ChangeLog
- dh_installdocs
- dh_installexamples
-# dh_installmenu
-# dh_installdebconf
-# dh_installlogrotate
-# dh_installemacsen
-# dh_installpam
-# dh_installmime
-# dh_python
-# dh_installinit
-# dh_installcron
-# dh_installinfo
- dh_installman
- dh_link
- dh_strip
- dh_compress
- dh_fixperms
-# dh_perl
- dh_makeshlibs
- dh_installdeb
- dh_shlibdeps
- dh_gencontrol
- dh_md5sums
- dh_builddeb
-# Build architecture independant packages using the common target.
-binary-indep: build-indep install-indep
- $(MAKE) -f debian/rules DH_OPTIONS=-i binary-common
-
-# Build architecture dependant packages using the common target.
-binary-arch: build-arch install-arch
- $(MAKE) -f debian/rules DH_OPTIONS=-s binary-common
-
-binary: binary-arch binary-indep
-.PHONY: build clean binary-indep binary-arch binary install install-indep install-arch configure
diff --git a/tools/build-system/launchpad/sflphone-gnome/debian/substvars b/tools/build-system/launchpad/sflphone-gnome/debian/substvars
deleted file mode 100644
index 566a162f0d3708c2c131a6eff863df6727922259..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/sflphone-gnome/debian/substvars
+++ /dev/null
@@ -1 +0,0 @@
-plop=0.9.6
diff --git a/tools/build-system/launchpad/sflphone-kde/debian/changelog b/tools/build-system/launchpad/sflphone-kde/debian/changelog
deleted file mode 100644
index 7bc6d8ca8c1641c4a08d7c1e6d76ee20cddd6da4..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/sflphone-kde/debian/changelog
+++ /dev/null
@@ -1,180 +0,0 @@
-sflphone-kde (1.4.0) SYSTEM; urgency=low
- * See website at www.sflphone.org for more details
-
- -- Emmanuel Lepage Vallee <emmanuel.lepage@savoirfairelinux.com> Tue, 7 Jun 2012 11:38:30 -0500
-
-sflphone-kde (1.1.0-rc20120607~ppa1~SYSTEM) SYSTEM; urgency=low
-
- ** SNAPSHOT 1.1.0-rc20120607~ppa1~SYSTEM **
-
- * [ #12096 ] Implement more calls details, make it more scalable
- * [ #11290 ] Update french translation
- * [ #11859 ] Fix a regression in name conversion
- * [ #11850 ] Fix a little regression
- * [ #11859 ] Fix sorting by popularity
- * [ #11569 ] Fix filter
- * [ #12003 ] Add icon overlay for history item
- * [ #11886 ] Add basic reverse peer naming support
- * kde/callview: removed trailing whitespace
- * [ #12008 ] Implement GUI part
- * [ #11988 ] Move ringtone to a new tab
- * [ #11990 ] Fix history delegate width
- * [ #11990 ] Make call item height configurable
- * [ #7007 ] Fix many history bugs and add 'copy' contextual menu
- * [ #7007 ] Add more contextual options
- * [ #11963 ] Move ringtone to account
- * [ #7003 ] Implement more drag and drop
- * [ #11888 ] Add support for previous timestamp
- * [ #11887 ] Drop phonon dependency
- * [ #11887 ] Use daemon player instead of phonon
- * [ #11889 ] Fix a regression
- * [ #11889 ] Fix a regression
- * [ #11861 ] Fix memory leak
- * [ #11861 ] Little profiling
- * [ #11850 ] Reimplement most destructor
- * [ #11847 ] Move unregister back into SFLPhoneApplication
- * [ #11847 ] Fix the unregister signal
- * [ #11846 ] Fix the text message box behavior
- * [ #11845 ] Refactor menus, duplicate display dock options
- * [ #11731 ] Fix visual glitch when hang up, harden against 0
- participant conference
- * [ #11798 ] Implement basic signal handling, it does not seem to be
- enough, but it help
- * [ #11815 ] Make double click call instead of queing items
- * [ #11822 ] Add configuration options
- * [ #11814 ] Add selection again
- * [ #3912 ] Finally make double clicking work, it was the last client
- bug, the other one is in the daemon
- * Fix plasmoid
- * [ #11733 ] Update the state machine to handle conferences correctly,
- not just ignore them
- * * #11252: daemon: removed deprecated zrtp code
- * [ #11385 ] Removing account work again
- * [ #11435 ] Fix trasfer, it apparently never worked, there was a bug
- in the state machine path
- * [ #11621 ] Support conferences in init
- * [ #3905 ] Fix the oldest KDE open bug, edit the conference pixmap
- * [ #11573 ] Add paste option
- * [ #11574 ] Add a account status label in status bar
- * [ #11577 ] Add a timer label
- * [ #11572 ] Implement accent independent filter for history and
- bookmark
- * [ #11572 ] Implement accent independent filter for contact
- * [ #11576 ] Implement optional contact details
- * [ #10222 ] Change default caller name from Unknown to his/her phone
- number
- * [ #11337 ] Add/Restore keyboard call selection
- * [ #11337 ] Add more accessibility options
- * [ #11337 ] Add basic text to speech status for calls
- * [ #11290 ] Fix dynamic translation
- * [ #11290 ] Update french translation
- * [ #11290 ] Add missing i18n() call, it should be translatable again
- * Make dependencies check more enforced for submodules
- * * #11269: merged master into video
- * [ #11255 ] Code polishing
- * [ #10222 ] Call again on double click
- * [ #11219 ] Change license to LGPLv2 for libraries
- * [ #7022 ] Make the new plasmoid the default, remove the old one, end
- of an era
- * [ #7022 ] Merge the new plasmoid branch, the older one is still the
- default
- * [ v1.1 ] Update version number
- * [ #10724 ] Add more dataengine services
- * [ #10724 ] Add bookmarks to the dataengine
- * [ #10724 ] Add bookmarks to the dataengine
- * [ #10724 ] Make contact sorting work for some sorting type
- * [ #10724 ] Refactor sorting in the KDE library, implement contact in
- the dataengine
- * [ #10724 ] Move the dataengine to the new KDE lib
- * [ #10724 ] Refactor data engine and split akonadi support out of the
- client to make the dataengine more future proof
- * Fix missing phonon
- * Fix compilation warning on old GCC
- * [ #10222 ] Fix warnings
- * [ #10380 ] Can't send message without a call
- * [ #10222 ] Test configuration dialog, fix it, implement basic
- messaging
- * [ #10222 ] Fix saving default history sorting
- * [ #10222 ] Use categorized Tree for history
- * [ #10222 ] Fix contact sorting
- * [ #10222 ] Async update for bookmark dock
- * [ #10222 ] Fix issues with account config dialog
- * [ #10222 ] Use categorized views, fix bookmark
- * [ #10222 ] Fix minor itches and bugs
- * [ #10121 ] Sync the KDE with daemon, fix a few issues and implement
- a recorded call player
- * * #10018: renamed registration related keys in dbus
- * * #8542: removed trailing whitespace from tree
- * * #8357: gnome client now works with new dbus video API
- * * 8487: cleanup in dbus
- * * #8450: fixed confusion between expire and enable values
- * * #8435: Remove typos
- * cleanup in xml files
- * Revert "Merge branch 'master' of
- git+ssh://git.sflphone.org/var/repos/sflphone/git/sflphone"
- * * #7264: gnome client now saves/loads history as List of Dicts.
- * [ #7901 ] Cleanup the changes
- * [ #7901 ] Fix compiler warnings
- * [ #7901 ] Do not show the conf button when in the same conference
- * [ #7929 ] Add some client side work around until it is fixed
- * [ #7901 ] Cosmetic and bug fixes
- * [ #7901 ] It is far from beautiful, but it work in most cases
- * [ #7901 ] Partial rewrite of the drag and drop event, does not
- really work
- * [ #7899 #7900 ] Fix toolbar icons
- * Remove useless icon in toolbar
- * [ #7876 ] qDebug -> kDebug (to be able to disable them using KDE
- gui)
- * [ #7876 ] Remove unneeded comments
- * [ #7887 ] 7% faster load time
- * [ #7887 ] Fix warning
- * [ #7887 ] Prevent most useless object copy
- * [ #7874 ] Twice less lines, same result
- * [ #7876 ] Remove unused class attribute, rename all attribute use
- use m_ for private member, m_p for private pointer, m_s for private
- static and m_sp for private static pointer
- * [ #7876 ] Update copyright from 2010 to 2012 (next version wont be
- released until then, so why not doing it now)
- * [ #7876 ] Add some doxygen
- * [ #7876 ] Sort include by owner, fix license issues
- * [ #7876 ] Clean includes
- * [ #7876 ] Add box comment for file sections (getter, setter,
- mutator, slots)
- * [ #7863 ] It work, here we go again
- * [ #7876 ] Spring cleanup
- * * #7264: added getHistorySimple, which return a dict of history
- entries
-
- -- Julien Bonjean <julien.bonjean@savoirfairelinux.com> Thu, 07 Jun 2012 16:10:36 -0400
-
-sflphone-kde (1.1.0-rc20120607~ppa1~SYSTEM) SYSTEM; urgency=low
-
- ** SNAPSHOT 1.1.0-rc20120607~ppa1~SYSTEM **
-
- * [ #12096 ] Implement more calls details, make it more scalable
- * [ #11290 ] Update french translation
- * [ #11859 ] Fix a regression in name conversion
- * [ #11850 ] Fix a little regression
- * [ #11859 ] Fix sorting by popularity
- * [ #11569 ] Fix filter
- * [ #12003 ] Add icon overlay for history item
- * [ #11886 ] Add basic reverse peer naming support
- * kde/callview: removed trailing whitespace
- * [ #12008 ] Implement GUI part
- * [ #11988 ] Move ringtone to a new tab
- * [ #11990 ] Fix history delegate width
- * [ #11990 ] Make call item height configurable
- * [ #7007 ] Fix many history bugs and add 'copy' contextual menu
- * [ #7007 ] Add more contextual options
-
- -- Julien Bonjean <julien.bonjean@savoirfairelinux.com> Thu, 07 Jun 2012 15:52:52 -0400
-
-sflphone-kde (1.1.0) SYSTEM; urgency=low
-
- ** SNAPSHOT 1.0.0-rc20110930~ppa1~SYSTEM **
-
- * Improve accessibility
- * Improve usability/UX
- * Fix all know bugs
-
- -- Emmanuel Lepage Vallee <emmanuel.lepage@savoirfairelinux.com> Tue, 7 Jun 2012 11:38:30 -0500
diff --git a/tools/build-system/launchpad/sflphone-kde/debian/compat b/tools/build-system/launchpad/sflphone-kde/debian/compat
deleted file mode 100644
index 7ed6ff82de6bcc2a78243fc9c54d3ef5ac14da69..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/sflphone-kde/debian/compat
+++ /dev/null
@@ -1 +0,0 @@
-5
diff --git a/tools/build-system/launchpad/sflphone-kde/debian/control b/tools/build-system/launchpad/sflphone-kde/debian/control
deleted file mode 100644
index d07015dffd2c9ed2998e67d88aa2823da0360088..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/sflphone-kde/debian/control
+++ /dev/null
@@ -1,13 +0,0 @@
-Source: sflphone-kde
-Section: kde
-Priority: optional
-Maintainer: Emmanuel Milou <emmanuel.milou@savoirfairelinux.com>
-Build-Depends: debhelper (>= 5.0), cdbs, kdelibs5-dev, cmake, kdepimlibs5-dev, libboost-dev, libx11-dev, libqt4-opengl-dev,
-Homepage: http://www.sfphone.org/
-
-Package: sflphone-kde
-Architecture: any
-Depends: sflphone-daemon-video (=${source:Version}), ${shlibs:Depends}, ${misc:Depends}, libqt4-dbus, libqt4-opengl
-Provides: sflphone-client-kde
-Replaces: sflphone-client-kde
-Description:KDE client for the sflphone-daemon SIP/IAX softphone
diff --git a/tools/build-system/launchpad/sflphone-kde/debian/copyright b/tools/build-system/launchpad/sflphone-kde/debian/copyright
deleted file mode 100644
index 754aa06895bd918a26abd3834731ca29539eb1e1..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/sflphone-kde/debian/copyright
+++ /dev/null
@@ -1,8 +0,0 @@
-SFLPhone:
-
- (C) 2004-2012 Savoir-faire Linux <contact@savoirfairelinux.com>
-
-SFLPhone KDE:
-
- Copyright (C) 2008-2009 Savoir-faire Linux <jeremy.quentin@savoirfairelinux.com>
- Copyright (C) 2009-2012 Savoir-faire Linux <emmanuel.lepage@savoirfairelinux.com>
diff --git a/tools/build-system/launchpad/sflphone-kde/debian/menu b/tools/build-system/launchpad/sflphone-kde/debian/menu
deleted file mode 100644
index 293ff7265fbe53145b35c994ee8ab6b96bcfab3d..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/sflphone-kde/debian/menu
+++ /dev/null
@@ -1,6 +0,0 @@
-?package(filelight):needs="X11" \
- section="Applications/Multimedia" \
- hints="KDE, Phone, Sip,Call" \
- command="sflphone-kde" \
- title="SFLPhone-KDE" \
- longtitle="SFLPhone Client KDE: Enterprise class softphone for KDE"
diff --git a/tools/build-system/launchpad/sflphone-kde/debian/rules b/tools/build-system/launchpad/sflphone-kde/debian/rules
deleted file mode 100755
index 1c0258f9c2ade2ac5c1fd5ff2768787ac31510fa..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/sflphone-kde/debian/rules
+++ /dev/null
@@ -1,5 +0,0 @@
-#!/usr/bin/make -f
-
-include /usr/share/cdbs/1/rules/debhelper.mk
-include /usr/share/cdbs/1/class/cmake.mk
-
diff --git a/tools/build-system/launchpad/sflphone-kde/debian/source.backup/format b/tools/build-system/launchpad/sflphone-kde/debian/source.backup/format
deleted file mode 100644
index 163aaf8d82b6c54f23c45f32895dbdfdcc27b047..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/sflphone-kde/debian/source.backup/format
+++ /dev/null
@@ -1 +0,0 @@
-3.0 (quilt)
diff --git a/tools/build-system/launchpad/sflphone-plugins/debian/changelog b/tools/build-system/launchpad/sflphone-plugins/debian/changelog
deleted file mode 100644
index 2cccb5d4740be43989ac01a546b55f7bcd4409af..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/sflphone-plugins/debian/changelog
+++ /dev/null
@@ -1,3117 +0,0 @@
-sflphone-plugins (1.1.0-rc20120607~ppa1~SYSTEM) SYSTEM; urgency=low
-
- ** SNAPSHOT 1.1.0-rc20120607~ppa1~SYSTEM **
-
- * * #11208: bump version numbers for release 1.1.0
- * * #9381: use autoconf macros and AC_SEARCH_LIBS
- * * #9144: Fixes "Only <glib.h> can be included directly" error
- * #8449: Update version 1.0.2
- * * #8542: removed trailing whitespace from tree
-
- -- Julien Bonjean <julien.bonjean@savoirfairelinux.com> Thu, 07 Jun 2012 16:10:55 -0400
-
-sflphone-plugins (1.0.0-rc20110930~ppa1~SYSTEM) SYSTEM; urgency=low
-
- ** SNAPSHOT 1.0.0-rc20110930~ppa1~SYSTEM **
-
- * update kde .gitignore
- * Fix bug in volume widget
- * More polishing for release
- * Bump version to 1.0.0
- * [#7023] Add the ability to load an abstract contact backend in the
- library to resolve more data, polish code
- * [#7021] More cleanup for release
- * Cleanup
- * [#7021] Refactor KDE client dbus handling, add a missing call in
- daemon and port the DataEngine to the new API
- * Remove some annoying debug
- * merge language scripts
- * remove obsolete 'VERSION' files
- * update install instructions
- * Add missing translations to gnome
- * language update
- * Revert "Don't reference count DBus clients, exit core immediately
- when one of them request it"
- * Don't reference count DBus clients, exit core immediately when one
- of them request it
- * [7021] Add contact abstraction support
- * [#7121] Polishing library (over). Indentation, spacing and naming
- are now consistent
- * codecs: link to libccrtp, don't use logger
- * Fix a daemon bug
- * [#7038] Fix adding contact
- * * #7037 : stop audio stream after all calls have been hanged up
- * [#7025] Add full support for bookmark
- * SFLPhone KDE do not destroy history anymore
- * Fix config skeleton
- * Close the daemon once and for all, no more automatic respawning
- * Fix "unregistered account" bug (I hope so)
- * Close SFLPhone at the right place, it still respawn, I don't know
- why
- * Remove dead code
- * Fix regressions introduced in the last commit
- * Dead code elimination 1/3
- * Fix bug, add "add contact" option, fix warning
- * * #7019: Fix IAX codec negociation
- * Remove or comment unnecessary/unhelpful debug output
- * Fix "same as local" account setting, fix IP2IP LED color
- * Add support for some more advanced config options and add missing
- config dialog icons
- * Fix crash with noise suppressor
- * Alternative can now be selected from the call view context menu
- * Add drag and drop support, initial context menu and fix 3 bugs in
- the account dialog
- * Add basic history drag and drop support
- * Complete contact support is back
- * * #6991 : fix IAX problems
- * Fix IAX accounts being disabled by default
- * Revert "deb: forge -g flags for pjsip"
- * * #5884: Disable debug code in pjsip
- * echo suppressor : more assertions
- * Don't let the daemon think crypto is enabled when it's not
- * Simplify ToneList
- * Some progress on contact support
- * Remove unused getRegistrationCount()
- * remove annoying debug
- * revert SIP bit of e27e5c39bad27bae28f574eb2cba7717e8956229
- * Simplify CallManager::placeCallFirstAccount
- * Fix crash on hold
- * * #6905 : SIP refactor
- * gnome client: be sure key exchange is set correctly
- * Move code into createSipTransport
- * Fix account registration on start
- * ManagerImpl::registerAccounts(): simplify
- * * #5884: don't mess with pjsip threads in echo suppressor
- * * #6905 : simplify udp/stun/tls pjsip transport creation
- * Restore and improve support for Call history
- * fix launchpad build
- * SIPVoIPLink: simplify / refactor
- * Fix libwidget linking
- * SIP: simplify
- * IM : simplify
- * gnome: remove some debug
- * AudioRtpFactory::stop() cannot fail
- * * #6905: simplify SIP code
- * pjlib: fix build without SSLv2, fix warnings
- * Port history to the new syntax
- * Test a dock widget based implementation for contact and history
- * Disable SSLv2 support from pjsip and sflphone
- * deb: forge -g flags for pjsip
- * Fix deb packaging to get debug symbols
- * remove debug
- * pjproject: update to last stable release (1.10)
- * Require gtk >= 2.20 and glib >= 2.24
- * tlsadvanceddialog: simplify
- * * #6902 : fix errors spotted by -DGSEAL_ENABLE
- * Update daemon dbus XML and port KDE config backend from dbus to
- local
- * Remove unused but set variables
- * * #6929 : fix IM widget, cleanup
- * Unconditionally enable debug symbols
- * Should fix many KDE issues
- * * #6886 : hitting backspace on empty number have no side effects
- * * #6905 : fix AudioCodecFactory access in optimized builds (-O > 0)
- * Remove unsupported and broken jaunty/karmic packages
- * * #6902 : avoid using some gtk deprecated functions
- * Update dbus introspection files
- * * #6904: removed unused contactmanager
- * * #6903 : use correct dbus-cxx package name
- * * #6902: don't use individual gtk headers
- * Fix a segfault when config is not present
- * Merge latest (0.9.13) KDE code. This version is not yet ready for
- git master, but better than the previous one
- * addressbook : simplify
- * * #5659 : sflphone-plugins doesn't depend on libedataserverui
- * * #5659 : addressbook doesn't use libedataserverui
- * gnome client doesn't depend on evolution
- * * #5695: addressbook: simplify
- * * #5695: addressbook : remove AddrBookHandle from plugin
- * * #5695 : addressbook : remove unused stuff in the client
- * * #5695 : addressbook : remove unused stuff, use static mutex
- * gnome client doesn't use evolution
- * gnome: use proper API to set GTK_CAN_FOCUS
- * * #6897: removed unused focus state vars/callbacks
- * gnome: fix calls to sflphone_fill_codec_list_per_account
- * * #6623: gnome: don't leak in mainwindow
- * gnome: mainwindow whitespace cleanup
- * gnome: actions.c parameter doesn't have to be a double pointer
- * * #6895: fix memleaks, cleanup in accountconfigdialog
- * * #6893: fixes segfault in client on clean history
- * * #6894: fix leaks, cleanup in sflnotify
- * daemon: fixed prints in main
- * * #6892: simplify, fix leaks in dialpad
- * * #6887: audiopreference creates audio layer
- * * #6660: use const char * const, not std::string for globally
- visible constants
- * * #6852: Preferences now solely responsible for audiolayer creation.
- * * #6860: refactor uimanager, also fixes #6865
- * * #6853: hangup as soon as all digits have been deleted
- * * #6852: alsa: retry if device is busy
- * * #6852: audiolayer creation depends only on preference.audioApi
- * * #6850: gnome: fix build for gtk < 2.22.0
- * cleanup in iax
- * alsa: typo
- * pulse: if we can't peek in audio input, we can't drop samples
- * * #6849: show error window if codecs are missing, instead of dying
- * EchoCancel: unused, remove
- * * #6629 : use number of samples as arguments for audio filters
- * * #6629 : remove unused Algorithm interface
- * * #6629 : use helper to call alsa functions and display error msgs
- * Remove unused type
- * * #6841: fix some error handling
- * * #6629: simplify AlsaLayer::alsa_set_params()
- * Get gdk key definition from header
- * * #6828: Replace raw key codes by gdk defines
- * remove some debug, enhance some other
- * mainbuffer: simplify
- * * #6561 : fix phantom call after transfer
- * Conference Participant set : simplify
- * SIPCall: remove unused functions, make invite session public
- * * #6229 : remove malloc/free from pulse audio loop
- * * #6629 : simplify pulse callbacks
- * * #6629
- * Simplify widgets
- * * #6629 : keep the correct audio module when frequency changes
- * * #6751: fixed erroneous debug msgs
- * callable_obj.h: removed unneeded pthread header
- * alsalayer: cleanup
- * * #6629: Always restart audio driver when changing parameters (ALSA
- only)
- * gnome GUI: don't block in DBus signal errorAlert()
- * * #6629 : simplify AudioLayer creation
- * * #6629 : remove unused and unconfigurable frameSize from audiolayer
- * * #6629 : remove unused error message from audio layer
- * Fix logic error when switching audio API
- * Remove unused AudioProcessing class
- * AudioRtpRecordHandler::initNoiseSuppress() : use noiseSuppress
- directly
- * * #6629 : use DC blocker directly in audio layers
- * * #6629 : clean AudioLayer
- * * #6629 : don't store mainbuffer inside audiolayer
- * * #6629 : correct AudioLayer::notifyincomingCall()
- * * #6554: cleanup, refactoring in sipvoiplink
- * * #6554: cleanup in iaxvoiplink
- * * #6554: throw exception in getSIPCall if pointer is NULL
- * * #6554: make some methods of sipvoiplink static
- * * #6655: cleanup in managerimpl
- * * #6554: refactoring, fix memleaks in sipvoiplink
- * * #6478: remove throw specs, cleanup in voiplink
- * * #6629 : remove unused AudioDevice
- * * #6655: removed more dependencies from managerimpl
- * * #6744: simplified numbercleaner
- * conference : remove one prototype
- * * #6743: fix ip2ip
- * Don't give glib warnings if icons are not found
- * gnome: fixed includes
- * Codec.h: removed unused function
- * * #6742 : clean dbus & icons
- * * #6699: refactor/cleanup accounts
- * icons: cleanup
- * timer : use second precision, not millisecond
- * calltree_update_clock : use correct type, returns something
- * * #6737: fixed typo in dbus call
- * * #6737: removed tests for removed API
- * * #6737: dbus: fixed bug from merge
- * * #6737: cleanup in accountlist
- * * #6737: cleanup in dbus
- * * #6740 : fix history double free
- * * #6740 : remove time updating thread from calls
- * * #6737 : use c99 for client
- * * #6738 : make history loading faster
- * sipvoiplink : don't crash on transfers
- * fixed typo
- * Remove unused file
- * Don't build networkmanager.cpp at all if NM is disabled
- * _debug* -> _debug
- * * #6554 : simplify sipvoiplink
- * hudson: added -x to git clean command
- * added git clean to hudson script
- * audiocodecfactory: cleanup
- * * #6718: refactored setTlsSettings into SIPAccount
- * * #6718: removed more unused methods
- * * #6718: refactored confmanager code into sipaccount
- * remove unused functions
- * * #6718: confmanager: removed more unused methods
- * AudioCodecFactory : cleanup
- * #6697 : Turn callableElement struct into union
- * * #6718: confmanager: removed more unused methods
- * * #6718: confmanager: removed more unused methods
- * * #6718: removed unused dbus methods, refactoring
- * * #6699: accounts: cleanup/refactoring
- * * #6699: refactoring, cleanup in accounts
- * * #6699: more account cleanup
- * remove unused autoconf variable
- * * #6714: fixed hudson script
- * make distclean in hudson
- * added || exit 1 to run_tests.sh call
- * * #6714: fixed make distcheck for sflphone-plugins
- * * #6714: fixed make distcheck for gnome client
- * * #6714: fixed make distcheck for daemon
- * git: #6698 split the main .gitignore file
- * gnome: gpointer is already a pointer
- * gnome: calltab_init: use calloc instead of malloc
- * * #6699: more account cleanup
- * * #6699: cleanup account
- * * #6554 : more *voiplink cleanup
- * * #6558 : more sipvoiplink simplification
- * * #6558: saner loadSIPLocalIP prototype
- * gnome: #6623 clean calllists
- * * #6692: more audiolayer cleanup
- * * #6692: cleanup/refactoring in audiolayers
- * * #6692: more forward declarations, AudioThread->AlsaThread
- * * #6692: audiolayer cleanup
- * * #6692: alsalayer cleanup
- * * #6558 : remove account creator
- * * #6558 : clean sipvoiplink
- * * #6554 : cleanup sipvoiplink
- * audiortp: cleanup
- * * #6657 : fix launchpad builds for good
- * * #6675 : send RTP dtmf events only once
- * * #6655: more cleanup
- * AudioRtpSession::updateSessionMedia() : simplify
- * * #6655: more cleanup in managerimpl
- * * #6655: removed more code, cleanup
- * * #6655: more cleanup, fixed infinite loop
- * * #6655: removed more unused files
- * * #6655: removed unused mutex
- * * #6655 removed more unused code
- * * #6655: removed unused methods
- * * #6655: cleanup in main
- * * #6663: fixed segfault when off hold from transfer
- * * #6658: user's active codec selection is respected
- * * #6660: static global string should be static const char* const
- class member
- * * #6659: use g_strcmp0, not strcmp for vals that may be null
- * callable_obj: fix double free
- * calltree_display_call_info() : simplify
- * * #6657: Fix launchpad builds
- * Logger::log() : simplify
- * AudioRtpSession : privatize members
- * * #6655: more constness, cleaned up/simplified methods
- * * #6654: call DBus::_init_threading so that dbus-c++ to make it
- threadaware
- * set default credentials on account creation
- * AudioCodecFactory::scanCodecDirectory() : simplify and correct
- * * #6623: fixed typos
- * * #6623: fixed more leaks
- * * #6623: fixed more leaks
- * * #6623: fixed more leaks, don't print codec name if null
- * * #6623: more leaks fixed in client
- * * #6623: fix more leaks, fixed some warnings
- * * #6623: fixed leak in history
- * updated gitignore
- * initialize dbus dispatcher correctly
- * Fix tests, hudson doesn't have a dbus daemon running
- * remove unused code
- * removeCall() : simplify , fix leak
- * stopRtpThread() : simplify
- * *CurrentCall : simplify
- * Fix memleak
- * fix serialization of audio api (pulse / alsa)
- * account map : simplify
- * remove call from callmap before terminating it, avoid use after free
- * * #6630 : don't make DBusManager a singleton
- * call: return confID by value
- * add back history code deleted by error
- * history : reverse logic
- * simplify history serialization and remove some debug
- * remove annoying debug
- * * #6464 : replace cerr with _error
- * * #6464: replace cout with logger macros
- * replace printf() with logger macros
- * update .gitignore
- * remove unused function
- * update eclipse projects
- * uimanager_new() : simplify
- * rename directories
- * celt: simplify a bit
- * Fix CELT configure.ac test
- * * #6612 : template speex codecs
- * * #6623: refactored conference obj
- * * #6623: refactored callable object, removed leaks
- * * #6623: more cleanup, fix leaks, make global vars static and rename
- them
- * * #6623: calltree: fixed memleaks, simplified code.
- * audiolayer: init pointer members
- * manager: catch exception on invalid hangup
- * * #6623: don't leak on calls to create_new_call
- * * #6611 : clarify codecs prototypes
- * ringtones : .au and .ul files are both ulaw
- * * #6611 : make sure samplerate converters are called correctly
- * ManagerImpl::switchAudioManager() : simplify
- * * #6623: fixed more leaks
- * * #6623: fixed more leaks
- * * #6623: fixed more leaks
- * * #6623: fixed leak, line-endings in imwidget
- * * #6627: zero-initialize pointers if they're going to be deleted
- * * #6628: don't leak calls on exceptions
- * Revert "audiortp: call join after calling stop on RtpThread"
- * sflphone-client: more constness
- * audiortp: call join after calling stop on RtpThread
- * * #6625: return 0 on successful completion
- * * #6624: fix segfault on servercallfailure
- * * #6621: Fixed double free, unlock mutex in ManagerImpl::terminate
- * * #6220: remove audio stream when peer hangs up
- * * #6596: AudioSymmetricSession shouldn't self-delete
- * resampler: grow internal buffers dynamically
- * merge up and down sampling => resampling
- * Leave test directory unchanged when running make check
- * audio algorithms : remove unused prototype
- * ringtone: detect codec from file extension
- * *AudioFile : simplify
- * * #6596: create local SDP on the stack, not the heap
- * * #6596: don't call Ost::Thread::terminate from dtor
- * audiofile: cleanup (samplerate -> unsigned)
- * remove unused func
- * samplerateconverter: cleanup
- * RingBuffer::Put() : remove unused return value
- * MainBuffer::putData() : remove unused return argument
- * audiolayer::putMain() : remove unused func
- * AudioLayer::putUrgent() : remove unused return value
- * * #6618: delete any remaining ringbuffers in destructor
- * RingBuffer::availForPut() : remove
- * * #6617: return from main rather than calling exit
- * MainBuffer::availForPut(): remove
- * RingBuffer: simplify
- * alsa : remove write only variable
- * fix memcpy declaration
- * bcopy(src, dst) -> memcpy(dst, src)
- * RingBuffer::Get() : remove constant volume argument
- * return a copy of the call ID, not just a reference.
- * MainBuffer::getDataById() : remove volume argument (always 100)
- * MainBuffer::getData() : remove constant volume argument
- * RingBuffer::Put() : remove constant volume argument
- * MainBuffer::putData() : remove constant (=100) volume argument
- * audiolayer: remove constant _defaultvolume
- * AudioRtpRecordHandler / AudioRtpSession : simplify
- * mainbuffer: fix test
- * iaxvoiplink : simplify
- * sip registration callback: fix a dbus crash
- * MainBuffer: simplify
- * AudioRtpFactory: return cached type of rtp session. The rtp session
- can have disappeared if the call was put on hold
- * AudioRtpFactory: remove unused setters
- * Fix launchpad builds
- * * #6611 : remove unused bandwidth codec information
- * * #6611: AudioCodec: remove useless/unused setters
- * make sure buffer string is initialized correctly
- * * #6596: declare certain destructors virtual
- * audiolayer : cleanup
- * Simplify doc build rules
- * * #6270: don't build dbus-api doc with make, should require make all
- * configure.ac: cleanup
- * Remove copy of dbus-c++ from libs/
- * * #6596: stop clock thread when peer hangs up
- * removed unused Fmtp.h
- * * #6595: more logical initialization order
- * * #6600 : fix account creation
- * * #6601 : fix configure.ac tests
- * remove unused variable
- * Don't mix stack and heap based allocations
- * Fix copyright (2009, 2008, 2009 -> 2008, 2009)
- * Fix warnings found by clang
- * * #6595: fix initialization order for AudioRTP
- * * #6592: removed typedef std::string CallID
- * * #6586: implement local g_slist_free_full for older glib versions
- * * #6579: fix memory leaks in client (there's a lot left)
- * ShortcutPreferences::setShortcuts() : simplify
- * Fix merge
- * * #6548: remove call to non thread-safe strerror()
- * AudioRtpFactory: each instance is associated to exactly one SipCall
- * create_audiocodecs_configuration() : make static
- * * #6269 : refactor AudioRtpSession
- * Fix AudioSymmetricRtpSession.h inclusion guard (cherry picked from
- commit c3081dce1cc1370d6d3558a4c4ef5cfac0d21caf)
- * * #6269: Rename AudioRtpSession to AudioSymmetricRtpSession
- * * #6574: Don't exit when connection to pulseaudio server fails
- * accountconfigdialog.h : remove some stuff from header
- * * #6560: fix configuration test
- * Fix warning in test
- * * #6560: don't hide password entry in security tab
- * * #6560: set initial password for SIP accounts
- * * #6506: remove useless pointer indirection
- * * 6560: password is now specific to IAX accounts
- * * #6560 : actually use, store, restore, transmit SIP credentials
- * * #6560: YamlEmitter: serialize sequences
- * YamlEmitterException: typo
- * ManagerImpl::computeMd5HashFromCredential() : simplify, fix memleak
- * * #6561: invite_session_state_changed_cb() : simplify
- * * #6561: More useful debug in VoIPLink::removeCall
- * * #6561 : fix ghost call reappearing in GUI after transfer
- * while -> for (make the code smaller)
- * * #6558 : Account::loadConfig() : move IAX code to IAXAccount
- * IAXVoIPLink::getAccountPtr : simplify
- * * #6554 : access the SIPVoIPLink directly, not per account
- * SIPVoIPLink is instanciated only once and is not associated to a
- single account
- * yamlnode: use const references when possible (still some left to do)
- * Account::_accountID: constify
- * VoIPLink: simplify, remove unused method
- * hudson test : no need to call run_tests.sh anymore
- * Remove AccountID type and AccountNULL define
- * Make check runs the test (no need to call run_tests.sh manually
- anymore)
- * gnome GUI: Fix tests
- * Revert "Move registration information from SIPAccount to
- SIPVoIPLink"
- * * #6392: pluginmanagertest: fix warnings reported by valgrind
- * * #6547 : remove unused exceptions
- * * #6547: CallManagerException: use runtime exceptions
- * * #6547: InstantMessageException: use runtime exceptions
- * * #6547: do not throw exceptions if some settings are not present in
- config file
- * * #6547: YamlParserException: use runtime exceptions
- * * #6547: VoipLinkException: use runtime exceptions
- * * #6547: YamlEmitterException: use runtime exceptions
- * * #6547: DTMFException: use runtime exceptions
- * * #6547: AudioFile: use runtime exceptions
- * * 6547: AudioZRtpSession: remove impossible error case
- * * #6547 : AudioRtpSession: remove impossible error case
- * * #6547: AudioZrtp: use runtime exceptions
- * * #6408 : send authenticationUsername to GUI
- * * #6408 : store/restore authenticationUsername from config file
- * SIPAccount: simplify
- * Move registration information from SIPAccount to SIPVoIPLink
- * SIPAccount::getAccountDetails : simplify
- * * #6540: yaml parser: simplify
- * sdp.cpp : fix a warning
- * * #6540: yaml parser : remove std::string typedefs
- * * #6540: Simplify yaml unserialization
- * * #6540 : add a Conf::ScalarNode constructor for booleans
- * setAccountDetails(): simplify
- * * #6408: store authentication username in daemon
- * * #6408: Be able to set the authentication username in the GUI
- * * #6507 : do not crash if the program is not sflphoned
- * Fix tests
- * macroify SIPAccount::unserialize()
- * Move all .cpp files from sflphoned target to libsflphone.la, except
- main.c
- * main() : simplify, return positive error codes
- * * #6507 : find codecs dir in build directory
- * * #6392: Sdp: move clean functions to destructor
- * AlsaLayer::adjustVolume() : simplify
- * alsalayer : reduce indentation
- * malloc/free -> new/delete
- * malloc/free -> new[]/delete[]
- * malloc/free -> new/delete
- * AudioSrtpSession: simplify base64 encoding
- * * #6392: Initialize std::string from pj_str_t correctly
- * * #6392: AudioRtpSession: Initialize remote port
- * Audio settings : Initialize _echoCancelTailLength and
- _echoCancelDelay(0)
- * Initialize variable
- * YamlParserException : fix use of stack variable after it has been
- deallocated
- * * #6392: fix memory leak in history
- * * #6392 AudioCodec : fix memory leak
- * * #6392 : fix memory leak in sip account
- * * #6408: clean up sipaccount (cosmetics mostly)
- * sipaccount.cpp serialize() : reduce number of lines
- * * #6392: invalid memory access
- * * #6392 : fix invalid memory access
- * * #6479: merged useful code from MimeParameters into Codec interface
- * * #6462: fixed hangup on IP2IP call
- * added run_daemon.sh script
- * test: remove unused variable
- * Remove functions only used by a failing test (cherry picked from
- commit fcf718cb75de7f1882dc61c07bb8d300dfa10f85)
- * * #6360 : make client tests build (cherry picked from commit
- 028b2835f040e51ab8ab979b32732b07b8798fce)
- * * #6360 : fix warnings in check_global test (cherry picked from
- commit 9e2bd6a7496dd64f6f48595e385760019aab1193)
- * * 6360: updated API calls in tests, but they're not building yet
- (cherry picked from commit 548f6f0f919b43772a3e9c667e5e292791281795)
- * Fixed include in tests (cherry picked from commit
- aeadc7525c1e31f936670ac8b02f0bcf387c38a8)
- * Remove unused variables and functions
- * IAX: fix warnings (cherry picked from commit
- fd7a113a11cac2cd9a7c36929e88ad28195c4c35)
- * Remove unused DEBUG define which interferes with logger.h (cherry
- picked from commit b2f72b91d0f43cb1dd94d138882a8caa9c841c24)
- * * #6392: no need to check for account NULLity since it is
- dereferenced above
- * * #6392: fix a memory leak, replace by stack allocation
- * * #6392: remove a variable assignement which confuses cppcheck
- * process_conference_participant_from_serialized() : remove unused
- function
- * * #6392: s/free/g_free/
- * * #6392: fix a memory leak in abookfactory_load_module()
- * * #6392: remove generate_call_id() used only once
- * * #6392: fix memory leak (opendir() without closedir())
- * * #6392: AudioRecorder(): ensures mbuffer is set
- * Remove SFLPHONED_VERSION from global.h, use autoconf PACKAGE_VERSION
- * #6298: Cleanup
- * #6331: Fix deleting ringtone file after call have been answered
- * * #6330: merged user_cfg into headers
- * #6298: Fix conference recording file update at conference end
- * #6298: Fix record file name serialization for conference
- * * #6295: cleanup of codec hierarchy
- * #6298: Fix gtk warnings
- * * #6300: added script to run tests
- * #6109: Add recording playback for conference
- * * #6300: tests do not require an installed sflphone
- * * #6295: re-removed clone methods
- * #6109: Fix gtk_critical warnings for incoming calls
- * #6109: Fix GTK_CRITICAL warning
- * #6109: Fix icons when history is not activated
- * #6109: Fix warnings
- * #6109: Implement stop recorded file playback signal
- * Revert "* #6295: removed unused clone method"
- * * #6295: removed unused clone method
- * * #6296: removed non existant file from Makefile.am
- * #6109: Stop fileplayback for outgoing call
- * #6109: Implement stop recording playback button
- * Fix binding names errors in dbus introspection file
- * #6109: Implement playback recorded file callback in client
- * #6109: Store recorded file path on client side
- * #6109: Add dbus methods for call recording playback
- * * #6290: remove unused classes from utilspp
- * * #6288: cleanup sdp
- * * #6288: fix exception usage
- * * #6288: simplify SdpException
- * * #6288: cleanup in sdp.cpp/h
- * #6109: Only display playback button if record file is set and valid
- * * 6290: updated configure.ac to remove functor Makefile
- * * #6290, #6289: removed unused classes from utilspp, fixed make
- check
- * #6109: Add button for history playback of recorded file
- * * #6289: removed unused observer class
- * * #6282: forward declare sdpMedia in sdp.h
- * * #6281: renamed setCallAudioLocal->setCallMediaLocal
- * #6183: Handle conference with more tahn two calls
- * #6183: Fix history icons when calling back a conference from history
- * #6183: Fix icons inconsistencies in history for conference hang up
- * #6183: Fix toolbar actions when selecting a conference in history
- * #6183: Fix conference serialization
- * #6268: Serialize only calls
- * * #6269: removed useless type testing
- * ignore some files in test/
- * * #6268: Remove dead class AudioSymmetricRtpSession
- * #6251: Do not had history calls in calllist when loading history
- file
- * #6251: Fix insertion in history map in before saving history file in
- daemon
- * #6251: Fix history unit tests
- * #6251: Order the list before serailization, get rid of the hashtable
- in history
- * #6251: Implement history serialization using a list wether than a
- map
- * * #6253: remove external audioport from header, make all members
- private
- * * #6253: don't store external local audio port (used for NAT) in
- Call
- * #6251: Add start_time timestamp in history serialization
- * #6251: Fix call insertion in conference items
- * #6233: Fix serialized account list terminated with a ";" character
- * #6238: Fix draggable history calls into current calls
- * #6233: Fix toolbar updates
- * #6233: Fix history
- * * #6235: remove pyc files from git tree
- * #6233: Handle cases when one or manuy calls are unreachable in
- createConfFomrParticipantList
- * #6233: Handle wrong numbers in createConferenceFromParticipantList
- * #6231: Fix drag-n-drop issue
- * * #6173 : move sippxml in tools
- * #6231: Fix merging issue
- * #6183: Implement conference unserialize
- * * #6212: remove extraneous flags from globals.mak
- * #6183: Unserialize conference data in conference
- * #6183: Add account information in request for conference call from
- history
- * #5755: Add -ldl to liker in sflphone-client-gnome
- * #5755: Fix fedora 15 compilation issue
- * #6183: Serialize conference participant phone number and account
- * #6183: Add conference timestamp in serialization
- * * #6186: don't include global.h, just logger.h
- * #6183: Fix saving history to file
- * #6183: Fix removing call from calllist
- * * #6184: remove pointers to Manager from AudioRtpSessions
- * #6183: Calling calltree_add_call explicitely for history
- * #6183: Ability to store conference inside history tab queue
- * * 6181: remove unused API from sipcall
- * #6171: Implment nreCallCreated callback
- * #6167: Fix participant list NULL ending
- * #6149: First draft of conference creation from history
- * #6149: Fix multiple call/conf selection callbacks ...
- * #6129: Fix place_call function called twice for pressing enter
- action
- * #6129: Fix double click action for history
- * #6149: Add dbus call for creating conference from history
- * #6129: Fix placing call from history and addressbook (still need to
- fix icon)
- * * #6148: removed unused AudioRtpFactory constructor
- * * #6145: remove unused isAudioStarted
- * * #6145: remove unused isAudioStarted
- * #6129: Add conference into history, fix call/conference selection
- * * #6143: don't use getType outside of serialization methods
- * * #6132: forward declarations instead of includes
- * * #6132: add constness, remove redundant "inline" keywords
- * #6129: Add timestamp to conference object to order history entries
- * * #6128: remove unused forward declarations from header
- * * #6127: make noncopyable class actually noncopyable
- * * #6125: don't include AudioRtpFactory in sipcall.h
- * #6123: Fix alsa ringback audio file
- * #6123: Fix raw audio file loading problem
- * #6109: Fix daemon plugin manager unit test
- * #6109: Fix history manager unit tests
- * #6109: Recording filename in daemon and client for history items +
- serialization
- * #6109: Refactor AudioFile to play recorded call
- * * #6104: AudioCodec moved to sfl namespace
- * * #6099: remove active flags from codec classes
- * #6095: Add notification-daemon as a runtime dependencies for rpm
- packages
- * #6095: Fix fedora 15 compilation in MineParameters.h
- * #6095: Declare static variable explicitely for client
- * #6095: Add logs to build OSC build machine
- * * #6098: global variables should have file-scope to avoid name
- conflicts
- * #6095: Fix compilation error for Fedora 15
- * #6095: Update SFLphone version to 0.9.14
- * #6095: Add specification file in opensusse build service for
- sflphone-plugins
- * #6073: Fix sflphone-plugins build on launchpad
- * #6093: Rename CodecDescriptor for AudioCodecFactory
- * * #6089: fix warnings in make check
- * * #6086: renamed codecs methods to audio_codecs
- * * #6085: renamed codec related dbus calls to audio_codec
- * #6065: Remove g_print from client, use DEBUG instead
- * #6065: Add actions name for addressbook
- * * #6085: renamed codecs* widgets/functions audiocodecs*
- * #6065: Fix Addressbook runtime warnings
- * #6065: Replace Codecs tab for Audio in account preference dialog
- * #6065: Fix "transfert" typo
- * #6065: Fix addressbook action runtime warning in uimanager
- * * #6082: fixes make check by adding libcrypto libs to test
- dependencies
- * #6073: Rename plugin/addressbook folders for addressbook/evolution
- in sflphone-plugins
- * #6074: Removed AC_SUBST from configure.ac when using
- PKG_CHECK_MODULE
- * #6073: Fix sflphone-plugins package build
- * #6073: Fix sflphone-common build
- * #6065: Fix runtime gtk warning when initializing searchbar without
- addressbook
- * #6063: Fix mozilla-tellify gitignore
- * #6063: Remove stream copy file using ifdef macro
- * * #6012: fix make dist for sflphone-common
- * #6063: Update .gitignore file
- * #6058: Fix base64 encoding related warnings
- * #6056: Fix SdpException handling
- * #6055: Fix unknown pargma warning for gcc <= 4.5
- * * #5949: test gcc version before disabling unused-but-set warning
- * #6054: Fix addressbook plugin compilation warning
- * #6048: Fix uimanager static initialization
- * #6046: Fix addressbook factory static initialization of member
- addrbook
- * #5979: Fix implicit function declaration warning
- * #6042: Fixed discarding qualifier warnings in client
- * #6041: Fix instant messaging unhandled case warning
- * #5994: Implement set current addressbook name and search type in
- addressbook plugin
- * #5994: add rules for launchpad packaging of addressbook plugin
- * #5994: Fix addressbook plugin configuration loading
- * #6027: Fix addressbook enabled test from configuration
- * #6027: No need of gnomedoc related macros in addressbook plugin
- * #6027: Add NEWS file required for build
- * #6027: Add addressbook plugin autogen.sh script
- * #6027: Remove plugins from client
- * #6027: Add sflphone-plugins folder at project's root level
- * #5994: Move addressbook folder from contacts to plugin folder
- * * #6011: removed unused Makefiles
- * * #6010: remove unused headers
- * * #5952: fix "string constant to char*" warnings
- * * #6009 fixed warnings
- * * #6003: finished cleanup of account classes
- * * #6003, #6004: cleanup of account classes, defaultAccount no longer
- global
- * * #6000: fix memory leak of args object
- * * #5998: removed using namespace std from networkmanager
- * * #5998: removed "using namespace std" from ZrtpSessionCallback
- * * #5998: removed using namespacestd from AudioZrtpSession.h
- * * #5998: remove "using namespace std" from auriorecord.h and
- MimeParameters.h
- * * #5998: remove using namespace std in main
- * * #5998: removed "using namespace std" from logger
- * * #5949: test gcc version before disabling unused-but-set warning
- * #5994: Installation of addressbook plugin
- * #5979: Implement codec full addressbook search from plugin
- * #5979: Implement addressbook factory and plugin
- * * #5981: unused webwidget removed
- * #5966: Account config synchronization fix (for stun)
- * #5954: Handle media name exception
- * #5954: Fix audio codec name display in client
- * #5954: Clean up getSessionMedia methods
- * * #5957: getRecordingSmplRate returns a value
- * #5954: Clean up getCurrentCodec methods
- * * #5950: remove "converting to non-pointer type 'int' from NULL"
- warnings
- * #5915: Full gain control version
- * * #5949: remove more unused variable warnings
- * * #5949: remove unused/unused-but-set variable warnings
- * * #5949: show_preferences_dialog returns a success value
- * * #5946: cleanup of include directives, undefined function
- * * #5515: comment out SSLv2 calls in pjsip
- * #5915: Implement different slope for attack tme and release time for
- gain control
- * #5915: use only one input signal for gain control (removed output
- buffer)
- * #5921: Fix no audio after holding a conference
- * #5916: Add gaincontrol files
- * #5916: Implement FFMPEG/CCRTP video streaming prototype
- * #5903: Fix call transfer during a conference
- * #5915: implement rms detector, first order averager, limiter for
- gain control
- * #5914: Fix call transfer when no notification request is required
- * #5899: Fix conference right-click segfault
- * #5884: temporary fix segfault in pjsip memory pool
- * #5883: Fix compilation issues on maverick and lucid
- * #5755: Fix fedora 15 compilation without patching ccrtp
- * [#5855] Make echo canceller optional
- * #5855: Fix echo suppression activation/deactivation
- * #5855: Implement pjsip echo canceller
- * #5814: Speex initialization function uses samples, not bytes
- * #5814: Test using more unbalanced signals
- * #5814: Fix buffer size for long echo length or long echo delay
- * #5814: Adjust level for echo cancellation at runtime
- * #5814: Process noise reduction before echo cancelling
- * #5814: Implement speex post echo canceller processing
- * #5814: Dump echo cancel file to disk
- * #5814: Add parameters for echo cancel
- * #5809: Add configuration parameters
- * #5809: Implement speex echo canceller in audio rtp session
- * #5814: Code cleanup
- * #5814: Fix conf creation with several incomming ringing calls
- * #5814: Fix conf creation segfault when dragging a call on hold on a
- ringing call
- * #5809: Added unit test for echo cancellation and implemented
- "process" virtual method
- * #5709: Add always recording option in configuration
- * #5709: Add always recording option in audio conference panel
- * #5709: Add core functionnality for always recording (missing config
- options)
- * #5769: Fix conference participant handling (detach/attach) and hold
- actions
- * #5747: Fix recording icons and state for conference when adding new
- participant
- * #5769: Code cleanup
- * #5769: Fix hangup unsent calls
- * #5769: Fix remove/add additional participant to conference
- * 5769: Several fixes concerning confererence handling
- * #5769: Fix compilation error
- * [#5769] Fix audio streams binding in main buffer
- * #5769: Removed access to audio mixer from audio layer
- * #5765: Fix audio crash for illformated wavefiles
- * #5765: Add maximum iteration for finding fmt and data "chunck"
- * #5589: Fix compilation of libnotify under
- * #5757: Fix abort signal when receiving INFO
- * #5747: Add usersDetached.svg
- * #5747: Handle offhold action for recording conference
- * #5747: Fix off hold action for conferences
- * #5747: Implement update conference in record action in calltree
- * #5747: Add new icons for recording conferences
- * #5747: Add recording state for conferences
- * [#5738] Remove getAudioDriver call from manager (replace by
- _audiodriver var)
- * [#5738] Refactor mutex protecting audiolayer
- * [#5737] Fix HD conference recording
- * [#5730] Fix start audio session after changing sampling rate
- * [#5714] Fix enter keyboard event for addressbbok and history
- * [5695] Fix addressbook combo box update when no addressbook selected
- * [#5695] Fix addressbook initialization and search bar update
- * [#5695] Add mutex for books_data in addressbook to protect async
- calls
- * [#5695] Get back addressbook open from uri
- * [#5695] Fix absolute addressbook URI for local addressbooks
- * [#5695] Implement libebook 3.0 interface
- * [#5571] Better logic for hangup (for case where call have not been
- sent yet)
- * [#5571] Update error handling in voip links
- * [#5571] Fix compile time warnings
- * [#5696] Fix installation dependencies for Natty
- * [#5669] Add mention that sflphone.org is for testing only
- * [#5693] Add natty in teh dput.conf file
- * [#5690] Remove not useful logs
- * [#5670] Use dynamic payload type for rtp dtmf
- * [#5668] Clean up sflphone configuration logging
- * [#5668] Fix hook checkbox configuration update
- * [#5666] Fix unit tests
- * [#5666] Manage event subscription
- * [#5666] Emit bye request when subscription is terminated
- * [#5666] Bye request should be sent after event subscription
- notification is done on transfer
- * [#5666] Make reinvite method static (to be called in pjsip
- callbacks)
- * [#5666] Hangup Call in manager for AccountNULL and IP2IP
- * [#5589] Use PKG_CHECK_MODULE for every client's dependencies
- * [#5623] Enlarge initial size of pjsip memory pool for calls (16k)
- * [#5564] Fix audio recording resampling for g722
- * [#5571] Move attribute handling for onhold/offhold actions in SDP
- session
- * [#5571] Codec negotiation refactored and unittested
- * [#5571] Implement tests
- * [#5571] Implement pjsip negociator
- * [#5571] Fix unit tests
- * [#5571] Add Fmtp.h to repository
- * [#5571] Integrate mime types and codec factory
- * [#5571] Handle exception when SDP negotiation fails
- * [#5570] Add sflphoned-sample.yml in repository
- * [#5564]: Implement stereo to mono mixing for rigntone
- * [#5342] Update audio stream initialization
- * [#5514] Restore test ni historytest suite
- * [#5514] Fix
- * [#5514] Disable test_create_history_path
- * [#5514] use pulseaudio in sample config file
- * [#5514] Fix test: load history from file
- * [#5514] Do not use X
- * [#5513] Make unit tests compile successfully
- * [#3947] Enable unit tests in Jenkins
- * [#5454] Fix build system to handle new version number
- * [#5454] Update languages from launchpad
- * [#5454] Add --without-celt in OpenSuse build service
- * [#5454] Change version number
- * [#5331] Added first SDP session tests
- * [#5273] Update nightly build version tags to conform dpkg rules
- * [#5211] Refactor send register method for iaxvoiplink and
- sipvoiplink
- * [#3950] Remove call being transfered from calltree
- * [#5211] Use appropriate memory pool for transport selector
- * [#5211] Fix strict aliasing rules warning in pjsip
- * [#5211] Bring back pjsip shutting down sleep to 1000 ms
- * [#5211] Fix registration callback segfault when closing the
- application
- * [#5211] Use the dialog memory pool for Route header in INVITE
- request
- * [#5211] Add temporary memory pool for findLocalAddressFromUri and
- findLocalPortFromUri
- * [#5211] Use individual memory pool for dtmfs
- * [#5211] SipVoipLink refactoring
- * [#3950] Attended transfer for conference calls
- * [#5284] Fix DNS resolution for Route with specified port number
- * [#5284] Some code cleanup
- * [#3947] Fix typo in hudson script
- * [#5284] Added sip route to REGISTER, INVITE, BYE request, plus DNS
- resolution
- * [#5266] Use RTP dtmf as default
- * [#5284] Added pjsip_process_route_set after setting routes in regc
- structure
- * [#5286] Fix parsing error due to long configuration file (removed
- max event)
- * [#5286] Fix false test in configuration emmiter
- * [#5286] Code cleanup
- * [#5286] Updated exception handling in configuration system
- * [#4969] Fix put SRTP call on hold
- * [#3950] Add debug messages
- * [#3950] Ability to perform an attended transfer
- * [#5276] Fix initialization problem in g722
- * [#3950] Add replace header in SIPVoIPLink::transferWithReplaces
- method
- * [#3950] Implemented attended method in SIPVoIPLink
- * [#3950] Cleanup transaction request received callback
- * [#3950] Implement dummy attended transfer in gnome-client
- * [#5249] Fix audio samplerate update algorithm for g722
- * [#5249] Fix uninitialized variable used in conditional jumps
- * [#5249] Fix conditional jump error in audiolayer (uninitialized
- value)
- * [#5267] Use autoconf 2.65 as a requirement (instead of 2.67)
- * [#5267] Restore manual pjsip configuration and compilation
- * [#5267] Autodetect celt version (0.9.1, 0.7.1)
- * [#5267] Fix deprecated macros in gnome client configure.ac
- * [#5267] Update configuration for libcelt-dev
- * [#5267] Fix build autoconf and automake
- * [#5227] Deactivate automatic call to astyle after compilation
- * [#5242] Hangup every calls before leaving
- * [#5237] Will now nightly-build for natty, Karmic deprecated
- * [#5229] Use inner class for rtp thread instead of inheritance
- * [#5211] Move mainbuffer unbind call in rtp final method
- * [#5211] Initialize sip call memory pool using 16 kb
- * [#5211] Use call memory pool in session reinvite
- * [#5211] Add debug messages
- * [#5211] Use and internal pool for calls
- * [#5211] Reduce pjsip memory pool usage for stateless error messages
- * [#5211] Refactor call deletion
- * [#5212]
- * [#5208] Refactor codec management for accounts
- * [#5168] Remove printf from codec's encode & decode method
- * [#5168] Fix celt compilation on launchpad
- * [#5168] Fix sflphoned compilation warnings in audiocodec.h
- * [#[#5168] Must keep the g722 specific RTP rate to avoid incoming
- packet timeout
- * [#5168] Fix static/dynamic payload rtp session update
- * [#5168] Throw SIPVoipLink Error if codec not instantiated in new
- outgoing call
- * [#5168] Fix dynamic/static codec payload type ambiguity
- * [#5169] Fix doubled IP2IP profile when no config file
- * [#4867] Add gtkinfobar in configuration panel
- * [#4867] Disable input/output/ringtone selection when using default
- alsa plugin
- * [#4952] Patches for possible buffer overflows
- * [$4885] Fix schemas problem
- * [#4885] sflphone-client-gnome.schemas not present during build
- * [#4885] Add gconf shemas directories in opensuse build system
- * [#4885] Add file/folder ownership for opensuse-factory build system
- * [#4906] Fix opensuse-factory build
- * [#4885] Update name dependency for libedataserver
- * [#4885] Fix non-void function without return in dbus-c++
- * [#4895] Update language translation
- * [#4896] Update session timestamp when updating media
- * [#4896] Reapply RTP hack for G722 payload type
- * [#4896] Update recording sampling rate when updating codec
- * [#4897] Save codecs in config for each configuration changes
- * [#4895] Do not save config when sflphone quit
- * [#4885] Update date for copyright
- * [#4885] Deactivate siptest that require more than one sipp instance
- * [#4879] Remove inmcoming call notification from IAX
- * [#4885] Some cleanup
- * [#4874] Add setCancel immediate/deffered for ost::Thread
- * [#4879] Fix incoming call notification
- * [#4878] Set keyboard focus on searchbar when selecting addressbook
- * [#4874] Fixed compilation warning
- * [#4874] Fixed compilation warning in sipvoiplink
- * [#4874] Fix compile time warning in RTP record handler
- * [#4874] Fix conditional jump in SDP
- * [#4874] Fix conditional jump based on uninitialized value
- * [#4874] Store call id within rtp thread context
- * [#4874] Fixed conditional jump based on uninitialised value in
- conference
- * [#4871] Fix default account fetching
- * [#4870] Delete RTP session when Refusing an incoming call
- * Restore IP to IP call
- * [#4857] Fix audio codec negotiation problem
- * [#3947] Adjust ressources allocated to compilation
- * [#3947] Disable unit tests in Hudson
- * [#4305] Free mutex only when really quiting SFLphone
- * [#4859] Update copyright to 2011 in every source file
- * [#3218] Character '.' stripped by the caller engine
- * [#4854] Fix typos, desktop entry
- * [#4847] Apply RTP modification to ZRTP session
- * [#4852] Update Karmic and Lucid dependencies
- * [#4852] Add Libedataserver and libedataserverui as gnome client
- dependencies
- * [#4852] Add authentication mechanism for EDS
- * [#4851] Fix segfault when closing pulseaudio layer too rapidly
- * [#4808] Some otehr cleanup
- * [#4808] Made some cleanup
- * [#4808] Added mutex in rtp session for codecs and noise process
- * [#4847] Update audio processing when updating RTP media
- * [#4842] Add support for linking with gold/ld --no-add-needed
- * [#4808] Make update g722 related static/dynamic payload logic
- * [#4827] Upper limit on the number of contacts to import from EDS is
- hard-coded to 500
- * [#4808] Fix put call on/off hold
- * [#4808] Implement early RTP start for incoming calls
- * [#4808] Audio stream is no longer start within RTP session.
- * [#4808] Removed coupling between audio layer and and RTP session
- * [#4702] Start audio rtp session as soon as it is created
- * [#4702] Init timestamp to 0
- * #4702: Send RTP packets immediately, no need of outgoing queue
- * [#4784] Update dbus-c++ version from gitorious
- * [#4702] Update RTP timeouts
- * [#4702] Lengthen RTP timeouts
- * [PATCH] Fixed compatibility with old libtool versions.
- * [PATCH] Accept older libebook (Maemo 5 has 1.4.2)
- * [PATCH] Fixed double-free error in preferences dialog
- * [PATCH] Fixed building of sflphone-common on Maemo5
- * [PATCH] Improved Gnome client initialization error handling. 1. It
- no longer segfaults when sflphoned isn't available. 2. User is
- provided with GUI error dialog.
- * [PATCH] Improved autogen.sh scripts 1. They do not require bash
- anymore 2. Added workaround for Debian bug #565663 3. Replaced
- manual autotools invocations with single autoreconf call 4. Non-zero
- return status on failure
- * Revert "[#4468] libtool <= 2.2 doesn't have LT_INIT macro so
- AC_PROG_LIBTOOL should be used instead."
- * Revert "[#4468] Libebook 1.4 is sufficient"
- * Revert "[#4468] Apply big path on dbus communication system"
- * [#4468] Apply big path on dbus communication system
- * [#4468] Libebook 1.4 is sufficient
- * [#4468] libtool <= 2.2 doesn't have LT_INIT macro so AC_PROG_LIBTOOL
- should be used instead.
- * [#4639] Fix determining default addressbook if this property is not
- set in gconf
- * [#4639] Fix memory leaks in Addressbook
- * [#4637] Fix opening default addressbook at sflphone init
- * [#4622] Free yaml events while parsing configuration file
- * [#4623] Fix conditional jumps based on uninitialized variable
- * [#4622] Fix leaks in yaml serialization engine
- * [#4616] Fix addressbook warnings
- * [#4514] Adjust RTP timestamp
- * #4527: Rename Karmic libyaml and Celt package in debian control file
- * #4495: Rework addressbook opening loop
- * [#4524] Increment RTP count when sending data
- * [#4524] DO NOT start RTP session twice
- * [#4367] Use PKG_CHECK_MODULE for celt
- * [#4367] Fedora package celt as celt (not libcelt)
- * [#4367] Astyling
- * [#4367] Update .po files
- * [#4367] Fix segfault in gensin
- * [#4354] Make celt a direct dependency on launchpad opensuse build
- service
- * [#4367] Make celt a required package, option --without-celt valid
- * [#4367] Fix zrtp timestamping error
- * [#4367] Fix audio zrtp timing
- * [#4367] Dispatch ZRTP packets
- * [#4367] Fix segfault when unloading account map
- * [#4367] Fix zrtp session
- * [#4367] Implement on packet receive
- * [#4367] use symetric audio rtp session, not dual
- * [#4367] Reduce packet receive/sent timeout
- * [#4367] Reduce RTP timeouts
- * [#4367] Move speaker data receive
- * [#4367] Move speaker data receive
- * [#4367] Move receive speaker data method
- * [#4367] Remove debug in rtp session
- * [#4367] Fix g722 codec clock rate
- * [#4367] Fix noise suppression initialization
- * [#4367] Fix segfault in RTP mic fadein method
- * [#4367] Refactor mic data encoding in rtp session
- * [#4367] Implement RTP main loop
- * [#4367] Fix compilation problem
- * [#4367] Fix AudioRtpclass using TRTPSessionBase
- * [#4367] Fix AudioRtpSession putDtmfEvent shadowing
- * [#4367] Fix AudioRtpSession putDtmfEvent shadowing
- * [#4367] Refactor RTP session (phase 2)
- * [#4367] Refactor RTP session (phase 1)
- * [#4367] Remove Redeclaration of SymetricAudioRtpSession in
- rtpfactory
- * [#4265] Add continue statement in for loop for invalid addressbook
- * [#4261] Makes addressbook initialization more robust
- * [#4257] Add maverick in build system
- * [#4233] Add sdp related unit tests
- * [#4233] Add condition and signal in two incoming call test
- * [#4243] Fix segfault in AudioSrtpSession
- * [#4243] Fix memory leak in AudioSrtpSession
- * [#4243] Make audio srtp optional in for incoming call
- * [#4243] Add boolean variable to make sure remote crypto context
- initialized only once
- * [#4243] Add documentation to AudioSrtpSession
- * [#4243] Use 80 bits authentication tags by default
- * [#4243] Init audio srtp remote crypto context in
- call_on_media_update
- * [#4243] Move SDP negotiastion in mod_on_rx_request
- * [#4243] Implement initLocalCryptoInfo to be called at different
- momment
- * [#4243] Init init local crypto context in when initializing audiortp
- * [#4243] Change key length according to sdes negociation
- * [#4243] Associate callid to accountid for incoming calls
- * [#4242] Fix no SDES keys in IP2IP calls
- * [#4242] Fix no SDES keys in IP2IP calls
- * [#4233] Test for call on/off hold
- * [#4233] Add two incoming call test
- * [#4233]
- * [#4233] Add 2 outgoing simultaneous call unit tests
-
- -- Julien Bonjean <julien.bonjean@savoirfairelinux.com> Fri, 30 Sep 2011 13:57:10 -0400
-
-sflphone-plugins (0.9.7~rc1~ppa1~SYSTEM) SYSTEM; urgency=low
-
- ** 0.9.7~rc1~ppa1~SYSTEM **
-
- * [#2462] Set explicitly the transport on incoming call too
- * [#2462] fix typo
- * [#2462] Use different address for SDP and call IP
- * [#2462] Use published address in SIP-SDP
- * [#2181] Fixed changelog files
- * [#2181] Updated spec file
- * [#2402] Fix pointer to int conversion warning (atoi)
- * [#2402] Remove daemon warnings, make indent
- * [#2459] Make sure the stream is opened when the call is answered
- * [#2402] Add conference related picture in documentation
- * [#2443] Not much ...
- * [#2399] Fix dialing display problem
- * [#2450] Fix incoming call already in conference crash
- * [#2399] Display peer name on the first line and peer number on the
- second
- * [#2450] Handle 403 FORBIDDEN when refused
- * [#2447] Bind offHold/onHold actions to button in gtk client
- * [#2447] Bind hangup action to button for conference
- * [#2447] Add conference action in gtk client's ToolBar
- * [#2381] Disable the password hashing in config file
- * [#2402] Cleanup
- * [#2366] Set callback to null when deleting Pulseaudio streams
- * [#1313] Fix main buffer unit test
- * [#1313] Fix audio layer unit test
- * [#2315] Hide pw in security tab, display when editing, sync with
- basic tab
- * [#1313] UnitTest change AudioRtpSession for AudioSymetricRtpSession
- instance
- * [#2402] Code cleanup
- * [#2444] Add debug to catch occasional crash when loading client's
- config
- * [#2444] Add debug info to catch occasional crash when loading config
- dialog
- * [#2402] Restore Call menu translations
- * [#2403] Use the published address if checked in GUI
- * [#2442] Add protection test in sdp
- * [#1841] Reapply pjsip patch concerning DNS SRV resolution
- * [#2384] Tags incoming call as direct SIP call, if applicable
- * [#2402] Change the monkey face
- * [#2315] Enable user to display password in clear text
- * [#2434] Force optimization level at 2
- * [#2284] Fix dbus_get_all_ip_interface compilation warnings
- * [#2431] Popup main window on incoming if applicable
- * [$2402] Fix simple warnings
- * [#2402] Fix implicit variable init order in LibraryManagerException
- * [#2402] Fixing implicit variable initialization warnings in
- AudioRtpSession
- * [#2402] Revert atoi change, fixing codec list doubled entries
- * [#2402] Fix gpointer to gint conversion
- * [#2402] Fix pointer casting to integer different size warning in
- codec list
- * [#2402] Fix warning discarting qualifiers from pointer target
- * [#2402] Fix gtk tree view assignement from incompatible type warning
- * [#1669] Fix audio recording folder utf-8 non compatibility issue
- * [#2414] Clean up debugs
- * [#2414] Use transport set in iptoip Account and update it frm
- preference
- * [#2348] Use macro N_() to mark ui.xml strings as translatable
- * [#2414] Rename getSipAddress/setSipAddress functions
- * [#2407] Fix volume controls display
- * [#2407] Fixes dialpad
- * [#2383] Set ip to ip config when clicking apply button
- * [#2404] Update call-to script - Maxime Chambreuil
- * [#2405] Client handles unknown call in current state as well
- * [#2383] Add DBUS signal to send IPtoIP local address and port as
- string
- * [#2383] Add Ip to IP config change apply call back
- * Clonflict
- * [#2402] Code cleanup
- * [#2383] Do the same for IPtoIP (init localn ip with first in the
- list)
- * [#2383] Use first interface in the list if local addresss is not
- defined
- * [#2403] Clean up unuseful addresses/ports
- * [#2403] Use the IP profile SIP port as global SIP port
- * [#2383] Fix dbus_get_all_ip_interface warnings
- * [#2383] Take into account sameAsLocal when loading published address
- * [#2383] Tsake into account sameAsLocal option when saving published
- address
- * [#2383] Update local ip address in ip to ip config
- * [#2383] Save ip 2 ip local port in config
- * [#2406] Update toolbar at startup
- * [#2284] Remove redefinition warnings + speex warnings
- * [#2383] Fix security table in account config
- * [#2383] Save ip 2 ip network interface parameters in config
- * [#2403] Restore sip transport selector
- * [#2383] Fix filling the Localt IP Address on account creation
- * [#2383] Fix Gtk-Critical when checking STUN
- * [#2383] Fix reopening account configuration display issue
- * [#2383] Load IPtoIP local address and port in preference iptoiptab
- * [#2383] Add LocalAddress and Localport in Preference IpToIp tab
- * [#2403] Use the address and port associated to the account as often
- as possible
- * [#1753] Removed pjsip generated files
- * [#1753] Removed remaining milenage lib references
- * [#2383] Add _publishedSameasLocal variable in sipaccount
- * [#2383] Add PUBLISHED_SAMEAS_LOCAL variable in config
- * [#2383] Fix stun set active or not when opening config
- * [#2181] Added RPM 64bits dbus patch
- * [#2402] Code indentation
- * [#2313] Force $(HOME).cache directory creation at startup
- * [#2383] Separate network interface and published address in account
- config
- * [#2400] Change dbus service installation path to libdir
- * [#2382] Move TLS related published address options in security tab
- * [#2382] Indent accountconfigdialog.c
- * [#2181] Install libdbus-c++ in $pkglib instead of $lib
- * [#1753] Remove ILBC code and disable it by default in the configure
- * [#1753] Remove milenage directory
- * [#2382] Fix switching interaface instabilities
- * [#2396] Save local ip in account creation wizard
- * [#2284] Remove warning on hold
- * [#2387] Fixes history searching and filtering
- * [#1215] Add samplerate display in the GUI
- * [#1663] Voicemail icon reflects voice messages
- * [#2395] Fix account registration ( specifically with callcentric)
- * [#2386] Strip "sip:" on incoming call, fixing history call back
- * [#2181] Updated spec files
- * [#1215] Display codec name in calltree instead of status bar
- * [#2390] Move back nbCalls and stopStream higher in refuseCall
- * [#2392] Fix ringtone during call in IAX
- * [#2391] Stop audio streams when there is 0 calls only
- * [#2391] Add debug when call state is not valid
- * [#2390] Clear returns in IAXvoipLink::sendAudioFromMic() method
- * [#2380] Fixing IncomingCallNotification not regular
- * [#2339] Query conference at client startup
- * [#2339] Working conference querying at startup
- * [#2339] Add conference in call tree
- * [#2339] Primitives to query conferences at client startup
- * [#2320] Add account selection in history
- * [#2355] Temporary solution: do not delete pointer when removing
- account
- * [#2380] Change algorithm in AudioRtp to trigger an
- IncomingCallNotification
- * [#2274] Comment sdebug in MainBuffer flush method
- * [#2274] Add flushMain() in ManagerImpl::addStream
- * [#2274] Add getBufferID() method in ring buffer
- * [#2274] Fix warning, comment debug in ringbuffer's flush method
- * [#2274] Use AudioLayer flushMain() and flushUrgent() in ALSA
- * [#2274] Clean up unused variable warning
- * [#2274] Protect minbudffer pointer on flushing
- * [#2274] Fix playATone method which writing empty buffer in urgent
- ringbuffer
- * [#2274] Use audio layer flushUrgent and flushMain in createStreams
- * [#2274] Use flush audio calls from audiolayer
- * [#2274] Flush when peer answered call
- * [#2375] Flush main buffer in iax when answering a call
- * [#2274] Parse displayname using c++ string method
- * [#2375] Flush main buffer when off holding calls
- * [#2375] Flush main buffer mon RTP startup
- * [#2376] Use now Pulseaudio module-cork-music-on-phone
- * Updated OSC packaging
-
- -- Julien Bonjean <julien.bonjean@savoirfairelinux.com> Fri, 20 Nov 2009 13:59:02 -0500
-
-sflphone-plugins (0.9.7~beta~ppa1~SYSTEM) SYSTEM; urgency=low
-
- ** 0.9.7~beta~ppa1~SYSTEM **
-
- * [#1933] Cleanup debug
- * [#1933] Clean up debug
- * Fix mic
- * [#1933] Set the IAx format earlier
- * [#1933] Move IAX sendAudioFromMic outside if (call) statement
- * [#1933] Fix startstream when offhold in iax and add debug concerning
- codec neg.
- * [#2371] sflphone_notify_voice_mail: minor gettext message formatting
- cleanup
- * [#2371] select_account_cb: properly gettextize status message
- * [#2371] show_account_list_config_dialog: properly gettextize status
- message
- * INSTALL: Minor tidyup of core install guide
- * Add /sflphone-client-gnome/src/icons/Makefile to .gitignore
- * [#2181] Updated OpenSUSE files (tmp)
- * [#1933] Add debug for codec negociation for iax
- * [#1933] Get rid of getMicAvail and getMicData in audiolayer (not
- used anymore)
- * [#1933] Add "audio codec not determined" error in IAX
- * [#1933] Test flush data
- * [#1933] Do not need to start audio stream in iax anymore
- * [#1933] Protecting pointer
- * [#2284] Remove more compilation/execution warnings
- * [#2284] Cleanup debug in client, use DEBUG instead of g_print
- * [#2284] Clean up uimanager
- * [#2370] Remove warnings
- * [#2366] Clean up other debug
- * [#2366] Clean up debug
- * [#2366] Call pa_xfree explicitely in writeToSpeaker
- * [#2284] Remove address book warnings
- * [#2365] Fixes bad cast
- * [#2352] Fix continuous ringing when peer hangup and call not yet
- answered
- * [#2181] Added version support
- * [#2181] Fixed some minor issues
- * [#2360] Moved MainBuffer from AudioLayer to ManagerImpl
- * [#2352] Makes getMainBuffer() everywhere
- * [#2352] Use 50 sec latency on pulseaudio stream creation
- * [#2352] Add alsa debug
- * [#2359] Update repository documentation
- * [#2354] Move pulseaudio disconnectAudioStream after stopping main
- loop
- * [#2352] Adjust nb byte copied in pulseaudio according to
- writeableSize
- * [#2352] Specify pulseaudio tlength parameters using pa_usec_to_bytes
- * [#2322] Convert italian translation to UTF-8
- * [#2357] Fixes window size
- * [#2357] Display only actionnable tool item
- * [#2333] Update streams parameters
- * [#2347] Use GNOME user settings for Menu and Toolbar appareance
- * [#2349] Load/Save properly audio params
- * [#2322] Update translations from Launchpad
- * [#2181] Added Francois Marier script
- * [#2350] Remove non-valid test
- * [#2181] Updated launchpad packaging
- * [#2333] Fix Pulseaudio Capture
- * [#2333] Use pulseaudio ADJUST_LATENCY flag and ALSA RT-SCHEDULING
- * [#2333] Pulseaudio Interpolate timing
- * [#2333] Change (again) Pulseaudio settings to fit logiteck usb hdw
- requirement
- * [#2333] Adjust pulseaudio fragment size to 4096 (max sflphone's
- frames per buffer)
- * [#2284] Remove recurrent compilation warning (g++ linker problem)
- * [#2333] Safer Audiostream parameters
- * [#2333] Fix alsa playback to reduce underrun
- * [#2333] Better audiostream parameters
- * [#2181] Updated version management
- * [#2333] Exclusive test in playback loop
- * [#2181] Updated build system
- * [#2333] Less underrun with these value
- * [#2333] Update playback audiostream parameters
- * [#2333] Lengthen the audio buffer reduce number of underrun in
- pulseaudio
- * [#2333] Add ALSA recovery functions for underrun (begin)
- * [#2333] Add pa_stream_trigger in pulse audio underrun callabck
- * [#2048] Reduce prebuffering in pulseaudio (which affect incomming
- calls' plbck)
- * [#2316] Do not display any icons to the right on the history tab
- * [#2333] Comment pa_stream_trigger in pulseaudio underrun
- * [#2333] Modify pulseaudio streams parameters
- * [#2318] Fix transfer tool button double signal
- * [#2181] Updated
- * [#2333] Fix ALSA ringtone
- * [#2333] Flush all main buffer before starting audio
- * [#2333] Open/Close Alsa thread between calls while there is no audio
- * [#2333] Add debug message and test condition on starting playback
- and capture
- * [#2181] Fixed gnome client makefile
- * [#2181] Updated
- * [#2308] Remove getTelephoneTone debug
- * [#2308] Change plughw for default in ALSA
- * [#2308] Oups, forgot to change function name in audiolayertest.cpp
- * [#2308] Cleanup in pulseaudio code (debug, function name)
- * [#2308] Fix pulseaudio stream closing assertion failure
- * [#2308] Moved pulseaudio mainloop locking from AudioStream
- disconnect stream
- * [2308] Fix latency at the beginning of a call, when playing DTMF and
- wehn starting tone
- * [#2181] Updated karmic
- * [#2317] [#2319] Fix address book toggle button contextual behaviour
- * [#2308] Stop stream when refusing a call
- * [#2308] Stop pulseaudio stream when peer hungup
- * [#2308] Fix tone and ringtone
- * [#2312] Display the STUN entry widget when opening the tab
- * [#2308] Implement two different callbacks for capture/playback in
- pulseaudio
- * [#2309] Open/close pulseaudio connections in startStream/stopStream
- * [2308] Leave pulseaudio stream running, do not cork/uncork them
- anymore
- * [#2295] Set gtk file chooser to None if nothing is set in
- configuration
- * [#1976] Add codec and conference documentation
- * [#2209] Fix recording in regard of resamling
- * [#2297] Update .gitignore
- * [#2297] Update translation files
- * [#2297] Add reference to our coding standards
- * [#2297] Remove old docbook code
- * [#2296] Reinit tls account settings after modification
- * [#2253] Add DcBlocker class to remove capture's dc offset
- * [#2034] Fixes for TLS transport to initialize
- * [#2284] Add silent build rule + client clean warnings
- * [#2274] Fix unserialize history items in cilent at startup
- * [#2274] Complete display name parsing and displaying
- * [#2274] Parse the Display Name in sip INVITE message
- * [#2050] Fix capture volume control in ALSA
- * [#1970] Volume controls disable when using pulseaudio
- * [#1970] Disable volume controls when using pulseaudio
- * [#2277] Fix direct ip2ip ZRTP enabling/disabling in ip2ip
- preferences
- * [#2181] Added launchpad debian files
- * [#2181] Added spec files for OSC
- * [#2274] Set display name for "Contact" sip header as the hostname
- * [#2181] Fixed daemon issues
- * [#2181] Fixed gnome client issues
- * [#1976] Remove warnings - need to fix the transfer
- * [#2006] Add init is_rec variable in ManagerImpl
- * [#2006] Update codec display on call selection
- * [#2006] Restore double click actions in history and contact calltree
- (GTK)
- * [#2176] use XDG_CACHE_HOME when initializing sfl.zid file
- * [#1976] Fix calltree switching from history
- * [#2209] (Re)Fix cache for zid
- * [#2209] Clean up debug messages
- * [#2209] Clean debug messages
- * [#2209] Fix trasnfering a call during a conference
- * [#2209] Speex decode must return the number of bytes
- * [#2209] Change frameSize speex 32kHz
- * [#2209] Fix speex codec framesize
- * [#2209] Reinit converterSamplingRate in RTP sessions
- * [#2209] Change speex ultra wide band framesize
- * [#1747] Add pixmap data
- * [#2252] Fix Receiving a server error 488 crashes the callee
- * [#2209] Fix iax low rate packate sending
- * [#2209] Clean up debug messages
- * [#2209] Add resampling changes for IAX
- * [#2209] Clean up resampling code
- * [#2209] Fix latency introduced by pulseaudio
- * [#2209] Fix initialization of mainbuffer's internal sampling rate
- * [#2176] Fix upsampling buffer size in audiolayer
- * [#2209] Add dynamic converter sampling rate in audiortp sessions
- * [#1747] Fixes runtime warnings
- * [#1747] Remove from repo
- * [#1747] register our icons to be used as stock icons
- * [#2209] Fix number of byte in alsa's write to speaker
- * [#2209] Fix putting non-resampled data in RTP's mainbuffer
- * [#2209] Add alsa resampler
- * [#2209] Add a samplerate converter in PulseLayer
- * [#2209] Add mainbuffer's internal sampling rate and flushall method
- * [#2176] Add mainbuffer stateInfo debug method
- * [#2209] Resampling is optimal using SRC_LINEAR not SRC_FASTEST
- * [#2176] Remove debug recordings
- * [#2176] Fix Holding a conference participant on new calls
- * [#2224] Add confID in callable object
- * [#2176] Fix putting onhold a call participating to a conference when
- pressing new call
- * [#2176] Reset auidio buffers when adding streams (rtp, audiolayer)
- * [#1976] Use xml to describe toolbars - Add a naviguation toolbar
- * [#2176] Remove conference default_id in joinParticipant
- * [#2176] Display error message in alsa's snd_pcm_avail_update call
- * [#2176] Alsa mic avail data debug
- * [#2176] Add some debug message for mic loss problem
- * [#2176] Flush mic ring buffer when offholding a call
- * [#2176] Reset ringbuffers' readpointer when adding main participant
- * [#2176] Fix getAvailData algorithm
- * [#2176] Reset ringbuffer's readpointer when adding a new participant
- to a conference
- * [#1744] Regex object renamed to Pattern. Previous attempt at
- providing
- * [#2176] Fix detach main participant problem when adding new one
- * [#1976] Use right domain to translate
- * [#1976] Add xml menu description
- * [#2176] Store a list of confernece participant in client
- * [#2176] Fix add participant, joinparticipant methods
- * [#2181] Do not install dbus-c++ headers + add return value
- * [#2176] Fix minor call handling instabilities
- * [#2174] Fix incoming IP call contact address
- * [#2211] Add test to protect NULL pointer
- * [#1163] Add Advanced account configuration section
- * [#2176] Add some usefull comments and debugging info
- * [#2176] Add conditions to display security icons in conference
- * [#2176] Fix detaching one participant while keeping communication to
- others
- * [#2176] Reenable userActive.svg in call tree
- * [#2176] Make user active blue (not red)
- * [#2176] Fix user active picture
- * [#2176] Fix "hidden" merge conflict in sipvoiplink
- * [#2176] Remove iax audio stream on peer hungup
- * [#2174] Multiple UDP transports functional (TESTED with 2 accounts
- and 3 calls)
- * [#2176] Fix fix audio stream binding in iax
- * [#2174] Create a default UDP transport + use tp selector for dialogs
- also
- * [#2176] Register iax audio stream in mainbuffer
- * [#2176] Fix getAudioCodecName in IAXvoipLink
- * [#2176] Fix iax account init
- * [#2176] Handle multiple account using the same sip transport
- * [#2165] Add .png files
- * [#2176] Small fixes concerning dtmf
- * [#2176] Fix make uninstall in codecs
- * [#2174] remove stund makefile generation
- * [#2176] Add conference lock
- * [#2174] Add transport selector for multiple accounts
- * [#2176] Change userActive picture from red to blue
- * [#2176] Fix security pixbuff in calltree
- * [#2176] Replace sfl.zid in .cache/sflphone instead of .sflphone
- * [#2176] Fix add call description
- * [#2176] Remove detach button from toolbar
- * [#2176] Fix calltree call description state and state code in
- conferences
- * [#2176] Fix pulse audio double free
- * [#2176] Fix conference selection
- * [#2174] Clean up - remove stun settings in client network
- configuration panel
- * [#2174] Remove voviva stun code
- * [#2174] Rsolve STUN with pjsip - DO NOT WORK
- * [#2165] Add user svg
- * [#2165] Debugging sip call failed
- * [#929] Link against uuid if installed
- * Oops
- * Fixed bugs related to libsexy (with GTK < 2.16)
- * [#929] Remove uuid-dev dependency in the core
- * [#2165] Debugging no negociated codecs at communicatio start
- * [#2165] Fix calltree bug (gtktreestore instead of gtkliststore)
- * [#2165] Fix several merge problems
- * Updated opensuse packaging script
- * [#1163] Add missing figures
- * [#1163] Update INSTALL file
- * [#2165] Fix IAX
- * [#2165] Add recordabe interface
- * [#2165] Finish recording refactoring for call (not for conference)
- * [#2165] Enable speaker recording for two different calls
- simultanously
- * [#2165] Implement call recording using the Recordable interface
- * [#2165] Add get and set to AudioLayer's audio recorder
- * [#2165] Add class recordable from which inherit call and conference
- * [#2006] Fix G722 and Speex 8khz codec conferencing
- * [#2006] add recording of audio buffers
- * [#1163] Add general settings section
- * [#1163] Fixes makefile error
- * [#2006] Fix some minor issues
- * [#2006] Drag a conference call on another conference call
- (difference conferences)
- * [#2006] Fix dragging a conference on itself
- * [#1744] Integrating some of the needed regular expression patterns
- in order
- * COmplete call features
- * [#1744] Added support for named subgroup in the Regex object. Also,
- new
- * [#1744] Adds thread safety features, compile() and setPattern()
- methods to the Regex class.
- * [#1744] Fix inconsistency in the finditer method from the last
- commit.
- * [#1744] Added regex pattern object built on top of libpcre. To be
- used
- * [#1744] Initial commit towards implementing RFC4568. Unimplemented
- in the
- * [#2157] Hide "security" and "advanced" tabs for IAX under account
- * [#1163] Add call features section
- * [#2006] Add joinConference capabilities
- * [#2006] Add dbus joinConference signal
- * [#2006] Drag a conference call onto a conference to add it
- * [#1163] Add addressbook section
- * [#2006] Drag a conference call onto a single call to create a
- conference
- * [#2006] Expand rows automatically
- * [#2006] Add minimal multiple conference handling
- * [#2006] Add atached/detached conference icons
- * [#2006] Add function processRemainingParticipant
- * [#2006] Deep refactoring, fix hangup bug
- * [#1163] Update documentation - Accounts part
- * [#1976] Integrate user doc to gnome client build system
- * [#2122] Remove double inclusion in dbus-c++/src/Makefile.am
- * Remove pjproject version number
- * [#2006] Fix peerHungup
- * [#1976] Make Yelp accessible from the GNOME client (need to install
- the sflphone.xml first)
- * [#2006] Fix multiconferencing hangup
- * [#2006] Fix hangup calls in a conference
- * [#2150] Make IAx2 reappear
- * [#2006] Fix detach participant on multiple call
- * [#2006] Can remove rining call from a conference
- * [#2006] Reinit confID when removing a participant
- * [#2006] Remove get isCurrentCAll in hangup/peerhungup (SipVoipLink)
- * [#2006] Fix refuse call
- * [#2006] Fix answerring incoming call
- * [#2006] Refactor conference's participant list
- * [#2101] Re-integrate test compilation in main build system
- * [#2101] Make the test directory compile
- * [#2136] Restore history functionality
- * [#2006] Fix binding main participant to himself
- * [#2006] Fix add current/incoming/onHold participant to an existing
- conference
- * [#2006] Fix add incoming calls to an already created conference
- * [#2006] Fix remove stream
- * [#2006] Fix detachParticipant/removeParticipant switchCall ids
- * [#2006] Fix adding a call in conference having state "CURRENT"
- * [#2006] Remove/add main participant from conferences
- * [#2006] Hold/unHold conference
- * [#2006] Detach a partcipant from drag n drop
- * [#2006] Hangup a conference
- * [#2006] Add hold/unhold conference dbus messages
- * [#2034] gtk-ui fix under the "basic" tab.
- * [#2006] Fix dragging calls on conference calls
- * [#2006] Fix detach participant from a conference
- * [#2034] Added default message is status bar under the account config
- dialog
- * [#2112] Fix a crashed caused when a non-md5 password was sent to
- pjsip.
- * [#2006] Detach participant by ID
- * [#2006] Fix addParticipant method in managerImpl to handle
- incoming/answered calls
- * [#2006] Add addParticipant method in managerimpl and related dbus
- messages
- * [#2111] Added the ability to configure zrtp on sip.sflphone.org from
- * [#2106] Fixed problem in the account assistant under gtk-ui. Also,
- assistant.c
- * [#2006] Fix dragging a conference call on another conference call
- (same conference)
- * [#1904] Small UI fix. Assistant was moved from "Call" to "Edit"
- menu.
- * [#1904] Fix a wrong label under gtk-ui.
- * [#2034] Renaming and source code splitting.
- * [#2034] Status bar added to account window to better reflect the
- registration
- * [#2006] Make calltree_remove_call recursive (for GtkTreeStore)
- * [#1110] Small gtk-UI fix in the account window (alignment).
- * [#2006] Fix remove conference, display children which are still
- active
- * [#2006] Recursive function call in calltree_update_call
- * [#2006] Add multilayered capabilities to calltree (GtkTreeStore)
- * [#2006] Implement remove conference in calltree
- * [#2034] Now useless as Direct Ip calls settings moved under
- Preferences.
- * [#2034] Edit/add buttons were set insensitive all the time under
- gtk-ui.
- * [#1887] Information about the state of the current SIP call is
- displayed
- * [#2006] Add call tree remove callback
- * [#2006] Fix create_conference function
- * [#2006] Update conference_added_cb to add new conference to the list
- * [#812] Added new tab under GTK-ui Preferences. Moving Direct Ip
- Calls from
- * [#2121] Disable temporarily test compilation
- * [#2006] Fix conferencelist to handle conference_obj_t instead of
- gchar
- * [#2006] Add conference_obj structure
- * [#2121] Update version
- * [#2006] Fix conference selection
- * [#2101] Use the new source tree to fetch the right object files
- * [#2006] Add conference in calltree
- * [#2006] Add Dbus signal conference added/removed/changed
- * [#2006] Add getConferenceDetails call on dbus
- * [#1904] Registration expire now appears as a spin box under gtk-ui.
- * [#812] Fixing a segmentation fault caused by a non-existing account
- ID
- * [#2006] Add getConfList method over dbus
- * [#2006] Add a conferencelist data structure in client-gnome
- * [#812] Defaults value are now sent if a non-existing account is
- requested
- * [#2006] Add sflphone action sflphone_join_participant
- * [#2006] Fix buffer read pointer problem deletion
- * [pjsip] Attempt at fixing via header incompatibility with
- Freeswitch.
- * [#1797] forget something
- * [#2006] Add call new state conferencing in deamon
- * [#2006] Remove addParticipant method for conference, use
- joinParticipant only
- * [#1163] Update INSTALL documentation
- * [#812] Msec/sec values were not taken into account.
- * [#1797] Make pjproject-1.4 compile
- * [#2006] Add Detach participant method
- * [#2006] Dragndrop fully functional with INCOMING and HOLD call
- * [#1797] Add pjproject-1.4
- * [#1797] Remove pjproject-1.0.3
- * [#2006] Get call state in conference related function
- * [#2006] Add joinParticipant (conference) method in ManagerImpl
- * [#2006] Add joinConference DBUS message
- * [#2006] Store the previously selected call_id on dragndrop
- * [#2006] Fix GValue pointer unref in selection callback
- * [#2006] Store dragged call_id
- * [#2006] Update drag_data_received_cb callback to manipulate CallIDs
- * [#2006] Add dragndrop signals
- * [#2006] Set calltree reordable
- * [#812] Adds the ability to create a TLS listener in case the user
- requests
- * [#812] Adds the ability to configure local/published address from
- * [#1883] Move switchCall in onHoldCall function
- * [#812] Deals with the published address/port problem when
- integrating TLS.
- * [#1883] Switch call id in managerimpl when peerHungUp
- * [#1883] Switch call id before hangup
- * [#1883] Add usefull and permanent debug info for conference
- cretion/deletion
- * [#812] Fix various segmentation faults related to Direct IP kind of
- calls.
- * [#1883] Fix deletion of std::map elements using iterators
- * [#2014] Add libzrtpcpp build dependency
- * [#1883] Still some for loop test ambiguity (while loop instead)
- * [#1883] Fix for loop initial test ambiguity (use while loop instead)
- * [#1883] We must discard data in urgent ring buffer if data is get in
- mainbuf
- * [#1883] Fix availForGet same id for ringbuffer and readpointer
- * [#812] Match "sips" as a Direct IP Call when the user enter a sip
- uri
- * [#812] Fix segmentation fault related to SIP URI creation.
- * [#812] Towards integrating multiple tls listeners at the same time.
- This
- * [#1883] Add debug messages in conference and fix mainbufferTest
- * [#812] gkt-ui fix. Private key must be fed as a filename and not as-
- is.
- * [#812] TLS integration within sipvoiplink and pjsip. Also,
- configure.ac
- * [#1883] Fix Alsa/Pulse mallocation
- * [#1883] Fix data corruption in AudioRtp's micData buffer
- * [#812] Full dbus integration for all the tls related options under
- gtk-ui.
- * [#1883] Fix memory leaks in audiortp session
- * [#1883] Fix mem leaks in audio rtp
- * [#812] Fix setAccountDetails where TLS_ENABLE was set to the value
- * [#812] Small gtk-ui fix.
- * [#811][#812] Small gtk-ui fix.
- * [#812] Introduced a mechanism for configuration files that makes
- possible
- * [#812] New dbus bindings added. Also, configuration compliance was
- enforced
- * [#1881] Remove default buffer from MainBuffer (update unit-tests)
- * [#1881] Add ring buffer read pointer tests
- * [#1883] Fix issues in ringbuffer reader pointers
- * [#2034] Implementing a new configuration dialogue for TLS transport
- settings
- * [#1883] Add some usefull debug and safety checks
- * [#2028] Notify the client with libnotify when the zrtp negotiation
- failed.
- * [#811] Harmless no to throw an exception, an makes the application
- less
- * [#2028] A minidialog is showed to the user under sflphone-client-
- gnome
- * Removed useless file.
- * Ignoring Makefile in src/widget
- * [#2027] Fix segmentation fault when showMessage callback is called
- after
- * [#2026] keyExchange was set to ZRTP instead of "1"
- * [#2024] Fix the wrong summary at the end of the assistant.
- * [#1883] Fix mnagerimpl conference map insertion
- * [#1883] Add Mutexes in MainBuffer
- * [#811] Gtk ui was not presenting the right information about zrtp
- for
- * [#2023] security icons were not installed in sflphone-client-gnome.
- * [#2021] Fix a mistake in the readme from sflphone-common that gives
- wrong
- * [#811] The current SRTP mode was not properly displayed for the
- IP2IP
- * [#1743] Re-implementation of the "automatically remove error dialogs
- [...]"
- * [#2017] [#2019] Fix the inability to dial a number and place a
- registered
- * [#811] Final re-integration of ZRTP support in the main branch from
- 0.9.6
- * [#1883] Fix map insertion methods
- * [#811] Combo box now is now set to the active key exchange method
- * [#811] ZRTP options now configurable back again from the Gtk UI.
- IP2IP
- * Updated hostname for git clone
- * [#1883] Add minimal functionalities to create a conference
- * [#811] re-integration of all the methods and signals on dbus.
- ManagerImpl
- * [#811] Got out of a precarious position were nothing would compile.
- * [#1976] Build documentation squeleton with docbook
- * [#1883] Add sflphone-client "addParticipant" button for conference
- * [#1994] Better organize the source directory structure. New
- subdirectories
- * [#1883] Add a simple Conference class
- * [#1882] Use static audio buffer in Pulse and ALSA layer (instead of
- malloc)
- * [#811] First commit toward re-integration and refactoring of ZRTP
- * [#1882] Flush RTP ring buffer before entering mainloop
- * [#1882] Fixed MainBuffer::UnBinCallID() in case there is no
- ringbuffer
- * [#1882] Test (and fixe) high level conference and mixing
- functionalities
- * [#1772] Apply patch to compile on fedora (sent by Marcin
- Zajączkowski <mszpak@wp.pl>)
- * [#1882] Update Bind, unBind call_id in MainBuffer
- * [#1959] This adds the ability to store password as an MD5 Hash in
- the
- * [#1538] Fixes rules compilation
- * [#1930][#1931] Fixed a mistake (again) related to index and
- credential count
- * [#1753] Remove ILBC from pjproject - Hacks in pjsip
- * [#1930][#1931] Credential was not selected properly using realm
- * [#1882] Finilize multiple reading pointer in RingBuffer
- * [#1538] Remove configure from autogen.sh to respect debian upstream
- authors policy
- * [#1773] Remove generated files from repo
- * [#1791] Use XDG_CACHE_HOME to save pid file
- * [#1791] Fixes path to save history
- * [#1791] Fix debian installation scripts
- * [#1930][#1931] Settings are now taken into account in the server.
- * [#1882] Add ringbuffer default ring buffer pointer in methods
- involving mStart
- * [#1882] Add default ringbuffer pointer
- * [#1882] Add RingBuffer multiple read pointer basic functionnalities
- * [#1882] Fix MainBuffer flushData unit test
- * [#1930][#1931] Ability to save and retreive the configuration from
- * [#1882] Added Multiple CallID mapping to MainBuffer
- * [#1791] Not much
- * [#1791] If XDG env variables are not null but empty, use default
- ones
- * [#1791] Make XDG_CONFIG_HOME writable
- * [#1930][#1931] Partial commit. Not working yet. Cannot delete
- account
- * [#1881] Fixed alsa capture latency problem
- * [#1881] Fixed Alsa capture temporarily
- * [#1930] [#1931] Partial unbroken commit providing the ability to
- * [#1881] MainBuffer implemented in AudioLayer/AudioRTP
- * [#1881] Add discard and flush unit-tests
- * [#1881] Add discard and flush functionnalites to MainRingBuffer
- * [#1881] Add availForGet in MainBuffer
- * [#1881] Add availForPut function to MainBuffer
- * [#1880] Remove AudioRTP* pointer from SipVoIP (reapered while
- merging master)
- * [#1881] Add a map between call id and coresponding ring buffer
- * [#1855] Refresh pot file and upload on Launchpad
- * [#1881] MainBuffe now robust to false ids on getData and putData
- * [#1881] Fix big big big memory leak
- * [#1881] Add getData and putData to mainBuffer
- * [#1881] Unit-test basic ring buffer functionnaities
- * [#1881] Add class MainBuffer and basic buffer creation unit-tests
- * [#1880] Fix call transfer (step2) issues
- * [#1880] Moved AudioRtp* pointer from SIPVoIPLink to SIPCall class
- * [#1791] Add postinst script to keep user data when migrating
- config/history file
- * [#1797] Make pjsip compile
- * [#1777] Code indentation
- * [#1791] Use XDG_DATA_HOME and XDG_CONFIG_HOME for sflphonedrc and
- history + unit tests
- * [#1746] Useless space does not appear anymore when volume sliders
- and
- * [#1643] GtkCheckMenuItem is used instead of icons for elements in
- the
- * [#1110] [#1668] STUN parameters are now located in the preferences,
- under
-
- -- Julien Bonjean <julien.bonjean@savoirfairelinux.com> Fri, 06 Nov 2009 11:20:01 -0500
-
-sflphone-plugins (0.9.6-SYSTEM) SYSTEM; urgency=low
-
- ** 0.9.6 **
-
- * Documentation on echo test
- * [redmine_down] codec names not displayed in total
- * [redmine_down] crash when hanging up a dialing call because tries to
- add it to history whereas no starttime
- * [#1927] alternate every time screen changed to call history
- * [#1886] clean code
- * [#1886] debug messages when loading history removed
- * [redmine_down] sflphone-kde icons
- * [#1855] Update language files
- * [#1502] Update version number
- * [redmine_down] setHistory at close
- * [#redmine_down] Handle PJ_DECLINE_SC as failure
- * [#1923] Fix segmentation fault when adding a new account
- * [#1923] Check on iterator before setting the config
- * [#1904] Added mnemonic to tabs in sflphone-client-gnome.
- * [#1905] The daemon was not sending the currentSelectedCodec signal
- on dbus when answering a call.
- * [#1922] Default values set to all account details
- * [#1886] Spinbox reg expire enables apply, and address book is not
- visible when disabled
- * [#1905] Bug fix for segmentation fault caused by an empty string,
- * [#1910] Warnings in test directory
- * [#1919] Error fixed
- * [#1855] Update russian translation - Hussein Abdallah
- * [#1910] Remove files
- * [#1919] fixed
- * [#1777] Code indentation
- * [#1918] fixed
- * [#1917] fixed
- * [#1910] Remove warnings compilation in src
- * [#1886] removed AccountListModel in configskeleton
- * [#1914]
- * [#1911] check previous and new port
- * [#1910] Remove compilation warnings in src/dbus and src/history
- * [#1910] Remove compilation warnings in src/audio
- * [1855] Update german translation - Sven Werlen
- * [#1909] removed
- * [#1906] Done
- * [#1904] The registration expire value is now configurable from the
- * Cleaned up debug messages.
- * [#1886] separated initCallItem in two functions
- * [#1886] reversed error in commit
- * [#1886] clean debug
- * [#1886] changed Name of classes and files
- * [#1886] clean
- * [#1870] In call_state_cb (dbus.c:126), _time_stop was overridden by
- the actual time.
- * [#1884] Added some new gpg flags to prevent tty warnings
- * [#1886] Clean audio config dialog
- * [#1886] No more compile warnings. + 1 comm
- * [#1872] Check if the user input is smaller than PJ_MAX_HOSTNAME.
- * [#1886]
- * [#1785] Fixed build when no new commit
- * [#1852] If chosen by the user, the hostname can now be solved and
- used
- * [#1871] * and # inverted back
- * [#1869] Conditional compilation that checks if
- * [#1309] removed test in main
- * [#1425] Put actions in SFLPhone window class instead of ui view,
- made a separate toolbar for screens.
-
- -- SFLphone Automatic Build System <team@sflphone.org> Mon, 27 Jul 2009 09:53:19 -0400
-
-sflphone-plugins (0.9.6~rc2-SYSTEM) SYSTEM; urgency=low
-
- ** 0.9.6~rc2 **
-
- * [#1755] Remove generated file
- * [#1753] restore ilbc ...
- * [#1866] Methods getSipPort and setSipPort now have an effect on the
- * [#1753] make pjsip compile without ilbc. Use ./autogen.sh --disable-
- ilbc-codec
- * [#1855] Fix error in russian translation
- * [#1805] Remove the old flawed signal mechanism which was failing in
- * [#1855] Refresh translation
- * Spanish translation finished + po README files updated + echo's in
- copy-in-clients
- * [#1850] Yun made the chinese HK-CN translation
- * [#1848] Fix transfer interface bug
- * [#1862] At install, kde client installs only french translation file
- * [#1841] A new fallback mechanism was added to the internal resolver
- in PJSIP.
- * Started AccountList model/view
- * [#1855] Remove po subdir in Makefile.am
- * [#1855] Fix typo error in sflphone-client-gnome
- * [#1855] Do not generate Makefile in sflphone-common/po
- * [#1855] Copy translation files into both clients dirs
- * [#1855] Remove po dir from sflphone-common
- * Comments added
- * [#1860] mailbox->voicemail...
- * make scripts executable
- * [#1855] French translation
- * [#1855] Chinese zh_HK partially filled...
- * [#1859] An unnamed pipe monitored by poll() was added. When we want
- to
- * [#1855] Sven completed the first part of the german translation
- * [#1855] Cantonese manually filled for already translated, almost
- equal strings
- * [#1855] Merge russian translation
- * [#1855] Spanish manually filled for already translated, almost equal
- strings
- * [#1855] Update german translation in ./lang/de
- * [#1858] This problem was fixed by removing a useless line in
- * [#1855] merged existing translations in lang/ sflphone.po's
- * [#1842] [#1843] An attempt at improving the expected behaviour that
- can't
- * [#1855] added po folder in gnome client and scripts for copying from
- common lang folder to clients
- * [#1853] Edit before call does nothing on call history
- * Put most language entries possible in common. From 300 to 250
- entries. Stays underscores problem. Scripts for copy in clients.
- * commit to merge master
- * [#1825] Changed "Bad authentification" to "Authentication Failed".
- * common po files
- * [#1753] Remove ILBC from pjproject
-
- -- SFLphone Automatic Build System <team@sflphone.org> Fri, 17 Jul 2009 19:12:58 -0400
-
-sflphone-plugins (0.9.6~rc1-SYSTEM) SYSTEM; urgency=low
-
- ** 0.9.6~rc1 **
-
- * Update some version number
- * [#1792] Creates .sflphone directory with permission 600. Also,
- "chmod 600" after
- * [#1810] GUI is now notified that the call failed. Also, a segfault
- was
- * [#1816] Address book search disabled when disabled address book and
- enabled it back plus button stays triggered
- * codeclistmodel + asynchronous loading of address book +
- enable/disable address book
- * [#1810] Now checking SDP answer after 200 OK. Still need to
- implement full
- * [#1794] Can't use the interface during a call
- * Updated translation files
- * Russian translation integrated
- * Codec list model/view started.
- * [#1807] Add configure.ac in pjproject-1.0.3
- * [#1787] closeRtpSession added in some places where it should have
- been
- * Use Item class for contacts and accounts
- * Comments + clean code
- * [#1794] Improved debug messages
- * [#1805] Replaced the old and unreliable mecanism that was was
- waiting for
- * [#1794] Can't use the interface during a call
- * [#1787] For those cases where no registered SIP account is
- configured
- * [#1797] Make pjsip compile
- * [#1787] Minor changes. Removed useless commented line. Changed order
- of
- * [#1777] Code indentation
- * [#1797] Update package generation with new pjsip version
- * [#1798] Does not hang up when the call is building up
- * [#1797] Update .gitignore with new pjsip version
- * [#1797] Remove generated files from repo
- * [#1797] Main build system now uses pjproject-1.0.3
- * [#1797] Add pjproject-1.0.3
- * [#1797] Remove pjproject-1.0.2
- * [#1796] Computing time optimization (samplerate conversion)
- * [#1787] _audiortp->start() moved away from offhold(),
- SIPCallAnswered()
- * [#1312] Added new states for calls initialized by other clients
- * [#1795] Crashes when adding a new account, checking it and applying
- * [#1782] Missing icons
- * [#1793] KDE client compilation problem
- * Fake ringtone files can no longer be set.
- * indentation
- * [#1312] Able to fetch to differentiate incoming/ringing call state
- * [#1784] Use DESTDIR variable in po Makefile - fix language file
- installation
- * [#1785] Fixed typo
- * [#1785] Fixed changelog update
- * [#1759] ./autogen.sh --prefix=/usr --with-debug to use optimization
- level 0
- * [#1773] Changed snapshot naming convention
- * [#1773] Removed gpg agent use, added repository cache cleaning
- * [#1759] Use optimization level 0 for repository, 2 for packages
- * [#1777] Code indentation/formatting
- * Translated new features in french
- * [#1785] Added missing changelog entry
- * [#1781] Window title is SFLPhone
- * [#1777] Add code indentation/formatting in the buil system
- * [#1774] Can't set voicemail number in KDE account creation wizard
- * [#1775] Can't modify account information for account created with
- the wizard
- * [#1771] Add a "Default" button in context menu to disable chosen
- prior account
- * [#1705]
- * [#1224] Remove generated file from the repo
- * [#1224] Remove generated file from the repo
- * [#1762] distclean target should remove kconfig generated files
- (settings.h, settings.cpp). Rename them?
- * [#1761] clear history button should really clear history
- * Dialpad works.
- * Implemented Dialpad widget instead of building it in main view.
- * Removed last occurence of the old config dialog, that made the build
- crash.
- * [#1755] Do not consider G722 as a dynamic payload elsewhere than in
- RTP layer
- * [#1753] Remove ilbc Makefile generation
- * [#1756] Implement a kde configuration dialog with kconfig xt and
- kconfigdialog class
- * [#1755] fix audiocodec folder parsing problem
- * [#1450] Reinit timestamp comparison in RTP, create session in
- newOutgoingCall
- * [#1753] Remove milenage third party code from pjsip
- * New Config Dialog integrated in GUI.(without codecs)
- * [#1753] Remove ILBC codec
- * kconfig started, tr2i18n -> i18n, icons folder, accountList changed
- * [#1705] Fixed Audio RTP thread creation/start
- * [#1714] Fix codec negociation result handling
- * [#1678] Fix audiortp payload setting
- * [#1678] Put bac putData method in rtp
- * [#1669] gtk_file_chooser_get_filename() support UTF-8 by default
- * [#1735] Add conditions to sdp update call if call declined
- * [#1737] substr of recordings destination folder to remove "file://"
- should be done in client rather than in daemon
- * [#1731] Enlarge audio stream buffer size
- * [#1714] Missing true
- * [#1317] Fixed Mandriva timeout
- * [#1317] Changed tag convention
- * [#1317] Cleaned git-dch
-
- -- SFLphone Automatic Build System <team@sflphone.org> Fri, 10 Jul 2009 15:50:26 -0400
-
-sflphone-plugins (0.9.6~beta-SYSTEM) SYSTEM; urgency=low
-
- ** 0.9.6~beta **
-
- * spec files for mandriva and opensuse updated with buildrequires
- libqt4-dev >=4.3
- * [#1700] Cannot build on ubuntu 8.10 and a few other distribs
- * [#1502] Update version number where applicable
- * [#1642] Update client icons
- * [#1450] Clean up useless debug and comments in sipvoiplink and
- audiortp
- * [#1450] Remove Semaphore object in AudioRtp thread deletion
- * [#1450] Audio RTP init now synchronized with Sip/SDP
- * [#1693] kde client crashes when changing codecs order/activation
- * [#1450] Deep refactoring of audiortp
- * [#1450] setRtpSessionRemoteIp
- * [#1689] getCallList at start
- * [#1224] Change path in package files
- * [#1450] Audio RTP initialized only once, payload and remote ip set
- at runtime
- * [#1450] Add setRtpSessionMedia and setRtpSessionRemoteIp address
- * [#1642] Make GNOME GUI fresher and younger ;)
- * [#1686] Status bar displaying used account
- * added sflphone-kde icon so that it compiles
- * [#1659] Ending a call causes the daemon to crash
- * corrected introspection XMLs, po files...
- * [#1211] g722 media descriptor in codecDescriptor
- * [#1310] Install sflphoned in $(prefix)/lib/sflphone
- * [#1502] Do not install test binaries and dbus utilitaries
- * [#1224] hack for pjsip build system!
- * [#1224] Remove pjsip binaries from repo
- * [#1224] Upgrade to pjsip 1.0.2
- * [#1658] About SFLphone (bugs)
- * [#1658] About SFLphone
- * [#1660] Displaying all dialed numbers in a call
- * Tested status bar.
- * [#790] Optimize pulse audio streams parameters
- * [#1678] Some usefull debug messages for mutex/semaphore deadlock
- problem
- * [#1669] Add/remove some usefull/unusefull debug
- * [#1665] Fix latency related to pulse audio stream openning/closing
- * [#1457] Make the menus and panels accessible in french
- * [#1457] Improve broken keyboard accessibility in menus and conf
- panels
- * [#961] Instanciate only once the searchbar icons
- * [#961] Restore transfer fonction
- * [#961] Filter on the history type OK
- * [#961] Fix compilation problems on hardy/intrepid
- * [#1157] Commit missing files
- * [#790] Reduce number of start/stop streams call on pulse audio
- * [#1639] kde client crashes when no account registered
- * [#1620] Fix the searchbar
- * [#1620] Get back caltree as it was during gtkcritical area
- * [#1620] Add history filter reinit function
- * [#1335] Add a missing label in address book preferences
- * [#1561] Update russian translation - Hussein Abdallah
- * [#1605] Fix edit menu french translation
- * [#961] Enable to search in the history according to the call type
- * [#1449] Searchbar does not work anymore
- * [#961] Add popup menu on the entry primary icon for history
- * [#1317] Fixed KDE client package dependency
- * [#936] speex 32 khz integration completed
- * [#936] Use 320 frame size
- * [#936] Test using a frame size at 320 smpls
- * [#1214] Enable / Disable history
- * [#1607] Fix compilation problem for ubuntu 8.10 (libsexy)
- * [#1313] Implement processDataEncode processDataDecode in audiortp
- * [#1613] codec list order can't be set
- * Better handling of localisation + added languages + corrected
- warnings + begginning of new config dialog with kconfig + 14px
- account leds
- * [#1214] Save and load history according to the limit timestamp +
- unit tests
- * [1609] Fix call number copy/paste feature
- * [1607] Restore clear action icon in searchbar
- * [#936] Try to decode using 1280 samples
- * [#936] Add some debug
- * [#936] Add .cpp file
- * [#936] Oops Forgot speex 32 khz
- * [#1214] Add configuration panel for history + D-Bus calls
- * [#1313] Test rtp thread function, frame size, nbbytes, resampling
- * [#790] Flush audio data before closing audio streams
- * [#1214] History displays local time
- * [#1214] Skip empty field on display
- * [#1214] Associate an account to an history entry
- * [#1342] Get addressbook options sensitive/non-sensitive
- * [#1211] Clean up and comments
- * [#1211] Get back to 20 ms framesize
- * [#1211] Use sendImmediate instead of putData in RTP
- * [#1211] Fix nb byte available in RTP
- * [#1211] Clear condition on maxNbSamples in RTP
- * [#1211] Fix max byte available in RTP session
- * [#1211] G722: Use 160 samples per frame instead of 320
- * [#1211] Test using a dynamic payload
- * [#1211] Test using a dynamic payload type
- * [#1211] Rename size variable (nb_samples, nb_bytes)
- * [#1211] Test g722 ip-to-ip sending twice the data lenth
- * [#1211] Test g722 ip-to-ip
- * [#1214] Do not select an history item by default at startup
- * [#1214] Remove some compilation warnings
- * [#1214] Handle empty field - remove g_print
- * [#1214] Add each history item only once
- * [#1214] Handle call timestamps properlier
- * [#1214] Do not need timestamp files anymore
- * [#1214] Use the saved date for history entry
- * Clean up
- * [#1214] Client doesn't crash if the D-Bus call fails
- * [#1214] Client is able to save its history - still some glitches
- * [#1211] Forgot 16000 for g722
- * [#1211] G722 initialization
- * [#1214] Save name/number, successfully load the history if no fields
- are empty
- * [#1499] Fixed destination directory bug
- * [#1214] Restore all the functionalities; peer name/number way more
- easy to handle !!
- * [#1214] Add callable_object instead of call_t, refactoring
- * [#1211] Test with polycom soundstation 16000
- * [#1211] Remove C like inline function in g722 codec
- * [#1342] Finalize gnome client preference window formating
- * [#1214] Retrieve the history when the gnome client startsup
- * [#1306] Implement localization for KDE client
- * [#1593] enable accounts apply button when account checked/unchecked
- * [#1214] Implement the dbus calls on server side
- * [#1214] Add serialized/unserialized functions to pass data on DBUS
- * [#1342] Formating gnome client configuration windows
- * [#1214] Save sucessfully a map of history items
- * [#1499] Removed multiple jobs compilation for KDE client (2)
- * [#1214] Load history from file into memory, add unit tests
- * [#1534] Throws a length_error exception in case URL exceeds
- std::string max_size
- * [#1499] Removed multiple jobs compilation for KDE client
- * [#1565] make account leds smaller
- * [1430] Fix dbus debug
- * [#1562] crashes when trying to change item of a call of state "OVER"
- * [#1116] Fix compilation bug
- * [#1317] Added mandriva and opensuse-11 64 bits
- * [#1108] Add messges in main window concerning transfer success
- failure
- * [#1116] Fix compilation problems
- * [#1211] g722 Makefile
- * [#1108] Client side transferFailed/trasferSucceded signals handling
- * [#1211] G722 mostly completed,
- * [#1555] make bigger toolbar (24x24)
- * [#1551] remove default mailbox number in wizard and disable mailbox
- button when first account doesn't have mailbox number
- * [#1342] Re-add sflphone manpages
- * [#1116] Fix compilation on non-jaunty distros
- * [#1317] Fixed opensuse startup sleep
- * [#1108] Add a signal in the client to notify successful or failed
- transfer
- * [#1108] Dbus signals concerning call transfer success/failure
- * [#1317] Added opensuse to automatic build system
- * [#1223] Fix manpages bug
- * [#1060] german translation glitch
- * Clean up some gnome client warnings
- * [#1547] replace ugly account leds by beautiful icons
- * [#1548] add close button that hides windowand just hide on clicking
- the cross
- * [#1549] put introspec XMLs in the client's source
- * [#1312] Implement getCallList D-BUS method
- * [#1116] Clear text in history and contacts
- * [#1499] KDE integration
- * [#1469] Modify header linkers in dbus-c++'s Makefile.am's
- * [#1469] Remove examples folder from dbus-c++
- * [#1214] History integration in build system; unit test squeleton
- * [#1317] Cleaning
- * [#1469] Remove configure stuff in dbus-c++
- * [#1469] Add unofficial mainline dbus-c++
- * [#1469] Remove dbus-c++ from freedesktop
- * [#1430] Bring account changed signal/callback back to normal
- * [#1060] Update german translation - Sven Werlen
- * [#1430] Add marshaller one string define
- * [#1430] Send account change signal broadcast using account id
- * [#1430] Remove condition on setRegistrationState, cause stun to
- crash
- * [#1317] Centralized version handling
- * [#1317] Fixed version number on sfl-git-dch
- * [#1317] Refactoring for new distributions
- * [#1215] Fix account order at startup if latency
- * [#1088] Restore sip dns srv
- * [#1214] Add squeleton for history manager
- * [#1430] Add accout id to accout changed method
- * [#1430] No connectionStatusNotification (account changed) if no
- changes
- * [#1538] Add COPYING file
- * [#1430] Add audio rtp thread tests
- * [#1317] Changed version detection
- * [#1538] Document license in libs/stund
- * [#1317] Added version files
- * [#1538] Apply François patches - debian packages
- * [#1317] Updated spec files
- * add files
- * [#1538] Apply François patches - debian packages
- * [#1535] Change program file structure (directory src...)
- * [#1317] Updated build system scripts
- * [#1317] Cleaning
- * [#1317] Copied introspect files to gnome client
- * [#1317] Added opensuse to build-system : first-shot
- * [#1317] Remove spec files from configure
- * [#1317] Added missing prefix
- * removed debug for daemon account fix
- * [#1430] Add a connection reference which most likely belong to
- libdbus
- * [#1430] Use shared connection instead of private
- * make daemon find the account, added userMatch
- * Clean code, add comments...
- * [#1317] Fixed packaging rules
- * [#1317] Updated autogen
- * Updated autogen.sh for pjsip
- * [#1526] Set accounts order
- * [#1317] Fixed pjsip lib dirs
- * [#1317] Updated debian packaging for new pjsip configuration script
- * [#1317] Switch to autogenerated guess and sub files
- * [#1317] Updated pjsip inclusion in build system
- * [#1317] Replaced pjsip guess and sub files
- * [#1317] Fixed compilation issues on opensuse 11
- * [#1505] account list seem to crash the application when clicking
- Apply very fast...
- * [#1456] Add a flag to be replaced in the control files
- * [#1456] Added version dependancy handling
- * put account alias in AccountWidgetItem rather than in the item with
- " " before.
- * [#1034] The KDE client should start sflphoned if it is not started
- * [#1500] Handle options for notifications and display on incoming
- call.
- * [#1443] Client should not crash when receive an unexpected
- stateChanged signal
- * [#1403] Do not stop the notification anymore
- * [#1456] Added version dependancy handling
- * [#1426] Daemon crashes when get alsa plugin
- * [#1422] Improved error messages
- * commit for merge
- * [#1424] Change logo in tray icon and put a different one when
- incoming call
- * [#1425] first part done, window title...
- * [#1413] add manpages creating and installing in build system
- * [#1417] The client should start the account creation wizard if
- started for the first time (if config file doesn't exist)
- * [#1421] Make volume bars horizontal when dialpad is hidden.
- * Changed main window title and fixed a mistake in sflphone_const.h
- * [#1412] make debian package building work
- * changelog changed.
- * Changed addAccount method in gnome client.
- * Debian and man folders added.
- * [#1388] Change project name from sflphone_kde to sflphone-client-kde
- * Better handle of kabc check.
- * [#1351] Automatic generation of dbus interfaces in makefile
- generated by cmake
- * [#1307] Implement "edit before call" in history and address book.
- * [#1344] change action_call label in call history from "call" to
- "call back".
- * [#1308] Implement Hook feature in kde client
- * Improved build system.
- * #1219 : Add address book configuration page
- * Better handling of registration to the daemon.
- * #1039 : Add tray icon in kde.
- * Issue no 1216 : Double click on item in history or address book
- causes call.
- * display peer name in call list and call history when called from
- address book.
- * Address book functionnal with photo displayed.
- * Help menu kde available but actions disappeared. All fonctions in
- view.
- * Address book functionnal but ugly and making its own sort in the
- complete address book.
- * Account choice on right click, clean out includes, page address
- book, fixed bugs...
- * Wizard, double click, context menu...
- * Removed sflphone_kde.kdevelop.filelist
- * Added account creation wizard and translated interface in english.
- * Transfer functionnal but ugly.
- * transfer not functionnal
- * Bug fixed : unholding (UNHOLD_CURRENT, UNHOLD_RECORD)
- * Commit functional for push. With install.sh
- * Before merge.
- * Problem with enable accounts. Account display increased.
- * Functional with codec order working , playDTMF.
- * Commit functional.
- * sflphone_kde/build added in .gitignore.
- * complete commit for checkout previous.
- * Commit before checkout previous version to check the display
- bug(little font everywhere...)
- * Functionnal client. Rest : history icons, config icons and
- functionalities
- * commit before merge asavard for isRecording.
- * Call and Automate fusion done and seems to work.
- * Commiting before putting Automate class in Call class.
- * Functionnal main window without recording, history, voicemail, kio
- widgets.
- * client kde avec kdevelop.
- * Config Dialog almost finished.
- * Base of QT client
-
- -- SFLphone Automatic Build System <team@sflphone.org> Tue, 23 Jun 2009 11:13:42 -0400
-
-sflphone-plugins (0.9.5-SYSTEM) SYSTEM; urgency=low
-
- ** 0.9.5 release **
-
- * [#1060] FIx bug in chinese translation
- * [#1313] git add rtpTest.cpp rtpTest.h
- * [#1313] Add init/close rtp tests
- * [#1313] Basic instanciation of the rtp layer
- * [#1449] Gtk-Critical concerning history filters and new calls
- * [#1400] Make the match with the hostname instead of username
- * [#1324] Change status bar label for "Using %s (%s)"
- * [#1403] Icon size: 60x60 px
- * [#1403] Do not remove notification, improve icon quality
- * [#1403] Add smaller icon for gnome notifications
- * [#1403] Prevent crash when hangup && no notification
- * [#1403] Remove all actions on notifications; code refactoring
- * [#1451] Use stun.sflphone.org as default STUN server
- * [#1060] New po files - need to be translated
- * [#1060] Update french translation - Rebuild template file
- * [#1456] Add a flag to be replaced in the control files
- * [#1454] Make cppunit optional; remove from build deps in control
- files
- * [#1401] Add libexpat1-dev dependency in control files
- * [#1448] Take off these ugly debug messages
- * [#1448] fixed getTelephoneTone and getTelephoneFile() called
- repeatedly
- * [#1406] add liblog4c-dev in build-depends
- * [#1409] Restore .desktop icon
-
- -- SFLphone Automatic Build System <team@sflphone.org> Mon, 25 May 2009 11:34:48 -0400
-
-sflphone-plugins (0.9.5-SYSTEM~rc2) SYSTEM; urgency=low
-
- ** 0.9.5 rc2 **
-
- * [#1422] Improved error message
- * [#1402] Fix pjsip build
- * [#1404] Clear GTK-Critical Bug at client startup
- * [#1422] Added automatic VM shutdown when building on more than one
- VM
- * [#1422] Fixed some issues with new changelog generation script
- * [#1422] Moved distribution update to specific file
- * [#1422] Dropped git-dch, replace by home made implementation
- * [#1402] Fix pjsip build
- * [#1404] Clear GTK-Critical Bug at client startup
- * Changes for name based dbus connection
- * Clean changelogs
- * [#1343] Gnome: Implement a callback system to handle focus on
- different widgets
- * Debus Session
- * Refactoring Python code, PEP8
- * [#1430] Get back dbus_g_proxy_new_for_name
- * [#1430] Get back DBUS_BUS_SESSION type
- * [#1430] Dbus fixed owner message binding
- * Second test with DBUS owner
- * [#1404] Gnome -> Preferences -> Hooks
- * [#1404] Gnome -> Preferences -> Recordings
- * [#1404] Call History
- * [#1404] Gnome -> Preferences -> Address Book
- * [#1404] IF the first notification option disable the second
- notification
- * Dbus with fixed owner does not automatically start the deamon
- * Add codec debug tests in pysflphone
- * [#1407] Some print info
- * [#1407] Add a scenario to pick_up action
- * Test client dbus connection to a fixed owner
- * Add python dbus test suite
- * [#1161] Modified version handling in build system
- * [#1314] Test pulse audio and audio streams connect and disconnect
- * [#1402] Add info message after configure
- * [#1402] Build the daemon with the local pjsip library (vs the
- installed one)
- * [#1009] Fix Codec Sampling Rate set to zeros
- * [#1314] Add mutex to pulse layer audio streams
- * [#1314] Refactoring pulseaudio stream to test connect disconnect
- * [#1314] Refactoring of pulselayer to test conect/disconnect
- * Add debug messages in debus calls concerning account
- * [#1314] Add some return values to audio init functions
- * [#1406] add liblog4c-dev in build-depends
- * [#1409] Restore .desktop icon
- * Bug #1405: Fix strings as requested.
- * Bug #1404: Fix strings in preferences panel.
-
- -- SFLphone Automatic Build System <team@sflphone.org> Tue, 19 May 2009 12:08:18 -0400
-
-sflphone-plugins (0.9.5-0ubuntu1~rc1) SYSTEM; urgency=low
-
- [ SFLphone Project ]
- * [#1262] Updated changelogs for version 0.9.5-0ubuntu1 Snapshot 2009-
- 05-05
-
- [ Emmanuel Milou ]
- * Add some python CLI client code; not really functional
- * [#1108] Fix peerHungup method for IP to IP call
-
- [ Alexandre Savard ]
- * [#1108] Correct setting of SIP contact for direct IP call
- * [#1108] SIP user agent handles incoming REFER
-
- [ Emmanuel Milou ]
- * Remove website from repository
- * Update translation
-
- [ Alexandre Savard ]
- * Sflphone icon's tooltip changed for "configured" instead of
- "registered"
-
- [ Emmanuel Milou ]
- * Update translation
-
- [ Sflphone Project ]
-
- -- Sflphone Project <sflphone@mtl.savoirfairelinux.net> Tue, 05 May 2009 19:16:13 -0400
-
-sflphone-plugins (0.9.5-0ubuntu1~beta) SYSTEM; urgency=low
-
- [ Julien Bonjean ]
- * Updated Eclipse stuff
- * Improved addressbook config window
- * Added sflphone Eclipse stuff
- * Implemented addressbook list server side
- * Moved dbus stuff in dbus directory
- * Updated addressbook configuration
-
- [ Emmanuel Milou ]
- * Remove unuseful installation scripts. Use apt-get build-dep sflphone
- instead
- * fix bug #1090
-
- [ Alexandre Savard ]
- * defining speex 16khz
-
- [ Emmanuel Milou ]
- * Remove unuseful file from build system
- * Start dns srv resolver
-
- [ Alexandre Savard ]
- * Basic ogg/vorbis initialization
-
- [ Emmanuel Milou ]
- * Handle incoming IP-to-IP invite correctly
-
- [ Alexandre Savard ]
- * speex wideband 16000
-
- [ Emmanuel Milou ]
- * Better handling of incoming IP to IP call
- * DNS SRV resolution functional
- * Implement IAX2 incoming URL
- * Allow user to make IP call without any accounts configured
- * Add a contextual menu to edit a number from the contacts tab
- * Add comments, tooltip and new button to the contextual menu
- * add delete event, migrate to GTK 2.16 for sexy icons
- * Resolve ticket #1118
- * Update suse spec file
- * Add phone number cleanup functions, unit tests and panel
- configuration
- * Add pertinent test that fails
- * fix dependencies for suse package
- * Add contextual edit menu in history - #1120
-
- [ Alexandre Savard ]
- * Temporary comit: make speex wideband (16 khz)
- * Temporary: shared object for speex narrow band
- * Temporary: speex narrowband and wideband coexist
-
- [ Julien Bonjean ]
- * Fixed bug when no book selected
- * Fixed addressbook related compilation warnings
- * Fixed GTK client remaining compilation warnings
- * Fixed segfault when book removed since last sflphone run
- * Fixed bug when book is unreachable (ldap error)
-
- [ Alexandre Savard ]
- * Fix codec list in audio config window
- * Active/inactive speex codec by payload
-
- [ Julien Bonjean ]
- * Updated gitignore
- * Added some comments
-
- [ Emmanuel Milou ]
- * Add callto: handler script for browsers and al.
- * Integrate test compilation in the daemon build-system
-
- [ Julien Bonjean ]
- * Fixed g_object_unref warning for pixbuf
- * Cleaned too verbose output
- * Fixed toolbar update warning
- * Added support for asynchornous books open (first shot)
-
- [ Emmanuel Milou ]
- * Add a DBus call to fetch the call details from a call ID - Ticket
- #928
-
- [ Julien Bonjean ]
- * Improved async open books
- * Fixed bug #1139
-
- [ Emmanuel Milou ]
- * Add a way to save account order
- * commit missing files
-
- [ Julien Bonjean ]
- * Introduced log4c (ticket #1162)
-
- [ Emmanuel Milou ]
- * Load/save account order functionnal - ticket #813
-
- [ Alexandre Savard ]
- * Add CELT codec (#1143)
- * Make celt frame size 256 (*1143)
-
- [ Julien Bonjean ]
- * Switched everything to log4c (ticket #1162)
- * Updated eclipse settings
-
- [ Emmanuel Milou ]
- * Restore adding account - ticket #1172
- * Add liblog4c dependecy - ticket #1179
-
- [ Alexandre Savard ]
- * Double maxAvailByte for frame size in rtp (#1143)
-
- [ Emmanuel Milou ]
- * Add User-Agent SIP header - Ticket #1173
-
- [ Julien Bonjean ]
- * Fixed autoresize issue (#708)
-
- [ Emmanuel Milou ]
- * Remove libcppuint dependency for the debian packages
- * Look for libsexy only if gtk version < 2.16 - Ticket #1116
- * Remove libsexy dependency for jaunty. ticket #1116
-
- [ Julien Bonjean ]
- * Introduced unit tests (#1146)
- * Updated gitignore
- * Fixed Makefile (#1146)
-
- [ Emmanuel Milou ]
- * [TICKET #1112] Add a test on the voice buffer to send through iax
- packets
- * Remove doublon in dependencies
- * Remove warnings from the client test framework
- * Update version number to 0.9.5~beta
- * Update build-package script
- * Add check dependency in build-deps control file field
- * Create debian files for the new sflphone-client-gnome
- * [TICKET #1212] Add Replaces field in control files
- * [TICKET #1212] Fix manpages installation path
- * [TICKET #1212] Add maintainer scripts to create alternatives
- * [#1212] Update the manpages generation - edit preinst maintainer
- script
- * [#1212] Fix reference error in manpage
- * [#1212] Add missing files on the client side
- * [#1212] Fix debian docs files - no TODO file
- * [1212] Fix manpage creation problem
- * [#1220] Generate client-side glue files and marshaller at
- compilation time
- * [#1220] Generate server-side glue files at compilation time
- * [#1212] Change binary name to sflphone-client-gnome
- * [#1212] Update .gitignore to fit the new working tree
- * [#1220] Explicitly generate glue files before building the library
- * [#1220] Compile dbus directory before audio
- * [#1212] Create sflphone-common at the root of the repository
- * [#1212] Re-add pjproject
- * [#1212] Remove Makefile from repo
- * [#1220] Fix Makefile.am
- * [#1212] New working directory functional
- * [#1212] Update .gitignore
- * [#1212] Hack to make pjsip compile..
- * [#1220] Use non-installed binary for dbusxx-xml2cpp
- * [#1212] Add descriptive files, remove unuseful scripts from tools/
-
- [ Alexandre Savard ]
- * Restore speex codecs
- * add frame size for celt (#1143)
- * add framesize to codec, independant from audiolayer (#1143)
- * use codec frame size in rtp (#1143)
- * compute fixed_codec_framesize (#1143)
- * do not resample if not required (#1143)
- * add condition on resampling for decoder (#1143)
- * add a condition on bytesAvail == 0 from mic data
- * no maximum in rtp decode (#1143)
- * compute maximum for decoding (#1143)
-
- [ Emmanuel Milou ]
- * [#1146] Implement unitary tests on the client-side
-
- [ Alexandre Savard ]
- * use float instead of int to compute max nb of sample (#1143)
- * add nbSampleMax for unresampled data (#1143)
- * make thread sleep during 5 ms insead of 20 (#1143)
- * use unix usleep (#1143)
- * 50 usecond thread!!!!! (#1143)
- * try with the smallest compression (#1143)
- * use timer set at framesize (#1143)
-
- [ Emmanuel Milou ]
- * [#1161] Restore changelog version
-
- [ Alexandre Savard ]
- * Remove celt stuff
-
- [ Emmanuel Milou ]
- * [#1161] Update changelog
- * [#1220] Add Conflicts: sflphone in debian control files
- * [#1179] Add liblog4c3 runtime dependency
- * [#1212] FIx typo error in dependency list for itnrepid
- * [#1212] FIx .desktop file to point on the right exec
- * [#1212] Modify changelog replacing tag
-
- [ Sflphone Project ]
- * "[#1262] Updated changelogs for version 0.9.5-0ubuntu1~beta"
-
- [ Emmanuel Milou ]
- * [#1212] restore changelogs
-
- [ Sflphone Project ]
- * [#1262] Updated changelogs for version 0.9.5-0ubuntu1 Snapshot 2009-
- 04-27
-
- [ Emmanuel Milou ]
- * [#1212] restore changelogs
-
- [ Sflphone Project ]
- * [#1262] Updated changelogs for version 0.9.5-0ubuntu1~beta
-
- [ Emmanuel Milou ]
- * [#1212] restore changelogs
-
- [ Sflphone Project ]
-
- -- Sflphone Project <sflphone@mtl.savoirfairelinux.net> Mon, 27 Apr 2009 17:00:03 -0400
-
-sflphone-plugins (0.9.4-0ubuntu2) SYSTEM; urgency=low
-
- [ Alexandre Savard ]
- * Restore speex and GSM detection
-
- [ Emmanuel Milou ]
- * Fix bug #1090
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Wed, 8 Apr 2009 11:29:15 -0500
-
-sflphone (0.9.4-0ubuntu1) SYSTEM; urgency=low
-
- [ Emmanuel Milou ]
- * Integrate DBus-c++ and libiax2 in the main build system
- * Clean up in the working repository
- * Reorder hooks configuration panel
- * Protect case when no codecs are active
- * Fix some return values
- * Add unitary tests for the hook manager (premisces)
-
- [Yun Liu]
- * Update chinese translation
-
- [Sven Werlen]
- * Update german translation
-
- [Hussein Abdallah]
- * Update russian translation
-
- [Maxime Chambreuil]
- * Update spanish translation
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Fri, 3 Apr 2009 18:29:15 -0500
-
-
-sflphone (0.9.4-rc1) SYSTEM; urgency=low
-
- [ Emmanuel Milou ]
- * Fix bug while trying to hold/unhold several simultaneous call
- * Improve address book build system
- * Implement SIP url popup on incoming call
- * Improve GTK+ panel configuration
- [ Julien Bonjean ]
- * GTK+ client refactoring
- * GTK+ clean up
- * Address book improvment
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Fri, 27 Mar 2009 18:29:15 -0500
-
-sflphone (0.9.4-0beta1) SYSTEM; urgency=low
-
- [ Alexandre Savard ]
- * Display codec used during conversation on the GUI
- * Enable/disable STUN parameters at runtime
- * Refactor search bar use
- [ Emmanuel Milou ]
- * Build system fixes
- * Implement SIP re-invite
- * Implement IP to IP call
- [ Julien Bonjean ]
- * Integrate GNOME address book based on evolution data server
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Fri, 20 Mar 2009 18:29:15 -0500
-
-
-sflphone (0.9.3-0ubuntu3) SYSTEM; urgency=low
-
- [ Alexandre Savard ]
- * Both playback and record streams in PA_STREAM_CORKED (pulseaudio)
- * Use PLUGHW device for ALSA capture
- * Functional IAX and SIP recording for voicemail
- * Use the less CPU-consuming interpolator algorithm for resampling
- * Display in GTK GUI the codec used in conversation
- * GTK GUI use ASCII instread of utf-8
- * Add record menus in GTK GUI
- * Put on hold when dialing a new number
- * AccountID's are saved in the history
-
- [ Emmanuel Milou ]
- * Integrate DBUS C++, libiax2 in the git repository
- * Update website
- * Use libspeexdsp only if available on the system
- * Updated .gitignore file
-
- [Cyrille Béraud]
- * Account assistant manager improvment
- * Add an email request when creating a new account to receive voicemails
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Sat, 14 Feb 2009 13:29:15 -0500
-
-sflphone (0.9.3-0ubuntu2) SYSTEM; urgency=low
-
- [ Emmanuel Milou ]
- * Add compilation note in README
- * Use default ALSA plugin for capture
- * Fix the ALSA capture problem one more time
- * Clean up debug messages in dbus.c
- * Add libspeexdsp dependency
- * Remove implicit declaration compilation warnings
- * Fix links in the website, add release note
- * Change capture for the website front page
- * Add alsa devel dependency in build-depends control file field
- * Clean up, indentation, try to handle latency problems in iax/pulseaudio
- * Remove pjsip generated files from the repo
- * Use the previous declared curAlias function in accountwindow
- * Fix bug in history call duration when the call fails
- * Remove runtime warning in the GTK+ client
- * Add librsvg2-common dependency to load SVG under KDE
- * Refresh .gitignore
- * Update locales files + french translation
- * Add configuration panel for future noise reduction
- * Add configuration panel for audio record module
- * Daemon less verbose; accounts don't try to access STUn options anymore
- * Fix typo in configwindow
- * Add content in the official website
- * use a GTK_STOCK icon for the record button
- * Complete description text in the assistant manager
- * Add libtool flags in client configure.ac
- * Remove unuseful dependency (snd)
- * Fix SIP transfer problems
- * Remove previous version of PJSIP from the repo
- * Upgrade PJSIP to version 1.0.1
- * Add the new website source in the repository
- * Use libspeexdsp for silence detection only if available
-
- [ Loïc Faure-Lacroix ]
- * Ajout du logo gpl3
- * Ajout des images
- * Ajout de la section screenshot pour le site
- * Ajout du favicon dans le header
- * Modification des cartes
-
- [ Alexandre Savard ]
- * Clean up <speex/libspeexdsp>
- * Small cleanup
- * Save Wave fixed
- * Fix new call button when recording
- * libspeexdsp added
- * Recording: default home folder at startup
- * Minor changes to config window
- * IAX recording fixed
- * Set / get recording path, still need some GTK for client
- * AudioRecord file name format
- * Now recording in HOME folder
-
- [ Cyrille Béraud ]
- * Fix bug in reqaccount.c
-
- [ Maxime Chambreuil ]
- * Update spanish translation
-
- [Yun Liu ]
- * Update chinese translation
-
- [ Hussein Abdallah ]
- * Update russian translation
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Sat, 14 Feb 2009 13:29:15 -0500
-
-sflphone (0.9.3-0ubuntu1) SYSTEM; urgency=low
-
- * Remove debug
- * Join thread before leaving
- * Fix implicit declaration in reqaccount
- * Add REST code to build the request to server
- * Fix GValue initialization warnings
- * Update version number, fix implicit declaration, fix GTK markup
- warnings
- * Apply patch to create custom SIP account from our own server
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Fri, 06 Feb 2009 19:17:32 -0500
-
-sflphone (0.9.2-2ubuntu9) SYSTEM; urgency=low
-
- [ Alexandre Savard ]
- * Speex audio codec preprocessing initialization
- * peer hung up segmentation fault solved
- * Stop recording when transfering
- * Terminate only one call
- * Add isRecording() function
- * Fix call_icon GTK client
- * Fix SIPCallClose() function, recorded file now close properly
- * Function terminateSIPCall added in sipvoiplink and managerimpl
- * Fix thread destructor
- * setRecordingOption function implement in audiorecord
- * Record now implemented in Call class
- * Record interface complete (on hold erase previous recording)
- * Added recButton in client
- * Added: record button related icons
- * Record button added
- * Overload AudioRecord::recData to get mic and speaker data mixed
- * Recording now in audiortp::run() method
- * Audio recording working in AudioRTP: receiveSessionForSpeaker
- * Open/close a wave file when pulse audio stream start/stop
-
- [ Emmanuel Milou ]
- * Fix path for GTK+ icons; clean up
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Thu, 05 Feb 2009 18:27:53 -0500
-
-sflphone (0.9.2-2ubuntu8) SYSTEM; urgency=low
-
- [ Emmanuel Milou ]
- * Update changelogs
- * Fix bug in merge and in Makefile.am
- * Terminate only one call
- * Disable PJsip shutdown when changing STUN parameters
- * Function terminateSIPCall added in sipvoiplink and managerimpl
- * Add a timer to the alsa thread to not jam the CPU load
- * Fix bug in sipvoiplink.cpp
- * Clean shutdown of pulseaudio on quiting
- * Fix DTMF at first start with Pulseaudio
- * Remove zeroconf from the build system
- * Add a library manager + exception handling
- * Clean up in the working directory
- * Better handling of capture XRUNs
- * Restore mic adjust volume on ALSA layer
- * Protect device ALSA operation if not opened
- * Fix the switching layer bug
- * Use dynamic_cast<> to use audiolayer-specific methods
- * Open the audio devices only once at startup
- * Refactoring of the ALSA part
- * Functional plug-in manager
- * Use a C++ thread to handle tones and DTMF in ALSA
- * Restore IAXVoIPLink, restore Mutex
- * Make the plugins registering against the plugin manager
- * Migrate to 1->N relationship between voiplink and accounts
- * API plugin for registration
- * Use C++ thread in SIP, move everything in sipvoiplink
- * Complete singleton pattern for the plugin manager
- * Add -Wno-return-type compilation flag to remove warnings; Update
- version number in configure.ac
- * Add the dynamic loading for the plugin framework; integate unittest
-
- [ Yun Liu ]
- * Update rpm spec file
- * modify build package script and spec file for suse
-
- [ Alexandre Savard ]
- * Add audiorecorder plugin and testaudiorecorder
- * Add audio Recording class, edit global.h
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Wed, 04 Feb 2009 14:00:30 -0500
-
-sflphone (0.9.2-2ubuntu7) SYSTEM; urgency=low
-
- [ Emmanuel Milou ]
- * Update changelog to 0.9.2-6
- * Fix some dbus-glib implementation details on the client side
- * Init history after dbus initialization
- * Add error checking in useragent; Clean sipvoiplink
- * Prevent crash when trying to call an empty number
- * Set the volume of the playback stream to PA_VOLUME_NORM at startup
- * Fix GTK+ generic value double initialization
- * Fix jaunty control file dependency problems
- * Fix jaunty control file dependency problems
-
- [ Yun Liu ]
- * Fix bug ticket # 137
- * Tolerant to gsm library of OpenSuse 11
-
- [ Sven Werlen ]
- * Update german translation
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Fri, 23 Jan 2009 17:48:13 -0500
-
-sflphone (0.9.2-2ubuntu6) SYSTEM; urgency=low
-
- [ Emmanuel Milou ]
- * Migrate STUN configuration to the main config window
- * Update french translation
- * Other tiny memory leaks
- * Fix memory leak in sampleconverter.cpp
- * Generate packages from the release branch
- * update the build package script
- * modify the control files with architecture=any
- * Remove valgring uninitialized value
- * IAX and SIP use the same global variables to set account
- configuration ; fix broken code
-
- [ Maxime Chambreuil ]
- * Update spanish translation
-
- [ Hussein Abdallah ]
- * Update russian translation
-
- [ Yun Liu ]
- * Update translation files
- * Fix the bug when user uncheck the account which fails in the
- previous registration
- * Add stun error status
- * Fix bug ticket #143
- * Script for auto-install dependencies
- * Fix bug ticket #140
- * Fix bug ticket 141
- * Fix the reregister process when user change the details of an
- account
-
- -- Emmanuel Milou <manu@sulfur.inside.savoirfairelinux.net> Fri, 16 Jan 2009 18:19:05 -0500
-
-sflphone (0.9.2-2ubuntu5) SYSTEM; urgency=low
-
- * Fix memory leak in the pulseaudio callback
- * Update debian package generation script
- * Warnings removal in GTK+ client
- * Clean adjust volume method in alsalayer
- * Plug the sflphone playback volume control to the pulseaudio volume
- manager
- * Display the date in history according to the current locale
- * Generate the changelog according to the git commit messages
- * Complete header in chinese translation file
- * Use the right gpg key to sign the packages
- * add debian jaunty jackalope support
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Wed, 14 Jan 2009 21:17:20 -0500
-
-sflphone (0.9.2-2ubuntu4) SYSTEM; urgency=low
-
- [ Emmanuel Milou ]
- * add german translation
-
- [ Yun Liu ]
- * Fix GUI crash in Ubuntu8.10 64bit system
-
- -- Yun Liu <yun.liu@savoirfairelinux.com> Thu, 08 Jan 2009 13:08:51 -0500
-
-sflphone (0.9.2-2ubuntu3) SYSTEM; urgency=low
-
- [ Emmanuel Milou ]
- * The main thread synchronizes the ringtone thread
- * disable custom ringtone for the ALSA layer
- * Fix the Makefile.am in man directory, add a SEE ALSO section
-
- [ Yun Liu ]
- * Fix daemon crash caused by the previous patch ( for bug ticket #129)
-
- -- Yun Liu <yun.liu@savoirfairelinux.com> Tue, 06 Jan 2009 16:18:38 -0500
-
-sflphone (0.9.2-2ubuntu2) SYSTEM; urgency=low
-
- * Fix bug ticket #129
-
- -- Yun Liu <yun.liu@savoirfairelinux.com> Wed, 5 Jan 2009 15:54:53 -0500
-
-sflphone (0.9.2-2ubuntu1) SYSTEM; urgency=low
-
- * Migrate from eXosip library to pjsip
- * Add multiple SIP accounts support
- * Fix ringtones problems
- * Add a pulseaudio support
- * Improve audio quality with ALSA
- * Add chinese translation
- * Improve spanish translation
- * Migrate to a maintained C++ DBus bindings
- * Clean and improve the build system
- * Add build-dependency on Perl because we need pod2man to generate manpages
-
- -- Yun Liu <yun.liu@savoirfairelinux.com> Wed, 26 Nov 2008 09:47:53 -0500
-
-sflphone (0.9.1) unstable; urgency=low
- * Add a search tool in the history
- * Migrate some gtk_entry_new to sexy_icon_entry_new
- * Bug fix (Ticket #78): The voicemail password isn't displayed anymore in
- the history tab
- * Add the SIP registration expire value in the user file.
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Thu, 22 May 2008 11:14:25 -0500
-
-sflphone (0.9.0) unstable; urgency=low
- * Add history features
- * Call date
- * Call duration
- * Mouse events in the history tab
- * Smooth switch from the history tab to the calls tab
- * Remove most of GTK-Critical warnings
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Wed, 13 May 2008 16:58:25 -0500
-
-sflphone (0.9-2008-06-06) unstable; urgency=low
- * Audio bug correction: capture stopped after a few minutes of conversation
- with USB Plantronics sound card
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Tue, 06 May 2008 16:58:25 -0500
-
-sflphone (0.9-2008-05-06) unstable; urgency=low
- * Bug correction: account creation with the assistant
- * GTK+ warnings removal
- * libnotify warnings removal
- * Remove aliasing on the SFLphone logo
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Mon, 05 May 2008 16:58:25 -0500
-
-sflphone (0.9) unstable; urgency=low
- * Clean dependencies ( removal of libboost )
- * Several GTK improvement and updates
- -account window
- -configuration window
- * Migrate from GtkCheckMenuItem to GtkImageMenuItem
- * ALSA standard I/O transfers: MMAP instead of R/W
- * Fix speex audio quality
- * IAX2 protocol
- -Fix hold/unhold situation
- -Add on hold music
- * SIP protocol
- -Ringtone on incoming call
- -Fix transfer situation
- * Add desktop notification ( libnotify )
- * Improve the system tray icon behaviour
- * Improve registration error handling
- * Register/unregister from the account window takes effect without starting back SFLphone
- * Compilation warnings removal
- * Call history
- * Add an account configuration wizard
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Wed, 30 Apr 2008 16:58:25 -0500
-
-sflphone (0.8.2) unstable; urgency=low
- * Internationalization of the GTK GUI
- * English / French
- * STUN support
- * Slight modifications of the graphical interface ( tooltips, dialpad, ...)
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Fri, 21 Mar 2008 11:37:53 -0500
diff --git a/tools/build-system/launchpad/sflphone-plugins/debian/compat b/tools/build-system/launchpad/sflphone-plugins/debian/compat
deleted file mode 100644
index 7ed6ff82de6bcc2a78243fc9c54d3ef5ac14da69..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/sflphone-plugins/debian/compat
+++ /dev/null
@@ -1 +0,0 @@
-5
diff --git a/tools/build-system/launchpad/sflphone-plugins/debian/control b/tools/build-system/launchpad/sflphone-plugins/debian/control
deleted file mode 100644
index 8e52e26ae28c1633dd5f17e88a2602ec1fa0aa66..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/sflphone-plugins/debian/control
+++ /dev/null
@@ -1,20 +0,0 @@
-Source: sflphone-plugins
-Maintainer: SavoirFaireLinux Inc <julien.bonjean@savoirfairelinux.com>
-Section: gnome
-Priority: optional
-Build-Depends: debhelper, libgcc1, autoconf, automake, libtool, libgconf2-dev, libgtk-3-dev, libebook1.2-dev, libedataserver1.2-dev
-Standards-Version: 3.7.3
-
-Package: sflphone-plugins
-Priority: optional
-Architecture: any
-Depends: sflphone-gnome (=${source:Version}) | sflphone-gnome-video (=${source:Version}), ${shlibs:Depends}, ${misc:Depends}
-Replaces: sflphone
-Conflicts: sflphone
-Homepage: http://www.sflphone.org
-Description: Evolution addressbook plugin for SFLphone
- Integrate evolution addressbook functionality to SFLphone.
- SFLphone is meant to be a robust enterprise-class desktop phone.
- SFLphone is released under the GNU General Public License.
- SFLphone is being developed by the global community, and maintained by
- Savoir-faire Linux, a Montreal, Quebec, Canada-based Linux consulting company.
diff --git a/tools/build-system/launchpad/sflphone-plugins/debian/copyright b/tools/build-system/launchpad/sflphone-plugins/debian/copyright
deleted file mode 100644
index 9bed104ba8882532610d33e5b8374a074736207a..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/sflphone-plugins/debian/copyright
+++ /dev/null
@@ -1,28 +0,0 @@
-This package was debianized by Alexandre Savard <alexandre.savard@savoirfairelinux.com> on
-Tue, Jun 2011 09:47:53 -0500.
-
-It was downloaded from the git repository of SFLphone: git://sflphone.org/git/sflphone.git
-
-Upstream Author: Alexandre Savard <alexandre.savard@savoirfairelinux.com>
-
-Copyright:
-
-Savoir-faire Linux Inc.
-
-License:
-
-This software is copyright (c) 2004-2016 Savoir-faire Linux Inc.
-
-You are free to distribute this software under the terms of
-the GNU General Public License version 3.
-On Debian systems, the complete text of the GNU General Public
-License can be found in the file `/usr/share/common-licenses/GPL'.
-
-This program is distributed in the hope that it will be useful,
-but WITHOUT ANY WARRANTY; without even the implied warranty of
-MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
-GNU General Public License for more details.
-
-You should have received a copy of the GNU General Public License
-along with this program; if not, write to the Free Software
-Foundation, Inc., 51 Franklyn St, Fifth Floor, Boston, MA 02110-1301, USA.
diff --git a/tools/build-system/launchpad/sflphone-plugins/debian/cron.d b/tools/build-system/launchpad/sflphone-plugins/debian/cron.d
deleted file mode 100644
index d11e61177739b56bce3aac6de6483b48e797a258..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/sflphone-plugins/debian/cron.d
+++ /dev/null
@@ -1,4 +0,0 @@
-#
-# Regular cron jobs for the sflphone package
-#
-0 4 * * * root sflphone_maintenance
diff --git a/tools/build-system/launchpad/sflphone-plugins/debian/dirs b/tools/build-system/launchpad/sflphone-plugins/debian/dirs
deleted file mode 100644
index e8b33c539969a5aa8fc5164dca26de981e35eb12..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/sflphone-plugins/debian/dirs
+++ /dev/null
@@ -1,3 +0,0 @@
-usr/lib
-usr/lib/sflphone
-usr/lib/sflphone/plugins
diff --git a/tools/build-system/launchpad/sflphone-plugins/debian/docs b/tools/build-system/launchpad/sflphone-plugins/debian/docs
deleted file mode 100644
index 9830da213fdb4baf4d68538e8c8e490248e209e1..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/sflphone-plugins/debian/docs
+++ /dev/null
@@ -1,5 +0,0 @@
-NEWS
-README
-ChangeLog
-AUTHORS
-
diff --git a/tools/build-system/launchpad/sflphone-plugins/debian/manpages b/tools/build-system/launchpad/sflphone-plugins/debian/manpages
deleted file mode 100644
index e69de29bb2d1d6434b8b29ae775ad8c2e48c5391..0000000000000000000000000000000000000000
diff --git a/tools/build-system/launchpad/sflphone-plugins/debian/rules b/tools/build-system/launchpad/sflphone-plugins/debian/rules
deleted file mode 100755
index 0395b7ec3590b4ca440b986c447412fcf032feb4..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/sflphone-plugins/debian/rules
+++ /dev/null
@@ -1,116 +0,0 @@
-#!/usr/bin/make -f
-# -*- makefile -*-
-# Sample debian/rules that uses debhelper.
-# This file was originally written by Joey Hess and Craig Small.
-# As a special exception, when this file is copied by dh-make into a
-# dh-make output file, you may use that output file without restriction.
-# This special exception was added by Craig Small in version 0.37 of dh-make.
-
-# Uncomment this to turn on verbose mode.
-#export DH_VERBOSE=1
-export DH_OPTIONS
-
-package=sflphone-plugins
-
-CXX = gcc-4.0
-CFLAGS = -Wall -g
-
-configure: configure-stamp
-configure-stamp:
- dh_testdir
- # Add here commands to configure the package.
- ./autogen.sh
- ./configure --prefix=/usr
- touch configure-stamp
-
-
-#Architecture
-build: build-arch build-indep
-
-build-arch: build-arch-stamp
-build-arch-stamp: configure-stamp
-
- # Add here commands to compile the arch part of the package.
- $(MAKE)
- touch $@
-
-build-indep: build-indep-stamp
-build-indep-stamp: configure-stamp
-
- # Add here commands to compile the indep part of the package.
- #$(MAKE) doc
- touch $@
-clean:
- dh_testdir
- dh_testroot
- rm -f build-arch-stamp build-indep-stamp configure-stamp
- # Add here commands to clean up after the build process.
- [ ! -f Makefile ] || $(MAKE) distclean
-
-ifneq "$(wildcard /usr/share/misc/config.sub)" ""
- cp -f /usr/share/misc/config.sub config.sub
-endif
-ifneq "$(wildcard /usr/share/misc/config.guess)" ""
- cp -f /usr/share/misc/config.guess config.guess
-endif
- dh_clean
-
-install: install-indep install-arch
-install-indep:
- dh_testdir
- dh_testroot
- dh_clean -k -i
- dh_installdirs -i
- # Add here commands to install the package into debian/sflphone.
-
-install-arch:
- dh_testdir
- dh_testroot
- dh_clean -k -s
- dh_installdirs -s
- # Add here commands to install the arch part of the package into
- # debian/tmp.
- $(MAKE) DESTDIR=$(CURDIR)/debian/$(package) install
- dh_install -s
-# Must not depend on anything. This is to be called by
-# binary-arch/binary-indep
-# in another 'make' thread.
-
-binary-common:
- dh_testdir
- dh_testroot
- dh_installchangelogs ChangeLog
- dh_installdocs
- dh_installexamples
-# dh_installmenu
-# dh_installdebconf
-# dh_installlogrotate
-# dh_installemacsen
-# dh_installpam
-# dh_installmime
-# dh_python
-# dh_installinit
-# dh_installcron
-# dh_installinfo
- dh_installman
- dh_link
- dh_strip
- dh_compress
- dh_fixperms
-# dh_perl
- dh_makeshlibs
- dh_installdeb
- dh_shlibdeps
- dh_gencontrol
- dh_md5sums
- dh_builddeb
-# Build architecture independant packages using the common target.
-binary-indep: build-indep install-indep
- $(MAKE) -f debian/rules DH_OPTIONS=-i binary-common
-
-# Build architecture dependant packages using the common target.
-binary-arch: build-arch install-arch
- $(MAKE) -f debian/rules DH_OPTIONS=-s binary-common
-
-binary: binary-arch binary-indep
-.PHONY: build clean binary-indep binary-arch binary install install-indep install-arch configure
diff --git a/tools/build-system/launchpad/sflphone-plugins/debian/substvars b/tools/build-system/launchpad/sflphone-plugins/debian/substvars
deleted file mode 100644
index 566a162f0d3708c2c131a6eff863df6727922259..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/sflphone-plugins/debian/substvars
+++ /dev/null
@@ -1 +0,0 @@
-plop=0.9.6
diff --git a/tools/build-system/launchpad/sflphone-video/debian/changelog b/tools/build-system/launchpad/sflphone-video/debian/changelog
deleted file mode 100644
index 19ebbcbbb64b5a494fd299f9ef92537b1550bede..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/sflphone-video/debian/changelog
+++ /dev/null
@@ -1,3585 +0,0 @@
-sflphone-video (1.1.0-rc20120607~ppa1~SYSTEM) SYSTEM; urgency=low
-
- ** SNAPSHOT 1.1.0-rc20120607~ppa1~SYSTEM **
-
- * * #12071: audiopreferences: fix make check
- * * #12071: cleanup
- * #12071: Lower expat dependency version
- * * #12085: alsa: fix ringtone update bug, cleanup
- * #12071: Fix pulseaudio compilation error
- * #12071: Make pulseaudio optional at configuration time
- * * #12091: daemon: issue warning if falling back to ALSA
- * video: fixed make check
- * *#12085: alsa: don't segfault on snd_pcm_avail_update error
- * #12070: Add --without-pulse option
- * * #12055: video: fix build for older libav
- * [ #11886 ] cleanup
- * [ #11886 ] Add basic reverse peer naming support
- * [ #12008 ] Implement GUI part
- * * #12012: video: fix some regressions
- * * #12002: yamlparser: don't wipe out config if going from normal
- build to --enable-video
- * [ #12008 ] Add ConfigurationManager::getRingtoneList()
- * * #12002: video: fix config file serialization/deserialization
- * * #11987: managerimpl: fix bugs in conference when removing
- participants
- * * #11987: manager: fixed transfer from conference
- * * #11987: mainbuffer: cleanup logging
- * * #11979: pulse: fixed mismatched device list
- * * #11971: audiolayer: fix bugs with getDeviceList
- * * #11960: manager: validate conference earlier when processing
- participants
- * * #11960: manager: fixed segfault on transfer from conference
- * * #11966: IP2IP: make alias consistently IP2IP
- * * #11965: sipvoiplink: add more error checking in SDP negotiation
- * * #11964: mainbuffer/ringbuffer: cleanup API
- * sdp: remove unused variable warning
- * * #11941: video: fix deprecated libav_api warnings
- * * #11949: pulselayer: fix bug in getDeviceList
- * video: whitespace fixes
- * * #11951: video: fixed threading issues for ucommon Thread
- * [#11848] Properly disable testPulseConnect
- * [#11848] Disable pulseConnect test
- * sdp: cleanup
- * sdp: cleanup
- * * #11860: mainbuffer: remove dead and/or buggy code
- * * #11851: sdp: fixed gcc type narrowing warnings
- * * #11851: audiostream: fixed gcc type narrowing warnings
- * * #11841: don't put code with side effects in assert()
- * * #11840: audiortp: remove some global symbols/variables
- * * #11828: audiofile: fix broken build
- * * #11828: audioloop: don't shadow sampleRate variable in derived
- classes
- * * #11818: gnome: stop daemon on SIGTERM, SIGINT or SIGHUP
- * * #11499: daemon should also quit gracefully on SIGHUP
- * * #11813: daemon configure should fail if expat is not installed
- * #10304: updatePlaybackScale dbus method uses int 32 bit for size and
- position (allow for 24 days long recording playback)
- * #10304: Add time lable for seekslider
- * * #11780: sip: don't use abort or leak calls on error and don't
- restrict SDP size to 1000 bytes in transaction_request_cb
- * * #11735: daemon: added timestamp start to call details
- * * #10304: historyitem: added operator > defined in terms of operator
- <
- * * #11252: historyitem: added missing unistd.h header
- * * #11252: daemon: removed deprecated zrtp code
- * * #11728: yaml: check that nodes are valid before using them.
- * * #10797: send DTMF over RTP as per RFC2833
- * * #11706: managerimpl: added unsetCurrentCall method
- * * #11698: daemon: fix build for c++11
- * * #10304: historyitem: file_exists need not be a member method
- * * #11499: managerimpl: don't crash if signal and dbus try and finish
- the manager at the same time
- * #10304: Prevent from storing removed files in history
- * * #11499: daemon: Exit cleanly on SIGINT or SIGTERM
- * * #10226: audiocodecfactory: use array instead of vector for codec
- name lookup
- * * #10226: audiocodecfactory: make codec loading stricter
- * #10304: RCecale positions and size values for playback recording
- * #11530: Make sure that only appropriate configuration option are
- parsed for IP2IP calls
- * * 11480: video: disabled by default
- * * #11459: history: protect historyitems vector with mutex
- * * #11448: fix video preferences for empty camera list
- * #10304: Implemented playback seek in gnome client
- * * #11269: video: fix codec per account management
- * #10304: Implemented playback scale in gnome client
- * Fix includes for gcc 4.7
- * * #11269: make clearer distinction between codecs and audiocodecs
- * * #11269: merged master into video
- * * #10296: managerimpl: more usage of getCallFromCallID
- * * #10296: verify that calls exists before trying to join them in a
- conference
- * * #11208: bump version numbers for release 1.1.0
- * managerimpl: rename ManagerImpl::serialize/unserialize ->
- join_string/split_string
- * Fix warnings in resampler test
- * * #10732: sipvoiplink: fix code that validates IP address
- * Add historyChanged signal, better than managing it client side
- * #10795: fix sipaccount deserialisation broken
- * #10736: implement getConferenceId dbus method given a call id
- * #10736: do not use iterator in daemon when joining conferences
- * #10736: Fix joining conferences in daemon
- * * #10736: gnome: fix crash on restart with active conference
- * managerimpl: removed unused pulselayer.h header
- * Save history everytime it change, prevent the file never to be saved
- in some senario (SIG, crash, ASSERT, etc)
- * * #10320: manager: check that participants are unique before joining
- * #10335: Add a noise suppressor for incoming rtp streams
- * * #10322: sip: registration state should not be always set to
- ErrorAuth on error
- * #10220: Fix recording thread does not exit when hanging up while
- recording
- * * #9903: fix includes for new ccrtp
- * #10230: Get back default mainbuffer sampling rate to 8kHz, no need
- of decoding noise suppressor
- * #10230: Use a different samplerate converter for rtp encoding and
- decoding
- * * #9903: create DynamicPayloadFormat on stack, initialize earlier
- * * #9903: audiorecorder: initialize buffer to silence, not random
- data
- * #10230: Test for triangular and sine signals
- * #10230: Add resampling unit test
- * * #10230: DTMF sample rate should come from main buffer, it should
- not be hardcoded
- * * #10213: audiolayer: create samplerateconverter on the stack
- * * #10213: audiolayer: cleanup
- * * #10213: increase resample buffer size, and check output size when
- resampling
- * * #10095: sipvoiplink: check pointers before using them
- * #9981: IP2IP calls based on ip address instead of sip:
- * * #10213: speex codecs should initialize their own parameters
- * * #10213: account: removed redundant cast
- * * #9832: removed extra printf
- * * #10172: include -sflphone in recording file name
- * * #9832: cleanup logging in tests
- * * #10096: srtp: use vectors to simplify key/salt manipulation
- * #10096: add case for non-srtp calls
- * [ #10121 ] Sync the KDE with daemon, fix a few issues and implement
- a recorded call player
- * #9980: make keep registration optional as there is different
- behavior on different registrar
- * #10096: use c++ arrays to store keys in srtp sesssion
- * * #10018: renamed registration related keys in dbus
- * #10096: Fix onhold/offhold srtp
- * * #8586: fixed make distcheck
- * * #9832: logger: don't hide logging if NDEBUG is present
- * #10096: Reinit crypto context when required on INVITE request
- * * #10111: Fixes segfault on empty config file
- * #100096: Set in/out queue crypto context at initialization, not when
- starting the thread
- * #10096: Update srtp key generation when holding/unholding a call
- * * #9831: logger: removed extraneous carriage-return character
- * * #10095: sipvoiplink: validate pointers before using them
- * * #10094: renamed config/config.{h,cpp} config/sfl_config.{h,cpp}
- * * #10090: fix segfault in transaction_state_changed_cb
- * * #9832: audio_rtp_record_handler: cleanup logging
- * * #9832: pulse: cleanup logging
- * * #9832: cleanup logging
- * * #9832: cleanup logging
- * * #9832: dbus: fix logging
- * * #9832: config: cleanup logging
- * * #9832: remove unused header
- * * #9832: manager: fix logging
- * * #9832: audio: fix logging
- * * #9832: audio: fix logging
- * * #9832: zrtp: cleanup logging
- * * #9832: AudioZRTPSession: cleanup logging
- * * #9832: AudioSRTPSession: fix logging
- * * #9832: cleanup logging
- * * #9832: AudioRtpSession: cleanup logging
- * * #9832: AudioRtpFactory: cleanup logging
- * * #9832: codecs: fix logging
- * * #9832: alsa: fix logging
- * * #9832: audio: clean up logging
- * * #9832: AudioRecord: cleanup logging
- * * #9832: Fix logging in Manager
- * * #9832: new logging macros
- * * #9979: ulaw: fixed unused var warning
- * * #10039: sipvoiplink: use references to avoid unnecessary parameter
- validation
- * * #10039: Fixed segfault on failed registration
- * * #9979: codecs: fixed unused variable warnings
- * * #9979: Don't do runtime assertions on data.
- * * #10016: SDP: removed verbose debuggin
- * #10016: Crypto context deletion are now managed inside the library
- * * #10016: srtp: cleanup
- * * #10016: SDES: fix uninitialized value bug, use const char*
- * * 100016: don't double free crypto contexts, and don't improperly
- copy CryptoSuiteDefinitions
- * * #100016: cleanup crypto contexts in audio_srtp_session
- * * #9979: removed unused methods from audicodec
- * * #9979: ulaw: normalize types
- * * #9979: cleanup
- * * #9979: Alaw: cleanup
- * * #9979: removed duplicate/superfluous code and type issues from
- g722
- * * #9979: AudioRtpRecord: let AudioRtpRecord handle fadeIn internally
- * #9980: Fix registration timer and transport shutdown on 401, default
- registration timer to 3600
- * * #9979: use std::tr1::array instead of plain array for audio
- buffers
- * * #9969: set loose routing param when creating route set
- * #9975: Fix account registration status display
- * * #9969: SIP: initialize body earlier
- * * #9969: sip: get received and rport fields if present in OK
- * * #9968: fixed segfault in transaction callback
- * #9898: make sure account are unregistered when sflphone quit, add
- timeout on pending transaction
- * #9910: fix contact header in outgoing request if via parameter are
- present
- * * #9910: SIP: use rport from VIA header if present
- * * #9910: SipTransport: pass parameters by const reference
- * #9910: fix sending call with new transport
- * yaml: remove verbose debug messages
- * * #9911: sipvoiplink: fixed "unused variable" warning
- * #9910: create new udp transport to fix registration failure with 606
- error & received parameter
- * * #9910: SIP: use pjsip error codes instead of magic numbers
- * * #9911: SIP Transports must be cached by IP:port
- * * #9910: SIP: cleanup
- * * #9905: SipTransport: address has to stay on stack to be valid
- * #9910: Update parse received parameter on 606 registration error
- * * #9911: simplify network manager state reporting
- * #9902: Fix SIPTest for IP to IP call
- * #9911: Fix network manager crashes
- * #9902: Move logic for ip2ip call in SIPVoIPLink
- * #9902: Move logic for ip2ip call in SIPVoIPLink
- * * #9910: fix 606 error code nomenclature
- * * #9905: fixed address initialization in createUdpTransport
- * * #9903: cleanup
- * #9902: Log failure cause when new outgoing call fail
- * * #9898: properly initialize ports
- * #9898: Unregister account when leaving sflphone
- * iax: create iaxvoiplink on stack
- * account: removed unused methods
- * * #9847: don't use assertions for input coming from DBus
- * * #9897: audiorecord cleanup
- * * #9897: audiorecord: cleanup, removed unused methods
- * #9897: Initialize and fallback recording path in home directory if
- not valid
- * * #9871: SipTransport: hide more implementation
- * * #9871: SipTransport: refactor SIP transport creation
- * * #9871: disable STUN for account if STUN setup failed
- * * #9847: check pointer before using it
- * #9871: Fallback on normal upd transport when stun resolution fails
- * Revert "#9871: Fallback on normal upd transport when stun resolution
- fails"
- * #9871: Fallback on normal upd transport when stun resolution fails
- * pulse: cleanup
- * * #9847: removed outdated README file
- * * #9847: use references instead of pointers where possible
- * * #9847: pass call by reference where possible
- * * #9847: audiolayer: fixed typo
- * * #9847: SIPVoipLink: gracefully handle invalid pointers
- * * #9847: check that transport is initialized
- * * #9847: SDP: avoid buffer overflow
- * * #9847: fixed segfault on bad call invite
- * * #9847: SDP: don't use assertions for runtime errors
- * * #9847: handle invalid remote session gracefully
- * * #9851: fixed segfault on stun socket cleanup
- * * #8586: fixed warnings
- * * #9849: added missing sstream header
- * #9623: add required TLS certificates for testing purpose
- * #9623: fixed tls registration
- * #9623: fix storing tls port in config for normal account
- * #9623: Allow all account to change tls listener port (not only
- IP2IP)
- * #9623: Allow for changing interface / port for tls transport
- * #9623: Open TLS listener on selected interface
- * #9833: remove unused debug
- * #9833: handlingEvents_ must be initialized to true when starting iax
- thread
- * #9830: Remove create_route_set from sipvoiplink
- * #9831: Fix sip transport port number
- * #9830: move sip header parsing function in sip_utils
- * Revert "* #8586: don't restore and save test files"
- * * #8586: fixed make distcheck
- * #9623: fixed tls registration
- * * #8586: don't restore and save test files
- * * #8586: refactored yaml code
- * #9623: fix storing tls port in config for normal account
- * #9623: Allow all account to change tls listener port (not only
- IP2IP)
- * #9623: Allow for changing interface / port for tls transport
- * #9623: Open TLS listener on selected interface
- * * #8586: added missing tests
- * #9833: remove unused debug
- * #9833: handlingEvents_ must be initialized to true when starting iax
- thread
- * #9830: Remove create_route_set from sipvoiplink
- * #9831: Fix sip transport port number
- * #9830: move sip header parsing function in sip_utils
- * * #8977: removed unnecessary AC_CANONICAL macros from configure.ac
- * * #8977: use actual PJSIP linking flags from pjproject/build.mak
- * * #9774: sipvoiplink's destructor should not be public
- * dbus: cleanup
- * * #9774: make sure sipvoiplink is destroyed before accounts are
- unloaded
- * * #9777: don't use deprecated auto_ptr
- * * #9778: removed AC_CHECK_FUNCS calls
- * * #9782: fix warnings in tests
- * * #9782: sip/sdp: fix emptiness checks
- * * #9782: sdes_negotiator: fix iterator usage and set dangling
- pointers to 0
- * * #9782: initialize all vars in iaxvoiplink
- * * #9782: use fstreams instead of fscanf
- * * #9782: yamlnode: fixed iterator usage
- * * #9782: yamlemitter: fix iterator usage
- * * #9782: yamlnode: make some methods const
- * * #9782: initialize all member vars in constructor
- * * #9782: Tone::interpolate should be const
- * * #9782: mainbuffer: get rid of unused vars
- * * #9782: GainControl::limit should be const
- * * #9782: fix ARRAYSIZE check
- * * #9782: use nanosleep instead of usleep
- * * #9782: fixed "inefficient emptiness test" cppcheck warning
- * * #9782: initialize dcblockers vars in constructor
- * * #9779: dropped CELT support
- * * #9750: moved sfl_data_format.h -> sfl_types.h
- * * #9750: refactored global.h
- * * #9736: restored command line options to daemon
- * tests: cleanup
- * * #8586: make distcheck was missing a header
- * tests: cleanup
- * * #9731: use all caps for application-wide constants
- * tests: cleanup
- * * #9730: cc++: enforce better checks in headers
- * * #9730: builds against libccrtp1
- * * #9572: sipvoiplink: fixed typo
- * * #9572: fixed threading issues with ccrtp2
- * * #9572: manager: pass config filename by const reference
- * * #9572: Replace utilspp singleton implementation
- * * #9571: regenerated config.{guess,sub} file to fix FTBFS on
- armel/armhf.
- * * #9665: siptransport: fixed udp_transport_start calls
- * #9620 Add test SIP account in configuration sample (test/sflphoned-
- sample.yml)
- * * #9641: audiortp: Fixed CryptoContext management
- * * #9641: fixed another memory leak in audio_srtp_session
- * * #9641: audiosrtpsession: fixed memory leak, simplified memory
- management
- * * #9641: avoid dynamic memory allocs/raw pointer usage in audio rtp
- stack
- * * #9641: get rid of getType/RtpMethod logic
- * fixed typo
- * #9572: make sflphone compile with libccrtp 2
- * * #9490: fixed registration state change callback that was crashing
- client
- * * #9547: fixed warnings in SipTransport header
- * #9547: Add SipTransport class
- * #9547: Extract all the transport layer from SIPVoIPLink to new
- SipTransport Class
- * #9547: Destroy the STUN resolver in Transport shutdown
- * sipvoiplink: removed erroneous FIXME
- * sipvoiplink: cleanup
- * #9547: Destroy the STUN resolver if server name change
- * sipvoiplink: fix warning about variable shadowing
- * #8320: Rename declared exception to avoid parameter shadowing
- * #8320: Send signal to client on stun failure
- * #8320: Use the same API for all transport creation (UDP, STUN, TLS)
- * * #9509: use vector for credential info
- * * #9508: fixes segfault in manager by changing order in which
- destructors are called
- * #8320: add dbus signal for stun failure
- * #8320: Use two different variables for status and return statement
- in stun's on_status_cb
- * * #9490: removed resolve_once parameter that was causing a segfault
- * #8320: make the retransmission callback to be rescheduled on error
- * HookPreference: cleanup
- * daemon: hookpreference: cleanup
- * iaxvoiplink: terminate() doesn't have to be virtual
- * sipvoiplink: functions need not be static if they are in an
- anonymous namespace
- * * #9037: moved CHECK macro into separate header
- * * #9037: cleanup error handling/checking in video threads
- * * #9037: video: cleanup
- * * #9037: only signal receiving_video_event for rtp sessions
- * * #9037: shared memory moved out of video_receive_thread
- * * #9381: daemon: fixed make check for video
- * * #9381: YAML_LIBS must be explicitly set in AC_SEARCH_LIBS macro
- * * #9381: reverted yaml check
- * * #9381: fix celt plugin compilation on fedora
- * * #9381: use PKG_CHECK_MODULES to test for yaml
- * * #9381: use autoconf macros and AC_SEARCH_LIBS
- * * #9381: use AC_SEARCH_LIBS, AC_CHECK_LIB
- * ringtonetest: cleanup
- * configurationtest: cleanup
- * instantmessagingtest.cpp: cleanup
- * mainbuffertest: cleanup
- * tests: cleanup
- * #8320: Make sure stun keep alive is enabled
- * call: push answer logic into call classes
- * sipaccount: simplify IP2IP code
- * sipaccount: avoid segfault if sipaccount is NULL
- * sipaccount: cleanup
- * #8084: Fix get sip header segfault when stun transport selected
- * * #9037: created shared_memory class
- * #8084: Init stun port with default valueas defined by RFC 3489
- * #9046: Move IP2IP_PROFILE global definition inside SIPAccount class
- * #9045: fix Changing the account expire is not taken applied in
- daemon
- * vidoe_receive_thread: cleanup
- * * #8968: suppress unusedFunction warnings for functions that are
- actually used
- * * #8821: fixed unit tests
- * #8821: Renamed account map keys for consistency
- * * #8968: audiorecord: added debug, clarified wave header creation
- * * #8968: added debug message to get rid of "unused struct member"
- warning
- * * #8968: manager: create History on the stack
- * * #9026: sfl::InstantMessaging is now a namespace
- * * #8698: managerimpl: removed unused method isWaitingCall
- * * #9008: don't include yaml headers in serializable.h
- * * #9008: cleanup account map initialization
- * refactor accountmap initialization
- * #9020: fix config file not generated when no account created
- * * #8968: audiorecord: removed unused getSndSamplingRate
- * * #8968: config: removed getConfigTreeItemIntValue
- * * #8968: recordable: removed unused getRecFileId
- * * #8968: removed unused Codec::getMimeType method
- * * #8968: manage lifetime of IMModule with auto_ptr
- * * #8968: history: removed unused method
- * * #8968: managerimpl: removed unused method
- * * #8968: config: removed unused methods
- * * #8968: managerimpl: removed unused getConfigBool/Int methods
- * * #8968: networkmanager: cleanup
- * * #8968: managerimpl: removed unused getConfig
- * * #8968: managerimpl: Manage telephoneTone_ with auto_ptr.
- * * #8968: history: fix memory leak upon exception
- * * #8968: AudioFile: initialize filepath earlier
- * * #8968: audiofile: fix memory leak on exception
- * * #8968: audiocodec: removed unused getChannel method
- * * #8968: use auto_ptr for dtmfKey
- * * #8968: use vector instead of dynamically allocated int array
- * * #8968: sdp.h: pass paramter by reference
- * * #8968: sipvoiplink: avoid C-style pointer casting
- * * #8968: yaml: avoid C-style pointer casts
- * * #8968: pulselayer: avoid C-style pointer casting
- * * #8968: recordable: removed unused getRecordingSmplRate method
- * * #8968: alsalayer: use preincrement for iterators
- * * #8968: config: removed unused method saveConfigTree
- * * #8968: mainbuffer: preincrement iterators
- * * #8977: history: added #include <fstream>
- * * #8968: don't leak memory on exception
- * * #8968: pulselayer: avoid C-style pointer casting
- * * #8968: Member variables must be initialized in AudioSrtpSession
- constructor
- * * #8968: fix potential memory leak in audiorecord
- * * #8968: Pass function parameter 'item' by const reference.
- * * #8969: fixed memory leaks in sdes_negotiator
- * video: cleanup
- * * #8940: removed video test source for now
- * #8763 Fix doxygen generation
- * #8763 Generate Doxygen with Hudson
- * * #8940: videosendthread: cleanup
- * sipvoiplink: cleanup
- * fileutils: cleanup
- * #8335 Fix default transport initialization on 5062, 5064
- * #8762: update mute for mic only, fix remove slide for pulseaudio
- * #8672: Add linear to decibel conversion functions in audio layer
- * #8672: Implement audio gain management in pulseaudio
- * #8671: Move audio gain management in audiolayer
- * * #8542: create symbolic link properly
- * * #8613: make check should fail early if another sflphone is running
- * #8449: Update version 1.0.2
- * * #8545: fixed error case
- * * #8545: fixed broken ringtone
- * * #8586: fixed make dist
- * sipvoiplink: use static_cast instead of reinterpret_cast if possible
- * * #8542: test for .git existence before moving pre-commit hook
- * * #8542: autogen.sh should not require git
- * eventthread: cleanup
- * * #8542: removed trailing whitespace from tree
- * * #8357: added disable video option to client
- * siptest: cleanup
- * * #8521: use avcodec_open2 instead of deprecated avcodec_open, if
- available
-
- -- Julien Bonjean <julien.bonjean@savoirfairelinux.com> Thu, 07 Jun 2012 16:08:15 -0400
-
-sflphone-common (1.0.0-rc20110930~ppa1~SYSTEM) SYSTEM; urgency=low
-
- ** SNAPSHOT 1.0.0-rc20110930~ppa1~SYSTEM **
-
- * update kde .gitignore
- * Fix bug in volume widget
- * More polishing for release
- * Bump version to 1.0.0
- * [#7023] Add the ability to load an abstract contact backend in the
- library to resolve more data, polish code
- * [#7021] More cleanup for release
- * Cleanup
- * [#7021] Refactor KDE client dbus handling, add a missing call in
- daemon and port the DataEngine to the new API
- * Remove some annoying debug
- * merge language scripts
- * remove obsolete 'VERSION' files
- * update install instructions
- * Add missing translations to gnome
- * language update
- * Revert "Don't reference count DBus clients, exit core immediately
- when one of them request it"
- * Don't reference count DBus clients, exit core immediately when one
- of them request it
- * [7021] Add contact abstraction support
- * [#7121] Polishing library (over). Indentation, spacing and naming
- are now consistent
- * codecs: link to libccrtp, don't use logger
- * Fix a daemon bug
- * [#7038] Fix adding contact
- * * #7037 : stop audio stream after all calls have been hanged up
- * [#7025] Add full support for bookmark
- * SFLPhone KDE do not destroy history anymore
- * Fix config skeleton
- * Close the daemon once and for all, no more automatic respawning
- * Fix "unregistered account" bug (I hope so)
- * Close SFLPhone at the right place, it still respawn, I don't know
- why
- * Remove dead code
- * Fix regressions introduced in the last commit
- * Dead code elimination 1/3
- * Fix bug, add "add contact" option, fix warning
- * * #7019: Fix IAX codec negociation
- * Remove or comment unnecessary/unhelpful debug output
- * Fix "same as local" account setting, fix IP2IP LED color
- * Add support for some more advanced config options and add missing
- config dialog icons
- * Fix crash with noise suppressor
- * Alternative can now be selected from the call view context menu
- * Add drag and drop support, initial context menu and fix 3 bugs in
- the account dialog
- * Add basic history drag and drop support
- * Complete contact support is back
- * * #6991 : fix IAX problems
- * Fix IAX accounts being disabled by default
- * Revert "deb: forge -g flags for pjsip"
- * * #5884: Disable debug code in pjsip
- * echo suppressor : more assertions
- * Don't let the daemon think crypto is enabled when it's not
- * Simplify ToneList
- * Some progress on contact support
- * Remove unused getRegistrationCount()
- * remove annoying debug
- * revert SIP bit of e27e5c39bad27bae28f574eb2cba7717e8956229
- * Simplify CallManager::placeCallFirstAccount
- * Fix crash on hold
- * * #6905 : SIP refactor
- * gnome client: be sure key exchange is set correctly
- * Move code into createSipTransport
- * Fix account registration on start
- * ManagerImpl::registerAccounts(): simplify
- * * #5884: don't mess with pjsip threads in echo suppressor
- * * #6905 : simplify udp/stun/tls pjsip transport creation
- * Restore and improve support for Call history
- * fix launchpad build
- * SIPVoIPLink: simplify / refactor
- * Fix libwidget linking
- * SIP: simplify
- * IM : simplify
- * gnome: remove some debug
- * AudioRtpFactory::stop() cannot fail
- * * #6905: simplify SIP code
- * pjlib: fix build without SSLv2, fix warnings
- * Port history to the new syntax
- * Test a dock widget based implementation for contact and history
- * Disable SSLv2 support from pjsip and sflphone
- * deb: forge -g flags for pjsip
- * Fix deb packaging to get debug symbols
- * remove debug
- * pjproject: update to last stable release (1.10)
- * Require gtk >= 2.20 and glib >= 2.24
- * tlsadvanceddialog: simplify
- * * #6902 : fix errors spotted by -DGSEAL_ENABLE
- * Update daemon dbus XML and port KDE config backend from dbus to
- local
- * Remove unused but set variables
- * * #6929 : fix IM widget, cleanup
- * Unconditionally enable debug symbols
- * Should fix many KDE issues
- * * #6886 : hitting backspace on empty number have no side effects
- * * #6905 : fix AudioCodecFactory access in optimized builds (-O > 0)
- * Remove unsupported and broken jaunty/karmic packages
- * * #6902 : avoid using some gtk deprecated functions
- * Update dbus introspection files
- * * #6904: removed unused contactmanager
- * * #6903 : use correct dbus-cxx package name
- * * #6902: don't use individual gtk headers
- * Fix a segfault when config is not present
- * Merge latest (0.9.13) KDE code. This version is not yet ready for
- git master, but better than the previous one
- * addressbook : simplify
- * * #5659 : sflphone-plugins doesn't depend on libedataserverui
- * * #5659 : addressbook doesn't use libedataserverui
- * gnome client doesn't depend on evolution
- * * #5695: addressbook: simplify
- * * #5695: addressbook : remove AddrBookHandle from plugin
- * * #5695 : addressbook : remove unused stuff in the client
- * * #5695 : addressbook : remove unused stuff, use static mutex
- * gnome client doesn't use evolution
- * gnome: use proper API to set GTK_CAN_FOCUS
- * * #6897: removed unused focus state vars/callbacks
- * gnome: fix calls to sflphone_fill_codec_list_per_account
- * * #6623: gnome: don't leak in mainwindow
- * gnome: mainwindow whitespace cleanup
- * gnome: actions.c parameter doesn't have to be a double pointer
- * * #6895: fix memleaks, cleanup in accountconfigdialog
- * * #6893: fixes segfault in client on clean history
- * * #6894: fix leaks, cleanup in sflnotify
- * daemon: fixed prints in main
- * * #6892: simplify, fix leaks in dialpad
- * * #6887: audiopreference creates audio layer
- * * #6660: use const char * const, not std::string for globally
- visible constants
- * * #6852: Preferences now solely responsible for audiolayer creation.
- * * #6860: refactor uimanager, also fixes #6865
- * * #6853: hangup as soon as all digits have been deleted
- * * #6852: alsa: retry if device is busy
- * * #6852: audiolayer creation depends only on preference.audioApi
- * * #6850: gnome: fix build for gtk < 2.22.0
- * cleanup in iax
- * alsa: typo
- * pulse: if we can't peek in audio input, we can't drop samples
- * * #6849: show error window if codecs are missing, instead of dying
- * EchoCancel: unused, remove
- * * #6629 : use number of samples as arguments for audio filters
- * * #6629 : remove unused Algorithm interface
- * * #6629 : use helper to call alsa functions and display error msgs
- * Remove unused type
- * * #6841: fix some error handling
- * * #6629: simplify AlsaLayer::alsa_set_params()
- * Get gdk key definition from header
- * * #6828: Replace raw key codes by gdk defines
- * remove some debug, enhance some other
- * mainbuffer: simplify
- * * #6561 : fix phantom call after transfer
- * Conference Participant set : simplify
- * SIPCall: remove unused functions, make invite session public
- * * #6229 : remove malloc/free from pulse audio loop
- * * #6629 : simplify pulse callbacks
- * * #6629
- * Simplify widgets
- * * #6629 : keep the correct audio module when frequency changes
- * * #6751: fixed erroneous debug msgs
- * callable_obj.h: removed unneeded pthread header
- * alsalayer: cleanup
- * * #6629: Always restart audio driver when changing parameters (ALSA
- only)
- * gnome GUI: don't block in DBus signal errorAlert()
- * * #6629 : simplify AudioLayer creation
- * * #6629 : remove unused and unconfigurable frameSize from audiolayer
- * * #6629 : remove unused error message from audio layer
- * Fix logic error when switching audio API
- * Remove unused AudioProcessing class
- * AudioRtpRecordHandler::initNoiseSuppress() : use noiseSuppress
- directly
- * * #6629 : use DC blocker directly in audio layers
- * * #6629 : clean AudioLayer
- * * #6629 : don't store mainbuffer inside audiolayer
- * * #6629 : correct AudioLayer::notifyincomingCall()
- * * #6554: cleanup, refactoring in sipvoiplink
- * * #6554: cleanup in iaxvoiplink
- * * #6554: throw exception in getSIPCall if pointer is NULL
- * * #6554: make some methods of sipvoiplink static
- * * #6655: cleanup in managerimpl
- * * #6554: refactoring, fix memleaks in sipvoiplink
- * * #6478: remove throw specs, cleanup in voiplink
- * * #6629 : remove unused AudioDevice
- * * #6655: removed more dependencies from managerimpl
- * * #6744: simplified numbercleaner
- * conference : remove one prototype
- * * #6743: fix ip2ip
- * Don't give glib warnings if icons are not found
- * gnome: fixed includes
- * Codec.h: removed unused function
- * * #6742 : clean dbus & icons
- * * #6699: refactor/cleanup accounts
- * icons: cleanup
- * timer : use second precision, not millisecond
- * calltree_update_clock : use correct type, returns something
- * * #6737: fixed typo in dbus call
- * * #6737: removed tests for removed API
- * * #6737: dbus: fixed bug from merge
- * * #6737: cleanup in accountlist
- * * #6737: cleanup in dbus
- * * #6740 : fix history double free
- * * #6740 : remove time updating thread from calls
- * * #6737 : use c99 for client
- * * #6738 : make history loading faster
- * sipvoiplink : don't crash on transfers
- * fixed typo
- * Remove unused file
- * Don't build networkmanager.cpp at all if NM is disabled
- * _debug* -> _debug
- * * #6554 : simplify sipvoiplink
- * hudson: added -x to git clean command
- * added git clean to hudson script
- * audiocodecfactory: cleanup
- * * #6718: refactored setTlsSettings into SIPAccount
- * * #6718: removed more unused methods
- * * #6718: refactored confmanager code into sipaccount
- * remove unused functions
- * * #6718: confmanager: removed more unused methods
- * AudioCodecFactory : cleanup
- * #6697 : Turn callableElement struct into union
- * * #6718: confmanager: removed more unused methods
- * * #6718: confmanager: removed more unused methods
- * * #6718: removed unused dbus methods, refactoring
- * * #6699: accounts: cleanup/refactoring
- * * #6699: refactoring, cleanup in accounts
- * * #6699: more account cleanup
- * remove unused autoconf variable
- * * #6714: fixed hudson script
- * make distclean in hudson
- * added || exit 1 to run_tests.sh call
- * * #6714: fixed make distcheck for sflphone-plugins
- * * #6714: fixed make distcheck for gnome client
- * * #6714: fixed make distcheck for daemon
- * git: #6698 split the main .gitignore file
- * gnome: gpointer is already a pointer
- * gnome: calltab_init: use calloc instead of malloc
- * * #6699: more account cleanup
- * * #6699: cleanup account
- * * #6554 : more *voiplink cleanup
- * * #6558 : more sipvoiplink simplification
- * * #6558: saner loadSIPLocalIP prototype
- * gnome: #6623 clean calllists
- * * #6692: more audiolayer cleanup
- * * #6692: cleanup/refactoring in audiolayers
- * * #6692: more forward declarations, AudioThread->AlsaThread
- * * #6692: audiolayer cleanup
- * * #6692: alsalayer cleanup
- * * #6558 : remove account creator
- * * #6558 : clean sipvoiplink
- * * #6554 : cleanup sipvoiplink
- * audiortp: cleanup
- * * #6657 : fix launchpad builds for good
- * * #6675 : send RTP dtmf events only once
- * * #6655: more cleanup
- * AudioRtpSession::updateSessionMedia() : simplify
- * * #6655: more cleanup in managerimpl
- * * #6655: removed more code, cleanup
- * * #6655: more cleanup, fixed infinite loop
- * * #6655: removed more unused files
- * * #6655: removed unused mutex
- * * #6655 removed more unused code
- * * #6655: removed unused methods
- * * #6655: cleanup in main
- * * #6663: fixed segfault when off hold from transfer
- * * #6658: user's active codec selection is respected
- * * #6660: static global string should be static const char* const
- class member
- * * #6659: use g_strcmp0, not strcmp for vals that may be null
- * callable_obj: fix double free
- * calltree_display_call_info() : simplify
- * * #6657: Fix launchpad builds
- * Logger::log() : simplify
- * AudioRtpSession : privatize members
- * * #6655: more constness, cleaned up/simplified methods
- * * #6654: call DBus::_init_threading so that dbus-c++ to make it
- threadaware
- * set default credentials on account creation
- * AudioCodecFactory::scanCodecDirectory() : simplify and correct
- * * #6623: fixed typos
- * * #6623: fixed more leaks
- * * #6623: fixed more leaks
- * * #6623: fixed more leaks, don't print codec name if null
- * * #6623: more leaks fixed in client
- * * #6623: fix more leaks, fixed some warnings
- * * #6623: fixed leak in history
- * updated gitignore
- * initialize dbus dispatcher correctly
- * Fix tests, hudson doesn't have a dbus daemon running
- * remove unused code
- * removeCall() : simplify , fix leak
- * stopRtpThread() : simplify
- * *CurrentCall : simplify
- * Fix memleak
- * fix serialization of audio api (pulse / alsa)
- * account map : simplify
- * remove call from callmap before terminating it, avoid use after free
- * * #6630 : don't make DBusManager a singleton
- * call: return confID by value
- * add back history code deleted by error
- * history : reverse logic
- * simplify history serialization and remove some debug
- * remove annoying debug
- * * #6464 : replace cerr with _error
- * * #6464: replace cout with logger macros
- * replace printf() with logger macros
- * update .gitignore
- * remove unused function
- * update eclipse projects
- * uimanager_new() : simplify
- * rename directories
- * celt: simplify a bit
- * Fix CELT configure.ac test
- * * #6612 : template speex codecs
- * * #6623: refactored conference obj
- * * #6623: refactored callable object, removed leaks
- * * #6623: more cleanup, fix leaks, make global vars static and rename
- them
- * * #6623: calltree: fixed memleaks, simplified code.
- * audiolayer: init pointer members
- * manager: catch exception on invalid hangup
- * * #6623: don't leak on calls to create_new_call
- * * #6611 : clarify codecs prototypes
- * ringtones : .au and .ul files are both ulaw
- * * #6611 : make sure samplerate converters are called correctly
- * ManagerImpl::switchAudioManager() : simplify
- * * #6623: fixed more leaks
- * * #6623: fixed more leaks
- * * #6623: fixed more leaks
- * * #6623: fixed leak, line-endings in imwidget
- * * #6627: zero-initialize pointers if they're going to be deleted
- * * #6628: don't leak calls on exceptions
- * Revert "audiortp: call join after calling stop on RtpThread"
- * sflphone-client: more constness
- * audiortp: call join after calling stop on RtpThread
- * * #6625: return 0 on successful completion
- * * #6624: fix segfault on servercallfailure
- * * #6621: Fixed double free, unlock mutex in ManagerImpl::terminate
- * * #6220: remove audio stream when peer hangs up
- * * #6596: AudioSymmetricSession shouldn't self-delete
- * resampler: grow internal buffers dynamically
- * merge up and down sampling => resampling
- * Leave test directory unchanged when running make check
- * audio algorithms : remove unused prototype
- * ringtone: detect codec from file extension
- * *AudioFile : simplify
- * * #6596: create local SDP on the stack, not the heap
- * * #6596: don't call Ost::Thread::terminate from dtor
- * audiofile: cleanup (samplerate -> unsigned)
- * remove unused func
- * samplerateconverter: cleanup
- * RingBuffer::Put() : remove unused return value
- * MainBuffer::putData() : remove unused return argument
- * audiolayer::putMain() : remove unused func
- * AudioLayer::putUrgent() : remove unused return value
- * * #6618: delete any remaining ringbuffers in destructor
- * RingBuffer::availForPut() : remove
- * * #6617: return from main rather than calling exit
- * MainBuffer::availForPut(): remove
- * RingBuffer: simplify
- * alsa : remove write only variable
- * fix memcpy declaration
- * bcopy(src, dst) -> memcpy(dst, src)
- * RingBuffer::Get() : remove constant volume argument
- * return a copy of the call ID, not just a reference.
- * MainBuffer::getDataById() : remove volume argument (always 100)
- * MainBuffer::getData() : remove constant volume argument
- * RingBuffer::Put() : remove constant volume argument
- * MainBuffer::putData() : remove constant (=100) volume argument
- * audiolayer: remove constant _defaultvolume
- * AudioRtpRecordHandler / AudioRtpSession : simplify
- * mainbuffer: fix test
- * iaxvoiplink : simplify
- * sip registration callback: fix a dbus crash
- * MainBuffer: simplify
- * AudioRtpFactory: return cached type of rtp session. The rtp session
- can have disappeared if the call was put on hold
- * AudioRtpFactory: remove unused setters
- * Fix launchpad builds
- * * #6611 : remove unused bandwidth codec information
- * * #6611: AudioCodec: remove useless/unused setters
- * make sure buffer string is initialized correctly
- * * #6596: declare certain destructors virtual
- * audiolayer : cleanup
- * Simplify doc build rules
- * * #6270: don't build dbus-api doc with make, should require make all
- * configure.ac: cleanup
- * Remove copy of dbus-c++ from libs/
- * * #6596: stop clock thread when peer hangs up
- * removed unused Fmtp.h
- * * #6595: more logical initialization order
- * * #6600 : fix account creation
- * * #6601 : fix configure.ac tests
- * remove unused variable
- * Don't mix stack and heap based allocations
- * Fix copyright (2009, 2008, 2009 -> 2008, 2009)
- * Fix warnings found by clang
- * * #6595: fix initialization order for AudioRTP
- * * #6592: removed typedef std::string CallID
- * * #6586: implement local g_slist_free_full for older glib versions
- * * #6579: fix memory leaks in client (there's a lot left)
- * ShortcutPreferences::setShortcuts() : simplify
- * Fix merge
- * * #6548: remove call to non thread-safe strerror()
- * AudioRtpFactory: each instance is associated to exactly one SipCall
- * create_audiocodecs_configuration() : make static
- * * #6269 : refactor AudioRtpSession
- * Fix AudioSymmetricRtpSession.h inclusion guard (cherry picked from
- commit c3081dce1cc1370d6d3558a4c4ef5cfac0d21caf)
- * * #6269: Rename AudioRtpSession to AudioSymmetricRtpSession
- * * #6574: Don't exit when connection to pulseaudio server fails
- * accountconfigdialog.h : remove some stuff from header
- * * #6560: fix configuration test
- * Fix warning in test
- * * #6560: don't hide password entry in security tab
- * * #6560: set initial password for SIP accounts
- * * #6506: remove useless pointer indirection
- * * 6560: password is now specific to IAX accounts
- * * #6560 : actually use, store, restore, transmit SIP credentials
- * * #6560: YamlEmitter: serialize sequences
- * YamlEmitterException: typo
- * ManagerImpl::computeMd5HashFromCredential() : simplify, fix memleak
- * * #6561: invite_session_state_changed_cb() : simplify
- * * #6561: More useful debug in VoIPLink::removeCall
- * * #6561 : fix ghost call reappearing in GUI after transfer
- * while -> for (make the code smaller)
- * * #6558 : Account::loadConfig() : move IAX code to IAXAccount
- * IAXVoIPLink::getAccountPtr : simplify
- * * #6554 : access the SIPVoIPLink directly, not per account
- * SIPVoIPLink is instanciated only once and is not associated to a
- single account
- * yamlnode: use const references when possible (still some left to do)
- * Account::_accountID: constify
- * VoIPLink: simplify, remove unused method
- * hudson test : no need to call run_tests.sh anymore
- * Remove AccountID type and AccountNULL define
- * Make check runs the test (no need to call run_tests.sh manually
- anymore)
- * gnome GUI: Fix tests
- * Revert "Move registration information from SIPAccount to
- SIPVoIPLink"
- * * #6392: pluginmanagertest: fix warnings reported by valgrind
- * * #6547 : remove unused exceptions
- * * #6547: CallManagerException: use runtime exceptions
- * * #6547: InstantMessageException: use runtime exceptions
- * * #6547: do not throw exceptions if some settings are not present in
- config file
- * * #6547: YamlParserException: use runtime exceptions
- * * #6547: VoipLinkException: use runtime exceptions
- * * #6547: YamlEmitterException: use runtime exceptions
- * * #6547: DTMFException: use runtime exceptions
- * * #6547: AudioFile: use runtime exceptions
- * * 6547: AudioZRtpSession: remove impossible error case
- * * #6547 : AudioRtpSession: remove impossible error case
- * * #6547: AudioZrtp: use runtime exceptions
- * * #6408 : send authenticationUsername to GUI
- * * #6408 : store/restore authenticationUsername from config file
- * SIPAccount: simplify
- * Move registration information from SIPAccount to SIPVoIPLink
- * SIPAccount::getAccountDetails : simplify
- * * #6540: yaml parser: simplify
- * sdp.cpp : fix a warning
- * * #6540: yaml parser : remove std::string typedefs
- * * #6540: Simplify yaml unserialization
- * * #6540 : add a Conf::ScalarNode constructor for booleans
- * setAccountDetails(): simplify
- * * #6408: store authentication username in daemon
- * * #6408: Be able to set the authentication username in the GUI
- * * #6507 : do not crash if the program is not sflphoned
- * Fix tests
- * macroify SIPAccount::unserialize()
- * Move all .cpp files from sflphoned target to libsflphone.la, except
- main.c
- * main() : simplify, return positive error codes
- * * #6507 : find codecs dir in build directory
- * * #6392: Sdp: move clean functions to destructor
- * AlsaLayer::adjustVolume() : simplify
- * alsalayer : reduce indentation
- * malloc/free -> new/delete
- * malloc/free -> new[]/delete[]
- * malloc/free -> new/delete
- * AudioSrtpSession: simplify base64 encoding
- * * #6392: Initialize std::string from pj_str_t correctly
- * * #6392: AudioRtpSession: Initialize remote port
- * Audio settings : Initialize _echoCancelTailLength and
- _echoCancelDelay(0)
- * Initialize variable
- * YamlParserException : fix use of stack variable after it has been
- deallocated
- * * #6392: fix memory leak in history
- * * #6392 AudioCodec : fix memory leak
- * * #6392 : fix memory leak in sip account
- * * #6408: clean up sipaccount (cosmetics mostly)
- * sipaccount.cpp serialize() : reduce number of lines
- * * #6392: invalid memory access
- * * #6392 : fix invalid memory access
- * * #6479: merged useful code from MimeParameters into Codec interface
- * * #6462: fixed hangup on IP2IP call
- * added run_daemon.sh script
- * test: remove unused variable
- * Remove functions only used by a failing test (cherry picked from
- commit fcf718cb75de7f1882dc61c07bb8d300dfa10f85)
- * * #6360 : make client tests build (cherry picked from commit
- 028b2835f040e51ab8ab979b32732b07b8798fce)
- * * #6360 : fix warnings in check_global test (cherry picked from
- commit 9e2bd6a7496dd64f6f48595e385760019aab1193)
- * * 6360: updated API calls in tests, but they're not building yet
- (cherry picked from commit 548f6f0f919b43772a3e9c667e5e292791281795)
- * Fixed include in tests (cherry picked from commit
- aeadc7525c1e31f936670ac8b02f0bcf387c38a8)
- * Remove unused variables and functions
- * IAX: fix warnings (cherry picked from commit
- fd7a113a11cac2cd9a7c36929e88ad28195c4c35)
- * Remove unused DEBUG define which interferes with logger.h (cherry
- picked from commit b2f72b91d0f43cb1dd94d138882a8caa9c841c24)
- * * #6392: no need to check for account NULLity since it is
- dereferenced above
- * * #6392: fix a memory leak, replace by stack allocation
- * * #6392: remove a variable assignement which confuses cppcheck
- * process_conference_participant_from_serialized() : remove unused
- function
- * * #6392: s/free/g_free/
- * * #6392: fix a memory leak in abookfactory_load_module()
- * * #6392: remove generate_call_id() used only once
- * * #6392: fix memory leak (opendir() without closedir())
- * * #6392: AudioRecorder(): ensures mbuffer is set
- * Remove SFLPHONED_VERSION from global.h, use autoconf PACKAGE_VERSION
- * #6298: Cleanup
- * #6331: Fix deleting ringtone file after call have been answered
- * * #6330: merged user_cfg into headers
- * #6298: Fix conference recording file update at conference end
- * #6298: Fix record file name serialization for conference
- * * #6295: cleanup of codec hierarchy
- * #6298: Fix gtk warnings
- * * #6300: added script to run tests
- * #6109: Add recording playback for conference
- * * #6300: tests do not require an installed sflphone
- * * #6295: re-removed clone methods
- * #6109: Fix gtk_critical warnings for incoming calls
- * #6109: Fix GTK_CRITICAL warning
- * #6109: Fix icons when history is not activated
- * #6109: Fix warnings
- * #6109: Implement stop recorded file playback signal
- * Revert "* #6295: removed unused clone method"
- * * #6295: removed unused clone method
- * * #6296: removed non existant file from Makefile.am
- * #6109: Stop fileplayback for outgoing call
- * #6109: Implement stop recording playback button
- * Fix binding names errors in dbus introspection file
- * #6109: Implement playback recorded file callback in client
- * #6109: Store recorded file path on client side
- * #6109: Add dbus methods for call recording playback
- * * #6290: remove unused classes from utilspp
- * * #6288: cleanup sdp
- * * #6288: fix exception usage
- * * #6288: simplify SdpException
- * * #6288: cleanup in sdp.cpp/h
- * #6109: Only display playback button if record file is set and valid
- * * 6290: updated configure.ac to remove functor Makefile
- * * #6290, #6289: removed unused classes from utilspp, fixed make
- check
- * #6109: Add button for history playback of recorded file
- * * #6289: removed unused observer class
- * * #6282: forward declare sdpMedia in sdp.h
- * * #6281: renamed setCallAudioLocal->setCallMediaLocal
- * #6183: Handle conference with more tahn two calls
- * #6183: Fix history icons when calling back a conference from history
- * #6183: Fix icons inconsistencies in history for conference hang up
- * #6183: Fix toolbar actions when selecting a conference in history
- * #6183: Fix conference serialization
- * #6268: Serialize only calls
- * * #6269: removed useless type testing
- * ignore some files in test/
- * * #6268: Remove dead class AudioSymmetricRtpSession
- * #6251: Do not had history calls in calllist when loading history
- file
- * #6251: Fix insertion in history map in before saving history file in
- daemon
- * #6251: Fix history unit tests
- * #6251: Order the list before serailization, get rid of the hashtable
- in history
- * #6251: Implement history serialization using a list wether than a
- map
- * * #6253: remove external audioport from header, make all members
- private
- * * #6253: don't store external local audio port (used for NAT) in
- Call
- * #6251: Add start_time timestamp in history serialization
- * #6251: Fix call insertion in conference items
- * #6233: Fix serialized account list terminated with a ";" character
- * #6238: Fix draggable history calls into current calls
- * #6233: Fix toolbar updates
- * #6233: Fix history
- * * #6235: remove pyc files from git tree
- * #6233: Handle cases when one or manuy calls are unreachable in
- createConfFomrParticipantList
- * #6233: Handle wrong numbers in createConferenceFromParticipantList
- * #6231: Fix drag-n-drop issue
- * * #6173 : move sippxml in tools
- * #6231: Fix merging issue
- * #6183: Implement conference unserialize
- * * #6212: remove extraneous flags from globals.mak
- * #6183: Unserialize conference data in conference
- * #6183: Add account information in request for conference call from
- history
- * #5755: Add -ldl to liker in sflphone-client-gnome
- * #5755: Fix fedora 15 compilation issue
- * #6183: Serialize conference participant phone number and account
- * #6183: Add conference timestamp in serialization
- * * #6186: don't include global.h, just logger.h
- * #6183: Fix saving history to file
- * #6183: Fix removing call from calllist
- * * #6184: remove pointers to Manager from AudioRtpSessions
- * #6183: Calling calltree_add_call explicitely for history
- * #6183: Ability to store conference inside history tab queue
- * * 6181: remove unused API from sipcall
- * #6171: Implment nreCallCreated callback
- * #6167: Fix participant list NULL ending
- * #6149: First draft of conference creation from history
- * #6149: Fix multiple call/conf selection callbacks ...
- * #6129: Fix place_call function called twice for pressing enter
- action
- * #6129: Fix double click action for history
- * #6149: Add dbus call for creating conference from history
- * #6129: Fix placing call from history and addressbook (still need to
- fix icon)
- * * #6148: removed unused AudioRtpFactory constructor
- * * #6145: remove unused isAudioStarted
- * * #6145: remove unused isAudioStarted
- * #6129: Add conference into history, fix call/conference selection
- * * #6143: don't use getType outside of serialization methods
- * * #6132: forward declarations instead of includes
- * * #6132: add constness, remove redundant "inline" keywords
- * #6129: Add timestamp to conference object to order history entries
- * * #6128: remove unused forward declarations from header
- * * #6127: make noncopyable class actually noncopyable
- * * #6125: don't include AudioRtpFactory in sipcall.h
- * #6123: Fix alsa ringback audio file
- * #6123: Fix raw audio file loading problem
- * #6109: Fix daemon plugin manager unit test
- * #6109: Fix history manager unit tests
- * #6109: Recording filename in daemon and client for history items +
- serialization
- * #6109: Refactor AudioFile to play recorded call
- * * #6104: AudioCodec moved to sfl namespace
- * * #6099: remove active flags from codec classes
- * #6095: Add notification-daemon as a runtime dependencies for rpm
- packages
- * #6095: Fix fedora 15 compilation in MineParameters.h
- * #6095: Declare static variable explicitely for client
- * #6095: Add logs to build OSC build machine
- * * #6098: global variables should have file-scope to avoid name
- conflicts
- * #6095: Fix compilation error for Fedora 15
- * #6095: Update SFLphone version to 0.9.14
- * #6095: Add specification file in opensusse build service for
- sflphone-plugins
- * #6073: Fix sflphone-plugins build on launchpad
- * #6093: Rename CodecDescriptor for AudioCodecFactory
- * * #6089: fix warnings in make check
- * * #6086: renamed codecs methods to audio_codecs
- * * #6085: renamed codec related dbus calls to audio_codec
- * #6065: Remove g_print from client, use DEBUG instead
- * #6065: Add actions name for addressbook
- * * #6085: renamed codecs* widgets/functions audiocodecs*
- * #6065: Fix Addressbook runtime warnings
- * #6065: Replace Codecs tab for Audio in account preference dialog
- * #6065: Fix "transfert" typo
- * #6065: Fix addressbook action runtime warning in uimanager
- * * #6082: fixes make check by adding libcrypto libs to test
- dependencies
- * #6073: Rename plugin/addressbook folders for addressbook/evolution
- in sflphone-plugins
- * #6074: Removed AC_SUBST from configure.ac when using
- PKG_CHECK_MODULE
- * #6073: Fix sflphone-plugins package build
- * #6073: Fix sflphone-common build
- * #6065: Fix runtime gtk warning when initializing searchbar without
- addressbook
- * #6063: Fix mozilla-tellify gitignore
- * #6063: Remove stream copy file using ifdef macro
- * * #6012: fix make dist for sflphone-common
- * #6063: Update .gitignore file
- * #6058: Fix base64 encoding related warnings
- * #6056: Fix SdpException handling
- * #6055: Fix unknown pargma warning for gcc <= 4.5
- * * #5949: test gcc version before disabling unused-but-set warning
- * #6054: Fix addressbook plugin compilation warning
- * #6048: Fix uimanager static initialization
- * #6046: Fix addressbook factory static initialization of member
- addrbook
- * #5979: Fix implicit function declaration warning
- * #6042: Fixed discarding qualifier warnings in client
- * #6041: Fix instant messaging unhandled case warning
- * #5994: Implement set current addressbook name and search type in
- addressbook plugin
- * #5994: add rules for launchpad packaging of addressbook plugin
- * #5994: Fix addressbook plugin configuration loading
- * #6027: Fix addressbook enabled test from configuration
- * #6027: No need of gnomedoc related macros in addressbook plugin
- * #6027: Add NEWS file required for build
- * #6027: Add addressbook plugin autogen.sh script
- * #6027: Remove plugins from client
- * #6027: Add sflphone-plugins folder at project's root level
- * #5994: Move addressbook folder from contacts to plugin folder
- * * #6011: removed unused Makefiles
- * * #6010: remove unused headers
- * * #5952: fix "string constant to char*" warnings
- * * #6009 fixed warnings
- * * #6003: finished cleanup of account classes
- * * #6003, #6004: cleanup of account classes, defaultAccount no longer
- global
- * * #6000: fix memory leak of args object
- * * #5998: removed using namespace std from networkmanager
- * * #5998: removed "using namespace std" from ZrtpSessionCallback
- * * #5998: removed using namespacestd from AudioZrtpSession.h
- * * #5998: remove "using namespace std" from auriorecord.h and
- MimeParameters.h
- * * #5998: remove using namespace std in main
- * * #5998: removed "using namespace std" from logger
- * * #5949: test gcc version before disabling unused-but-set warning
- * #5994: Installation of addressbook plugin
- * #5979: Implement codec full addressbook search from plugin
- * #5979: Implement addressbook factory and plugin
- * * #5981: unused webwidget removed
- * #5966: Account config synchronization fix (for stun)
- * #5954: Handle media name exception
- * #5954: Fix audio codec name display in client
- * #5954: Clean up getSessionMedia methods
- * * #5957: getRecordingSmplRate returns a value
- * #5954: Clean up getCurrentCodec methods
- * * #5950: remove "converting to non-pointer type 'int' from NULL"
- warnings
- * #5915: Full gain control version
- * * #5949: remove more unused variable warnings
- * * #5949: remove unused/unused-but-set variable warnings
- * * #5949: show_preferences_dialog returns a success value
- * * #5946: cleanup of include directives, undefined function
- * * #5515: comment out SSLv2 calls in pjsip
- * #5915: Implement different slope for attack tme and release time for
- gain control
- * #5915: use only one input signal for gain control (removed output
- buffer)
- * #5921: Fix no audio after holding a conference
- * #5916: Add gaincontrol files
- * #5916: Implement FFMPEG/CCRTP video streaming prototype
- * #5903: Fix call transfer during a conference
- * #5915: implement rms detector, first order averager, limiter for
- gain control
- * #5914: Fix call transfer when no notification request is required
- * #5899: Fix conference right-click segfault
- * #5884: temporary fix segfault in pjsip memory pool
- * #5883: Fix compilation issues on maverick and lucid
- * #5755: Fix fedora 15 compilation without patching ccrtp
- * [#5855] Make echo canceller optional
- * #5855: Fix echo suppression activation/deactivation
- * #5855: Implement pjsip echo canceller
- * #5814: Speex initialization function uses samples, not bytes
- * #5814: Test using more unbalanced signals
- * #5814: Fix buffer size for long echo length or long echo delay
- * #5814: Adjust level for echo cancellation at runtime
- * #5814: Process noise reduction before echo cancelling
- * #5814: Implement speex post echo canceller processing
- * #5814: Dump echo cancel file to disk
- * #5814: Add parameters for echo cancel
- * #5809: Add configuration parameters
- * #5809: Implement speex echo canceller in audio rtp session
- * #5814: Code cleanup
- * #5814: Fix conf creation with several incomming ringing calls
- * #5814: Fix conf creation segfault when dragging a call on hold on a
- ringing call
- * #5809: Added unit test for echo cancellation and implemented
- "process" virtual method
- * #5709: Add always recording option in configuration
- * #5709: Add always recording option in audio conference panel
- * #5709: Add core functionnality for always recording (missing config
- options)
- * #5769: Fix conference participant handling (detach/attach) and hold
- actions
- * #5747: Fix recording icons and state for conference when adding new
- participant
- * #5769: Code cleanup
- * #5769: Fix hangup unsent calls
- * #5769: Fix remove/add additional participant to conference
- * 5769: Several fixes concerning confererence handling
- * #5769: Fix compilation error
- * [#5769] Fix audio streams binding in main buffer
- * #5769: Removed access to audio mixer from audio layer
- * #5765: Fix audio crash for illformated wavefiles
- * #5765: Add maximum iteration for finding fmt and data "chunck"
- * #5589: Fix compilation of libnotify under
- * #5757: Fix abort signal when receiving INFO
- * #5747: Add usersDetached.svg
- * #5747: Handle offhold action for recording conference
- * #5747: Fix off hold action for conferences
- * #5747: Implement update conference in record action in calltree
- * #5747: Add new icons for recording conferences
- * #5747: Add recording state for conferences
- * [#5738] Remove getAudioDriver call from manager (replace by
- _audiodriver var)
- * [#5738] Refactor mutex protecting audiolayer
- * [#5737] Fix HD conference recording
- * [#5730] Fix start audio session after changing sampling rate
- * [#5714] Fix enter keyboard event for addressbbok and history
- * [5695] Fix addressbook combo box update when no addressbook selected
- * [#5695] Fix addressbook initialization and search bar update
- * [#5695] Add mutex for books_data in addressbook to protect async
- calls
- * [#5695] Get back addressbook open from uri
- * [#5695] Fix absolute addressbook URI for local addressbooks
- * [#5695] Implement libebook 3.0 interface
- * [#5571] Better logic for hangup (for case where call have not been
- sent yet)
- * [#5571] Update error handling in voip links
- * [#5571] Fix compile time warnings
- * [#5696] Fix installation dependencies for Natty
- * [#5669] Add mention that sflphone.org is for testing only
- * [#5693] Add natty in teh dput.conf file
- * [#5690] Remove not useful logs
- * [#5670] Use dynamic payload type for rtp dtmf
- * [#5668] Clean up sflphone configuration logging
- * [#5668] Fix hook checkbox configuration update
- * [#5666] Fix unit tests
- * [#5666] Manage event subscription
- * [#5666] Emit bye request when subscription is terminated
- * [#5666] Bye request should be sent after event subscription
- notification is done on transfer
- * [#5666] Make reinvite method static (to be called in pjsip
- callbacks)
- * [#5666] Hangup Call in manager for AccountNULL and IP2IP
- * [#5589] Use PKG_CHECK_MODULE for every client's dependencies
- * [#5623] Enlarge initial size of pjsip memory pool for calls (16k)
- * [#5564] Fix audio recording resampling for g722
- * [#5571] Move attribute handling for onhold/offhold actions in SDP
- session
- * [#5571] Codec negotiation refactored and unittested
- * [#5571] Implement tests
- * [#5571] Implement pjsip negociator
- * [#5571] Fix unit tests
- * [#5571] Add Fmtp.h to repository
- * [#5571] Integrate mime types and codec factory
- * [#5571] Handle exception when SDP negotiation fails
- * [#5570] Add sflphoned-sample.yml in repository
- * [#5564]: Implement stereo to mono mixing for rigntone
- * [#5342] Update audio stream initialization
- * [#5514] Restore test ni historytest suite
- * [#5514] Fix
- * [#5514] Disable test_create_history_path
- * [#5514] use pulseaudio in sample config file
- * [#5514] Fix test: load history from file
- * [#5514] Do not use X
- * [#5513] Make unit tests compile successfully
- * [#3947] Enable unit tests in Jenkins
- * [#5454] Fix build system to handle new version number
- * [#5454] Update languages from launchpad
- * [#5454] Add --without-celt in OpenSuse build service
- * [#5454] Change version number
- * [#5331] Added first SDP session tests
- * [#5273] Update nightly build version tags to conform dpkg rules
- * [#5211] Refactor send register method for iaxvoiplink and
- sipvoiplink
- * [#3950] Remove call being transfered from calltree
- * [#5211] Use appropriate memory pool for transport selector
- * [#5211] Fix strict aliasing rules warning in pjsip
- * [#5211] Bring back pjsip shutting down sleep to 1000 ms
- * [#5211] Fix registration callback segfault when closing the
- application
- * [#5211] Use the dialog memory pool for Route header in INVITE
- request
- * [#5211] Add temporary memory pool for findLocalAddressFromUri and
- findLocalPortFromUri
- * [#5211] Use individual memory pool for dtmfs
- * [#5211] SipVoipLink refactoring
- * [#3950] Attended transfer for conference calls
- * [#5284] Fix DNS resolution for Route with specified port number
- * [#5284] Some code cleanup
- * [#3947] Fix typo in hudson script
- * [#5284] Added sip route to REGISTER, INVITE, BYE request, plus DNS
- resolution
- * [#5266] Use RTP dtmf as default
- * [#5284] Added pjsip_process_route_set after setting routes in regc
- structure
- * [#5286] Fix parsing error due to long configuration file (removed
- max event)
- * [#5286] Fix false test in configuration emmiter
- * [#5286] Code cleanup
- * [#5286] Updated exception handling in configuration system
- * [#4969] Fix put SRTP call on hold
- * [#3950] Add debug messages
- * [#3950] Ability to perform an attended transfer
- * [#5276] Fix initialization problem in g722
- * [#3950] Add replace header in SIPVoIPLink::transferWithReplaces
- method
- * [#3950] Implemented attended method in SIPVoIPLink
- * [#3950] Cleanup transaction request received callback
- * [#3950] Implement dummy attended transfer in gnome-client
- * [#5249] Fix audio samplerate update algorithm for g722
- * [#5249] Fix uninitialized variable used in conditional jumps
- * [#5249] Fix conditional jump error in audiolayer (uninitialized
- value)
- * [#5267] Use autoconf 2.65 as a requirement (instead of 2.67)
- * [#5267] Restore manual pjsip configuration and compilation
- * [#5267] Autodetect celt version (0.9.1, 0.7.1)
- * [#5267] Fix deprecated macros in gnome client configure.ac
- * [#5267] Update configuration for libcelt-dev
- * [#5267] Fix build autoconf and automake
- * [#5227] Deactivate automatic call to astyle after compilation
- * [#5242] Hangup every calls before leaving
- * [#5237] Will now nightly-build for natty, Karmic deprecated
- * [#5229] Use inner class for rtp thread instead of inheritance
- * [#5211] Move mainbuffer unbind call in rtp final method
- * [#5211] Initialize sip call memory pool using 16 kb
- * [#5211] Use call memory pool in session reinvite
- * [#5211] Add debug messages
- * [#5211] Use and internal pool for calls
- * [#5211] Reduce pjsip memory pool usage for stateless error messages
- * [#5211] Refactor call deletion
- * [#5212]
- * [#5208] Refactor codec management for accounts
- * [#5168] Remove printf from codec's encode & decode method
- * [#5168] Fix celt compilation on launchpad
- * [#5168] Fix sflphoned compilation warnings in audiocodec.h
- * [#[#5168] Must keep the g722 specific RTP rate to avoid incoming
- packet timeout
- * [#5168] Fix static/dynamic payload rtp session update
- * [#5168] Throw SIPVoipLink Error if codec not instantiated in new
- outgoing call
- * [#5168] Fix dynamic/static codec payload type ambiguity
- * [#5169] Fix doubled IP2IP profile when no config file
- * [#4867] Add gtkinfobar in configuration panel
- * [#4867] Disable input/output/ringtone selection when using default
- alsa plugin
- * [#4952] Patches for possible buffer overflows
- * [$4885] Fix schemas problem
- * [#4885] sflphone-client-gnome.schemas not present during build
- * [#4885] Add gconf shemas directories in opensuse build system
- * [#4885] Add file/folder ownership for opensuse-factory build system
- * [#4906] Fix opensuse-factory build
- * [#4885] Update name dependency for libedataserver
- * [#4885] Fix non-void function without return in dbus-c++
- * [#4895] Update language translation
- * [#4896] Update session timestamp when updating media
- * [#4896] Reapply RTP hack for G722 payload type
- * [#4896] Update recording sampling rate when updating codec
- * [#4897] Save codecs in config for each configuration changes
- * [#4895] Do not save config when sflphone quit
- * [#4885] Update date for copyright
- * [#4885] Deactivate siptest that require more than one sipp instance
- * [#4879] Remove inmcoming call notification from IAX
- * [#4885] Some cleanup
- * [#4874] Add setCancel immediate/deffered for ost::Thread
- * [#4879] Fix incoming call notification
- * [#4878] Set keyboard focus on searchbar when selecting addressbook
- * [#4874] Fixed compilation warning
- * [#4874] Fixed compilation warning in sipvoiplink
- * [#4874] Fix compile time warning in RTP record handler
- * [#4874] Fix conditional jump in SDP
- * [#4874] Fix conditional jump based on uninitialized value
- * [#4874] Store call id within rtp thread context
- * [#4874] Fixed conditional jump based on uninitialised value in
- conference
- * [#4871] Fix default account fetching
- * [#4870] Delete RTP session when Refusing an incoming call
- * Restore IP to IP call
- * [#4857] Fix audio codec negotiation problem
- * [#3947] Adjust ressources allocated to compilation
- * [#3947] Disable unit tests in Hudson
- * [#4305] Free mutex only when really quiting SFLphone
- * [#4859] Update copyright to 2011 in every source file
- * [#3218] Character '.' stripped by the caller engine
- * [#4854] Fix typos, desktop entry
- * [#4847] Apply RTP modification to ZRTP session
- * [#4852] Update Karmic and Lucid dependencies
- * [#4852] Add Libedataserver and libedataserverui as gnome client
- dependencies
- * [#4852] Add authentication mechanism for EDS
- * [#4851] Fix segfault when closing pulseaudio layer too rapidly
- * [#4808] Some otehr cleanup
- * [#4808] Made some cleanup
- * [#4808] Added mutex in rtp session for codecs and noise process
- * [#4847] Update audio processing when updating RTP media
- * [#4842] Add support for linking with gold/ld --no-add-needed
- * [#4808] Make update g722 related static/dynamic payload logic
- * [#4827] Upper limit on the number of contacts to import from EDS is
- hard-coded to 500
- * [#4808] Fix put call on/off hold
- * [#4808] Implement early RTP start for incoming calls
- * [#4808] Audio stream is no longer start within RTP session.
- * [#4808] Removed coupling between audio layer and and RTP session
- * [#4702] Start audio rtp session as soon as it is created
- * [#4702] Init timestamp to 0
- * #4702: Send RTP packets immediately, no need of outgoing queue
- * [#4784] Update dbus-c++ version from gitorious
- * [#4702] Update RTP timeouts
- * [#4702] Lengthen RTP timeouts
- * [PATCH] Fixed compatibility with old libtool versions.
- * [PATCH] Accept older libebook (Maemo 5 has 1.4.2)
- * [PATCH] Fixed double-free error in preferences dialog
- * [PATCH] Fixed building of sflphone-common on Maemo5
- * [PATCH] Improved Gnome client initialization error handling. 1. It
- no longer segfaults when sflphoned isn't available. 2. User is
- provided with GUI error dialog.
- * [PATCH] Improved autogen.sh scripts 1. They do not require bash
- anymore 2. Added workaround for Debian bug #565663 3. Replaced
- manual autotools invocations with single autoreconf call 4. Non-zero
- return status on failure
- * Revert "[#4468] libtool <= 2.2 doesn't have LT_INIT macro so
- AC_PROG_LIBTOOL should be used instead."
- * Revert "[#4468] Libebook 1.4 is sufficient"
- * Revert "[#4468] Apply big path on dbus communication system"
- * [#4468] Apply big path on dbus communication system
- * [#4468] Libebook 1.4 is sufficient
- * [#4468] libtool <= 2.2 doesn't have LT_INIT macro so AC_PROG_LIBTOOL
- should be used instead.
- * [#4639] Fix determining default addressbook if this property is not
- set in gconf
- * [#4639] Fix memory leaks in Addressbook
- * [#4637] Fix opening default addressbook at sflphone init
- * [#4622] Free yaml events while parsing configuration file
- * [#4623] Fix conditional jumps based on uninitialized variable
- * [#4622] Fix leaks in yaml serialization engine
- * [#4616] Fix addressbook warnings
- * [#4514] Adjust RTP timestamp
- * #4527: Rename Karmic libyaml and Celt package in debian control file
- * #4495: Rework addressbook opening loop
- * [#4524] Increment RTP count when sending data
- * [#4524] DO NOT start RTP session twice
- * [#4367] Use PKG_CHECK_MODULE for celt
- * [#4367] Fedora package celt as celt (not libcelt)
- * [#4367] Astyling
- * [#4367] Update .po files
- * [#4367] Fix segfault in gensin
- * [#4354] Make celt a direct dependency on launchpad opensuse build
- service
- * [#4367] Make celt a required package, option --without-celt valid
- * [#4367] Fix zrtp timestamping error
- * [#4367] Fix audio zrtp timing
- * [#4367] Dispatch ZRTP packets
- * [#4367] Fix segfault when unloading account map
- * [#4367] Fix zrtp session
- * [#4367] Implement on packet receive
- * [#4367] use symetric audio rtp session, not dual
- * [#4367] Reduce packet receive/sent timeout
- * [#4367] Reduce RTP timeouts
- * [#4367] Move speaker data receive
- * [#4367] Move speaker data receive
- * [#4367] Move receive speaker data method
- * [#4367] Remove debug in rtp session
- * [#4367] Fix g722 codec clock rate
- * [#4367] Fix noise suppression initialization
- * [#4367] Fix segfault in RTP mic fadein method
- * [#4367] Refactor mic data encoding in rtp session
- * [#4367] Implement RTP main loop
- * [#4367] Fix compilation problem
- * [#4367] Fix AudioRtpclass using TRTPSessionBase
- * [#4367] Fix AudioRtpSession putDtmfEvent shadowing
- * [#4367] Fix AudioRtpSession putDtmfEvent shadowing
- * [#4367] Refactor RTP session (phase 2)
- * [#4367] Refactor RTP session (phase 1)
- * [#4367] Remove Redeclaration of SymetricAudioRtpSession in
- rtpfactory
- * [#4265] Add continue statement in for loop for invalid addressbook
- * [#4261] Makes addressbook initialization more robust
- * [#4257] Add maverick in build system
- * [#4233] Add sdp related unit tests
- * [#4233] Add condition and signal in two incoming call test
- * [#4243] Fix segfault in AudioSrtpSession
- * [#4243] Fix memory leak in AudioSrtpSession
- * [#4243] Make audio srtp optional in for incoming call
- * [#4243] Add boolean variable to make sure remote crypto context
- initialized only once
- * [#4243] Add documentation to AudioSrtpSession
- * [#4243] Use 80 bits authentication tags by default
- * [#4243] Init audio srtp remote crypto context in
- call_on_media_update
- * [#4243] Move SDP negotiastion in mod_on_rx_request
- * [#4243] Implement initLocalCryptoInfo to be called at different
- momment
- * [#4243] Init init local crypto context in when initializing audiortp
- * [#4243] Change key length according to sdes negociation
- * [#4243] Associate callid to accountid for incoming calls
- * [#4242] Fix no SDES keys in IP2IP calls
- * [#4242] Fix no SDES keys in IP2IP calls
- * [#4233] Test for call on/off hold
- * [#4233] Add two incoming call test
- * [#4233]
- * [#4233] Add 2 outgoing simultaneous call unit tests
-
- -- Julien Bonjean <julien.bonjean@savoirfairelinux.com> Fri, 30 Sep 2011 13:51:04 -0400
-
-sflphone-common (0.9.7~rc1~ppa1~SYSTEM) SYSTEM; urgency=low
-
- ** 0.9.7~rc1~ppa1~SYSTEM **
-
- * [#2462] Set explicitly the transport on incoming call too
- * [#2462] fix typo
- * [#2462] Use different address for SDP and call IP
- * [#2462] Use published address in SIP-SDP
- * [#2181] Fixed changelog files
- * [#2181] Updated spec file
- * [#2402] Fix pointer to int conversion warning (atoi)
- * [#2402] Remove daemon warnings, make indent
- * [#2459] Make sure the stream is opened when the call is answered
- * [#2402] Add conference related picture in documentation
- * [#2443] Not much ...
- * [#2399] Fix dialing display problem
- * [#2450] Fix incoming call already in conference crash
- * [#2399] Display peer name on the first line and peer number on the
- second
- * [#2450] Handle 403 FORBIDDEN when refused
- * [#2447] Bind offHold/onHold actions to button in gtk client
- * [#2447] Bind hangup action to button for conference
- * [#2447] Add conference action in gtk client's ToolBar
- * [#2381] Disable the password hashing in config file
- * [#2402] Cleanup
- * [#2366] Set callback to null when deleting Pulseaudio streams
- * [#1313] Fix main buffer unit test
- * [#1313] Fix audio layer unit test
- * [#2315] Hide pw in security tab, display when editing, sync with
- basic tab
- * [#1313] UnitTest change AudioRtpSession for AudioSymetricRtpSession
- instance
- * [#2402] Code cleanup
- * [#2444] Add debug to catch occasional crash when loading client's
- config
- * [#2444] Add debug info to catch occasional crash when loading config
- dialog
- * [#2402] Restore Call menu translations
- * [#2403] Use the published address if checked in GUI
- * [#2442] Add protection test in sdp
- * [#1841] Reapply pjsip patch concerning DNS SRV resolution
- * [#2384] Tags incoming call as direct SIP call, if applicable
- * [#2402] Change the monkey face
- * [#2315] Enable user to display password in clear text
- * [#2434] Force optimization level at 2
- * [#2284] Fix dbus_get_all_ip_interface compilation warnings
- * [#2431] Popup main window on incoming if applicable
- * [$2402] Fix simple warnings
- * [#2402] Fix implicit variable init order in LibraryManagerException
- * [#2402] Fixing implicit variable initialization warnings in
- AudioRtpSession
- * [#2402] Revert atoi change, fixing codec list doubled entries
- * [#2402] Fix gpointer to gint conversion
- * [#2402] Fix pointer casting to integer different size warning in
- codec list
- * [#2402] Fix warning discarting qualifiers from pointer target
- * [#2402] Fix gtk tree view assignement from incompatible type warning
- * [#1669] Fix audio recording folder utf-8 non compatibility issue
- * [#2414] Clean up debugs
- * [#2414] Use transport set in iptoip Account and update it frm
- preference
- * [#2348] Use macro N_() to mark ui.xml strings as translatable
- * [#2414] Rename getSipAddress/setSipAddress functions
- * [#2407] Fix volume controls display
- * [#2407] Fixes dialpad
- * [#2383] Set ip to ip config when clicking apply button
- * [#2404] Update call-to script - Maxime Chambreuil
- * [#2405] Client handles unknown call in current state as well
- * [#2383] Add DBUS signal to send IPtoIP local address and port as
- string
- * [#2383] Add Ip to IP config change apply call back
- * Clonflict
- * [#2402] Code cleanup
- * [#2383] Do the same for IPtoIP (init localn ip with first in the
- list)
- * [#2383] Use first interface in the list if local addresss is not
- defined
- * [#2403] Clean up unuseful addresses/ports
- * [#2403] Use the IP profile SIP port as global SIP port
- * [#2383] Fix dbus_get_all_ip_interface warnings
- * [#2383] Take into account sameAsLocal when loading published address
- * [#2383] Tsake into account sameAsLocal option when saving published
- address
- * [#2383] Update local ip address in ip to ip config
- * [#2383] Save ip 2 ip local port in config
- * [#2406] Update toolbar at startup
- * [#2284] Remove redefinition warnings + speex warnings
- * [#2383] Fix security table in account config
- * [#2383] Save ip 2 ip network interface parameters in config
- * [#2403] Restore sip transport selector
- * [#2383] Fix filling the Localt IP Address on account creation
- * [#2383] Fix Gtk-Critical when checking STUN
- * [#2383] Fix reopening account configuration display issue
- * [#2383] Load IPtoIP local address and port in preference iptoiptab
- * [#2383] Add LocalAddress and Localport in Preference IpToIp tab
- * [#2403] Use the address and port associated to the account as often
- as possible
- * [#1753] Removed pjsip generated files
- * [#1753] Removed remaining milenage lib references
- * [#2383] Add _publishedSameasLocal variable in sipaccount
- * [#2383] Add PUBLISHED_SAMEAS_LOCAL variable in config
- * [#2383] Fix stun set active or not when opening config
- * [#2181] Added RPM 64bits dbus patch
- * [#2402] Code indentation
- * [#2313] Force $(HOME).cache directory creation at startup
- * [#2383] Separate network interface and published address in account
- config
- * [#2400] Change dbus service installation path to libdir
- * [#2382] Move TLS related published address options in security tab
- * [#2382] Indent accountconfigdialog.c
- * [#2181] Install libdbus-c++ in $pkglib instead of $lib
- * [#1753] Remove ILBC code and disable it by default in the configure
- * [#1753] Remove milenage directory
- * [#2382] Fix switching interaface instabilities
- * [#2396] Save local ip in account creation wizard
- * [#2284] Remove warning on hold
- * [#2387] Fixes history searching and filtering
- * [#1215] Add samplerate display in the GUI
- * [#1663] Voicemail icon reflects voice messages
- * [#2395] Fix account registration ( specifically with callcentric)
- * [#2386] Strip "sip:" on incoming call, fixing history call back
- * [#2181] Updated spec files
- * [#1215] Display codec name in calltree instead of status bar
- * [#2390] Move back nbCalls and stopStream higher in refuseCall
- * [#2392] Fix ringtone during call in IAX
- * [#2391] Stop audio streams when there is 0 calls only
- * [#2391] Add debug when call state is not valid
- * [#2390] Clear returns in IAXvoipLink::sendAudioFromMic() method
- * [#2380] Fixing IncomingCallNotification not regular
- * [#2339] Query conference at client startup
- * [#2339] Working conference querying at startup
- * [#2339] Add conference in call tree
- * [#2339] Primitives to query conferences at client startup
- * [#2320] Add account selection in history
- * [#2355] Temporary solution: do not delete pointer when removing
- account
- * [#2380] Change algorithm in AudioRtp to trigger an
- IncomingCallNotification
- * [#2274] Comment sdebug in MainBuffer flush method
- * [#2274] Add flushMain() in ManagerImpl::addStream
- * [#2274] Add getBufferID() method in ring buffer
- * [#2274] Fix warning, comment debug in ringbuffer's flush method
- * [#2274] Use AudioLayer flushMain() and flushUrgent() in ALSA
- * [#2274] Clean up unused variable warning
- * [#2274] Protect minbudffer pointer on flushing
- * [#2274] Fix playATone method which writing empty buffer in urgent
- ringbuffer
- * [#2274] Use audio layer flushUrgent and flushMain in createStreams
- * [#2274] Use flush audio calls from audiolayer
- * [#2274] Flush when peer answered call
- * [#2375] Flush main buffer in iax when answering a call
- * [#2274] Parse displayname using c++ string method
- * [#2375] Flush main buffer when off holding calls
- * [#2375] Flush main buffer mon RTP startup
- * [#2376] Use now Pulseaudio module-cork-music-on-phone
- * Updated OSC packaging
-
- -- Julien Bonjean <julien.bonjean@savoirfairelinux.com> Fri, 20 Nov 2009 14:00:02 -0500
-
-sflphone-common (0.9.7~beta~ppa1~SYSTEM) SYSTEM; urgency=low
-
- ** 0.9.7~beta~ppa1~SYSTEM **
-
- * [#1933] Cleanup debug
- * [#1933] Clean up debug
- * Fix mic
- * [#1933] Set the IAx format earlier
- * [#1933] Move IAX sendAudioFromMic outside if (call) statement
- * [#1933] Fix startstream when offhold in iax and add debug concerning
- codec neg.
- * [#2371] sflphone_notify_voice_mail: minor gettext message formatting
- cleanup
- * [#2371] select_account_cb: properly gettextize status message
- * [#2371] show_account_list_config_dialog: properly gettextize status
- message
- * INSTALL: Minor tidyup of core install guide
- * Add /sflphone-client-gnome/src/icons/Makefile to .gitignore
- * [#2181] Updated OpenSUSE files (tmp)
- * [#1933] Add debug for codec negociation for iax
- * [#1933] Get rid of getMicAvail and getMicData in audiolayer (not
- used anymore)
- * [#1933] Add "audio codec not determined" error in IAX
- * [#1933] Test flush data
- * [#1933] Do not need to start audio stream in iax anymore
- * [#1933] Protecting pointer
- * [#2284] Remove more compilation/execution warnings
- * [#2284] Cleanup debug in client, use DEBUG instead of g_print
- * [#2284] Clean up uimanager
- * [#2370] Remove warnings
- * [#2366] Clean up other debug
- * [#2366] Clean up debug
- * [#2366] Call pa_xfree explicitely in writeToSpeaker
- * [#2284] Remove address book warnings
- * [#2365] Fixes bad cast
- * [#2352] Fix continuous ringing when peer hangup and call not yet
- answered
- * [#2181] Added version support
- * [#2181] Fixed some minor issues
- * [#2360] Moved MainBuffer from AudioLayer to ManagerImpl
- * [#2352] Makes getMainBuffer() everywhere
- * [#2352] Use 50 sec latency on pulseaudio stream creation
- * [#2352] Add alsa debug
- * [#2359] Update repository documentation
- * [#2354] Move pulseaudio disconnectAudioStream after stopping main
- loop
- * [#2352] Adjust nb byte copied in pulseaudio according to
- writeableSize
- * [#2352] Specify pulseaudio tlength parameters using pa_usec_to_bytes
- * [#2322] Convert italian translation to UTF-8
- * [#2357] Fixes window size
- * [#2357] Display only actionnable tool item
- * [#2333] Update streams parameters
- * [#2347] Use GNOME user settings for Menu and Toolbar appareance
- * [#2349] Load/Save properly audio params
- * [#2322] Update translations from Launchpad
- * [#2181] Added Francois Marier script
- * [#2350] Remove non-valid test
- * [#2181] Updated launchpad packaging
- * [#2333] Fix Pulseaudio Capture
- * [#2333] Use pulseaudio ADJUST_LATENCY flag and ALSA RT-SCHEDULING
- * [#2333] Pulseaudio Interpolate timing
- * [#2333] Change (again) Pulseaudio settings to fit logiteck usb hdw
- requirement
- * [#2333] Adjust pulseaudio fragment size to 4096 (max sflphone's
- frames per buffer)
- * [#2284] Remove recurrent compilation warning (g++ linker problem)
- * [#2333] Safer Audiostream parameters
- * [#2333] Fix alsa playback to reduce underrun
- * [#2333] Better audiostream parameters
- * [#2181] Updated version management
- * [#2333] Exclusive test in playback loop
- * [#2181] Updated build system
- * [#2333] Less underrun with these value
- * [#2333] Update playback audiostream parameters
- * [#2333] Lengthen the audio buffer reduce number of underrun in
- pulseaudio
- * [#2333] Add ALSA recovery functions for underrun (begin)
- * [#2333] Add pa_stream_trigger in pulse audio underrun callabck
- * [#2048] Reduce prebuffering in pulseaudio (which affect incomming
- calls' plbck)
- * [#2316] Do not display any icons to the right on the history tab
- * [#2333] Comment pa_stream_trigger in pulseaudio underrun
- * [#2333] Modify pulseaudio streams parameters
- * [#2318] Fix transfer tool button double signal
- * [#2181] Updated
- * [#2333] Fix ALSA ringtone
- * [#2333] Flush all main buffer before starting audio
- * [#2333] Open/Close Alsa thread between calls while there is no audio
- * [#2333] Add debug message and test condition on starting playback
- and capture
- * [#2181] Fixed gnome client makefile
- * [#2181] Updated
- * [#2308] Remove getTelephoneTone debug
- * [#2308] Change plughw for default in ALSA
- * [#2308] Oups, forgot to change function name in audiolayertest.cpp
- * [#2308] Cleanup in pulseaudio code (debug, function name)
- * [#2308] Fix pulseaudio stream closing assertion failure
- * [#2308] Moved pulseaudio mainloop locking from AudioStream
- disconnect stream
- * [2308] Fix latency at the beginning of a call, when playing DTMF and
- wehn starting tone
- * [#2181] Updated karmic
- * [#2317] [#2319] Fix address book toggle button contextual behaviour
- * [#2308] Stop stream when refusing a call
- * [#2308] Stop pulseaudio stream when peer hungup
- * [#2308] Fix tone and ringtone
- * [#2312] Display the STUN entry widget when opening the tab
- * [#2308] Implement two different callbacks for capture/playback in
- pulseaudio
- * [#2309] Open/close pulseaudio connections in startStream/stopStream
- * [2308] Leave pulseaudio stream running, do not cork/uncork them
- anymore
- * [#2295] Set gtk file chooser to None if nothing is set in
- configuration
- * [#1976] Add codec and conference documentation
- * [#2209] Fix recording in regard of resamling
- * [#2297] Update .gitignore
- * [#2297] Update translation files
- * [#2297] Add reference to our coding standards
- * [#2297] Remove old docbook code
- * [#2296] Reinit tls account settings after modification
- * [#2253] Add DcBlocker class to remove capture's dc offset
- * [#2034] Fixes for TLS transport to initialize
- * [#2284] Add silent build rule + client clean warnings
- * [#2274] Fix unserialize history items in cilent at startup
- * [#2274] Complete display name parsing and displaying
- * [#2274] Parse the Display Name in sip INVITE message
- * [#2050] Fix capture volume control in ALSA
- * [#1970] Volume controls disable when using pulseaudio
- * [#1970] Disable volume controls when using pulseaudio
- * [#2277] Fix direct ip2ip ZRTP enabling/disabling in ip2ip
- preferences
- * [#2181] Added launchpad debian files
- * [#2181] Added spec files for OSC
- * [#2274] Set display name for "Contact" sip header as the hostname
- * [#2181] Fixed daemon issues
- * [#2181] Fixed gnome client issues
- * [#1976] Remove warnings - need to fix the transfer
- * [#2006] Add init is_rec variable in ManagerImpl
- * [#2006] Update codec display on call selection
- * [#2006] Restore double click actions in history and contact calltree
- (GTK)
- * [#2176] use XDG_CACHE_HOME when initializing sfl.zid file
- * [#1976] Fix calltree switching from history
- * [#2209] (Re)Fix cache for zid
- * [#2209] Clean up debug messages
- * [#2209] Clean debug messages
- * [#2209] Fix trasnfering a call during a conference
- * [#2209] Speex decode must return the number of bytes
- * [#2209] Change frameSize speex 32kHz
- * [#2209] Fix speex codec framesize
- * [#2209] Reinit converterSamplingRate in RTP sessions
- * [#2209] Change speex ultra wide band framesize
- * [#1747] Add pixmap data
- * [#2252] Fix Receiving a server error 488 crashes the callee
- * [#2209] Fix iax low rate packate sending
- * [#2209] Clean up debug messages
- * [#2209] Add resampling changes for IAX
- * [#2209] Clean up resampling code
- * [#2209] Fix latency introduced by pulseaudio
- * [#2209] Fix initialization of mainbuffer's internal sampling rate
- * [#2176] Fix upsampling buffer size in audiolayer
- * [#2209] Add dynamic converter sampling rate in audiortp sessions
- * [#1747] Fixes runtime warnings
- * [#1747] Remove from repo
- * [#1747] register our icons to be used as stock icons
- * [#2209] Fix number of byte in alsa's write to speaker
- * [#2209] Fix putting non-resampled data in RTP's mainbuffer
- * [#2209] Add alsa resampler
- * [#2209] Add a samplerate converter in PulseLayer
- * [#2209] Add mainbuffer's internal sampling rate and flushall method
- * [#2176] Add mainbuffer stateInfo debug method
- * [#2209] Resampling is optimal using SRC_LINEAR not SRC_FASTEST
- * [#2176] Remove debug recordings
- * [#2176] Fix Holding a conference participant on new calls
- * [#2224] Add confID in callable object
- * [#2176] Fix putting onhold a call participating to a conference when
- pressing new call
- * [#2176] Reset auidio buffers when adding streams (rtp, audiolayer)
- * [#1976] Use xml to describe toolbars - Add a naviguation toolbar
- * [#2176] Remove conference default_id in joinParticipant
- * [#2176] Display error message in alsa's snd_pcm_avail_update call
- * [#2176] Alsa mic avail data debug
- * [#2176] Add some debug message for mic loss problem
- * [#2176] Flush mic ring buffer when offholding a call
- * [#2176] Reset ringbuffers' readpointer when adding main participant
- * [#2176] Fix getAvailData algorithm
- * [#2176] Reset ringbuffer's readpointer when adding a new participant
- to a conference
- * [#1744] Regex object renamed to Pattern. Previous attempt at
- providing
- * [#2176] Fix detach main participant problem when adding new one
- * [#1976] Use right domain to translate
- * [#1976] Add xml menu description
- * [#2176] Store a list of confernece participant in client
- * [#2176] Fix add participant, joinparticipant methods
- * [#2181] Do not install dbus-c++ headers + add return value
- * [#2176] Fix minor call handling instabilities
- * [#2174] Fix incoming IP call contact address
- * [#2211] Add test to protect NULL pointer
- * [#1163] Add Advanced account configuration section
- * [#2176] Add some usefull comments and debugging info
- * [#2176] Add conditions to display security icons in conference
- * [#2176] Fix detaching one participant while keeping communication to
- others
- * [#2176] Reenable userActive.svg in call tree
- * [#2176] Make user active blue (not red)
- * [#2176] Fix user active picture
- * [#2176] Fix "hidden" merge conflict in sipvoiplink
- * [#2176] Remove iax audio stream on peer hungup
- * [#2174] Multiple UDP transports functional (TESTED with 2 accounts
- and 3 calls)
- * [#2176] Fix fix audio stream binding in iax
- * [#2174] Create a default UDP transport + use tp selector for dialogs
- also
- * [#2176] Register iax audio stream in mainbuffer
- * [#2176] Fix getAudioCodecName in IAXvoipLink
- * [#2176] Fix iax account init
- * [#2176] Handle multiple account using the same sip transport
- * [#2165] Add .png files
- * [#2176] Small fixes concerning dtmf
- * [#2176] Fix make uninstall in codecs
- * [#2174] remove stund makefile generation
- * [#2176] Add conference lock
- * [#2174] Add transport selector for multiple accounts
- * [#2176] Change userActive picture from red to blue
- * [#2176] Fix security pixbuff in calltree
- * [#2176] Replace sfl.zid in .cache/sflphone instead of .sflphone
- * [#2176] Fix add call description
- * [#2176] Remove detach button from toolbar
- * [#2176] Fix calltree call description state and state code in
- conferences
- * [#2176] Fix pulse audio double free
- * [#2176] Fix conference selection
- * [#2174] Clean up - remove stun settings in client network
- configuration panel
- * [#2174] Remove voviva stun code
- * [#2174] Rsolve STUN with pjsip - DO NOT WORK
- * [#2165] Add user svg
- * [#2165] Debugging sip call failed
- * [#929] Link against uuid if installed
- * Oops
- * Fixed bugs related to libsexy (with GTK < 2.16)
- * [#929] Remove uuid-dev dependency in the core
- * [#2165] Debugging no negociated codecs at communicatio start
- * [#2165] Fix calltree bug (gtktreestore instead of gtkliststore)
- * [#2165] Fix several merge problems
- * Updated opensuse packaging script
- * [#1163] Add missing figures
- * [#1163] Update INSTALL file
- * [#2165] Fix IAX
- * [#2165] Add recordabe interface
- * [#2165] Finish recording refactoring for call (not for conference)
- * [#2165] Enable speaker recording for two different calls
- simultanously
- * [#2165] Implement call recording using the Recordable interface
- * [#2165] Add get and set to AudioLayer's audio recorder
- * [#2165] Add class recordable from which inherit call and conference
- * [#2006] Fix G722 and Speex 8khz codec conferencing
- * [#2006] add recording of audio buffers
- * [#1163] Add general settings section
- * [#1163] Fixes makefile error
- * [#2006] Fix some minor issues
- * [#2006] Drag a conference call on another conference call
- (difference conferences)
- * [#2006] Fix dragging a conference on itself
- * [#1744] Integrating some of the needed regular expression patterns
- in order
- * COmplete call features
- * [#1744] Added support for named subgroup in the Regex object. Also,
- new
- * [#1744] Adds thread safety features, compile() and setPattern()
- methods to the Regex class.
- * [#1744] Fix inconsistency in the finditer method from the last
- commit.
- * [#1744] Added regex pattern object built on top of libpcre. To be
- used
- * [#1744] Initial commit towards implementing RFC4568. Unimplemented
- in the
- * [#2157] Hide "security" and "advanced" tabs for IAX under account
- * [#1163] Add call features section
- * [#2006] Add joinConference capabilities
- * [#2006] Add dbus joinConference signal
- * [#2006] Drag a conference call onto a conference to add it
- * [#1163] Add addressbook section
- * [#2006] Drag a conference call onto a single call to create a
- conference
- * [#2006] Expand rows automatically
- * [#2006] Add minimal multiple conference handling
- * [#2006] Add atached/detached conference icons
- * [#2006] Add function processRemainingParticipant
- * [#2006] Deep refactoring, fix hangup bug
- * [#1163] Update documentation - Accounts part
- * [#1976] Integrate user doc to gnome client build system
- * [#2122] Remove double inclusion in dbus-c++/src/Makefile.am
- * Remove pjproject version number
- * [#2006] Fix peerHungup
- * [#1976] Make Yelp accessible from the GNOME client (need to install
- the sflphone.xml first)
- * [#2006] Fix multiconferencing hangup
- * [#2006] Fix hangup calls in a conference
- * [#2150] Make IAx2 reappear
- * [#2006] Fix detach participant on multiple call
- * [#2006] Can remove rining call from a conference
- * [#2006] Reinit confID when removing a participant
- * [#2006] Remove get isCurrentCAll in hangup/peerhungup (SipVoipLink)
- * [#2006] Fix refuse call
- * [#2006] Fix answerring incoming call
- * [#2006] Refactor conference's participant list
- * [#2101] Re-integrate test compilation in main build system
- * [#2101] Make the test directory compile
- * [#2136] Restore history functionality
- * [#2006] Fix binding main participant to himself
- * [#2006] Fix add current/incoming/onHold participant to an existing
- conference
- * [#2006] Fix add incoming calls to an already created conference
- * [#2006] Fix remove stream
- * [#2006] Fix detachParticipant/removeParticipant switchCall ids
- * [#2006] Fix adding a call in conference having state "CURRENT"
- * [#2006] Remove/add main participant from conferences
- * [#2006] Hold/unHold conference
- * [#2006] Detach a partcipant from drag n drop
- * [#2006] Hangup a conference
- * [#2006] Add hold/unhold conference dbus messages
- * [#2034] gtk-ui fix under the "basic" tab.
- * [#2006] Fix dragging calls on conference calls
- * [#2006] Fix detach participant from a conference
- * [#2034] Added default message is status bar under the account config
- dialog
- * [#2112] Fix a crashed caused when a non-md5 password was sent to
- pjsip.
- * [#2006] Detach participant by ID
- * [#2006] Fix addParticipant method in managerImpl to handle
- incoming/answered calls
- * [#2006] Add addParticipant method in managerimpl and related dbus
- messages
- * [#2111] Added the ability to configure zrtp on sip.sflphone.org from
- * [#2106] Fixed problem in the account assistant under gtk-ui. Also,
- assistant.c
- * [#2006] Fix dragging a conference call on another conference call
- (same conference)
- * [#1904] Small UI fix. Assistant was moved from "Call" to "Edit"
- menu.
- * [#1904] Fix a wrong label under gtk-ui.
- * [#2034] Renaming and source code splitting.
- * [#2034] Status bar added to account window to better reflect the
- registration
- * [#2006] Make calltree_remove_call recursive (for GtkTreeStore)
- * [#1110] Small gtk-UI fix in the account window (alignment).
- * [#2006] Fix remove conference, display children which are still
- active
- * [#2006] Recursive function call in calltree_update_call
- * [#2006] Add multilayered capabilities to calltree (GtkTreeStore)
- * [#2006] Implement remove conference in calltree
- * [#2034] Now useless as Direct Ip calls settings moved under
- Preferences.
- * [#2034] Edit/add buttons were set insensitive all the time under
- gtk-ui.
- * [#1887] Information about the state of the current SIP call is
- displayed
- * [#2006] Add call tree remove callback
- * [#2006] Fix create_conference function
- * [#2006] Update conference_added_cb to add new conference to the list
- * [#812] Added new tab under GTK-ui Preferences. Moving Direct Ip
- Calls from
- * [#2121] Disable temporarily test compilation
- * [#2006] Fix conferencelist to handle conference_obj_t instead of
- gchar
- * [#2006] Add conference_obj structure
- * [#2121] Update version
- * [#2006] Fix conference selection
- * [#2101] Use the new source tree to fetch the right object files
- * [#2006] Add conference in calltree
- * [#2006] Add Dbus signal conference added/removed/changed
- * [#2006] Add getConferenceDetails call on dbus
- * [#1904] Registration expire now appears as a spin box under gtk-ui.
- * [#812] Fixing a segmentation fault caused by a non-existing account
- ID
- * [#2006] Add getConfList method over dbus
- * [#2006] Add a conferencelist data structure in client-gnome
- * [#812] Defaults value are now sent if a non-existing account is
- requested
- * [#2006] Add sflphone action sflphone_join_participant
- * [#2006] Fix buffer read pointer problem deletion
- * [pjsip] Attempt at fixing via header incompatibility with
- Freeswitch.
- * [#1797] forget something
- * [#2006] Add call new state conferencing in deamon
- * [#2006] Remove addParticipant method for conference, use
- joinParticipant only
- * [#1163] Update INSTALL documentation
- * [#812] Msec/sec values were not taken into account.
- * [#1797] Make pjproject-1.4 compile
- * [#2006] Add Detach participant method
- * [#2006] Dragndrop fully functional with INCOMING and HOLD call
- * [#1797] Add pjproject-1.4
- * [#1797] Remove pjproject-1.0.3
- * [#2006] Get call state in conference related function
- * [#2006] Add joinParticipant (conference) method in ManagerImpl
- * [#2006] Add joinConference DBUS message
- * [#2006] Store the previously selected call_id on dragndrop
- * [#2006] Fix GValue pointer unref in selection callback
- * [#2006] Store dragged call_id
- * [#2006] Update drag_data_received_cb callback to manipulate CallIDs
- * [#2006] Add dragndrop signals
- * [#2006] Set calltree reordable
- * [#812] Adds the ability to create a TLS listener in case the user
- requests
- * [#812] Adds the ability to configure local/published address from
- * [#1883] Move switchCall in onHoldCall function
- * [#812] Deals with the published address/port problem when
- integrating TLS.
- * [#1883] Switch call id in managerimpl when peerHungUp
- * [#1883] Switch call id before hangup
- * [#1883] Add usefull and permanent debug info for conference
- cretion/deletion
- * [#812] Fix various segmentation faults related to Direct IP kind of
- calls.
- * [#1883] Fix deletion of std::map elements using iterators
- * [#2014] Add libzrtpcpp build dependency
- * [#1883] Still some for loop test ambiguity (while loop instead)
- * [#1883] Fix for loop initial test ambiguity (use while loop instead)
- * [#1883] We must discard data in urgent ring buffer if data is get in
- mainbuf
- * [#1883] Fix availForGet same id for ringbuffer and readpointer
- * [#812] Match "sips" as a Direct IP Call when the user enter a sip
- uri
- * [#812] Fix segmentation fault related to SIP URI creation.
- * [#812] Towards integrating multiple tls listeners at the same time.
- This
- * [#1883] Add debug messages in conference and fix mainbufferTest
- * [#812] gkt-ui fix. Private key must be fed as a filename and not as-
- is.
- * [#812] TLS integration within sipvoiplink and pjsip. Also,
- configure.ac
- * [#1883] Fix Alsa/Pulse mallocation
- * [#1883] Fix data corruption in AudioRtp's micData buffer
- * [#812] Full dbus integration for all the tls related options under
- gtk-ui.
- * [#1883] Fix memory leaks in audiortp session
- * [#1883] Fix mem leaks in audio rtp
- * [#812] Fix setAccountDetails where TLS_ENABLE was set to the value
- * [#812] Small gtk-ui fix.
- * [#811][#812] Small gtk-ui fix.
- * [#812] Introduced a mechanism for configuration files that makes
- possible
- * [#812] New dbus bindings added. Also, configuration compliance was
- enforced
- * [#1881] Remove default buffer from MainBuffer (update unit-tests)
- * [#1881] Add ring buffer read pointer tests
- * [#1883] Fix issues in ringbuffer reader pointers
- * [#2034] Implementing a new configuration dialogue for TLS transport
- settings
- * [#1883] Add some usefull debug and safety checks
- * [#2028] Notify the client with libnotify when the zrtp negotiation
- failed.
- * [#811] Harmless no to throw an exception, an makes the application
- less
- * [#2028] A minidialog is showed to the user under sflphone-client-
- gnome
- * Removed useless file.
- * Ignoring Makefile in src/widget
- * [#2027] Fix segmentation fault when showMessage callback is called
- after
- * [#2026] keyExchange was set to ZRTP instead of "1"
- * [#2024] Fix the wrong summary at the end of the assistant.
- * [#1883] Fix mnagerimpl conference map insertion
- * [#1883] Add Mutexes in MainBuffer
- * [#811] Gtk ui was not presenting the right information about zrtp
- for
- * [#2023] security icons were not installed in sflphone-client-gnome.
- * [#2021] Fix a mistake in the readme from sflphone-common that gives
- wrong
- * [#811] The current SRTP mode was not properly displayed for the
- IP2IP
- * [#1743] Re-implementation of the "automatically remove error dialogs
- [...]"
- * [#2017] [#2019] Fix the inability to dial a number and place a
- registered
- * [#811] Final re-integration of ZRTP support in the main branch from
- 0.9.6
- * [#1883] Fix map insertion methods
- * [#811] Combo box now is now set to the active key exchange method
- * [#811] ZRTP options now configurable back again from the Gtk UI.
- IP2IP
- * Updated hostname for git clone
- * [#1883] Add minimal functionalities to create a conference
- * [#811] re-integration of all the methods and signals on dbus.
- ManagerImpl
- * [#811] Got out of a precarious position were nothing would compile.
- * [#1976] Build documentation squeleton with docbook
- * [#1883] Add sflphone-client "addParticipant" button for conference
- * [#1994] Better organize the source directory structure. New
- subdirectories
- * [#1883] Add a simple Conference class
- * [#1882] Use static audio buffer in Pulse and ALSA layer (instead of
- malloc)
- * [#811] First commit toward re-integration and refactoring of ZRTP
- * [#1882] Flush RTP ring buffer before entering mainloop
- * [#1882] Fixed MainBuffer::UnBinCallID() in case there is no
- ringbuffer
- * [#1882] Test (and fixe) high level conference and mixing
- functionalities
- * [#1772] Apply patch to compile on fedora (sent by Marcin
- Zajączkowski <mszpak@wp.pl>)
- * [#1882] Update Bind, unBind call_id in MainBuffer
- * [#1959] This adds the ability to store password as an MD5 Hash in
- the
- * [#1538] Fixes rules compilation
- * [#1930][#1931] Fixed a mistake (again) related to index and
- credential count
- * [#1753] Remove ILBC from pjproject - Hacks in pjsip
- * [#1930][#1931] Credential was not selected properly using realm
- * [#1882] Finilize multiple reading pointer in RingBuffer
- * [#1538] Remove configure from autogen.sh to respect debian upstream
- authors policy
- * [#1773] Remove generated files from repo
- * [#1791] Use XDG_CACHE_HOME to save pid file
- * [#1791] Fixes path to save history
- * [#1791] Fix debian installation scripts
- * [#1930][#1931] Settings are now taken into account in the server.
- * [#1882] Add ringbuffer default ring buffer pointer in methods
- involving mStart
- * [#1882] Add default ringbuffer pointer
- * [#1882] Add RingBuffer multiple read pointer basic functionnalities
- * [#1882] Fix MainBuffer flushData unit test
- * [#1930][#1931] Ability to save and retreive the configuration from
- * [#1882] Added Multiple CallID mapping to MainBuffer
- * [#1791] Not much
- * [#1791] If XDG env variables are not null but empty, use default
- ones
- * [#1791] Make XDG_CONFIG_HOME writable
- * [#1930][#1931] Partial commit. Not working yet. Cannot delete
- account
- * [#1881] Fixed alsa capture latency problem
- * [#1881] Fixed Alsa capture temporarily
- * [#1930] [#1931] Partial unbroken commit providing the ability to
- * [#1881] MainBuffer implemented in AudioLayer/AudioRTP
- * [#1881] Add discard and flush unit-tests
- * [#1881] Add discard and flush functionnalites to MainRingBuffer
- * [#1881] Add availForGet in MainBuffer
- * [#1881] Add availForPut function to MainBuffer
- * [#1880] Remove AudioRTP* pointer from SipVoIP (reapered while
- merging master)
- * [#1881] Add a map between call id and coresponding ring buffer
- * [#1855] Refresh pot file and upload on Launchpad
- * [#1881] MainBuffe now robust to false ids on getData and putData
- * [#1881] Fix big big big memory leak
- * [#1881] Add getData and putData to mainBuffer
- * [#1881] Unit-test basic ring buffer functionnaities
- * [#1881] Add class MainBuffer and basic buffer creation unit-tests
- * [#1880] Fix call transfer (step2) issues
- * [#1880] Moved AudioRtp* pointer from SIPVoIPLink to SIPCall class
- * [#1791] Add postinst script to keep user data when migrating
- config/history file
- * [#1797] Make pjsip compile
- * [#1777] Code indentation
- * [#1791] Use XDG_DATA_HOME and XDG_CONFIG_HOME for sflphonedrc and
- history + unit tests
- * [#1746] Useless space does not appear anymore when volume sliders
- and
- * [#1643] GtkCheckMenuItem is used instead of icons for elements in
- the
- * [#1110] [#1668] STUN parameters are now located in the preferences,
- under
-
- -- Julien Bonjean <julien.bonjean@savoirfairelinux.com> Fri, 06 Nov 2009 11:23:15 -0500
-
-sflphone-common (0.9.6-SYSTEM) SYSTEM; urgency=low
-
- ** 0.9.6 **
-
- * Documentation on echo test
- * [redmine_down] codec names not displayed in total
- * [redmine_down] crash when hanging up a dialing call because tries to
- add it to history whereas no starttime
- * [#1927] alternate every time screen changed to call history
- * [#1886] clean code
- * [#1886] debug messages when loading history removed
- * [redmine_down] sflphone-kde icons
- * [#1855] Update language files
- * [#1502] Update version number
- * [redmine_down] setHistory at close
- * [#redmine_down] Handle PJ_DECLINE_SC as failure
- * [#1923] Fix segmentation fault when adding a new account
- * [#1923] Check on iterator before setting the config
- * [#1904] Added mnemonic to tabs in sflphone-client-gnome.
- * [#1905] The daemon was not sending the currentSelectedCodec signal
- on dbus when answering a call.
- * [#1922] Default values set to all account details
- * [#1886] Spinbox reg expire enables apply, and address book is not
- visible when disabled
- * [#1905] Bug fix for segmentation fault caused by an empty string,
- * [#1910] Warnings in test directory
- * [#1919] Error fixed
- * [#1855] Update russian translation - Hussein Abdallah
- * [#1910] Remove files
- * [#1919] fixed
- * [#1777] Code indentation
- * [#1918] fixed
- * [#1917] fixed
- * [#1910] Remove warnings compilation in src
- * [#1886] removed AccountListModel in configskeleton
- * [#1914]
- * [#1911] check previous and new port
- * [#1910] Remove compilation warnings in src/dbus and src/history
- * [#1910] Remove compilation warnings in src/audio
- * [1855] Update german translation - Sven Werlen
- * [#1909] removed
- * [#1906] Done
- * [#1904] The registration expire value is now configurable from the
- * Cleaned up debug messages.
- * [#1886] separated initCallItem in two functions
- * [#1886] reversed error in commit
- * [#1886] clean debug
- * [#1886] changed Name of classes and files
- * [#1886] clean
- * [#1870] In call_state_cb (dbus.c:126), _time_stop was overridden by
- the actual time.
- * [#1884] Added some new gpg flags to prevent tty warnings
- * [#1886] Clean audio config dialog
- * [#1886] No more compile warnings. + 1 comm
- * [#1872] Check if the user input is smaller than PJ_MAX_HOSTNAME.
- * [#1886]
- * [#1785] Fixed build when no new commit
- * [#1852] If chosen by the user, the hostname can now be solved and
- used
- * [#1871] * and # inverted back
- * [#1869] Conditional compilation that checks if
- * [#1309] removed test in main
- * [#1425] Put actions in SFLPhone window class instead of ui view,
- made a separate toolbar for screens.
-
- -- SFLphone Automatic Build System <team@sflphone.org> Mon, 27 Jul 2009 09:53:00 -0400
-
-sflphone-common (0.9.6~rc2-SYSTEM) SYSTEM; urgency=low
-
- ** 0.9.6~rc2 **
-
- * [#1755] Remove generated file
- * [#1753] restore ilbc ...
- * [#1866] Methods getSipPort and setSipPort now have an effect on the
- * [#1753] make pjsip compile without ilbc. Use ./autogen.sh --disable-
- ilbc-codec
- * [#1855] Fix error in russian translation
- * [#1805] Remove the old flawed signal mechanism which was failing in
- * [#1855] Refresh translation
- * Spanish translation finished + po README files updated + echo's in
- copy-in-clients
- * [#1850] Yun made the chinese HK-CN translation
- * [#1848] Fix transfer interface bug
- * [#1862] At install, kde client installs only french translation file
- * [#1841] A new fallback mechanism was added to the internal resolver
- in PJSIP.
- * Started AccountList model/view
- * [#1855] Remove po subdir in Makefile.am
- * [#1855] Fix typo error in sflphone-client-gnome
- * [#1855] Do not generate Makefile in sflphone-common/po
- * [#1855] Copy translation files into both clients dirs
- * [#1855] Remove po dir from sflphone-common
- * Comments added
- * [#1860] mailbox->voicemail...
- * make scripts executable
- * [#1855] French translation
- * [#1855] Chinese zh_HK partially filled...
- * [#1859] An unnamed pipe monitored by poll() was added. When we want
- to
- * [#1855] Sven completed the first part of the german translation
- * [#1855] Cantonese manually filled for already translated, almost
- equal strings
- * [#1855] Merge russian translation
- * [#1855] Spanish manually filled for already translated, almost equal
- strings
- * [#1855] Update german translation in ./lang/de
- * [#1858] This problem was fixed by removing a useless line in
- * [#1855] merged existing translations in lang/ sflphone.po's
- * [#1842] [#1843] An attempt at improving the expected behaviour that
- can't
- * [#1855] added po folder in gnome client and scripts for copying from
- common lang folder to clients
- * [#1853] Edit before call does nothing on call history
- * Put most language entries possible in common. From 300 to 250
- entries. Stays underscores problem. Scripts for copy in clients.
- * commit to merge master
- * [#1825] Changed "Bad authentification" to "Authentication Failed".
- * common po files
- * [#1753] Remove ILBC from pjproject
-
- -- SFLphone Automatic Build System <team@sflphone.org> Fri, 17 Jul 2009 19:12:44 -0400
-
-sflphone-common (0.9.6~rc1-SYSTEM) SYSTEM; urgency=low
-
- ** 0.9.6~rc1 **
-
- * Update some version number
- * [#1792] Creates .sflphone directory with permission 600. Also,
- "chmod 600" after
- * [#1810] GUI is now notified that the call failed. Also, a segfault
- was
- * [#1816] Address book search disabled when disabled address book and
- enabled it back plus button stays triggered
- * codeclistmodel + asynchronous loading of address book +
- enable/disable address book
- * [#1810] Now checking SDP answer after 200 OK. Still need to
- implement full
- * [#1794] Can't use the interface during a call
- * Updated translation files
- * Russian translation integrated
- * Codec list model/view started.
- * [#1807] Add configure.ac in pjproject-1.0.3
- * [#1787] closeRtpSession added in some places where it should have
- been
- * Use Item class for contacts and accounts
- * Comments + clean code
- * [#1794] Improved debug messages
- * [#1805] Replaced the old and unreliable mecanism that was was
- waiting for
- * [#1794] Can't use the interface during a call
- * [#1787] For those cases where no registered SIP account is
- configured
- * [#1797] Make pjsip compile
- * [#1787] Minor changes. Removed useless commented line. Changed order
- of
- * [#1777] Code indentation
- * [#1797] Update package generation with new pjsip version
- * [#1798] Does not hang up when the call is building up
- * [#1797] Update .gitignore with new pjsip version
- * [#1797] Remove generated files from repo
- * [#1797] Main build system now uses pjproject-1.0.3
- * [#1797] Add pjproject-1.0.3
- * [#1797] Remove pjproject-1.0.2
- * [#1796] Computing time optimization (samplerate conversion)
- * [#1787] _audiortp->start() moved away from offhold(),
- SIPCallAnswered()
- * [#1312] Added new states for calls initialized by other clients
- * [#1795] Crashes when adding a new account, checking it and applying
- * [#1782] Missing icons
- * [#1793] KDE client compilation problem
- * Fake ringtone files can no longer be set.
- * indentation
- * [#1312] Able to fetch to differentiate incoming/ringing call state
- * [#1784] Use DESTDIR variable in po Makefile - fix language file
- installation
- * [#1785] Fixed typo
- * [#1785] Fixed changelog update
- * [#1759] ./autogen.sh --prefix=/usr --with-debug to use optimization
- level 0
- * [#1773] Changed snapshot naming convention
- * [#1773] Removed gpg agent use, added repository cache cleaning
- * [#1759] Use optimization level 0 for repository, 2 for packages
- * [#1777] Code indentation/formatting
- * Translated new features in french
- * [#1785] Added missing changelog entry
- * [#1781] Window title is SFLPhone
- * [#1777] Add code indentation/formatting in the buil system
- * [#1774] Can't set voicemail number in KDE account creation wizard
- * [#1775] Can't modify account information for account created with
- the wizard
- * [#1771] Add a "Default" button in context menu to disable chosen
- prior account
- * [#1705]
- * [#1224] Remove generated file from the repo
- * [#1224] Remove generated file from the repo
- * [#1762] distclean target should remove kconfig generated files
- (settings.h, settings.cpp). Rename them?
- * [#1761] clear history button should really clear history
- * Dialpad works.
- * Implemented Dialpad widget instead of building it in main view.
- * Removed last occurence of the old config dialog, that made the build
- crash.
- * [#1755] Do not consider G722 as a dynamic payload elsewhere than in
- RTP layer
- * [#1753] Remove ilbc Makefile generation
- * [#1756] Implement a kde configuration dialog with kconfig xt and
- kconfigdialog class
- * [#1755] fix audiocodec folder parsing problem
- * [#1450] Reinit timestamp comparison in RTP, create session in
- newOutgoingCall
- * [#1753] Remove milenage third party code from pjsip
- * New Config Dialog integrated in GUI.(without codecs)
- * [#1753] Remove ILBC codec
- * kconfig started, tr2i18n -> i18n, icons folder, accountList changed
- * [#1705] Fixed Audio RTP thread creation/start
- * [#1714] Fix codec negociation result handling
- * [#1678] Fix audiortp payload setting
- * [#1678] Put bac putData method in rtp
- * [#1669] gtk_file_chooser_get_filename() support UTF-8 by default
- * [#1735] Add conditions to sdp update call if call declined
- * [#1737] substr of recordings destination folder to remove "file://"
- should be done in client rather than in daemon
- * [#1731] Enlarge audio stream buffer size
- * [#1714] Missing true
- * [#1317] Fixed Mandriva timeout
- * [#1317] Changed tag convention
- * [#1317] Cleaned git-dch
-
- -- SFLphone Automatic Build System <team@sflphone.org> Fri, 10 Jul 2009 15:49:56 -0400
-
-sflphone-common (0.9.6~beta-SYSTEM) SYSTEM; urgency=low
-
- ** 0.9.6~beta **
-
- * spec files for mandriva and opensuse updated with buildrequires
- libqt4-dev >=4.3
- * [#1700] Cannot build on ubuntu 8.10 and a few other distribs
- * [#1502] Update version number where applicable
- * [#1642] Update client icons
- * [#1450] Clean up useless debug and comments in sipvoiplink and
- audiortp
- * [#1450] Remove Semaphore object in AudioRtp thread deletion
- * [#1450] Audio RTP init now synchronized with Sip/SDP
- * [#1693] kde client crashes when changing codecs order/activation
- * [#1450] Deep refactoring of audiortp
- * [#1450] setRtpSessionRemoteIp
- * [#1689] getCallList at start
- * [#1224] Change path in package files
- * [#1450] Audio RTP initialized only once, payload and remote ip set
- at runtime
- * [#1450] Add setRtpSessionMedia and setRtpSessionRemoteIp address
- * [#1642] Make GNOME GUI fresher and younger ;)
- * [#1686] Status bar displaying used account
- * added sflphone-kde icon so that it compiles
- * [#1659] Ending a call causes the daemon to crash
- * corrected introspection XMLs, po files...
- * [#1211] g722 media descriptor in codecDescriptor
- * [#1310] Install sflphoned in $(prefix)/lib/sflphone
- * [#1502] Do not install test binaries and dbus utilitaries
- * [#1224] hack for pjsip build system!
- * [#1224] Remove pjsip binaries from repo
- * [#1224] Upgrade to pjsip 1.0.2
- * [#1658] About SFLphone (bugs)
- * [#1658] About SFLphone
- * [#1660] Displaying all dialed numbers in a call
- * Tested status bar.
- * [#790] Optimize pulse audio streams parameters
- * [#1678] Some usefull debug messages for mutex/semaphore deadlock
- problem
- * [#1669] Add/remove some usefull/unusefull debug
- * [#1665] Fix latency related to pulse audio stream openning/closing
- * [#1457] Make the menus and panels accessible in french
- * [#1457] Improve broken keyboard accessibility in menus and conf
- panels
- * [#961] Instanciate only once the searchbar icons
- * [#961] Restore transfer fonction
- * [#961] Filter on the history type OK
- * [#961] Fix compilation problems on hardy/intrepid
- * [#1157] Commit missing files
- * [#790] Reduce number of start/stop streams call on pulse audio
- * [#1639] kde client crashes when no account registered
- * [#1620] Fix the searchbar
- * [#1620] Get back caltree as it was during gtkcritical area
- * [#1620] Add history filter reinit function
- * [#1335] Add a missing label in address book preferences
- * [#1561] Update russian translation - Hussein Abdallah
- * [#1605] Fix edit menu french translation
- * [#961] Enable to search in the history according to the call type
- * [#1449] Searchbar does not work anymore
- * [#961] Add popup menu on the entry primary icon for history
- * [#1317] Fixed KDE client package dependency
- * [#936] speex 32 khz integration completed
- * [#936] Use 320 frame size
- * [#936] Test using a frame size at 320 smpls
- * [#1214] Enable / Disable history
- * [#1607] Fix compilation problem for ubuntu 8.10 (libsexy)
- * [#1313] Implement processDataEncode processDataDecode in audiortp
- * [#1613] codec list order can't be set
- * Better handling of localisation + added languages + corrected
- warnings + begginning of new config dialog with kconfig + 14px
- account leds
- * [#1214] Save and load history according to the limit timestamp +
- unit tests
- * [1609] Fix call number copy/paste feature
- * [1607] Restore clear action icon in searchbar
- * [#936] Try to decode using 1280 samples
- * [#936] Add some debug
- * [#936] Add .cpp file
- * [#936] Oops Forgot speex 32 khz
- * [#1214] Add configuration panel for history + D-Bus calls
- * [#1313] Test rtp thread function, frame size, nbbytes, resampling
- * [#790] Flush audio data before closing audio streams
- * [#1214] History displays local time
- * [#1214] Skip empty field on display
- * [#1214] Associate an account to an history entry
- * [#1342] Get addressbook options sensitive/non-sensitive
- * [#1211] Clean up and comments
- * [#1211] Get back to 20 ms framesize
- * [#1211] Use sendImmediate instead of putData in RTP
- * [#1211] Fix nb byte available in RTP
- * [#1211] Clear condition on maxNbSamples in RTP
- * [#1211] Fix max byte available in RTP session
- * [#1211] G722: Use 160 samples per frame instead of 320
- * [#1211] Test using a dynamic payload
- * [#1211] Test using a dynamic payload type
- * [#1211] Rename size variable (nb_samples, nb_bytes)
- * [#1211] Test g722 ip-to-ip sending twice the data lenth
- * [#1211] Test g722 ip-to-ip
- * [#1214] Do not select an history item by default at startup
- * [#1214] Remove some compilation warnings
- * [#1214] Handle empty field - remove g_print
- * [#1214] Add each history item only once
- * [#1214] Handle call timestamps properlier
- * [#1214] Do not need timestamp files anymore
- * [#1214] Use the saved date for history entry
- * Clean up
- * [#1214] Client doesn't crash if the D-Bus call fails
- * [#1214] Client is able to save its history - still some glitches
- * [#1211] Forgot 16000 for g722
- * [#1211] G722 initialization
- * [#1214] Save name/number, successfully load the history if no fields
- are empty
- * [#1499] Fixed destination directory bug
- * [#1214] Restore all the functionalities; peer name/number way more
- easy to handle !!
- * [#1214] Add callable_object instead of call_t, refactoring
- * [#1211] Test with polycom soundstation 16000
- * [#1211] Remove C like inline function in g722 codec
- * [#1342] Finalize gnome client preference window formating
- * [#1214] Retrieve the history when the gnome client startsup
- * [#1306] Implement localization for KDE client
- * [#1593] enable accounts apply button when account checked/unchecked
- * [#1214] Implement the dbus calls on server side
- * [#1214] Add serialized/unserialized functions to pass data on DBUS
- * [#1342] Formating gnome client configuration windows
- * [#1214] Save sucessfully a map of history items
- * [#1499] Removed multiple jobs compilation for KDE client (2)
- * [#1214] Load history from file into memory, add unit tests
- * [#1534] Throws a length_error exception in case URL exceeds
- std::string max_size
- * [#1499] Removed multiple jobs compilation for KDE client
- * [#1565] make account leds smaller
- * [1430] Fix dbus debug
- * [#1562] crashes when trying to change item of a call of state "OVER"
- * [#1116] Fix compilation bug
- * [#1317] Added mandriva and opensuse-11 64 bits
- * [#1108] Add messges in main window concerning transfer success
- failure
- * [#1116] Fix compilation problems
- * [#1211] g722 Makefile
- * [#1108] Client side transferFailed/trasferSucceded signals handling
- * [#1211] G722 mostly completed,
- * [#1555] make bigger toolbar (24x24)
- * [#1551] remove default mailbox number in wizard and disable mailbox
- button when first account doesn't have mailbox number
- * [#1342] Re-add sflphone manpages
- * [#1116] Fix compilation on non-jaunty distros
- * [#1317] Fixed opensuse startup sleep
- * [#1108] Add a signal in the client to notify successful or failed
- transfer
- * [#1108] Dbus signals concerning call transfer success/failure
- * [#1317] Added opensuse to automatic build system
- * [#1223] Fix manpages bug
- * [#1060] german translation glitch
- * Clean up some gnome client warnings
- * [#1547] replace ugly account leds by beautiful icons
- * [#1548] add close button that hides windowand just hide on clicking
- the cross
- * [#1549] put introspec XMLs in the client's source
- * [#1312] Implement getCallList D-BUS method
- * [#1116] Clear text in history and contacts
- * [#1499] KDE integration
- * [#1469] Modify header linkers in dbus-c++'s Makefile.am's
- * [#1469] Remove examples folder from dbus-c++
- * [#1214] History integration in build system; unit test squeleton
- * [#1317] Cleaning
- * [#1469] Remove configure stuff in dbus-c++
- * [#1469] Add unofficial mainline dbus-c++
- * [#1469] Remove dbus-c++ from freedesktop
- * [#1430] Bring account changed signal/callback back to normal
- * [#1060] Update german translation - Sven Werlen
- * [#1430] Add marshaller one string define
- * [#1430] Send account change signal broadcast using account id
- * [#1430] Remove condition on setRegistrationState, cause stun to
- crash
- * [#1317] Centralized version handling
- * [#1317] Fixed version number on sfl-git-dch
- * [#1317] Refactoring for new distributions
- * [#1215] Fix account order at startup if latency
- * [#1088] Restore sip dns srv
- * [#1214] Add squeleton for history manager
- * [#1430] Add accout id to accout changed method
- * [#1430] No connectionStatusNotification (account changed) if no
- changes
- * [#1538] Add COPYING file
- * [#1430] Add audio rtp thread tests
- * [#1317] Changed version detection
- * [#1538] Document license in libs/stund
- * [#1317] Added version files
- * [#1538] Apply François patches - debian packages
- * [#1317] Updated spec files
- * add files
- * [#1538] Apply François patches - debian packages
- * [#1535] Change program file structure (directory src...)
- * [#1317] Updated build system scripts
- * [#1317] Cleaning
- * [#1317] Copied introspect files to gnome client
- * [#1317] Added opensuse to build-system : first-shot
- * [#1317] Remove spec files from configure
- * [#1317] Added missing prefix
- * removed debug for daemon account fix
- * [#1430] Add a connection reference which most likely belong to
- libdbus
- * [#1430] Use shared connection instead of private
- * make daemon find the account, added userMatch
- * Clean code, add comments...
- * [#1317] Fixed packaging rules
- * [#1317] Updated autogen
- * Updated autogen.sh for pjsip
- * [#1526] Set accounts order
- * [#1317] Fixed pjsip lib dirs
- * [#1317] Updated debian packaging for new pjsip configuration script
- * [#1317] Switch to autogenerated guess and sub files
- * [#1317] Updated pjsip inclusion in build system
- * [#1317] Replaced pjsip guess and sub files
- * [#1317] Fixed compilation issues on opensuse 11
- * [#1505] account list seem to crash the application when clicking
- Apply very fast...
- * [#1456] Add a flag to be replaced in the control files
- * [#1456] Added version dependancy handling
- * put account alias in AccountWidgetItem rather than in the item with
- " " before.
- * [#1034] The KDE client should start sflphoned if it is not started
- * [#1500] Handle options for notifications and display on incoming
- call.
- * [#1443] Client should not crash when receive an unexpected
- stateChanged signal
- * [#1403] Do not stop the notification anymore
- * [#1456] Added version dependancy handling
- * [#1426] Daemon crashes when get alsa plugin
- * [#1422] Improved error messages
- * commit for merge
- * [#1424] Change logo in tray icon and put a different one when
- incoming call
- * [#1425] first part done, window title...
- * [#1413] add manpages creating and installing in build system
- * [#1417] The client should start the account creation wizard if
- started for the first time (if config file doesn't exist)
- * [#1421] Make volume bars horizontal when dialpad is hidden.
- * Changed main window title and fixed a mistake in sflphone_const.h
- * [#1412] make debian package building work
- * changelog changed.
- * Changed addAccount method in gnome client.
- * Debian and man folders added.
- * [#1388] Change project name from sflphone_kde to sflphone-client-kde
- * Better handle of kabc check.
- * [#1351] Automatic generation of dbus interfaces in makefile
- generated by cmake
- * [#1307] Implement "edit before call" in history and address book.
- * [#1344] change action_call label in call history from "call" to
- "call back".
- * [#1308] Implement Hook feature in kde client
- * Improved build system.
- * #1219 : Add address book configuration page
- * Better handling of registration to the daemon.
- * #1039 : Add tray icon in kde.
- * Issue no 1216 : Double click on item in history or address book
- causes call.
- * display peer name in call list and call history when called from
- address book.
- * Address book functionnal with photo displayed.
- * Help menu kde available but actions disappeared. All fonctions in
- view.
- * Address book functionnal but ugly and making its own sort in the
- complete address book.
- * Account choice on right click, clean out includes, page address
- book, fixed bugs...
- * Wizard, double click, context menu...
- * Removed sflphone_kde.kdevelop.filelist
- * Added account creation wizard and translated interface in english.
- * Transfer functionnal but ugly.
- * transfer not functionnal
- * Bug fixed : unholding (UNHOLD_CURRENT, UNHOLD_RECORD)
- * Commit functional for push. With install.sh
- * Before merge.
- * Problem with enable accounts. Account display increased.
- * Functional with codec order working , playDTMF.
- * Commit functional.
- * sflphone_kde/build added in .gitignore.
- * complete commit for checkout previous.
- * Commit before checkout previous version to check the display
- bug(little font everywhere...)
- * Functionnal client. Rest : history icons, config icons and
- functionalities
- * commit before merge asavard for isRecording.
- * Call and Automate fusion done and seems to work.
- * Commiting before putting Automate class in Call class.
- * Functionnal main window without recording, history, voicemail, kio
- widgets.
- * client kde avec kdevelop.
- * Config Dialog almost finished.
- * Base of QT client
-
- -- SFLphone Automatic Build System <team@sflphone.org> Tue, 23 Jun 2009 11:12:06 -0400
-
-sflphone-common (0.9.5-SYSTEM) SYSTEM; urgency=low
-
- ** 0.9.5 release **
-
- * [#1060] FIx bug in chinese translation
- * [#1313] git add rtpTest.cpp rtpTest.h
- * [#1313] Add init/close rtp tests
- * [#1313] Basic instanciation of the rtp layer
- * [#1449] Gtk-Critical concerning history filters and new calls
- * [#1400] Make the match with the hostname instead of username
- * [#1324] Change status bar label for "Using %s (%s)"
- * [#1403] Icon size: 60x60 px
- * [#1403] Do not remove notification, improve icon quality
- * [#1403] Add smaller icon for gnome notifications
- * [#1403] Prevent crash when hangup && no notification
- * [#1403] Remove all actions on notifications; code refactoring
- * [#1451] Use stun.sflphone.org as default STUN server
- * [#1060] New po files - need to be translated
- * [#1060] Update french translation - Rebuild template file
- * [#1456] Add a flag to be replaced in the control files
- * [#1454] Make cppunit optional; remove from build deps in control
- files
- * [#1401] Add libexpat1-dev dependency in control files
- * [#1448] Take off these ugly debug messages
- * [#1448] fixed getTelephoneTone and getTelephoneFile() called
- repeatedly
- * [#1406] add liblog4c-dev in build-depends
- * [#1409] Restore .desktop icon
-
- -- SFLphone Automatic Build System <team@sflphone.org> Mon, 25 May 2009 11:34:40 -0400
-
-sflphone-common (0.9.5-SYSTEM~rc2) SYSTEM; urgency=low
-
- ** 0.9.5 rc2 **
-
- * [#1422] Improved error message
- * [#1402] Fix pjsip build
- * [#1404] Clear GTK-Critical Bug at client startup
- * [#1422] Added automatic VM shutdown when building on more than one
- VM
- * [#1422] Fixed some issues with new changelog generation script
- * [#1422] Moved distribution update to specific file
- * [#1422] Dropped git-dch, replace by home made implementation
- * [#1402] Fix pjsip build
- * [#1404] Clear GTK-Critical Bug at client startup
- * Changes for name based dbus connection
- * Clean changelogs
- * [#1343] Gnome: Implement a callback system to handle focus on
- different widgets
- * Debus Session
- * Refactoring Python code, PEP8
- * [#1430] Get back dbus_g_proxy_new_for_name
- * [#1430] Get back DBUS_BUS_SESSION type
- * [#1430] Dbus fixed owner message binding
- * Second test with DBUS owner
- * [#1404] Gnome -> Preferences -> Hooks
- * [#1404] Gnome -> Preferences -> Recordings
- * [#1404] Call History
- * [#1404] Gnome -> Preferences -> Address Book
- * [#1404] IF the first notification option disable the second
- notification
- * Dbus with fixed owner does not automatically start the deamon
- * Add codec debug tests in pysflphone
- * [#1407] Some print info
- * [#1407] Add a scenario to pick_up action
- * Test client dbus connection to a fixed owner
- * Add python dbus test suite
- * [#1161] Modified version handling in build system
- * [#1314] Test pulse audio and audio streams connect and disconnect
- * [#1402] Add info message after configure
- * [#1402] Build the daemon with the local pjsip library (vs the
- installed one)
- * [#1009] Fix Codec Sampling Rate set to zeros
- * [#1314] Add mutex to pulse layer audio streams
- * [#1314] Refactoring pulseaudio stream to test connect disconnect
- * [#1314] Refactoring of pulselayer to test conect/disconnect
- * Add debug messages in debus calls concerning account
- * [#1314] Add some return values to audio init functions
- * [#1406] add liblog4c-dev in build-depends
- * [#1409] Restore .desktop icon
- * Bug #1405: Fix strings as requested.
- * Bug #1404: Fix strings in preferences panel.
-
- -- SFLphone Automatic Build System <team@sflphone.org> Tue, 19 May 2009 12:08:03 -0400
-
-sflphone-common (0.9.5-0ubuntu1~rc1) SYSTEM; urgency=low
-
- [ SFLphone Project ]
- * [#1262] Updated changelogs for version 0.9.5-0ubuntu1 Snapshot 2009-
- 05-05
-
- [ Emmanuel Milou ]
- * Add some python CLI client code; not really functional
- * [#1108] Fix peerHungup method for IP to IP call
-
- [ Alexandre Savard ]
- * [#1108] Correct setting of SIP contact for direct IP call
- * [#1108] SIP user agent handles incoming REFER
-
- [ Emmanuel Milou ]
- * Remove website from repository
- * Update translation
-
- [ Alexandre Savard ]
- * Sflphone icon's tooltip changed for "configured" instead of
- "registered"
-
- [ Emmanuel Milou ]
- * Update translation
-
- [ Sflphone Project ]
-
- -- Sflphone Project <sflphone@mtl.savoirfairelinux.net> Tue, 05 May 2009 19:16:09 -0400
-
-sflphone-common (0.9.5-0ubuntu1~beta) SYSTEM; urgency=low
-
- [ Julien Bonjean ]
- * Updated Eclipse stuff
- * Improved addressbook config window
- * Added sflphone Eclipse stuff
- * Implemented addressbook list server side
- * Moved dbus stuff in dbus directory
- * Updated addressbook configuration
-
- [ Emmanuel Milou ]
- * Remove unuseful installation scripts. Use apt-get build-dep sflphone
- instead
- * fix bug #1090
-
- [ Alexandre Savard ]
- * defining speex 16khz
-
- [ Emmanuel Milou ]
- * Remove unuseful file from build system
- * Start dns srv resolver
-
- [ Alexandre Savard ]
- * Basic ogg/vorbis initialization
-
- [ Emmanuel Milou ]
- * Handle incoming IP-to-IP invite correctly
-
- [ Alexandre Savard ]
- * speex wideband 16000
-
- [ Emmanuel Milou ]
- * Better handling of incoming IP to IP call
- * DNS SRV resolution functional
- * Implement IAX2 incoming URL
- * Allow user to make IP call without any accounts configured
- * Add a contextual menu to edit a number from the contacts tab
- * Add comments, tooltip and new button to the contextual menu
- * add delete event, migrate to GTK 2.16 for sexy icons
- * Resolve ticket #1118
- * Update suse spec file
- * Add phone number cleanup functions, unit tests and panel
- configuration
- * Add pertinent test that fails
- * fix dependencies for suse package
- * Add contextual edit menu in history - #1120
-
- [ Alexandre Savard ]
- * Temporary comit: make speex wideband (16 khz)
- * Temporary: shared object for speex narrow band
- * Temporary: speex narrowband and wideband coexist
-
- [ Julien Bonjean ]
- * Fixed bug when no book selected
- * Fixed addressbook related compilation warnings
- * Fixed GTK client remaining compilation warnings
- * Fixed segfault when book removed since last sflphone run
- * Fixed bug when book is unreachable (ldap error)
-
- [ Alexandre Savard ]
- * Fix codec list in audio config window
- * Active/inactive speex codec by payload
-
- [ Julien Bonjean ]
- * Updated gitignore
- * Added some comments
-
- [ Emmanuel Milou ]
- * Add callto: handler script for browsers and al.
- * Integrate test compilation in the daemon build-system
-
- [ Julien Bonjean ]
- * Fixed g_object_unref warning for pixbuf
- * Cleaned too verbose output
- * Fixed toolbar update warning
- * Added support for asynchornous books open (first shot)
-
- [ Emmanuel Milou ]
- * Add a DBus call to fetch the call details from a call ID - Ticket
- #928
-
- [ Julien Bonjean ]
- * Improved async open books
- * Fixed bug #1139
-
- [ Emmanuel Milou ]
- * Add a way to save account order
- * commit missing files
-
- [ Julien Bonjean ]
- * Introduced log4c (ticket #1162)
-
- [ Emmanuel Milou ]
- * Load/save account order functionnal - ticket #813
-
- [ Alexandre Savard ]
- * Add CELT codec (#1143)
- * Make celt frame size 256 (*1143)
-
- [ Julien Bonjean ]
- * Switched everything to log4c (ticket #1162)
- * Updated eclipse settings
-
- [ Emmanuel Milou ]
- * Restore adding account - ticket #1172
- * Add liblog4c dependecy - ticket #1179
-
- [ Alexandre Savard ]
- * Double maxAvailByte for frame size in rtp (#1143)
-
- [ Emmanuel Milou ]
- * Add User-Agent SIP header - Ticket #1173
-
- [ Julien Bonjean ]
- * Fixed autoresize issue (#708)
-
- [ Emmanuel Milou ]
- * Remove libcppuint dependency for the debian packages
- * Look for libsexy only if gtk version < 2.16 - Ticket #1116
- * Remove libsexy dependency for jaunty. ticket #1116
-
- [ Julien Bonjean ]
- * Introduced unit tests (#1146)
- * Updated gitignore
- * Fixed Makefile (#1146)
-
- [ Emmanuel Milou ]
- * [TICKET #1112] Add a test on the voice buffer to send through iax
- packets
- * Remove doublon in dependencies
- * Remove warnings from the client test framework
- * Update version number to 0.9.5~beta
- * Update build-package script
- * Add check dependency in build-deps control file field
- * Create debian files for the new sflphone-client-gnome
- * [TICKET #1212] Add Replaces field in control files
- * [TICKET #1212] Fix manpages installation path
- * [TICKET #1212] Add maintainer scripts to create alternatives
- * [#1212] Update the manpages generation - edit preinst maintainer
- script
- * [#1212] Fix reference error in manpage
- * [#1212] Add missing files on the client side
- * [#1212] Fix debian docs files - no TODO file
- * [1212] Fix manpage creation problem
- * [#1220] Generate client-side glue files and marshaller at
- compilation time
- * [#1220] Generate server-side glue files at compilation time
- * [#1212] Change binary name to sflphone-client-gnome
- * [#1212] Update .gitignore to fit the new working tree
- * [#1220] Explicitly generate glue files before building the library
- * [#1220] Compile dbus directory before audio
- * [#1212] Create sflphone-common at the root of the repository
- * [#1212] Re-add pjproject
- * [#1212] Remove Makefile from repo
- * [#1220] Fix Makefile.am
- * [#1212] New working directory functional
- * [#1212] Update .gitignore
- * [#1212] Hack to make pjsip compile..
- * [#1220] Use non-installed binary for dbusxx-xml2cpp
- * [#1212] Add descriptive files, remove unuseful scripts from tools/
-
- [ Alexandre Savard ]
- * Restore speex codecs
- * add frame size for celt (#1143)
- * add framesize to codec, independant from audiolayer (#1143)
- * use codec frame size in rtp (#1143)
- * compute fixed_codec_framesize (#1143)
- * do not resample if not required (#1143)
- * add condition on resampling for decoder (#1143)
- * add a condition on bytesAvail == 0 from mic data
- * no maximum in rtp decode (#1143)
- * compute maximum for decoding (#1143)
-
- [ Emmanuel Milou ]
- * [#1146] Implement unitary tests on the client-side
-
- [ Alexandre Savard ]
- * use float instead of int to compute max nb of sample (#1143)
- * add nbSampleMax for unresampled data (#1143)
- * make thread sleep during 5 ms insead of 20 (#1143)
- * use unix usleep (#1143)
- * 50 usecond thread!!!!! (#1143)
- * try with the smallest compression (#1143)
- * use timer set at framesize (#1143)
-
- [ Emmanuel Milou ]
- * [#1161] Restore changelog version
-
- [ Alexandre Savard ]
- * Remove celt stuff
-
- [ Emmanuel Milou ]
- * [#1161] Update changelog
- * [#1220] Add Conflicts: sflphone in debian control files
- * [#1179] Add liblog4c3 runtime dependency
- * [#1212] FIx typo error in dependency list for itnrepid
- * [#1212] FIx .desktop file to point on the right exec
- * [#1212] Modify changelog replacing tag
-
- [ Sflphone Project ]
- * "[#1262] Updated changelogs for version 0.9.5-0ubuntu1~beta"
-
- [ Emmanuel Milou ]
- * [#1212] restore changelogs
-
- [ Sflphone Project ]
- * [#1262] Updated changelogs for version 0.9.5-0ubuntu1 Snapshot 2009-
- 04-27
-
- [ Emmanuel Milou ]
- * [#1212] restore changelogs
-
- [ Sflphone Project ]
- * [#1262] Updated changelogs for version 0.9.5-0ubuntu1~beta
-
- [ Emmanuel Milou ]
- * [#1212] restore changelogs
-
- [ Sflphone Project ]
-
- -- Sflphone Project <sflphone@mtl.savoirfairelinux.net> Mon, 27 Apr 2009 16:57:00 -0400
-
-sflphone-common (0.9.4-0ubuntu2) SYSTEM; urgency=low
-
- [ Alexandre Savard ]
- * Restore speex and GSM detection
-
- [ Emmanuel Milou ]
- * Fix bug #1090
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Wed, 8 Apr 2009 11:29:15 -0500
-
-sflphone (0.9.4-0ubuntu1) SYSTEM; urgency=low
-
- [ Emmanuel Milou ]
- * Integrate DBus-c++ and libiax2 in the main build system
- * Clean up in the working repository
- * Reorder hooks configuration panel
- * Protect case when no codecs are active
- * Fix some return values
- * Add unitary tests for the hook manager (premisces)
-
- [Yun Liu]
- * Update chinese translation
-
- [Sven Werlen]
- * Update german translation
-
- [Hussein Abdallah]
- * Update russian translation
-
- [Maxime Chambreuil]
- * Update spanish translation
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Fri, 3 Apr 2009 18:29:15 -0500
-
-
-sflphone (0.9.4-rc1) SYSTEM; urgency=low
-
- [ Emmanuel Milou ]
- * Fix bug while trying to hold/unhold several simultaneous call
- * Improve address book build system
- * Implement SIP url popup on incoming call
- * Improve GTK+ panel configuration
- [ Julien Bonjean ]
- * GTK+ client refactoring
- * GTK+ clean up
- * Address book improvment
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Fri, 27 Mar 2009 18:29:15 -0500
-
-sflphone (0.9.4-0beta1) SYSTEM; urgency=low
-
- [ Alexandre Savard ]
- * Display codec used during conversation on the GUI
- * Enable/disable STUN parameters at runtime
- * Refactor search bar use
- [ Emmanuel Milou ]
- * Build system fixes
- * Implement SIP re-invite
- * Implement IP to IP call
- [ Julien Bonjean ]
- * Integrate GNOME address book based on evolution data server
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Fri, 20 Mar 2009 18:29:15 -0500
-
-
-sflphone (0.9.3-0ubuntu3) SYSTEM; urgency=low
-
- [ Alexandre Savard ]
- * Both playback and record streams in PA_STREAM_CORKED (pulseaudio)
- * Use PLUGHW device for ALSA capture
- * Functional IAX and SIP recording for voicemail
- * Use the less CPU-consuming interpolator algorithm for resampling
- * Display in GTK GUI the codec used in conversation
- * GTK GUI use ASCII instread of utf-8
- * Add record menus in GTK GUI
- * Put on hold when dialing a new number
- * AccountID's are saved in the history
-
- [ Emmanuel Milou ]
- * Integrate DBUS C++, libiax2 in the git repository
- * Update website
- * Use libspeexdsp only if available on the system
- * Updated .gitignore file
-
- [Cyrille Béraud]
- * Account assistant manager improvment
- * Add an email request when creating a new account to receive voicemails
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Sat, 14 Feb 2009 13:29:15 -0500
-
-sflphone (0.9.3-0ubuntu2) SYSTEM; urgency=low
-
- [ Emmanuel Milou ]
- * Add compilation note in README
- * Use default ALSA plugin for capture
- * Fix the ALSA capture problem one more time
- * Clean up debug messages in dbus.c
- * Add libspeexdsp dependency
- * Remove implicit declaration compilation warnings
- * Fix links in the website, add release note
- * Change capture for the website front page
- * Add alsa devel dependency in build-depends control file field
- * Clean up, indentation, try to handle latency problems in iax/pulseaudio
- * Remove pjsip generated files from the repo
- * Use the previous declared curAlias function in accountwindow
- * Fix bug in history call duration when the call fails
- * Remove runtime warning in the GTK+ client
- * Add librsvg2-common dependency to load SVG under KDE
- * Refresh .gitignore
- * Update locales files + french translation
- * Add configuration panel for future noise reduction
- * Add configuration panel for audio record module
- * Daemon less verbose; accounts don't try to access STUn options anymore
- * Fix typo in configwindow
- * Add content in the official website
- * use a GTK_STOCK icon for the record button
- * Complete description text in the assistant manager
- * Add libtool flags in client configure.ac
- * Remove unuseful dependency (snd)
- * Fix SIP transfer problems
- * Remove previous version of PJSIP from the repo
- * Upgrade PJSIP to version 1.0.1
- * Add the new website source in the repository
- * Use libspeexdsp for silence detection only if available
-
- [ Loïc Faure-Lacroix ]
- * Ajout du logo gpl3
- * Ajout des images
- * Ajout de la section screenshot pour le site
- * Ajout du favicon dans le header
- * Modification des cartes
-
- [ Alexandre Savard ]
- * Clean up <speex/libspeexdsp>
- * Small cleanup
- * Save Wave fixed
- * Fix new call button when recording
- * libspeexdsp added
- * Recording: default home folder at startup
- * Minor changes to config window
- * IAX recording fixed
- * Set / get recording path, still need some GTK for client
- * AudioRecord file name format
- * Now recording in HOME folder
-
- [ Cyrille Béraud ]
- * Fix bug in reqaccount.c
-
- [ Maxime Chambreuil ]
- * Update spanish translation
-
- [Yun Liu ]
- * Update chinese translation
-
- [ Hussein Abdallah ]
- * Update russian translation
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Sat, 14 Feb 2009 13:29:15 -0500
-
-sflphone (0.9.3-0ubuntu1) SYSTEM; urgency=low
-
- * Remove debug
- * Join thread before leaving
- * Fix implicit declaration in reqaccount
- * Add REST code to build the request to server
- * Fix GValue initialization warnings
- * Update version number, fix implicit declaration, fix GTK markup
- warnings
- * Apply patch to create custom SIP account from our own server
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Fri, 06 Feb 2009 19:17:32 -0500
-
-sflphone (0.9.2-2ubuntu9) SYSTEM; urgency=low
-
- [ Alexandre Savard ]
- * Speex audio codec preprocessing initialization
- * peer hung up segmentation fault solved
- * Stop recording when transfering
- * Terminate only one call
- * Add isRecording() function
- * Fix call_icon GTK client
- * Fix SIPCallClose() function, recorded file now close properly
- * Function terminateSIPCall added in sipvoiplink and managerimpl
- * Fix thread destructor
- * setRecordingOption function implement in audiorecord
- * Record now implemented in Call class
- * Record interface complete (on hold erase previous recording)
- * Added recButton in client
- * Added: record button related icons
- * Record button added
- * Overload AudioRecord::recData to get mic and speaker data mixed
- * Recording now in audiortp::run() method
- * Audio recording working in AudioRTP: receiveSessionForSpeaker
- * Open/close a wave file when pulse audio stream start/stop
-
- [ Emmanuel Milou ]
- * Fix path for GTK+ icons; clean up
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Thu, 05 Feb 2009 18:27:53 -0500
-
-sflphone (0.9.2-2ubuntu8) SYSTEM; urgency=low
-
- [ Emmanuel Milou ]
- * Update changelogs
- * Fix bug in merge and in Makefile.am
- * Terminate only one call
- * Disable PJsip shutdown when changing STUN parameters
- * Function terminateSIPCall added in sipvoiplink and managerimpl
- * Add a timer to the alsa thread to not jam the CPU load
- * Fix bug in sipvoiplink.cpp
- * Clean shutdown of pulseaudio on quiting
- * Fix DTMF at first start with Pulseaudio
- * Remove zeroconf from the build system
- * Add a library manager + exception handling
- * Clean up in the working directory
- * Better handling of capture XRUNs
- * Restore mic adjust volume on ALSA layer
- * Protect device ALSA operation if not opened
- * Fix the switching layer bug
- * Use dynamic_cast<> to use audiolayer-specific methods
- * Open the audio devices only once at startup
- * Refactoring of the ALSA part
- * Functional plug-in manager
- * Use a C++ thread to handle tones and DTMF in ALSA
- * Restore IAXVoIPLink, restore Mutex
- * Make the plugins registering against the plugin manager
- * Migrate to 1->N relationship between voiplink and accounts
- * API plugin for registration
- * Use C++ thread in SIP, move everything in sipvoiplink
- * Complete singleton pattern for the plugin manager
- * Add -Wno-return-type compilation flag to remove warnings; Update
- version number in configure.ac
- * Add the dynamic loading for the plugin framework; integate unittest
-
- [ Yun Liu ]
- * Update rpm spec file
- * modify build package script and spec file for suse
-
- [ Alexandre Savard ]
- * Add audiorecorder plugin and testaudiorecorder
- * Add audio Recording class, edit global.h
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Wed, 04 Feb 2009 14:00:30 -0500
-
-sflphone (0.9.2-2ubuntu7) SYSTEM; urgency=low
-
- [ Emmanuel Milou ]
- * Update changelog to 0.9.2-6
- * Fix some dbus-glib implementation details on the client side
- * Init history after dbus initialization
- * Add error checking in useragent; Clean sipvoiplink
- * Prevent crash when trying to call an empty number
- * Set the volume of the playback stream to PA_VOLUME_NORM at startup
- * Fix GTK+ generic value double initialization
- * Fix jaunty control file dependency problems
- * Fix jaunty control file dependency problems
-
- [ Yun Liu ]
- * Fix bug ticket # 137
- * Tolerant to gsm library of OpenSuse 11
-
- [ Sven Werlen ]
- * Update german translation
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Fri, 23 Jan 2009 17:48:13 -0500
-
-sflphone (0.9.2-2ubuntu6) SYSTEM; urgency=low
-
- [ Emmanuel Milou ]
- * Migrate STUN configuration to the main config window
- * Update french translation
- * Other tiny memory leaks
- * Fix memory leak in sampleconverter.cpp
- * Generate packages from the release branch
- * update the build package script
- * modify the control files with architecture=any
- * Remove valgring uninitialized value
- * IAX and SIP use the same global variables to set account
- configuration ; fix broken code
-
- [ Maxime Chambreuil ]
- * Update spanish translation
-
- [ Hussein Abdallah ]
- * Update russian translation
-
- [ Yun Liu ]
- * Update translation files
- * Fix the bug when user uncheck the account which fails in the
- previous registration
- * Add stun error status
- * Fix bug ticket #143
- * Script for auto-install dependencies
- * Fix bug ticket #140
- * Fix bug ticket 141
- * Fix the reregister process when user change the details of an
- account
-
- -- Emmanuel Milou <manu@sulfur.inside.savoirfairelinux.net> Fri, 16 Jan 2009 18:19:05 -0500
-
-sflphone (0.9.2-2ubuntu5) SYSTEM; urgency=low
-
- * Fix memory leak in the pulseaudio callback
- * Update debian package generation script
- * Warnings removal in GTK+ client
- * Clean adjust volume method in alsalayer
- * Plug the sflphone playback volume control to the pulseaudio volume
- manager
- * Display the date in history according to the current locale
- * Generate the changelog according to the git commit messages
- * Complete header in chinese translation file
- * Use the right gpg key to sign the packages
- * add debian jaunty jackalope support
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Wed, 14 Jan 2009 21:17:20 -0500
-
-sflphone (0.9.2-2ubuntu4) SYSTEM; urgency=low
-
- [ Emmanuel Milou ]
- * add german translation
-
- [ Yun Liu ]
- * Fix GUI crash in Ubuntu8.10 64bit system
-
- -- Yun Liu <yun.liu@savoirfairelinux.com> Thu, 08 Jan 2009 13:08:51 -0500
-
-sflphone (0.9.2-2ubuntu3) SYSTEM; urgency=low
-
- [ Emmanuel Milou ]
- * The main thread synchronizes the ringtone thread
- * disable custom ringtone for the ALSA layer
- * Fix the Makefile.am in man directory, add a SEE ALSO section
-
- [ Yun Liu ]
- * Fix daemon crash caused by the previous patch ( for bug ticket #129)
-
- -- Yun Liu <yun.liu@savoirfairelinux.com> Tue, 06 Jan 2009 16:18:38 -0500
-
-sflphone (0.9.2-2ubuntu2) SYSTEM; urgency=low
-
- * Fix bug ticket #129
-
- -- Yun Liu <yun.liu@savoirfairelinux.com> Wed, 5 Jan 2009 15:54:53 -0500
-
-sflphone (0.9.2-2ubuntu1) SYSTEM; urgency=low
-
- * Migrate from eXosip library to pjsip
- * Add multiple SIP accounts support
- * Fix ringtones problems
- * Add a pulseaudio support
- * Improve audio quality with ALSA
- * Add chinese translation
- * Improve spanish translation
- * Migrate to a maintained C++ DBus bindings
- * Clean and improve the build system
- * Add build-dependency on Perl because we need pod2man to generate manpages
-
- -- Yun Liu <yun.liu@savoirfairelinux.com> Wed, 26 Nov 2008 09:47:53 -0500
-
-sflphone (0.9.1) unstable; urgency=low
- * Add a search tool in the history
- * Migrate some gtk_entry_new to sexy_icon_entry_new
- * Bug fix (Ticket #78): The voicemail password isn't displayed anymore in
- the history tab
- * Add the SIP registration expire value in the user file.
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Thu, 22 May 2008 11:14:25 -0500
-
-sflphone (0.9.0) unstable; urgency=low
- * Add history features
- * Call date
- * Call duration
- * Mouse events in the history tab
- * Smooth switch from the history tab to the calls tab
- * Remove most of GTK-Critical warnings
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Wed, 13 May 2008 16:58:25 -0500
-
-sflphone (0.9-2008-06-06) unstable; urgency=low
- * Audio bug correction: capture stopped after a few minutes of conversation
- with USB Plantronics sound card
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Tue, 06 May 2008 16:58:25 -0500
-
-sflphone (0.9-2008-05-06) unstable; urgency=low
- * Bug correction: account creation with the assistant
- * GTK+ warnings removal
- * libnotify warnings removal
- * Remove aliasing on the SFLphone logo
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Mon, 05 May 2008 16:58:25 -0500
-
-sflphone (0.9) unstable; urgency=low
- * Clean dependencies ( removal of libboost )
- * Several GTK improvement and updates
- -account window
- -configuration window
- * Migrate from GtkCheckMenuItem to GtkImageMenuItem
- * ALSA standard I/O transfers: MMAP instead of R/W
- * Fix speex audio quality
- * IAX2 protocol
- -Fix hold/unhold situation
- -Add on hold music
- * SIP protocol
- -Ringtone on incoming call
- -Fix transfer situation
- * Add desktop notification ( libnotify )
- * Improve the system tray icon behaviour
- * Improve registration error handling
- * Register/unregister from the account window takes effect without starting back SFLphone
- * Compilation warnings removal
- * Call history
- * Add an account configuration wizard
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Wed, 30 Apr 2008 16:58:25 -0500
-
-sflphone (0.8.2) unstable; urgency=low
- * Internationalization of the GTK GUI
- * English / French
- * STUN support
- * Slight modifications of the graphical interface ( tooltips, dialpad, ...)
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Fri, 21 Mar 2008 11:37:53 -0500
diff --git a/tools/build-system/launchpad/sflphone-video/debian/control b/tools/build-system/launchpad/sflphone-video/debian/control
deleted file mode 100644
index 11e6259f6a9a52d2035dd74597b7dbc0bd75d820..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/sflphone-video/debian/control
+++ /dev/null
@@ -1,12 +0,0 @@
-Package: sflphone-video
-Section: gnome
-Priority: optional
-Architecture: all
-Depends: sflphone-daemon-video, sflphone-gnome-video
-Maintainer: Savoir-faire Linux Inc <emmanuel.milou@savoirfairelinux.com>
-Description: GNOME client for SFLphone, with video support
- Provide a GNOME client for SFLphone.
- SFLphone is meant to be a robust enterprise-class desktop phone.
- SFLphone is released under the GNU General Public License.
- SFLphone is being developed by the global community, and maintained by
- Savoir-faire Linux, a Montreal, Quebec, Canada-based Linux consulting company.
diff --git a/tools/build-system/launchpad/sflphone/debian/changelog b/tools/build-system/launchpad/sflphone/debian/changelog
deleted file mode 100644
index 78cc17db26580026556cd5c42e8213c39c364043..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/sflphone/debian/changelog
+++ /dev/null
@@ -1,3138 +0,0 @@
-sflphone (1.1.0-rc20120607~ppa1~SYSTEM) SYSTEM; urgency=low
-
- ** SNAPSHOT 1.1.0-rc20120607~ppa1~SYSTEM **
-
- * gnome: cleanup logging
- * * #12061: gnome: fix video codec list display
- * * #12050: gnome: fixed regression in active codec handling
- * #11732: Do not align menu with other interface component
- * * #11818: gnome: stop daemon on SIGTERM, SIGINT or SIGHUP
- * #10304: updatePlaybackScale dbus method uses int 32 bit for size and
- position (allow for 24 days long recording playback)
- * #10304: Add time lable for seekslider
- * * #10725: gnome: seekslider: fixed compiler warning due to missing
- header
- * * #11732: gnome:calltree: fix warning when searchbar is not created
- * * #11252: daemon: removed deprecated zrtp code
- * #10725: Consolidate test to determine if there is a recording,
- remove log on NULL pointer
- * #10725: Remove logging from stop callback in seekslider
- * #10725: No need of playback related callbacks in uimanager
- * #10725: Remove logging in seekslider when no selected call
- * #11732: Align Icons with dialpad in gnome client
- * #10304: Make sure that the playback won't be updated after reset
- * #10304: Prevent playback widget reset from stoping dtmf
- * #10304: Move Playback widget at the bottom of the main window
- * #10304: Fix logic to determine outgoing/incoming calls for recording
- icon
- * * #11685: gnome: fixed memory leaks in config menus
- * #10304: Fix reset seekslider that prevent scale to be updated at
- first read
-
- -- Julien Bonjean <julien.bonjean@savoirfairelinux.com> Thu, 07 Jun 2012 16:04:59 -0400
-
-sflphone (1.0.0-rc20110930~ppa1~SYSTEM) SYSTEM; urgency=low
-
- ** SNAPSHOT 1.0.0-rc20110930~ppa1~SYSTEM **
-
- * update kde .gitignore
- * Fix bug in volume widget
- * More polishing for release
- * Bump version to 1.0.0
- * [#7023] Add the ability to load an abstract contact backend in the
- library to resolve more data, polish code
- * [#7021] More cleanup for release
- * Cleanup
- * [#7021] Refactor KDE client dbus handling, add a missing call in
- daemon and port the DataEngine to the new API
- * Remove some annoying debug
- * merge language scripts
- * remove obsolete 'VERSION' files
- * update install instructions
- * Add missing translations to gnome
- * language update
- * Revert "Don't reference count DBus clients, exit core immediately
- when one of them request it"
- * Don't reference count DBus clients, exit core immediately when one
- of them request it
- * [7021] Add contact abstraction support
- * [#7121] Polishing library (over). Indentation, spacing and naming
- are now consistent
- * codecs: link to libccrtp, don't use logger
- * Fix a daemon bug
- * [#7038] Fix adding contact
- * * #7037 : stop audio stream after all calls have been hanged up
- * [#7025] Add full support for bookmark
- * SFLPhone KDE do not destroy history anymore
- * Fix config skeleton
- * Close the daemon once and for all, no more automatic respawning
- * Fix "unregistered account" bug (I hope so)
- * Close SFLPhone at the right place, it still respawn, I don't know
- why
- * Remove dead code
- * Fix regressions introduced in the last commit
- * Dead code elimination 1/3
- * Fix bug, add "add contact" option, fix warning
- * * #7019: Fix IAX codec negociation
- * Remove or comment unnecessary/unhelpful debug output
- * Fix "same as local" account setting, fix IP2IP LED color
- * Add support for some more advanced config options and add missing
- config dialog icons
- * Fix crash with noise suppressor
- * Alternative can now be selected from the call view context menu
- * Add drag and drop support, initial context menu and fix 3 bugs in
- the account dialog
- * Add basic history drag and drop support
- * Complete contact support is back
- * * #6991 : fix IAX problems
- * Fix IAX accounts being disabled by default
- * Revert "deb: forge -g flags for pjsip"
- * * #5884: Disable debug code in pjsip
- * echo suppressor : more assertions
- * Don't let the daemon think crypto is enabled when it's not
- * Simplify ToneList
- * Some progress on contact support
- * Remove unused getRegistrationCount()
- * remove annoying debug
- * revert SIP bit of e27e5c39bad27bae28f574eb2cba7717e8956229
- * Simplify CallManager::placeCallFirstAccount
- * Fix crash on hold
- * * #6905 : SIP refactor
- * gnome client: be sure key exchange is set correctly
- * Move code into createSipTransport
- * Fix account registration on start
- * ManagerImpl::registerAccounts(): simplify
- * * #5884: don't mess with pjsip threads in echo suppressor
- * * #6905 : simplify udp/stun/tls pjsip transport creation
- * Restore and improve support for Call history
- * fix launchpad build
- * SIPVoIPLink: simplify / refactor
- * Fix libwidget linking
- * SIP: simplify
- * IM : simplify
- * gnome: remove some debug
- * AudioRtpFactory::stop() cannot fail
- * * #6905: simplify SIP code
- * pjlib: fix build without SSLv2, fix warnings
- * Port history to the new syntax
- * Test a dock widget based implementation for contact and history
- * Disable SSLv2 support from pjsip and sflphone
- * deb: forge -g flags for pjsip
- * Fix deb packaging to get debug symbols
- * remove debug
- * pjproject: update to last stable release (1.10)
- * Require gtk >= 2.20 and glib >= 2.24
- * tlsadvanceddialog: simplify
- * * #6902 : fix errors spotted by -DGSEAL_ENABLE
- * Update daemon dbus XML and port KDE config backend from dbus to
- local
- * Remove unused but set variables
- * * #6929 : fix IM widget, cleanup
- * Unconditionally enable debug symbols
- * Should fix many KDE issues
- * * #6886 : hitting backspace on empty number have no side effects
- * * #6905 : fix AudioCodecFactory access in optimized builds (-O > 0)
- * Remove unsupported and broken jaunty/karmic packages
- * * #6902 : avoid using some gtk deprecated functions
- * Update dbus introspection files
- * * #6904: removed unused contactmanager
- * * #6903 : use correct dbus-cxx package name
- * * #6902: don't use individual gtk headers
- * Fix a segfault when config is not present
- * Merge latest (0.9.13) KDE code. This version is not yet ready for
- git master, but better than the previous one
- * addressbook : simplify
- * * #5659 : sflphone-plugins doesn't depend on libedataserverui
- * * #5659 : addressbook doesn't use libedataserverui
- * gnome client doesn't depend on evolution
- * * #5695: addressbook: simplify
- * * #5695: addressbook : remove AddrBookHandle from plugin
- * * #5695 : addressbook : remove unused stuff in the client
- * * #5695 : addressbook : remove unused stuff, use static mutex
- * gnome client doesn't use evolution
- * gnome: use proper API to set GTK_CAN_FOCUS
- * * #6897: removed unused focus state vars/callbacks
- * gnome: fix calls to sflphone_fill_codec_list_per_account
- * * #6623: gnome: don't leak in mainwindow
- * gnome: mainwindow whitespace cleanup
- * gnome: actions.c parameter doesn't have to be a double pointer
- * * #6895: fix memleaks, cleanup in accountconfigdialog
- * * #6893: fixes segfault in client on clean history
- * * #6894: fix leaks, cleanup in sflnotify
- * daemon: fixed prints in main
- * * #6892: simplify, fix leaks in dialpad
- * * #6887: audiopreference creates audio layer
- * * #6660: use const char * const, not std::string for globally
- visible constants
- * * #6852: Preferences now solely responsible for audiolayer creation.
- * * #6860: refactor uimanager, also fixes #6865
- * * #6853: hangup as soon as all digits have been deleted
- * * #6852: alsa: retry if device is busy
- * * #6852: audiolayer creation depends only on preference.audioApi
- * * #6850: gnome: fix build for gtk < 2.22.0
- * cleanup in iax
- * alsa: typo
- * pulse: if we can't peek in audio input, we can't drop samples
- * * #6849: show error window if codecs are missing, instead of dying
- * EchoCancel: unused, remove
- * * #6629 : use number of samples as arguments for audio filters
- * * #6629 : remove unused Algorithm interface
- * * #6629 : use helper to call alsa functions and display error msgs
- * Remove unused type
- * * #6841: fix some error handling
- * * #6629: simplify AlsaLayer::alsa_set_params()
- * Get gdk key definition from header
- * * #6828: Replace raw key codes by gdk defines
- * remove some debug, enhance some other
- * mainbuffer: simplify
- * * #6561 : fix phantom call after transfer
- * Conference Participant set : simplify
- * SIPCall: remove unused functions, make invite session public
- * * #6229 : remove malloc/free from pulse audio loop
- * * #6629 : simplify pulse callbacks
- * * #6629
- * Simplify widgets
- * * #6629 : keep the correct audio module when frequency changes
- * * #6751: fixed erroneous debug msgs
- * callable_obj.h: removed unneeded pthread header
- * alsalayer: cleanup
- * * #6629: Always restart audio driver when changing parameters (ALSA
- only)
- * gnome GUI: don't block in DBus signal errorAlert()
- * * #6629 : simplify AudioLayer creation
- * * #6629 : remove unused and unconfigurable frameSize from audiolayer
- * * #6629 : remove unused error message from audio layer
- * Fix logic error when switching audio API
- * Remove unused AudioProcessing class
- * AudioRtpRecordHandler::initNoiseSuppress() : use noiseSuppress
- directly
- * * #6629 : use DC blocker directly in audio layers
- * * #6629 : clean AudioLayer
- * * #6629 : don't store mainbuffer inside audiolayer
- * * #6629 : correct AudioLayer::notifyincomingCall()
- * * #6554: cleanup, refactoring in sipvoiplink
- * * #6554: cleanup in iaxvoiplink
- * * #6554: throw exception in getSIPCall if pointer is NULL
- * * #6554: make some methods of sipvoiplink static
- * * #6655: cleanup in managerimpl
- * * #6554: refactoring, fix memleaks in sipvoiplink
- * * #6478: remove throw specs, cleanup in voiplink
- * * #6629 : remove unused AudioDevice
- * * #6655: removed more dependencies from managerimpl
- * * #6744: simplified numbercleaner
- * conference : remove one prototype
- * * #6743: fix ip2ip
- * Don't give glib warnings if icons are not found
- * gnome: fixed includes
- * Codec.h: removed unused function
- * * #6742 : clean dbus & icons
- * * #6699: refactor/cleanup accounts
- * icons: cleanup
- * timer : use second precision, not millisecond
- * calltree_update_clock : use correct type, returns something
- * * #6737: fixed typo in dbus call
- * * #6737: removed tests for removed API
- * * #6737: dbus: fixed bug from merge
- * * #6737: cleanup in accountlist
- * * #6737: cleanup in dbus
- * * #6740 : fix history double free
- * * #6740 : remove time updating thread from calls
- * * #6737 : use c99 for client
- * * #6738 : make history loading faster
- * sipvoiplink : don't crash on transfers
- * fixed typo
- * Remove unused file
- * Don't build networkmanager.cpp at all if NM is disabled
- * _debug* -> _debug
- * * #6554 : simplify sipvoiplink
- * hudson: added -x to git clean command
- * added git clean to hudson script
- * audiocodecfactory: cleanup
- * * #6718: refactored setTlsSettings into SIPAccount
- * * #6718: removed more unused methods
- * * #6718: refactored confmanager code into sipaccount
- * remove unused functions
- * * #6718: confmanager: removed more unused methods
- * AudioCodecFactory : cleanup
- * #6697 : Turn callableElement struct into union
- * * #6718: confmanager: removed more unused methods
- * * #6718: confmanager: removed more unused methods
- * * #6718: removed unused dbus methods, refactoring
- * * #6699: accounts: cleanup/refactoring
- * * #6699: refactoring, cleanup in accounts
- * * #6699: more account cleanup
- * remove unused autoconf variable
- * * #6714: fixed hudson script
- * make distclean in hudson
- * added || exit 1 to run_tests.sh call
- * * #6714: fixed make distcheck for sflphone-plugins
- * * #6714: fixed make distcheck for gnome client
- * * #6714: fixed make distcheck for daemon
- * git: #6698 split the main .gitignore file
- * gnome: gpointer is already a pointer
- * gnome: calltab_init: use calloc instead of malloc
- * * #6699: more account cleanup
- * * #6699: cleanup account
- * * #6554 : more *voiplink cleanup
- * * #6558 : more sipvoiplink simplification
- * * #6558: saner loadSIPLocalIP prototype
- * gnome: #6623 clean calllists
- * * #6692: more audiolayer cleanup
- * * #6692: cleanup/refactoring in audiolayers
- * * #6692: more forward declarations, AudioThread->AlsaThread
- * * #6692: audiolayer cleanup
- * * #6692: alsalayer cleanup
- * * #6558 : remove account creator
- * * #6558 : clean sipvoiplink
- * * #6554 : cleanup sipvoiplink
- * audiortp: cleanup
- * * #6657 : fix launchpad builds for good
- * * #6675 : send RTP dtmf events only once
- * * #6655: more cleanup
- * AudioRtpSession::updateSessionMedia() : simplify
- * * #6655: more cleanup in managerimpl
- * * #6655: removed more code, cleanup
- * * #6655: more cleanup, fixed infinite loop
- * * #6655: removed more unused files
- * * #6655: removed unused mutex
- * * #6655 removed more unused code
- * * #6655: removed unused methods
- * * #6655: cleanup in main
- * * #6663: fixed segfault when off hold from transfer
- * * #6658: user's active codec selection is respected
- * * #6660: static global string should be static const char* const
- class member
- * * #6659: use g_strcmp0, not strcmp for vals that may be null
- * callable_obj: fix double free
- * calltree_display_call_info() : simplify
- * * #6657: Fix launchpad builds
- * Logger::log() : simplify
- * AudioRtpSession : privatize members
- * * #6655: more constness, cleaned up/simplified methods
- * * #6654: call DBus::_init_threading so that dbus-c++ to make it
- threadaware
- * set default credentials on account creation
- * AudioCodecFactory::scanCodecDirectory() : simplify and correct
- * * #6623: fixed typos
- * * #6623: fixed more leaks
- * * #6623: fixed more leaks
- * * #6623: fixed more leaks, don't print codec name if null
- * * #6623: more leaks fixed in client
- * * #6623: fix more leaks, fixed some warnings
- * * #6623: fixed leak in history
- * updated gitignore
- * initialize dbus dispatcher correctly
- * Fix tests, hudson doesn't have a dbus daemon running
- * remove unused code
- * removeCall() : simplify , fix leak
- * stopRtpThread() : simplify
- * *CurrentCall : simplify
- * Fix memleak
- * fix serialization of audio api (pulse / alsa)
- * account map : simplify
- * remove call from callmap before terminating it, avoid use after free
- * * #6630 : don't make DBusManager a singleton
- * call: return confID by value
- * add back history code deleted by error
- * history : reverse logic
- * simplify history serialization and remove some debug
- * remove annoying debug
- * * #6464 : replace cerr with _error
- * * #6464: replace cout with logger macros
- * replace printf() with logger macros
- * update .gitignore
- * remove unused function
- * update eclipse projects
- * uimanager_new() : simplify
- * rename directories
- * celt: simplify a bit
- * Fix CELT configure.ac test
- * * #6612 : template speex codecs
- * * #6623: refactored conference obj
- * * #6623: refactored callable object, removed leaks
- * * #6623: more cleanup, fix leaks, make global vars static and rename
- them
- * * #6623: calltree: fixed memleaks, simplified code.
- * audiolayer: init pointer members
- * manager: catch exception on invalid hangup
- * * #6623: don't leak on calls to create_new_call
- * * #6611 : clarify codecs prototypes
- * ringtones : .au and .ul files are both ulaw
- * * #6611 : make sure samplerate converters are called correctly
- * ManagerImpl::switchAudioManager() : simplify
- * * #6623: fixed more leaks
- * * #6623: fixed more leaks
- * * #6623: fixed more leaks
- * * #6623: fixed leak, line-endings in imwidget
- * * #6627: zero-initialize pointers if they're going to be deleted
- * * #6628: don't leak calls on exceptions
- * Revert "audiortp: call join after calling stop on RtpThread"
- * sflphone-client: more constness
- * audiortp: call join after calling stop on RtpThread
- * * #6625: return 0 on successful completion
- * * #6624: fix segfault on servercallfailure
- * * #6621: Fixed double free, unlock mutex in ManagerImpl::terminate
- * * #6220: remove audio stream when peer hangs up
- * * #6596: AudioSymmetricSession shouldn't self-delete
- * resampler: grow internal buffers dynamically
- * merge up and down sampling => resampling
- * Leave test directory unchanged when running make check
- * audio algorithms : remove unused prototype
- * ringtone: detect codec from file extension
- * *AudioFile : simplify
- * * #6596: create local SDP on the stack, not the heap
- * * #6596: don't call Ost::Thread::terminate from dtor
- * audiofile: cleanup (samplerate -> unsigned)
- * remove unused func
- * samplerateconverter: cleanup
- * RingBuffer::Put() : remove unused return value
- * MainBuffer::putData() : remove unused return argument
- * audiolayer::putMain() : remove unused func
- * AudioLayer::putUrgent() : remove unused return value
- * * #6618: delete any remaining ringbuffers in destructor
- * RingBuffer::availForPut() : remove
- * * #6617: return from main rather than calling exit
- * MainBuffer::availForPut(): remove
- * RingBuffer: simplify
- * alsa : remove write only variable
- * fix memcpy declaration
- * bcopy(src, dst) -> memcpy(dst, src)
- * RingBuffer::Get() : remove constant volume argument
- * return a copy of the call ID, not just a reference.
- * MainBuffer::getDataById() : remove volume argument (always 100)
- * MainBuffer::getData() : remove constant volume argument
- * RingBuffer::Put() : remove constant volume argument
- * MainBuffer::putData() : remove constant (=100) volume argument
- * audiolayer: remove constant _defaultvolume
- * AudioRtpRecordHandler / AudioRtpSession : simplify
- * mainbuffer: fix test
- * iaxvoiplink : simplify
- * sip registration callback: fix a dbus crash
- * MainBuffer: simplify
- * AudioRtpFactory: return cached type of rtp session. The rtp session
- can have disappeared if the call was put on hold
- * AudioRtpFactory: remove unused setters
- * Fix launchpad builds
- * * #6611 : remove unused bandwidth codec information
- * * #6611: AudioCodec: remove useless/unused setters
- * make sure buffer string is initialized correctly
- * * #6596: declare certain destructors virtual
- * audiolayer : cleanup
- * Simplify doc build rules
- * * #6270: don't build dbus-api doc with make, should require make all
- * configure.ac: cleanup
- * Remove copy of dbus-c++ from libs/
- * * #6596: stop clock thread when peer hangs up
- * removed unused Fmtp.h
- * * #6595: more logical initialization order
- * * #6600 : fix account creation
- * * #6601 : fix configure.ac tests
- * remove unused variable
- * Don't mix stack and heap based allocations
- * Fix copyright (2009, 2008, 2009 -> 2008, 2009)
- * Fix warnings found by clang
- * * #6595: fix initialization order for AudioRTP
- * * #6592: removed typedef std::string CallID
- * * #6586: implement local g_slist_free_full for older glib versions
- * * #6579: fix memory leaks in client (there's a lot left)
- * ShortcutPreferences::setShortcuts() : simplify
- * Fix merge
- * * #6548: remove call to non thread-safe strerror()
- * AudioRtpFactory: each instance is associated to exactly one SipCall
- * create_audiocodecs_configuration() : make static
- * * #6269 : refactor AudioRtpSession
- * Fix AudioSymmetricRtpSession.h inclusion guard (cherry picked from
- commit c3081dce1cc1370d6d3558a4c4ef5cfac0d21caf)
- * * #6269: Rename AudioRtpSession to AudioSymmetricRtpSession
- * * #6574: Don't exit when connection to pulseaudio server fails
- * accountconfigdialog.h : remove some stuff from header
- * * #6560: fix configuration test
- * Fix warning in test
- * * #6560: don't hide password entry in security tab
- * * #6560: set initial password for SIP accounts
- * * #6506: remove useless pointer indirection
- * * 6560: password is now specific to IAX accounts
- * * #6560 : actually use, store, restore, transmit SIP credentials
- * * #6560: YamlEmitter: serialize sequences
- * YamlEmitterException: typo
- * ManagerImpl::computeMd5HashFromCredential() : simplify, fix memleak
- * * #6561: invite_session_state_changed_cb() : simplify
- * * #6561: More useful debug in VoIPLink::removeCall
- * * #6561 : fix ghost call reappearing in GUI after transfer
- * while -> for (make the code smaller)
- * * #6558 : Account::loadConfig() : move IAX code to IAXAccount
- * IAXVoIPLink::getAccountPtr : simplify
- * * #6554 : access the SIPVoIPLink directly, not per account
- * SIPVoIPLink is instanciated only once and is not associated to a
- single account
- * yamlnode: use const references when possible (still some left to do)
- * Account::_accountID: constify
- * VoIPLink: simplify, remove unused method
- * hudson test : no need to call run_tests.sh anymore
- * Remove AccountID type and AccountNULL define
- * Make check runs the test (no need to call run_tests.sh manually
- anymore)
- * gnome GUI: Fix tests
- * Revert "Move registration information from SIPAccount to
- SIPVoIPLink"
- * * #6392: pluginmanagertest: fix warnings reported by valgrind
- * * #6547 : remove unused exceptions
- * * #6547: CallManagerException: use runtime exceptions
- * * #6547: InstantMessageException: use runtime exceptions
- * * #6547: do not throw exceptions if some settings are not present in
- config file
- * * #6547: YamlParserException: use runtime exceptions
- * * #6547: VoipLinkException: use runtime exceptions
- * * #6547: YamlEmitterException: use runtime exceptions
- * * #6547: DTMFException: use runtime exceptions
- * * #6547: AudioFile: use runtime exceptions
- * * 6547: AudioZRtpSession: remove impossible error case
- * * #6547 : AudioRtpSession: remove impossible error case
- * * #6547: AudioZrtp: use runtime exceptions
- * * #6408 : send authenticationUsername to GUI
- * * #6408 : store/restore authenticationUsername from config file
- * SIPAccount: simplify
- * Move registration information from SIPAccount to SIPVoIPLink
- * SIPAccount::getAccountDetails : simplify
- * * #6540: yaml parser: simplify
- * sdp.cpp : fix a warning
- * * #6540: yaml parser : remove std::string typedefs
- * * #6540: Simplify yaml unserialization
- * * #6540 : add a Conf::ScalarNode constructor for booleans
- * setAccountDetails(): simplify
- * * #6408: store authentication username in daemon
- * * #6408: Be able to set the authentication username in the GUI
- * * #6507 : do not crash if the program is not sflphoned
- * Fix tests
- * macroify SIPAccount::unserialize()
- * Move all .cpp files from sflphoned target to libsflphone.la, except
- main.c
- * main() : simplify, return positive error codes
- * * #6507 : find codecs dir in build directory
- * * #6392: Sdp: move clean functions to destructor
- * AlsaLayer::adjustVolume() : simplify
- * alsalayer : reduce indentation
- * malloc/free -> new/delete
- * malloc/free -> new[]/delete[]
- * malloc/free -> new/delete
- * AudioSrtpSession: simplify base64 encoding
- * * #6392: Initialize std::string from pj_str_t correctly
- * * #6392: AudioRtpSession: Initialize remote port
- * Audio settings : Initialize _echoCancelTailLength and
- _echoCancelDelay(0)
- * Initialize variable
- * YamlParserException : fix use of stack variable after it has been
- deallocated
- * * #6392: fix memory leak in history
- * * #6392 AudioCodec : fix memory leak
- * * #6392 : fix memory leak in sip account
- * * #6408: clean up sipaccount (cosmetics mostly)
- * sipaccount.cpp serialize() : reduce number of lines
- * * #6392: invalid memory access
- * * #6392 : fix invalid memory access
- * * #6479: merged useful code from MimeParameters into Codec interface
- * * #6462: fixed hangup on IP2IP call
- * added run_daemon.sh script
- * test: remove unused variable
- * Remove functions only used by a failing test (cherry picked from
- commit fcf718cb75de7f1882dc61c07bb8d300dfa10f85)
- * * #6360 : make client tests build (cherry picked from commit
- 028b2835f040e51ab8ab979b32732b07b8798fce)
- * * #6360 : fix warnings in check_global test (cherry picked from
- commit 9e2bd6a7496dd64f6f48595e385760019aab1193)
- * * 6360: updated API calls in tests, but they're not building yet
- (cherry picked from commit 548f6f0f919b43772a3e9c667e5e292791281795)
- * Fixed include in tests (cherry picked from commit
- aeadc7525c1e31f936670ac8b02f0bcf387c38a8)
- * Remove unused variables and functions
- * IAX: fix warnings (cherry picked from commit
- fd7a113a11cac2cd9a7c36929e88ad28195c4c35)
- * Remove unused DEBUG define which interferes with logger.h (cherry
- picked from commit b2f72b91d0f43cb1dd94d138882a8caa9c841c24)
- * * #6392: no need to check for account NULLity since it is
- dereferenced above
- * * #6392: fix a memory leak, replace by stack allocation
- * * #6392: remove a variable assignement which confuses cppcheck
- * process_conference_participant_from_serialized() : remove unused
- function
- * * #6392: s/free/g_free/
- * * #6392: fix a memory leak in abookfactory_load_module()
- * * #6392: remove generate_call_id() used only once
- * * #6392: fix memory leak (opendir() without closedir())
- * * #6392: AudioRecorder(): ensures mbuffer is set
- * Remove SFLPHONED_VERSION from global.h, use autoconf PACKAGE_VERSION
- * #6298: Cleanup
- * #6331: Fix deleting ringtone file after call have been answered
- * * #6330: merged user_cfg into headers
- * #6298: Fix conference recording file update at conference end
- * #6298: Fix record file name serialization for conference
- * * #6295: cleanup of codec hierarchy
- * #6298: Fix gtk warnings
- * * #6300: added script to run tests
- * #6109: Add recording playback for conference
- * * #6300: tests do not require an installed sflphone
- * * #6295: re-removed clone methods
- * #6109: Fix gtk_critical warnings for incoming calls
- * #6109: Fix GTK_CRITICAL warning
- * #6109: Fix icons when history is not activated
- * #6109: Fix warnings
- * #6109: Implement stop recorded file playback signal
- * Revert "* #6295: removed unused clone method"
- * * #6295: removed unused clone method
- * * #6296: removed non existant file from Makefile.am
- * #6109: Stop fileplayback for outgoing call
- * #6109: Implement stop recording playback button
- * Fix binding names errors in dbus introspection file
- * #6109: Implement playback recorded file callback in client
- * #6109: Store recorded file path on client side
- * #6109: Add dbus methods for call recording playback
- * * #6290: remove unused classes from utilspp
- * * #6288: cleanup sdp
- * * #6288: fix exception usage
- * * #6288: simplify SdpException
- * * #6288: cleanup in sdp.cpp/h
- * #6109: Only display playback button if record file is set and valid
- * * 6290: updated configure.ac to remove functor Makefile
- * * #6290, #6289: removed unused classes from utilspp, fixed make
- check
- * #6109: Add button for history playback of recorded file
- * * #6289: removed unused observer class
- * * #6282: forward declare sdpMedia in sdp.h
- * * #6281: renamed setCallAudioLocal->setCallMediaLocal
- * #6183: Handle conference with more tahn two calls
- * #6183: Fix history icons when calling back a conference from history
- * #6183: Fix icons inconsistencies in history for conference hang up
- * #6183: Fix toolbar actions when selecting a conference in history
- * #6183: Fix conference serialization
- * #6268: Serialize only calls
- * * #6269: removed useless type testing
- * ignore some files in test/
- * * #6268: Remove dead class AudioSymmetricRtpSession
- * #6251: Do not had history calls in calllist when loading history
- file
- * #6251: Fix insertion in history map in before saving history file in
- daemon
- * #6251: Fix history unit tests
- * #6251: Order the list before serailization, get rid of the hashtable
- in history
- * #6251: Implement history serialization using a list wether than a
- map
- * * #6253: remove external audioport from header, make all members
- private
- * * #6253: don't store external local audio port (used for NAT) in
- Call
- * #6251: Add start_time timestamp in history serialization
- * #6251: Fix call insertion in conference items
- * #6233: Fix serialized account list terminated with a ";" character
- * #6238: Fix draggable history calls into current calls
- * #6233: Fix toolbar updates
- * #6233: Fix history
- * * #6235: remove pyc files from git tree
- * #6233: Handle cases when one or manuy calls are unreachable in
- createConfFomrParticipantList
- * #6233: Handle wrong numbers in createConferenceFromParticipantList
- * #6231: Fix drag-n-drop issue
- * * #6173 : move sippxml in tools
- * #6231: Fix merging issue
- * #6183: Implement conference unserialize
- * * #6212: remove extraneous flags from globals.mak
- * #6183: Unserialize conference data in conference
- * #6183: Add account information in request for conference call from
- history
- * #5755: Add -ldl to liker in sflphone
- * #5755: Fix fedora 15 compilation issue
- * #6183: Serialize conference participant phone number and account
- * #6183: Add conference timestamp in serialization
- * * #6186: don't include global.h, just logger.h
- * #6183: Fix saving history to file
- * #6183: Fix removing call from calllist
- * * #6184: remove pointers to Manager from AudioRtpSessions
- * #6183: Calling calltree_add_call explicitely for history
- * #6183: Ability to store conference inside history tab queue
- * * 6181: remove unused API from sipcall
- * #6171: Implment nreCallCreated callback
- * #6167: Fix participant list NULL ending
- * #6149: First draft of conference creation from history
- * #6149: Fix multiple call/conf selection callbacks ...
- * #6129: Fix place_call function called twice for pressing enter
- action
- * #6129: Fix double click action for history
- * #6149: Add dbus call for creating conference from history
- * #6129: Fix placing call from history and addressbook (still need to
- fix icon)
- * * #6148: removed unused AudioRtpFactory constructor
- * * #6145: remove unused isAudioStarted
- * * #6145: remove unused isAudioStarted
- * #6129: Add conference into history, fix call/conference selection
- * * #6143: don't use getType outside of serialization methods
- * * #6132: forward declarations instead of includes
- * * #6132: add constness, remove redundant "inline" keywords
- * #6129: Add timestamp to conference object to order history entries
- * * #6128: remove unused forward declarations from header
- * * #6127: make noncopyable class actually noncopyable
- * * #6125: don't include AudioRtpFactory in sipcall.h
- * #6123: Fix alsa ringback audio file
- * #6123: Fix raw audio file loading problem
- * #6109: Fix daemon plugin manager unit test
- * #6109: Fix history manager unit tests
- * #6109: Recording filename in daemon and client for history items +
- serialization
- * #6109: Refactor AudioFile to play recorded call
- * * #6104: AudioCodec moved to sfl namespace
- * * #6099: remove active flags from codec classes
- * #6095: Add notification-daemon as a runtime dependencies for rpm
- packages
- * #6095: Fix fedora 15 compilation in MineParameters.h
- * #6095: Declare static variable explicitely for client
- * #6095: Add logs to build OSC build machine
- * * #6098: global variables should have file-scope to avoid name
- conflicts
- * #6095: Fix compilation error for Fedora 15
- * #6095: Update SFLphone version to 0.9.14
- * #6095: Add specification file in opensusse build service for
- sflphone-plugins
- * #6073: Fix sflphone-plugins build on launchpad
- * #6093: Rename CodecDescriptor for AudioCodecFactory
- * * #6089: fix warnings in make check
- * * #6086: renamed codecs methods to audio_codecs
- * * #6085: renamed codec related dbus calls to audio_codec
- * #6065: Remove g_print from client, use DEBUG instead
- * #6065: Add actions name for addressbook
- * * #6085: renamed codecs* widgets/functions audiocodecs*
- * #6065: Fix Addressbook runtime warnings
- * #6065: Replace Codecs tab for Audio in account preference dialog
- * #6065: Fix "transfert" typo
- * #6065: Fix addressbook action runtime warning in uimanager
- * * #6082: fixes make check by adding libcrypto libs to test
- dependencies
- * #6073: Rename plugin/addressbook folders for addressbook/evolution
- in sflphone-plugins
- * #6074: Removed AC_SUBST from configure.ac when using
- PKG_CHECK_MODULE
- * #6073: Fix sflphone-plugins package build
- * #6073: Fix sflphone-common build
- * #6065: Fix runtime gtk warning when initializing searchbar without
- addressbook
- * #6063: Fix mozilla-tellify gitignore
- * #6063: Remove stream copy file using ifdef macro
- * * #6012: fix make dist for sflphone-common
- * #6063: Update .gitignore file
- * #6058: Fix base64 encoding related warnings
- * #6056: Fix SdpException handling
- * #6055: Fix unknown pargma warning for gcc <= 4.5
- * * #5949: test gcc version before disabling unused-but-set warning
- * #6054: Fix addressbook plugin compilation warning
- * #6048: Fix uimanager static initialization
- * #6046: Fix addressbook factory static initialization of member
- addrbook
- * #5979: Fix implicit function declaration warning
- * #6042: Fixed discarding qualifier warnings in client
- * #6041: Fix instant messaging unhandled case warning
- * #5994: Implement set current addressbook name and search type in
- addressbook plugin
- * #5994: add rules for launchpad packaging of addressbook plugin
- * #5994: Fix addressbook plugin configuration loading
- * #6027: Fix addressbook enabled test from configuration
- * #6027: No need of gnomedoc related macros in addressbook plugin
- * #6027: Add NEWS file required for build
- * #6027: Add addressbook plugin autogen.sh script
- * #6027: Remove plugins from client
- * #6027: Add sflphone-plugins folder at project's root level
- * #5994: Move addressbook folder from contacts to plugin folder
- * * #6011: removed unused Makefiles
- * * #6010: remove unused headers
- * * #5952: fix "string constant to char*" warnings
- * * #6009 fixed warnings
- * * #6003: finished cleanup of account classes
- * * #6003, #6004: cleanup of account classes, defaultAccount no longer
- global
- * * #6000: fix memory leak of args object
- * * #5998: removed using namespace std from networkmanager
- * * #5998: removed "using namespace std" from ZrtpSessionCallback
- * * #5998: removed using namespacestd from AudioZrtpSession.h
- * * #5998: remove "using namespace std" from auriorecord.h and
- MimeParameters.h
- * * #5998: remove using namespace std in main
- * * #5998: removed "using namespace std" from logger
- * * #5949: test gcc version before disabling unused-but-set warning
- * #5994: Installation of addressbook plugin
- * #5979: Implement codec full addressbook search from plugin
- * #5979: Implement addressbook factory and plugin
- * * #5981: unused webwidget removed
- * #5966: Account config synchronization fix (for stun)
- * #5954: Handle media name exception
- * #5954: Fix audio codec name display in client
- * #5954: Clean up getSessionMedia methods
- * * #5957: getRecordingSmplRate returns a value
- * #5954: Clean up getCurrentCodec methods
- * * #5950: remove "converting to non-pointer type 'int' from NULL"
- warnings
- * #5915: Full gain control version
- * * #5949: remove more unused variable warnings
- * * #5949: remove unused/unused-but-set variable warnings
- * * #5949: show_preferences_dialog returns a success value
- * * #5946: cleanup of include directives, undefined function
- * * #5515: comment out SSLv2 calls in pjsip
- * #5915: Implement different slope for attack tme and release time for
- gain control
- * #5915: use only one input signal for gain control (removed output
- buffer)
- * #5921: Fix no audio after holding a conference
- * #5916: Add gaincontrol files
- * #5916: Implement FFMPEG/CCRTP video streaming prototype
- * #5903: Fix call transfer during a conference
- * #5915: implement rms detector, first order averager, limiter for
- gain control
- * #5914: Fix call transfer when no notification request is required
- * #5899: Fix conference right-click segfault
- * #5884: temporary fix segfault in pjsip memory pool
- * #5883: Fix compilation issues on maverick and lucid
- * #5755: Fix fedora 15 compilation without patching ccrtp
- * [#5855] Make echo canceller optional
- * #5855: Fix echo suppression activation/deactivation
- * #5855: Implement pjsip echo canceller
- * #5814: Speex initialization function uses samples, not bytes
- * #5814: Test using more unbalanced signals
- * #5814: Fix buffer size for long echo length or long echo delay
- * #5814: Adjust level for echo cancellation at runtime
- * #5814: Process noise reduction before echo cancelling
- * #5814: Implement speex post echo canceller processing
- * #5814: Dump echo cancel file to disk
- * #5814: Add parameters for echo cancel
- * #5809: Add configuration parameters
- * #5809: Implement speex echo canceller in audio rtp session
- * #5814: Code cleanup
- * #5814: Fix conf creation with several incomming ringing calls
- * #5814: Fix conf creation segfault when dragging a call on hold on a
- ringing call
- * #5809: Added unit test for echo cancellation and implemented
- "process" virtual method
- * #5709: Add always recording option in configuration
- * #5709: Add always recording option in audio conference panel
- * #5709: Add core functionnality for always recording (missing config
- options)
- * #5769: Fix conference participant handling (detach/attach) and hold
- actions
- * #5747: Fix recording icons and state for conference when adding new
- participant
- * #5769: Code cleanup
- * #5769: Fix hangup unsent calls
- * #5769: Fix remove/add additional participant to conference
- * 5769: Several fixes concerning confererence handling
- * #5769: Fix compilation error
- * [#5769] Fix audio streams binding in main buffer
- * #5769: Removed access to audio mixer from audio layer
- * #5765: Fix audio crash for illformated wavefiles
- * #5765: Add maximum iteration for finding fmt and data "chunck"
- * #5589: Fix compilation of libnotify under
- * #5757: Fix abort signal when receiving INFO
- * #5747: Add usersDetached.svg
- * #5747: Handle offhold action for recording conference
- * #5747: Fix off hold action for conferences
- * #5747: Implement update conference in record action in calltree
- * #5747: Add new icons for recording conferences
- * #5747: Add recording state for conferences
- * [#5738] Remove getAudioDriver call from manager (replace by
- _audiodriver var)
- * [#5738] Refactor mutex protecting audiolayer
- * [#5737] Fix HD conference recording
- * [#5730] Fix start audio session after changing sampling rate
- * [#5714] Fix enter keyboard event for addressbbok and history
- * [5695] Fix addressbook combo box update when no addressbook selected
- * [#5695] Fix addressbook initialization and search bar update
- * [#5695] Add mutex for books_data in addressbook to protect async
- calls
- * [#5695] Get back addressbook open from uri
- * [#5695] Fix absolute addressbook URI for local addressbooks
- * [#5695] Implement libebook 3.0 interface
- * [#5571] Better logic for hangup (for case where call have not been
- sent yet)
- * [#5571] Update error handling in voip links
- * [#5571] Fix compile time warnings
- * [#5696] Fix installation dependencies for Natty
- * [#5669] Add mention that sflphone.org is for testing only
- * [#5693] Add natty in teh dput.conf file
- * [#5690] Remove not useful logs
- * [#5670] Use dynamic payload type for rtp dtmf
- * [#5668] Clean up sflphone configuration logging
- * [#5668] Fix hook checkbox configuration update
- * [#5666] Fix unit tests
- * [#5666] Manage event subscription
- * [#5666] Emit bye request when subscription is terminated
- * [#5666] Bye request should be sent after event subscription
- notification is done on transfer
- * [#5666] Make reinvite method static (to be called in pjsip
- callbacks)
- * [#5666] Hangup Call in manager for AccountNULL and IP2IP
- * [#5589] Use PKG_CHECK_MODULE for every client's dependencies
- * [#5623] Enlarge initial size of pjsip memory pool for calls (16k)
- * [#5564] Fix audio recording resampling for g722
- * [#5571] Move attribute handling for onhold/offhold actions in SDP
- session
- * [#5571] Codec negotiation refactored and unittested
- * [#5571] Implement tests
- * [#5571] Implement pjsip negociator
- * [#5571] Fix unit tests
- * [#5571] Add Fmtp.h to repository
- * [#5571] Integrate mime types and codec factory
- * [#5571] Handle exception when SDP negotiation fails
- * [#5570] Add sflphoned-sample.yml in repository
- * [#5564]: Implement stereo to mono mixing for rigntone
- * [#5342] Update audio stream initialization
- * [#5514] Restore test ni historytest suite
- * [#5514] Fix
- * [#5514] Disable test_create_history_path
- * [#5514] use pulseaudio in sample config file
- * [#5514] Fix test: load history from file
- * [#5514] Do not use X
- * [#5513] Make unit tests compile successfully
- * [#3947] Enable unit tests in Jenkins
- * [#5454] Fix build system to handle new version number
- * [#5454] Update languages from launchpad
- * [#5454] Add --without-celt in OpenSuse build service
- * [#5454] Change version number
- * [#5331] Added first SDP session tests
- * [#5273] Update nightly build version tags to conform dpkg rules
- * [#5211] Refactor send register method for iaxvoiplink and
- sipvoiplink
- * [#3950] Remove call being transfered from calltree
- * [#5211] Use appropriate memory pool for transport selector
- * [#5211] Fix strict aliasing rules warning in pjsip
- * [#5211] Bring back pjsip shutting down sleep to 1000 ms
- * [#5211] Fix registration callback segfault when closing the
- application
- * [#5211] Use the dialog memory pool for Route header in INVITE
- request
- * [#5211] Add temporary memory pool for findLocalAddressFromUri and
- findLocalPortFromUri
- * [#5211] Use individual memory pool for dtmfs
- * [#5211] SipVoipLink refactoring
- * [#3950] Attended transfer for conference calls
- * [#5284] Fix DNS resolution for Route with specified port number
- * [#5284] Some code cleanup
- * [#3947] Fix typo in hudson script
- * [#5284] Added sip route to REGISTER, INVITE, BYE request, plus DNS
- resolution
- * [#5266] Use RTP dtmf as default
- * [#5284] Added pjsip_process_route_set after setting routes in regc
- structure
- * [#5286] Fix parsing error due to long configuration file (removed
- max event)
- * [#5286] Fix false test in configuration emmiter
- * [#5286] Code cleanup
- * [#5286] Updated exception handling in configuration system
- * [#4969] Fix put SRTP call on hold
- * [#3950] Add debug messages
- * [#3950] Ability to perform an attended transfer
- * [#5276] Fix initialization problem in g722
- * [#3950] Add replace header in SIPVoIPLink::transferWithReplaces
- method
- * [#3950] Implemented attended method in SIPVoIPLink
- * [#3950] Cleanup transaction request received callback
- * [#3950] Implement dummy attended transfer in gnome-client
- * [#5249] Fix audio samplerate update algorithm for g722
- * [#5249] Fix uninitialized variable used in conditional jumps
- * [#5249] Fix conditional jump error in audiolayer (uninitialized
- value)
- * [#5267] Use autoconf 2.65 as a requirement (instead of 2.67)
- * [#5267] Restore manual pjsip configuration and compilation
- * [#5267] Autodetect celt version (0.9.1, 0.7.1)
- * [#5267] Fix deprecated macros in gnome client configure.ac
- * [#5267] Update configuration for libcelt-dev
- * [#5267] Fix build autoconf and automake
- * [#5227] Deactivate automatic call to astyle after compilation
- * [#5242] Hangup every calls before leaving
- * [#5237] Will now nightly-build for natty, Karmic deprecated
- * [#5229] Use inner class for rtp thread instead of inheritance
- * [#5211] Move mainbuffer unbind call in rtp final method
- * [#5211] Initialize sip call memory pool using 16 kb
- * [#5211] Use call memory pool in session reinvite
- * [#5211] Add debug messages
- * [#5211] Use and internal pool for calls
- * [#5211] Reduce pjsip memory pool usage for stateless error messages
- * [#5211] Refactor call deletion
- * [#5212]
- * [#5208] Refactor codec management for accounts
- * [#5168] Remove printf from codec's encode & decode method
- * [#5168] Fix celt compilation on launchpad
- * [#5168] Fix sflphoned compilation warnings in audiocodec.h
- * [#[#5168] Must keep the g722 specific RTP rate to avoid incoming
- packet timeout
- * [#5168] Fix static/dynamic payload rtp session update
- * [#5168] Throw SIPVoipLink Error if codec not instantiated in new
- outgoing call
- * [#5168] Fix dynamic/static codec payload type ambiguity
- * [#5169] Fix doubled IP2IP profile when no config file
- * [#4867] Add gtkinfobar in configuration panel
- * [#4867] Disable input/output/ringtone selection when using default
- alsa plugin
- * [#4952] Patches for possible buffer overflows
- * [$4885] Fix schemas problem
- * [#4885] sflphone.schemas not present during build
- * [#4885] Add gconf shemas directories in opensuse build system
- * [#4885] Add file/folder ownership for opensuse-factory build system
- * [#4906] Fix opensuse-factory build
- * [#4885] Update name dependency for libedataserver
- * [#4885] Fix non-void function without return in dbus-c++
- * [#4895] Update language translation
- * [#4896] Update session timestamp when updating media
- * [#4896] Reapply RTP hack for G722 payload type
- * [#4896] Update recording sampling rate when updating codec
- * [#4897] Save codecs in config for each configuration changes
- * [#4895] Do not save config when sflphone quit
- * [#4885] Update date for copyright
- * [#4885] Deactivate siptest that require more than one sipp instance
- * [#4879] Remove inmcoming call notification from IAX
- * [#4885] Some cleanup
- * [#4874] Add setCancel immediate/deffered for ost::Thread
- * [#4879] Fix incoming call notification
- * [#4878] Set keyboard focus on searchbar when selecting addressbook
- * [#4874] Fixed compilation warning
- * [#4874] Fixed compilation warning in sipvoiplink
- * [#4874] Fix compile time warning in RTP record handler
- * [#4874] Fix conditional jump in SDP
- * [#4874] Fix conditional jump based on uninitialized value
- * [#4874] Store call id within rtp thread context
- * [#4874] Fixed conditional jump based on uninitialised value in
- conference
- * [#4871] Fix default account fetching
- * [#4870] Delete RTP session when Refusing an incoming call
- * Restore IP to IP call
- * [#4857] Fix audio codec negotiation problem
- * [#3947] Adjust ressources allocated to compilation
- * [#3947] Disable unit tests in Hudson
- * [#4305] Free mutex only when really quiting SFLphone
- * [#4859] Update copyright to 2011 in every source file
- * [#3218] Character '.' stripped by the caller engine
- * [#4854] Fix typos, desktop entry
- * [#4847] Apply RTP modification to ZRTP session
- * [#4852] Update Karmic and Lucid dependencies
- * [#4852] Add Libedataserver and libedataserverui as gnome client
- dependencies
- * [#4852] Add authentication mechanism for EDS
- * [#4851] Fix segfault when closing pulseaudio layer too rapidly
- * [#4808] Some otehr cleanup
- * [#4808] Made some cleanup
- * [#4808] Added mutex in rtp session for codecs and noise process
- * [#4847] Update audio processing when updating RTP media
- * [#4842] Add support for linking with gold/ld --no-add-needed
- * [#4808] Make update g722 related static/dynamic payload logic
- * [#4827] Upper limit on the number of contacts to import from EDS is
- hard-coded to 500
- * [#4808] Fix put call on/off hold
- * [#4808] Implement early RTP start for incoming calls
- * [#4808] Audio stream is no longer start within RTP session.
- * [#4808] Removed coupling between audio layer and and RTP session
- * [#4702] Start audio rtp session as soon as it is created
- * [#4702] Init timestamp to 0
- * #4702: Send RTP packets immediately, no need of outgoing queue
- * [#4784] Update dbus-c++ version from gitorious
- * [#4702] Update RTP timeouts
- * [#4702] Lengthen RTP timeouts
- * [PATCH] Fixed compatibility with old libtool versions.
- * [PATCH] Accept older libebook (Maemo 5 has 1.4.2)
- * [PATCH] Fixed double-free error in preferences dialog
- * [PATCH] Fixed building of sflphone-common on Maemo5
- * [PATCH] Improved Gnome client initialization error handling. 1. It
- no longer segfaults when sflphoned isn't available. 2. User is
- provided with GUI error dialog.
- * [PATCH] Improved autogen.sh scripts 1. They do not require bash
- anymore 2. Added workaround for Debian bug #565663 3. Replaced
- manual autotools invocations with single autoreconf call 4. Non-zero
- return status on failure
- * Revert "[#4468] libtool <= 2.2 doesn't have LT_INIT macro so
- AC_PROG_LIBTOOL should be used instead."
- * Revert "[#4468] Libebook 1.4 is sufficient"
- * Revert "[#4468] Apply big path on dbus communication system"
- * [#4468] Apply big path on dbus communication system
- * [#4468] Libebook 1.4 is sufficient
- * [#4468] libtool <= 2.2 doesn't have LT_INIT macro so AC_PROG_LIBTOOL
- should be used instead.
- * [#4639] Fix determining default addressbook if this property is not
- set in gconf
- * [#4639] Fix memory leaks in Addressbook
- * [#4637] Fix opening default addressbook at sflphone init
- * [#4622] Free yaml events while parsing configuration file
- * [#4623] Fix conditional jumps based on uninitialized variable
- * [#4622] Fix leaks in yaml serialization engine
- * [#4616] Fix addressbook warnings
- * [#4514] Adjust RTP timestamp
- * #4527: Rename Karmic libyaml and Celt package in debian control file
- * #4495: Rework addressbook opening loop
- * [#4524] Increment RTP count when sending data
- * [#4524] DO NOT start RTP session twice
- * [#4367] Use PKG_CHECK_MODULE for celt
- * [#4367] Fedora package celt as celt (not libcelt)
- * [#4367] Astyling
- * [#4367] Update .po files
- * [#4367] Fix segfault in gensin
- * [#4354] Make celt a direct dependency on launchpad opensuse build
- service
- * [#4367] Make celt a required package, option --without-celt valid
- * [#4367] Fix zrtp timestamping error
- * [#4367] Fix audio zrtp timing
- * [#4367] Dispatch ZRTP packets
- * [#4367] Fix segfault when unloading account map
- * [#4367] Fix zrtp session
- * [#4367] Implement on packet receive
- * [#4367] use symetric audio rtp session, not dual
- * [#4367] Reduce packet receive/sent timeout
- * [#4367] Reduce RTP timeouts
- * [#4367] Move speaker data receive
- * [#4367] Move speaker data receive
- * [#4367] Move receive speaker data method
- * [#4367] Remove debug in rtp session
- * [#4367] Fix g722 codec clock rate
- * [#4367] Fix noise suppression initialization
- * [#4367] Fix segfault in RTP mic fadein method
- * [#4367] Refactor mic data encoding in rtp session
- * [#4367] Implement RTP main loop
- * [#4367] Fix compilation problem
- * [#4367] Fix AudioRtpclass using TRTPSessionBase
- * [#4367] Fix AudioRtpSession putDtmfEvent shadowing
- * [#4367] Fix AudioRtpSession putDtmfEvent shadowing
- * [#4367] Refactor RTP session (phase 2)
- * [#4367] Refactor RTP session (phase 1)
- * [#4367] Remove Redeclaration of SymetricAudioRtpSession in
- rtpfactory
- * [#4265] Add continue statement in for loop for invalid addressbook
- * [#4261] Makes addressbook initialization more robust
- * [#4257] Add maverick in build system
- * [#4233] Add sdp related unit tests
- * [#4233] Add condition and signal in two incoming call test
- * [#4243] Fix segfault in AudioSrtpSession
- * [#4243] Fix memory leak in AudioSrtpSession
- * [#4243] Make audio srtp optional in for incoming call
- * [#4243] Add boolean variable to make sure remote crypto context
- initialized only once
- * [#4243] Add documentation to AudioSrtpSession
- * [#4243] Use 80 bits authentication tags by default
- * [#4243] Init audio srtp remote crypto context in
- call_on_media_update
- * [#4243] Move SDP negotiastion in mod_on_rx_request
- * [#4243] Implement initLocalCryptoInfo to be called at different
- momment
- * [#4243] Init init local crypto context in when initializing audiortp
- * [#4243] Change key length according to sdes negociation
- * [#4243] Associate callid to accountid for incoming calls
- * [#4242] Fix no SDES keys in IP2IP calls
- * [#4242] Fix no SDES keys in IP2IP calls
- * [#4233] Test for call on/off hold
- * [#4233] Add two incoming call test
- * [#4233]
- * [#4233] Add 2 outgoing simultaneous call unit tests
-
- -- Julien Bonjean <julien.bonjean@savoirfairelinux.com> Fri, 30 Sep 2011 13:44:57 -0400
-
-sflphone (0.9.7~rc1~ppa1~SYSTEM) SYSTEM; urgency=low
-
- ** 0.9.7~rc1~ppa1~SYSTEM **
-
- * [#2462] Set explicitly the transport on incoming call too
- * [#2462] fix typo
- * [#2462] Use different address for SDP and call IP
- * [#2462] Use published address in SIP-SDP
- * [#2181] Fixed changelog files
- * [#2181] Updated spec file
- * [#2402] Fix pointer to int conversion warning (atoi)
- * [#2402] Remove daemon warnings, make indent
- * [#2459] Make sure the stream is opened when the call is answered
- * [#2402] Add conference related picture in documentation
- * [#2443] Not much ...
- * [#2399] Fix dialing display problem
- * [#2450] Fix incoming call already in conference crash
- * [#2399] Display peer name on the first line and peer number on the
- second
- * [#2450] Handle 403 FORBIDDEN when refused
- * [#2447] Bind offHold/onHold actions to button in gtk client
- * [#2447] Bind hangup action to button for conference
- * [#2447] Add conference action in gtk client's ToolBar
- * [#2381] Disable the password hashing in config file
- * [#2402] Cleanup
- * [#2366] Set callback to null when deleting Pulseaudio streams
- * [#1313] Fix main buffer unit test
- * [#1313] Fix audio layer unit test
- * [#2315] Hide pw in security tab, display when editing, sync with
- basic tab
- * [#1313] UnitTest change AudioRtpSession for AudioSymetricRtpSession
- instance
- * [#2402] Code cleanup
- * [#2444] Add debug to catch occasional crash when loading client's
- config
- * [#2444] Add debug info to catch occasional crash when loading config
- dialog
- * [#2402] Restore Call menu translations
- * [#2403] Use the published address if checked in GUI
- * [#2442] Add protection test in sdp
- * [#1841] Reapply pjsip patch concerning DNS SRV resolution
- * [#2384] Tags incoming call as direct SIP call, if applicable
- * [#2402] Change the monkey face
- * [#2315] Enable user to display password in clear text
- * [#2434] Force optimization level at 2
- * [#2284] Fix dbus_get_all_ip_interface compilation warnings
- * [#2431] Popup main window on incoming if applicable
- * [$2402] Fix simple warnings
- * [#2402] Fix implicit variable init order in LibraryManagerException
- * [#2402] Fixing implicit variable initialization warnings in
- AudioRtpSession
- * [#2402] Revert atoi change, fixing codec list doubled entries
- * [#2402] Fix gpointer to gint conversion
- * [#2402] Fix pointer casting to integer different size warning in
- codec list
- * [#2402] Fix warning discarting qualifiers from pointer target
- * [#2402] Fix gtk tree view assignement from incompatible type warning
- * [#1669] Fix audio recording folder utf-8 non compatibility issue
- * [#2414] Clean up debugs
- * [#2414] Use transport set in iptoip Account and update it frm
- preference
- * [#2348] Use macro N_() to mark ui.xml strings as translatable
- * [#2414] Rename getSipAddress/setSipAddress functions
- * [#2407] Fix volume controls display
- * [#2407] Fixes dialpad
- * [#2383] Set ip to ip config when clicking apply button
- * [#2404] Update call-to script - Maxime Chambreuil
- * [#2405] Client handles unknown call in current state as well
- * [#2383] Add DBUS signal to send IPtoIP local address and port as
- string
- * [#2383] Add Ip to IP config change apply call back
- * Clonflict
- * [#2402] Code cleanup
- * [#2383] Do the same for IPtoIP (init localn ip with first in the
- list)
- * [#2383] Use first interface in the list if local addresss is not
- defined
- * [#2403] Clean up unuseful addresses/ports
- * [#2403] Use the IP profile SIP port as global SIP port
- * [#2383] Fix dbus_get_all_ip_interface warnings
- * [#2383] Take into account sameAsLocal when loading published address
- * [#2383] Tsake into account sameAsLocal option when saving published
- address
- * [#2383] Update local ip address in ip to ip config
- * [#2383] Save ip 2 ip local port in config
- * [#2406] Update toolbar at startup
- * [#2284] Remove redefinition warnings + speex warnings
- * [#2383] Fix security table in account config
- * [#2383] Save ip 2 ip network interface parameters in config
- * [#2403] Restore sip transport selector
- * [#2383] Fix filling the Localt IP Address on account creation
- * [#2383] Fix Gtk-Critical when checking STUN
- * [#2383] Fix reopening account configuration display issue
- * [#2383] Load IPtoIP local address and port in preference iptoiptab
- * [#2383] Add LocalAddress and Localport in Preference IpToIp tab
- * [#2403] Use the address and port associated to the account as often
- as possible
- * [#1753] Removed pjsip generated files
- * [#1753] Removed remaining milenage lib references
- * [#2383] Add _publishedSameasLocal variable in sipaccount
- * [#2383] Add PUBLISHED_SAMEAS_LOCAL variable in config
- * [#2383] Fix stun set active or not when opening config
- * [#2181] Added RPM 64bits dbus patch
- * [#2402] Code indentation
- * [#2313] Force $(HOME).cache directory creation at startup
- * [#2383] Separate network interface and published address in account
- config
- * [#2400] Change dbus service installation path to libdir
- * [#2382] Move TLS related published address options in security tab
- * [#2382] Indent accountconfigdialog.c
- * [#2181] Install libdbus-c++ in $pkglib instead of $lib
- * [#1753] Remove ILBC code and disable it by default in the configure
- * [#1753] Remove milenage directory
- * [#2382] Fix switching interaface instabilities
- * [#2396] Save local ip in account creation wizard
- * [#2284] Remove warning on hold
- * [#2387] Fixes history searching and filtering
- * [#1215] Add samplerate display in the GUI
- * [#1663] Voicemail icon reflects voice messages
- * [#2395] Fix account registration ( specifically with callcentric)
- * [#2386] Strip "sip:" on incoming call, fixing history call back
- * [#2181] Updated spec files
- * [#1215] Display codec name in calltree instead of status bar
- * [#2390] Move back nbCalls and stopStream higher in refuseCall
- * [#2392] Fix ringtone during call in IAX
- * [#2391] Stop audio streams when there is 0 calls only
- * [#2391] Add debug when call state is not valid
- * [#2390] Clear returns in IAXvoipLink::sendAudioFromMic() method
- * [#2380] Fixing IncomingCallNotification not regular
- * [#2339] Query conference at client startup
- * [#2339] Working conference querying at startup
- * [#2339] Add conference in call tree
- * [#2339] Primitives to query conferences at client startup
- * [#2320] Add account selection in history
- * [#2355] Temporary solution: do not delete pointer when removing
- account
- * [#2380] Change algorithm in AudioRtp to trigger an
- IncomingCallNotification
- * [#2274] Comment sdebug in MainBuffer flush method
- * [#2274] Add flushMain() in ManagerImpl::addStream
- * [#2274] Add getBufferID() method in ring buffer
- * [#2274] Fix warning, comment debug in ringbuffer's flush method
- * [#2274] Use AudioLayer flushMain() and flushUrgent() in ALSA
- * [#2274] Clean up unused variable warning
- * [#2274] Protect minbudffer pointer on flushing
- * [#2274] Fix playATone method which writing empty buffer in urgent
- ringbuffer
- * [#2274] Use audio layer flushUrgent and flushMain in createStreams
- * [#2274] Use flush audio calls from audiolayer
- * [#2274] Flush when peer answered call
- * [#2375] Flush main buffer in iax when answering a call
- * [#2274] Parse displayname using c++ string method
- * [#2375] Flush main buffer when off holding calls
- * [#2375] Flush main buffer mon RTP startup
- * [#2376] Use now Pulseaudio module-cork-music-on-phone
- * Updated OSC packaging
-
- -- Julien Bonjean <julien.bonjean@savoirfairelinux.com> Fri, 20 Nov 2009 13:59:02 -0500
-
-sflphone (0.9.7~beta~ppa1~SYSTEM) SYSTEM; urgency=low
-
- ** 0.9.7~beta~ppa1~SYSTEM **
-
- * [#1933] Cleanup debug
- * [#1933] Clean up debug
- * Fix mic
- * [#1933] Set the IAx format earlier
- * [#1933] Move IAX sendAudioFromMic outside if (call) statement
- * [#1933] Fix startstream when offhold in iax and add debug concerning
- codec neg.
- * [#2371] sflphone_notify_voice_mail: minor gettext message formatting
- cleanup
- * [#2371] select_account_cb: properly gettextize status message
- * [#2371] show_account_list_config_dialog: properly gettextize status
- message
- * INSTALL: Minor tidyup of core install guide
- * Add /sflphone/src/icons/Makefile to .gitignore
- * [#2181] Updated OpenSUSE files (tmp)
- * [#1933] Add debug for codec negociation for iax
- * [#1933] Get rid of getMicAvail and getMicData in audiolayer (not
- used anymore)
- * [#1933] Add "audio codec not determined" error in IAX
- * [#1933] Test flush data
- * [#1933] Do not need to start audio stream in iax anymore
- * [#1933] Protecting pointer
- * [#2284] Remove more compilation/execution warnings
- * [#2284] Cleanup debug in client, use DEBUG instead of g_print
- * [#2284] Clean up uimanager
- * [#2370] Remove warnings
- * [#2366] Clean up other debug
- * [#2366] Clean up debug
- * [#2366] Call pa_xfree explicitely in writeToSpeaker
- * [#2284] Remove address book warnings
- * [#2365] Fixes bad cast
- * [#2352] Fix continuous ringing when peer hangup and call not yet
- answered
- * [#2181] Added version support
- * [#2181] Fixed some minor issues
- * [#2360] Moved MainBuffer from AudioLayer to ManagerImpl
- * [#2352] Makes getMainBuffer() everywhere
- * [#2352] Use 50 sec latency on pulseaudio stream creation
- * [#2352] Add alsa debug
- * [#2359] Update repository documentation
- * [#2354] Move pulseaudio disconnectAudioStream after stopping main
- loop
- * [#2352] Adjust nb byte copied in pulseaudio according to
- writeableSize
- * [#2352] Specify pulseaudio tlength parameters using pa_usec_to_bytes
- * [#2322] Convert italian translation to UTF-8
- * [#2357] Fixes window size
- * [#2357] Display only actionnable tool item
- * [#2333] Update streams parameters
- * [#2347] Use GNOME user settings for Menu and Toolbar appareance
- * [#2349] Load/Save properly audio params
- * [#2322] Update translations from Launchpad
- * [#2181] Added Francois Marier script
- * [#2350] Remove non-valid test
- * [#2181] Updated launchpad packaging
- * [#2333] Fix Pulseaudio Capture
- * [#2333] Use pulseaudio ADJUST_LATENCY flag and ALSA RT-SCHEDULING
- * [#2333] Pulseaudio Interpolate timing
- * [#2333] Change (again) Pulseaudio settings to fit logiteck usb hdw
- requirement
- * [#2333] Adjust pulseaudio fragment size to 4096 (max sflphone's
- frames per buffer)
- * [#2284] Remove recurrent compilation warning (g++ linker problem)
- * [#2333] Safer Audiostream parameters
- * [#2333] Fix alsa playback to reduce underrun
- * [#2333] Better audiostream parameters
- * [#2181] Updated version management
- * [#2333] Exclusive test in playback loop
- * [#2181] Updated build system
- * [#2333] Less underrun with these value
- * [#2333] Update playback audiostream parameters
- * [#2333] Lengthen the audio buffer reduce number of underrun in
- pulseaudio
- * [#2333] Add ALSA recovery functions for underrun (begin)
- * [#2333] Add pa_stream_trigger in pulse audio underrun callabck
- * [#2048] Reduce prebuffering in pulseaudio (which affect incomming
- calls' plbck)
- * [#2316] Do not display any icons to the right on the history tab
- * [#2333] Comment pa_stream_trigger in pulseaudio underrun
- * [#2333] Modify pulseaudio streams parameters
- * [#2318] Fix transfer tool button double signal
- * [#2181] Updated
- * [#2333] Fix ALSA ringtone
- * [#2333] Flush all main buffer before starting audio
- * [#2333] Open/Close Alsa thread between calls while there is no audio
- * [#2333] Add debug message and test condition on starting playback
- and capture
- * [#2181] Fixed gnome client makefile
- * [#2181] Updated
- * [#2308] Remove getTelephoneTone debug
- * [#2308] Change plughw for default in ALSA
- * [#2308] Oups, forgot to change function name in audiolayertest.cpp
- * [#2308] Cleanup in pulseaudio code (debug, function name)
- * [#2308] Fix pulseaudio stream closing assertion failure
- * [#2308] Moved pulseaudio mainloop locking from AudioStream
- disconnect stream
- * [2308] Fix latency at the beginning of a call, when playing DTMF and
- wehn starting tone
- * [#2181] Updated karmic
- * [#2317] [#2319] Fix address book toggle button contextual behaviour
- * [#2308] Stop stream when refusing a call
- * [#2308] Stop pulseaudio stream when peer hungup
- * [#2308] Fix tone and ringtone
- * [#2312] Display the STUN entry widget when opening the tab
- * [#2308] Implement two different callbacks for capture/playback in
- pulseaudio
- * [#2309] Open/close pulseaudio connections in startStream/stopStream
- * [2308] Leave pulseaudio stream running, do not cork/uncork them
- anymore
- * [#2295] Set gtk file chooser to None if nothing is set in
- configuration
- * [#1976] Add codec and conference documentation
- * [#2209] Fix recording in regard of resamling
- * [#2297] Update .gitignore
- * [#2297] Update translation files
- * [#2297] Add reference to our coding standards
- * [#2297] Remove old docbook code
- * [#2296] Reinit tls account settings after modification
- * [#2253] Add DcBlocker class to remove capture's dc offset
- * [#2034] Fixes for TLS transport to initialize
- * [#2284] Add silent build rule + client clean warnings
- * [#2274] Fix unserialize history items in cilent at startup
- * [#2274] Complete display name parsing and displaying
- * [#2274] Parse the Display Name in sip INVITE message
- * [#2050] Fix capture volume control in ALSA
- * [#1970] Volume controls disable when using pulseaudio
- * [#1970] Disable volume controls when using pulseaudio
- * [#2277] Fix direct ip2ip ZRTP enabling/disabling in ip2ip
- preferences
- * [#2181] Added launchpad debian files
- * [#2181] Added spec files for OSC
- * [#2274] Set display name for "Contact" sip header as the hostname
- * [#2181] Fixed daemon issues
- * [#2181] Fixed gnome client issues
- * [#1976] Remove warnings - need to fix the transfer
- * [#2006] Add init is_rec variable in ManagerImpl
- * [#2006] Update codec display on call selection
- * [#2006] Restore double click actions in history and contact calltree
- (GTK)
- * [#2176] use XDG_CACHE_HOME when initializing sfl.zid file
- * [#1976] Fix calltree switching from history
- * [#2209] (Re)Fix cache for zid
- * [#2209] Clean up debug messages
- * [#2209] Clean debug messages
- * [#2209] Fix trasnfering a call during a conference
- * [#2209] Speex decode must return the number of bytes
- * [#2209] Change frameSize speex 32kHz
- * [#2209] Fix speex codec framesize
- * [#2209] Reinit converterSamplingRate in RTP sessions
- * [#2209] Change speex ultra wide band framesize
- * [#1747] Add pixmap data
- * [#2252] Fix Receiving a server error 488 crashes the callee
- * [#2209] Fix iax low rate packate sending
- * [#2209] Clean up debug messages
- * [#2209] Add resampling changes for IAX
- * [#2209] Clean up resampling code
- * [#2209] Fix latency introduced by pulseaudio
- * [#2209] Fix initialization of mainbuffer's internal sampling rate
- * [#2176] Fix upsampling buffer size in audiolayer
- * [#2209] Add dynamic converter sampling rate in audiortp sessions
- * [#1747] Fixes runtime warnings
- * [#1747] Remove from repo
- * [#1747] register our icons to be used as stock icons
- * [#2209] Fix number of byte in alsa's write to speaker
- * [#2209] Fix putting non-resampled data in RTP's mainbuffer
- * [#2209] Add alsa resampler
- * [#2209] Add a samplerate converter in PulseLayer
- * [#2209] Add mainbuffer's internal sampling rate and flushall method
- * [#2176] Add mainbuffer stateInfo debug method
- * [#2209] Resampling is optimal using SRC_LINEAR not SRC_FASTEST
- * [#2176] Remove debug recordings
- * [#2176] Fix Holding a conference participant on new calls
- * [#2224] Add confID in callable object
- * [#2176] Fix putting onhold a call participating to a conference when
- pressing new call
- * [#2176] Reset auidio buffers when adding streams (rtp, audiolayer)
- * [#1976] Use xml to describe toolbars - Add a naviguation toolbar
- * [#2176] Remove conference default_id in joinParticipant
- * [#2176] Display error message in alsa's snd_pcm_avail_update call
- * [#2176] Alsa mic avail data debug
- * [#2176] Add some debug message for mic loss problem
- * [#2176] Flush mic ring buffer when offholding a call
- * [#2176] Reset ringbuffers' readpointer when adding main participant
- * [#2176] Fix getAvailData algorithm
- * [#2176] Reset ringbuffer's readpointer when adding a new participant
- to a conference
- * [#1744] Regex object renamed to Pattern. Previous attempt at
- providing
- * [#2176] Fix detach main participant problem when adding new one
- * [#1976] Use right domain to translate
- * [#1976] Add xml menu description
- * [#2176] Store a list of confernece participant in client
- * [#2176] Fix add participant, joinparticipant methods
- * [#2181] Do not install dbus-c++ headers + add return value
- * [#2176] Fix minor call handling instabilities
- * [#2174] Fix incoming IP call contact address
- * [#2211] Add test to protect NULL pointer
- * [#1163] Add Advanced account configuration section
- * [#2176] Add some usefull comments and debugging info
- * [#2176] Add conditions to display security icons in conference
- * [#2176] Fix detaching one participant while keeping communication to
- others
- * [#2176] Reenable userActive.svg in call tree
- * [#2176] Make user active blue (not red)
- * [#2176] Fix user active picture
- * [#2176] Fix "hidden" merge conflict in sipvoiplink
- * [#2176] Remove iax audio stream on peer hungup
- * [#2174] Multiple UDP transports functional (TESTED with 2 accounts
- and 3 calls)
- * [#2176] Fix fix audio stream binding in iax
- * [#2174] Create a default UDP transport + use tp selector for dialogs
- also
- * [#2176] Register iax audio stream in mainbuffer
- * [#2176] Fix getAudioCodecName in IAXvoipLink
- * [#2176] Fix iax account init
- * [#2176] Handle multiple account using the same sip transport
- * [#2165] Add .png files
- * [#2176] Small fixes concerning dtmf
- * [#2176] Fix make uninstall in codecs
- * [#2174] remove stund makefile generation
- * [#2176] Add conference lock
- * [#2174] Add transport selector for multiple accounts
- * [#2176] Change userActive picture from red to blue
- * [#2176] Fix security pixbuff in calltree
- * [#2176] Replace sfl.zid in .cache/sflphone instead of .sflphone
- * [#2176] Fix add call description
- * [#2176] Remove detach button from toolbar
- * [#2176] Fix calltree call description state and state code in
- conferences
- * [#2176] Fix pulse audio double free
- * [#2176] Fix conference selection
- * [#2174] Clean up - remove stun settings in client network
- configuration panel
- * [#2174] Remove voviva stun code
- * [#2174] Rsolve STUN with pjsip - DO NOT WORK
- * [#2165] Add user svg
- * [#2165] Debugging sip call failed
- * [#929] Link against uuid if installed
- * Oops
- * Fixed bugs related to libsexy (with GTK < 2.16)
- * [#929] Remove uuid-dev dependency in the core
- * [#2165] Debugging no negociated codecs at communicatio start
- * [#2165] Fix calltree bug (gtktreestore instead of gtkliststore)
- * [#2165] Fix several merge problems
- * Updated opensuse packaging script
- * [#1163] Add missing figures
- * [#1163] Update INSTALL file
- * [#2165] Fix IAX
- * [#2165] Add recordabe interface
- * [#2165] Finish recording refactoring for call (not for conference)
- * [#2165] Enable speaker recording for two different calls
- simultanously
- * [#2165] Implement call recording using the Recordable interface
- * [#2165] Add get and set to AudioLayer's audio recorder
- * [#2165] Add class recordable from which inherit call and conference
- * [#2006] Fix G722 and Speex 8khz codec conferencing
- * [#2006] add recording of audio buffers
- * [#1163] Add general settings section
- * [#1163] Fixes makefile error
- * [#2006] Fix some minor issues
- * [#2006] Drag a conference call on another conference call
- (difference conferences)
- * [#2006] Fix dragging a conference on itself
- * [#1744] Integrating some of the needed regular expression patterns
- in order
- * COmplete call features
- * [#1744] Added support for named subgroup in the Regex object. Also,
- new
- * [#1744] Adds thread safety features, compile() and setPattern()
- methods to the Regex class.
- * [#1744] Fix inconsistency in the finditer method from the last
- commit.
- * [#1744] Added regex pattern object built on top of libpcre. To be
- used
- * [#1744] Initial commit towards implementing RFC4568. Unimplemented
- in the
- * [#2157] Hide "security" and "advanced" tabs for IAX under account
- * [#1163] Add call features section
- * [#2006] Add joinConference capabilities
- * [#2006] Add dbus joinConference signal
- * [#2006] Drag a conference call onto a conference to add it
- * [#1163] Add addressbook section
- * [#2006] Drag a conference call onto a single call to create a
- conference
- * [#2006] Expand rows automatically
- * [#2006] Add minimal multiple conference handling
- * [#2006] Add atached/detached conference icons
- * [#2006] Add function processRemainingParticipant
- * [#2006] Deep refactoring, fix hangup bug
- * [#1163] Update documentation - Accounts part
- * [#1976] Integrate user doc to gnome client build system
- * [#2122] Remove double inclusion in dbus-c++/src/Makefile.am
- * Remove pjproject version number
- * [#2006] Fix peerHungup
- * [#1976] Make Yelp accessible from the GNOME client (need to install
- the sflphone.xml first)
- * [#2006] Fix multiconferencing hangup
- * [#2006] Fix hangup calls in a conference
- * [#2150] Make IAx2 reappear
- * [#2006] Fix detach participant on multiple call
- * [#2006] Can remove rining call from a conference
- * [#2006] Reinit confID when removing a participant
- * [#2006] Remove get isCurrentCAll in hangup/peerhungup (SipVoipLink)
- * [#2006] Fix refuse call
- * [#2006] Fix answerring incoming call
- * [#2006] Refactor conference's participant list
- * [#2101] Re-integrate test compilation in main build system
- * [#2101] Make the test directory compile
- * [#2136] Restore history functionality
- * [#2006] Fix binding main participant to himself
- * [#2006] Fix add current/incoming/onHold participant to an existing
- conference
- * [#2006] Fix add incoming calls to an already created conference
- * [#2006] Fix remove stream
- * [#2006] Fix detachParticipant/removeParticipant switchCall ids
- * [#2006] Fix adding a call in conference having state "CURRENT"
- * [#2006] Remove/add main participant from conferences
- * [#2006] Hold/unHold conference
- * [#2006] Detach a partcipant from drag n drop
- * [#2006] Hangup a conference
- * [#2006] Add hold/unhold conference dbus messages
- * [#2034] gtk-ui fix under the "basic" tab.
- * [#2006] Fix dragging calls on conference calls
- * [#2006] Fix detach participant from a conference
- * [#2034] Added default message is status bar under the account config
- dialog
- * [#2112] Fix a crashed caused when a non-md5 password was sent to
- pjsip.
- * [#2006] Detach participant by ID
- * [#2006] Fix addParticipant method in managerImpl to handle
- incoming/answered calls
- * [#2006] Add addParticipant method in managerimpl and related dbus
- messages
- * [#2111] Added the ability to configure zrtp on sip.sflphone.org from
- * [#2106] Fixed problem in the account assistant under gtk-ui. Also,
- assistant.c
- * [#2006] Fix dragging a conference call on another conference call
- (same conference)
- * [#1904] Small UI fix. Assistant was moved from "Call" to "Edit"
- menu.
- * [#1904] Fix a wrong label under gtk-ui.
- * [#2034] Renaming and source code splitting.
- * [#2034] Status bar added to account window to better reflect the
- registration
- * [#2006] Make calltree_remove_call recursive (for GtkTreeStore)
- * [#1110] Small gtk-UI fix in the account window (alignment).
- * [#2006] Fix remove conference, display children which are still
- active
- * [#2006] Recursive function call in calltree_update_call
- * [#2006] Add multilayered capabilities to calltree (GtkTreeStore)
- * [#2006] Implement remove conference in calltree
- * [#2034] Now useless as Direct Ip calls settings moved under
- Preferences.
- * [#2034] Edit/add buttons were set insensitive all the time under
- gtk-ui.
- * [#1887] Information about the state of the current SIP call is
- displayed
- * [#2006] Add call tree remove callback
- * [#2006] Fix create_conference function
- * [#2006] Update conference_added_cb to add new conference to the list
- * [#812] Added new tab under GTK-ui Preferences. Moving Direct Ip
- Calls from
- * [#2121] Disable temporarily test compilation
- * [#2006] Fix conferencelist to handle conference_obj_t instead of
- gchar
- * [#2006] Add conference_obj structure
- * [#2121] Update version
- * [#2006] Fix conference selection
- * [#2101] Use the new source tree to fetch the right object files
- * [#2006] Add conference in calltree
- * [#2006] Add Dbus signal conference added/removed/changed
- * [#2006] Add getConferenceDetails call on dbus
- * [#1904] Registration expire now appears as a spin box under gtk-ui.
- * [#812] Fixing a segmentation fault caused by a non-existing account
- ID
- * [#2006] Add getConfList method over dbus
- * [#2006] Add a conferencelist data structure in client-gnome
- * [#812] Defaults value are now sent if a non-existing account is
- requested
- * [#2006] Add sflphone action sflphone_join_participant
- * [#2006] Fix buffer read pointer problem deletion
- * [pjsip] Attempt at fixing via header incompatibility with
- Freeswitch.
- * [#1797] forget something
- * [#2006] Add call new state conferencing in deamon
- * [#2006] Remove addParticipant method for conference, use
- joinParticipant only
- * [#1163] Update INSTALL documentation
- * [#812] Msec/sec values were not taken into account.
- * [#1797] Make pjproject-1.4 compile
- * [#2006] Add Detach participant method
- * [#2006] Dragndrop fully functional with INCOMING and HOLD call
- * [#1797] Add pjproject-1.4
- * [#1797] Remove pjproject-1.0.3
- * [#2006] Get call state in conference related function
- * [#2006] Add joinParticipant (conference) method in ManagerImpl
- * [#2006] Add joinConference DBUS message
- * [#2006] Store the previously selected call_id on dragndrop
- * [#2006] Fix GValue pointer unref in selection callback
- * [#2006] Store dragged call_id
- * [#2006] Update drag_data_received_cb callback to manipulate CallIDs
- * [#2006] Add dragndrop signals
- * [#2006] Set calltree reordable
- * [#812] Adds the ability to create a TLS listener in case the user
- requests
- * [#812] Adds the ability to configure local/published address from
- * [#1883] Move switchCall in onHoldCall function
- * [#812] Deals with the published address/port problem when
- integrating TLS.
- * [#1883] Switch call id in managerimpl when peerHungUp
- * [#1883] Switch call id before hangup
- * [#1883] Add usefull and permanent debug info for conference
- cretion/deletion
- * [#812] Fix various segmentation faults related to Direct IP kind of
- calls.
- * [#1883] Fix deletion of std::map elements using iterators
- * [#2014] Add libzrtpcpp build dependency
- * [#1883] Still some for loop test ambiguity (while loop instead)
- * [#1883] Fix for loop initial test ambiguity (use while loop instead)
- * [#1883] We must discard data in urgent ring buffer if data is get in
- mainbuf
- * [#1883] Fix availForGet same id for ringbuffer and readpointer
- * [#812] Match "sips" as a Direct IP Call when the user enter a sip
- uri
- * [#812] Fix segmentation fault related to SIP URI creation.
- * [#812] Towards integrating multiple tls listeners at the same time.
- This
- * [#1883] Add debug messages in conference and fix mainbufferTest
- * [#812] gkt-ui fix. Private key must be fed as a filename and not as-
- is.
- * [#812] TLS integration within sipvoiplink and pjsip. Also,
- configure.ac
- * [#1883] Fix Alsa/Pulse mallocation
- * [#1883] Fix data corruption in AudioRtp's micData buffer
- * [#812] Full dbus integration for all the tls related options under
- gtk-ui.
- * [#1883] Fix memory leaks in audiortp session
- * [#1883] Fix mem leaks in audio rtp
- * [#812] Fix setAccountDetails where TLS_ENABLE was set to the value
- * [#812] Small gtk-ui fix.
- * [#811][#812] Small gtk-ui fix.
- * [#812] Introduced a mechanism for configuration files that makes
- possible
- * [#812] New dbus bindings added. Also, configuration compliance was
- enforced
- * [#1881] Remove default buffer from MainBuffer (update unit-tests)
- * [#1881] Add ring buffer read pointer tests
- * [#1883] Fix issues in ringbuffer reader pointers
- * [#2034] Implementing a new configuration dialogue for TLS transport
- settings
- * [#1883] Add some usefull debug and safety checks
- * [#2028] Notify the client with libnotify when the zrtp negotiation
- failed.
- * [#811] Harmless no to throw an exception, an makes the application
- less
- * [#2028] A minidialog is showed to the user under sflphone-client-
- gnome
- * Removed useless file.
- * Ignoring Makefile in src/widget
- * [#2027] Fix segmentation fault when showMessage callback is called
- after
- * [#2026] keyExchange was set to ZRTP instead of "1"
- * [#2024] Fix the wrong summary at the end of the assistant.
- * [#1883] Fix mnagerimpl conference map insertion
- * [#1883] Add Mutexes in MainBuffer
- * [#811] Gtk ui was not presenting the right information about zrtp
- for
- * [#2023] security icons were not installed in sflphone.
- * [#2021] Fix a mistake in the readme from sflphone-common that gives
- wrong
- * [#811] The current SRTP mode was not properly displayed for the
- IP2IP
- * [#1743] Re-implementation of the "automatically remove error dialogs
- [...]"
- * [#2017] [#2019] Fix the inability to dial a number and place a
- registered
- * [#811] Final re-integration of ZRTP support in the main branch from
- 0.9.6
- * [#1883] Fix map insertion methods
- * [#811] Combo box now is now set to the active key exchange method
- * [#811] ZRTP options now configurable back again from the Gtk UI.
- IP2IP
- * Updated hostname for git clone
- * [#1883] Add minimal functionalities to create a conference
- * [#811] re-integration of all the methods and signals on dbus.
- ManagerImpl
- * [#811] Got out of a precarious position were nothing would compile.
- * [#1976] Build documentation squeleton with docbook
- * [#1883] Add sflphone-client "addParticipant" button for conference
- * [#1994] Better organize the source directory structure. New
- subdirectories
- * [#1883] Add a simple Conference class
- * [#1882] Use static audio buffer in Pulse and ALSA layer (instead of
- malloc)
- * [#811] First commit toward re-integration and refactoring of ZRTP
- * [#1882] Flush RTP ring buffer before entering mainloop
- * [#1882] Fixed MainBuffer::UnBinCallID() in case there is no
- ringbuffer
- * [#1882] Test (and fixe) high level conference and mixing
- functionalities
- * [#1772] Apply patch to compile on fedora (sent by Marcin
- Zajączkowski <mszpak@wp.pl>)
- * [#1882] Update Bind, unBind call_id in MainBuffer
- * [#1959] This adds the ability to store password as an MD5 Hash in
- the
- * [#1538] Fixes rules compilation
- * [#1930][#1931] Fixed a mistake (again) related to index and
- credential count
- * [#1753] Remove ILBC from pjproject - Hacks in pjsip
- * [#1930][#1931] Credential was not selected properly using realm
- * [#1882] Finilize multiple reading pointer in RingBuffer
- * [#1538] Remove configure from autogen.sh to respect debian upstream
- authors policy
- * [#1773] Remove generated files from repo
- * [#1791] Use XDG_CACHE_HOME to save pid file
- * [#1791] Fixes path to save history
- * [#1791] Fix debian installation scripts
- * [#1930][#1931] Settings are now taken into account in the server.
- * [#1882] Add ringbuffer default ring buffer pointer in methods
- involving mStart
- * [#1882] Add default ringbuffer pointer
- * [#1882] Add RingBuffer multiple read pointer basic functionnalities
- * [#1882] Fix MainBuffer flushData unit test
- * [#1930][#1931] Ability to save and retreive the configuration from
- * [#1882] Added Multiple CallID mapping to MainBuffer
- * [#1791] Not much
- * [#1791] If XDG env variables are not null but empty, use default
- ones
- * [#1791] Make XDG_CONFIG_HOME writable
- * [#1930][#1931] Partial commit. Not working yet. Cannot delete
- account
- * [#1881] Fixed alsa capture latency problem
- * [#1881] Fixed Alsa capture temporarily
- * [#1930] [#1931] Partial unbroken commit providing the ability to
- * [#1881] MainBuffer implemented in AudioLayer/AudioRTP
- * [#1881] Add discard and flush unit-tests
- * [#1881] Add discard and flush functionnalites to MainRingBuffer
- * [#1881] Add availForGet in MainBuffer
- * [#1881] Add availForPut function to MainBuffer
- * [#1880] Remove AudioRTP* pointer from SipVoIP (reapered while
- merging master)
- * [#1881] Add a map between call id and coresponding ring buffer
- * [#1855] Refresh pot file and upload on Launchpad
- * [#1881] MainBuffe now robust to false ids on getData and putData
- * [#1881] Fix big big big memory leak
- * [#1881] Add getData and putData to mainBuffer
- * [#1881] Unit-test basic ring buffer functionnaities
- * [#1881] Add class MainBuffer and basic buffer creation unit-tests
- * [#1880] Fix call transfer (step2) issues
- * [#1880] Moved AudioRtp* pointer from SIPVoIPLink to SIPCall class
- * [#1791] Add postinst script to keep user data when migrating
- config/history file
- * [#1797] Make pjsip compile
- * [#1777] Code indentation
- * [#1791] Use XDG_DATA_HOME and XDG_CONFIG_HOME for sflphonedrc and
- history + unit tests
- * [#1746] Useless space does not appear anymore when volume sliders
- and
- * [#1643] GtkCheckMenuItem is used instead of icons for elements in
- the
- * [#1110] [#1668] STUN parameters are now located in the preferences,
- under
-
- -- Julien Bonjean <julien.bonjean@savoirfairelinux.com> Fri, 06 Nov 2009 11:20:01 -0500
-
-sflphone (0.9.6-SYSTEM) SYSTEM; urgency=low
-
- ** 0.9.6 **
-
- * Documentation on echo test
- * [redmine_down] codec names not displayed in total
- * [redmine_down] crash when hanging up a dialing call because tries to
- add it to history whereas no starttime
- * [#1927] alternate every time screen changed to call history
- * [#1886] clean code
- * [#1886] debug messages when loading history removed
- * [redmine_down] sflphone-kde icons
- * [#1855] Update language files
- * [#1502] Update version number
- * [redmine_down] setHistory at close
- * [#redmine_down] Handle PJ_DECLINE_SC as failure
- * [#1923] Fix segmentation fault when adding a new account
- * [#1923] Check on iterator before setting the config
- * [#1904] Added mnemonic to tabs in sflphone.
- * [#1905] The daemon was not sending the currentSelectedCodec signal
- on dbus when answering a call.
- * [#1922] Default values set to all account details
- * [#1886] Spinbox reg expire enables apply, and address book is not
- visible when disabled
- * [#1905] Bug fix for segmentation fault caused by an empty string,
- * [#1910] Warnings in test directory
- * [#1919] Error fixed
- * [#1855] Update russian translation - Hussein Abdallah
- * [#1910] Remove files
- * [#1919] fixed
- * [#1777] Code indentation
- * [#1918] fixed
- * [#1917] fixed
- * [#1910] Remove warnings compilation in src
- * [#1886] removed AccountListModel in configskeleton
- * [#1914]
- * [#1911] check previous and new port
- * [#1910] Remove compilation warnings in src/dbus and src/history
- * [#1910] Remove compilation warnings in src/audio
- * [1855] Update german translation - Sven Werlen
- * [#1909] removed
- * [#1906] Done
- * [#1904] The registration expire value is now configurable from the
- * Cleaned up debug messages.
- * [#1886] separated initCallItem in two functions
- * [#1886] reversed error in commit
- * [#1886] clean debug
- * [#1886] changed Name of classes and files
- * [#1886] clean
- * [#1870] In call_state_cb (dbus.c:126), _time_stop was overridden by
- the actual time.
- * [#1884] Added some new gpg flags to prevent tty warnings
- * [#1886] Clean audio config dialog
- * [#1886] No more compile warnings. + 1 comm
- * [#1872] Check if the user input is smaller than PJ_MAX_HOSTNAME.
- * [#1886]
- * [#1785] Fixed build when no new commit
- * [#1852] If chosen by the user, the hostname can now be solved and
- used
- * [#1871] * and # inverted back
- * [#1869] Conditional compilation that checks if
- * [#1309] removed test in main
- * [#1425] Put actions in SFLPhone window class instead of ui view,
- made a separate toolbar for screens.
-
- -- SFLphone Automatic Build System <team@sflphone.org> Mon, 27 Jul 2009 09:53:19 -0400
-
-sflphone (0.9.6~rc2-SYSTEM) SYSTEM; urgency=low
-
- ** 0.9.6~rc2 **
-
- * [#1755] Remove generated file
- * [#1753] restore ilbc ...
- * [#1866] Methods getSipPort and setSipPort now have an effect on the
- * [#1753] make pjsip compile without ilbc. Use ./autogen.sh --disable-
- ilbc-codec
- * [#1855] Fix error in russian translation
- * [#1805] Remove the old flawed signal mechanism which was failing in
- * [#1855] Refresh translation
- * Spanish translation finished + po README files updated + echo's in
- copy-in-clients
- * [#1850] Yun made the chinese HK-CN translation
- * [#1848] Fix transfer interface bug
- * [#1862] At install, kde client installs only french translation file
- * [#1841] A new fallback mechanism was added to the internal resolver
- in PJSIP.
- * Started AccountList model/view
- * [#1855] Remove po subdir in Makefile.am
- * [#1855] Fix typo error in sflphone
- * [#1855] Do not generate Makefile in sflphone-common/po
- * [#1855] Copy translation files into both clients dirs
- * [#1855] Remove po dir from sflphone-common
- * Comments added
- * [#1860] mailbox->voicemail...
- * make scripts executable
- * [#1855] French translation
- * [#1855] Chinese zh_HK partially filled...
- * [#1859] An unnamed pipe monitored by poll() was added. When we want
- to
- * [#1855] Sven completed the first part of the german translation
- * [#1855] Cantonese manually filled for already translated, almost
- equal strings
- * [#1855] Merge russian translation
- * [#1855] Spanish manually filled for already translated, almost equal
- strings
- * [#1855] Update german translation in ./lang/de
- * [#1858] This problem was fixed by removing a useless line in
- * [#1855] merged existing translations in lang/ sflphone.po's
- * [#1842] [#1843] An attempt at improving the expected behaviour that
- can't
- * [#1855] added po folder in gnome client and scripts for copying from
- common lang folder to clients
- * [#1853] Edit before call does nothing on call history
- * Put most language entries possible in common. From 300 to 250
- entries. Stays underscores problem. Scripts for copy in clients.
- * commit to merge master
- * [#1825] Changed "Bad authentification" to "Authentication Failed".
- * common po files
- * [#1753] Remove ILBC from pjproject
-
- -- SFLphone Automatic Build System <team@sflphone.org> Fri, 17 Jul 2009 19:12:58 -0400
-
-sflphone (0.9.6~rc1-SYSTEM) SYSTEM; urgency=low
-
- ** 0.9.6~rc1 **
-
- * Update some version number
- * [#1792] Creates .sflphone directory with permission 600. Also,
- "chmod 600" after
- * [#1810] GUI is now notified that the call failed. Also, a segfault
- was
- * [#1816] Address book search disabled when disabled address book and
- enabled it back plus button stays triggered
- * codeclistmodel + asynchronous loading of address book +
- enable/disable address book
- * [#1810] Now checking SDP answer after 200 OK. Still need to
- implement full
- * [#1794] Can't use the interface during a call
- * Updated translation files
- * Russian translation integrated
- * Codec list model/view started.
- * [#1807] Add configure.ac in pjproject-1.0.3
- * [#1787] closeRtpSession added in some places where it should have
- been
- * Use Item class for contacts and accounts
- * Comments + clean code
- * [#1794] Improved debug messages
- * [#1805] Replaced the old and unreliable mecanism that was was
- waiting for
- * [#1794] Can't use the interface during a call
- * [#1787] For those cases where no registered SIP account is
- configured
- * [#1797] Make pjsip compile
- * [#1787] Minor changes. Removed useless commented line. Changed order
- of
- * [#1777] Code indentation
- * [#1797] Update package generation with new pjsip version
- * [#1798] Does not hang up when the call is building up
- * [#1797] Update .gitignore with new pjsip version
- * [#1797] Remove generated files from repo
- * [#1797] Main build system now uses pjproject-1.0.3
- * [#1797] Add pjproject-1.0.3
- * [#1797] Remove pjproject-1.0.2
- * [#1796] Computing time optimization (samplerate conversion)
- * [#1787] _audiortp->start() moved away from offhold(),
- SIPCallAnswered()
- * [#1312] Added new states for calls initialized by other clients
- * [#1795] Crashes when adding a new account, checking it and applying
- * [#1782] Missing icons
- * [#1793] KDE client compilation problem
- * Fake ringtone files can no longer be set.
- * indentation
- * [#1312] Able to fetch to differentiate incoming/ringing call state
- * [#1784] Use DESTDIR variable in po Makefile - fix language file
- installation
- * [#1785] Fixed typo
- * [#1785] Fixed changelog update
- * [#1759] ./autogen.sh --prefix=/usr --with-debug to use optimization
- level 0
- * [#1773] Changed snapshot naming convention
- * [#1773] Removed gpg agent use, added repository cache cleaning
- * [#1759] Use optimization level 0 for repository, 2 for packages
- * [#1777] Code indentation/formatting
- * Translated new features in french
- * [#1785] Added missing changelog entry
- * [#1781] Window title is SFLPhone
- * [#1777] Add code indentation/formatting in the buil system
- * [#1774] Can't set voicemail number in KDE account creation wizard
- * [#1775] Can't modify account information for account created with
- the wizard
- * [#1771] Add a "Default" button in context menu to disable chosen
- prior account
- * [#1705]
- * [#1224] Remove generated file from the repo
- * [#1224] Remove generated file from the repo
- * [#1762] distclean target should remove kconfig generated files
- (settings.h, settings.cpp). Rename them?
- * [#1761] clear history button should really clear history
- * Dialpad works.
- * Implemented Dialpad widget instead of building it in main view.
- * Removed last occurence of the old config dialog, that made the build
- crash.
- * [#1755] Do not consider G722 as a dynamic payload elsewhere than in
- RTP layer
- * [#1753] Remove ilbc Makefile generation
- * [#1756] Implement a kde configuration dialog with kconfig xt and
- kconfigdialog class
- * [#1755] fix audiocodec folder parsing problem
- * [#1450] Reinit timestamp comparison in RTP, create session in
- newOutgoingCall
- * [#1753] Remove milenage third party code from pjsip
- * New Config Dialog integrated in GUI.(without codecs)
- * [#1753] Remove ILBC codec
- * kconfig started, tr2i18n -> i18n, icons folder, accountList changed
- * [#1705] Fixed Audio RTP thread creation/start
- * [#1714] Fix codec negociation result handling
- * [#1678] Fix audiortp payload setting
- * [#1678] Put bac putData method in rtp
- * [#1669] gtk_file_chooser_get_filename() support UTF-8 by default
- * [#1735] Add conditions to sdp update call if call declined
- * [#1737] substr of recordings destination folder to remove "file://"
- should be done in client rather than in daemon
- * [#1731] Enlarge audio stream buffer size
- * [#1714] Missing true
- * [#1317] Fixed Mandriva timeout
- * [#1317] Changed tag convention
- * [#1317] Cleaned git-dch
-
- -- SFLphone Automatic Build System <team@sflphone.org> Fri, 10 Jul 2009 15:50:26 -0400
-
-sflphone (0.9.6~beta-SYSTEM) SYSTEM; urgency=low
-
- ** 0.9.6~beta **
-
- * spec files for mandriva and opensuse updated with buildrequires
- libqt4-dev >=4.3
- * [#1700] Cannot build on ubuntu 8.10 and a few other distribs
- * [#1502] Update version number where applicable
- * [#1642] Update client icons
- * [#1450] Clean up useless debug and comments in sipvoiplink and
- audiortp
- * [#1450] Remove Semaphore object in AudioRtp thread deletion
- * [#1450] Audio RTP init now synchronized with Sip/SDP
- * [#1693] kde client crashes when changing codecs order/activation
- * [#1450] Deep refactoring of audiortp
- * [#1450] setRtpSessionRemoteIp
- * [#1689] getCallList at start
- * [#1224] Change path in package files
- * [#1450] Audio RTP initialized only once, payload and remote ip set
- at runtime
- * [#1450] Add setRtpSessionMedia and setRtpSessionRemoteIp address
- * [#1642] Make GNOME GUI fresher and younger ;)
- * [#1686] Status bar displaying used account
- * added sflphone-kde icon so that it compiles
- * [#1659] Ending a call causes the daemon to crash
- * corrected introspection XMLs, po files...
- * [#1211] g722 media descriptor in codecDescriptor
- * [#1310] Install sflphoned in $(prefix)/lib/sflphone
- * [#1502] Do not install test binaries and dbus utilitaries
- * [#1224] hack for pjsip build system!
- * [#1224] Remove pjsip binaries from repo
- * [#1224] Upgrade to pjsip 1.0.2
- * [#1658] About SFLphone (bugs)
- * [#1658] About SFLphone
- * [#1660] Displaying all dialed numbers in a call
- * Tested status bar.
- * [#790] Optimize pulse audio streams parameters
- * [#1678] Some usefull debug messages for mutex/semaphore deadlock
- problem
- * [#1669] Add/remove some usefull/unusefull debug
- * [#1665] Fix latency related to pulse audio stream openning/closing
- * [#1457] Make the menus and panels accessible in french
- * [#1457] Improve broken keyboard accessibility in menus and conf
- panels
- * [#961] Instanciate only once the searchbar icons
- * [#961] Restore transfer fonction
- * [#961] Filter on the history type OK
- * [#961] Fix compilation problems on hardy/intrepid
- * [#1157] Commit missing files
- * [#790] Reduce number of start/stop streams call on pulse audio
- * [#1639] kde client crashes when no account registered
- * [#1620] Fix the searchbar
- * [#1620] Get back caltree as it was during gtkcritical area
- * [#1620] Add history filter reinit function
- * [#1335] Add a missing label in address book preferences
- * [#1561] Update russian translation - Hussein Abdallah
- * [#1605] Fix edit menu french translation
- * [#961] Enable to search in the history according to the call type
- * [#1449] Searchbar does not work anymore
- * [#961] Add popup menu on the entry primary icon for history
- * [#1317] Fixed KDE client package dependency
- * [#936] speex 32 khz integration completed
- * [#936] Use 320 frame size
- * [#936] Test using a frame size at 320 smpls
- * [#1214] Enable / Disable history
- * [#1607] Fix compilation problem for ubuntu 8.10 (libsexy)
- * [#1313] Implement processDataEncode processDataDecode in audiortp
- * [#1613] codec list order can't be set
- * Better handling of localisation + added languages + corrected
- warnings + begginning of new config dialog with kconfig + 14px
- account leds
- * [#1214] Save and load history according to the limit timestamp +
- unit tests
- * [1609] Fix call number copy/paste feature
- * [1607] Restore clear action icon in searchbar
- * [#936] Try to decode using 1280 samples
- * [#936] Add some debug
- * [#936] Add .cpp file
- * [#936] Oops Forgot speex 32 khz
- * [#1214] Add configuration panel for history + D-Bus calls
- * [#1313] Test rtp thread function, frame size, nbbytes, resampling
- * [#790] Flush audio data before closing audio streams
- * [#1214] History displays local time
- * [#1214] Skip empty field on display
- * [#1214] Associate an account to an history entry
- * [#1342] Get addressbook options sensitive/non-sensitive
- * [#1211] Clean up and comments
- * [#1211] Get back to 20 ms framesize
- * [#1211] Use sendImmediate instead of putData in RTP
- * [#1211] Fix nb byte available in RTP
- * [#1211] Clear condition on maxNbSamples in RTP
- * [#1211] Fix max byte available in RTP session
- * [#1211] G722: Use 160 samples per frame instead of 320
- * [#1211] Test using a dynamic payload
- * [#1211] Test using a dynamic payload type
- * [#1211] Rename size variable (nb_samples, nb_bytes)
- * [#1211] Test g722 ip-to-ip sending twice the data lenth
- * [#1211] Test g722 ip-to-ip
- * [#1214] Do not select an history item by default at startup
- * [#1214] Remove some compilation warnings
- * [#1214] Handle empty field - remove g_print
- * [#1214] Add each history item only once
- * [#1214] Handle call timestamps properlier
- * [#1214] Do not need timestamp files anymore
- * [#1214] Use the saved date for history entry
- * Clean up
- * [#1214] Client doesn't crash if the D-Bus call fails
- * [#1214] Client is able to save its history - still some glitches
- * [#1211] Forgot 16000 for g722
- * [#1211] G722 initialization
- * [#1214] Save name/number, successfully load the history if no fields
- are empty
- * [#1499] Fixed destination directory bug
- * [#1214] Restore all the functionalities; peer name/number way more
- easy to handle !!
- * [#1214] Add callable_object instead of call_t, refactoring
- * [#1211] Test with polycom soundstation 16000
- * [#1211] Remove C like inline function in g722 codec
- * [#1342] Finalize gnome client preference window formating
- * [#1214] Retrieve the history when the gnome client startsup
- * [#1306] Implement localization for KDE client
- * [#1593] enable accounts apply button when account checked/unchecked
- * [#1214] Implement the dbus calls on server side
- * [#1214] Add serialized/unserialized functions to pass data on DBUS
- * [#1342] Formating gnome client configuration windows
- * [#1214] Save sucessfully a map of history items
- * [#1499] Removed multiple jobs compilation for KDE client (2)
- * [#1214] Load history from file into memory, add unit tests
- * [#1534] Throws a length_error exception in case URL exceeds
- std::string max_size
- * [#1499] Removed multiple jobs compilation for KDE client
- * [#1565] make account leds smaller
- * [1430] Fix dbus debug
- * [#1562] crashes when trying to change item of a call of state "OVER"
- * [#1116] Fix compilation bug
- * [#1317] Added mandriva and opensuse-11 64 bits
- * [#1108] Add messges in main window concerning transfer success
- failure
- * [#1116] Fix compilation problems
- * [#1211] g722 Makefile
- * [#1108] Client side transferFailed/trasferSucceded signals handling
- * [#1211] G722 mostly completed,
- * [#1555] make bigger toolbar (24x24)
- * [#1551] remove default mailbox number in wizard and disable mailbox
- button when first account doesn't have mailbox number
- * [#1342] Re-add sflphone manpages
- * [#1116] Fix compilation on non-jaunty distros
- * [#1317] Fixed opensuse startup sleep
- * [#1108] Add a signal in the client to notify successful or failed
- transfer
- * [#1108] Dbus signals concerning call transfer success/failure
- * [#1317] Added opensuse to automatic build system
- * [#1223] Fix manpages bug
- * [#1060] german translation glitch
- * Clean up some gnome client warnings
- * [#1547] replace ugly account leds by beautiful icons
- * [#1548] add close button that hides windowand just hide on clicking
- the cross
- * [#1549] put introspec XMLs in the client's source
- * [#1312] Implement getCallList D-BUS method
- * [#1116] Clear text in history and contacts
- * [#1499] KDE integration
- * [#1469] Modify header linkers in dbus-c++'s Makefile.am's
- * [#1469] Remove examples folder from dbus-c++
- * [#1214] History integration in build system; unit test squeleton
- * [#1317] Cleaning
- * [#1469] Remove configure stuff in dbus-c++
- * [#1469] Add unofficial mainline dbus-c++
- * [#1469] Remove dbus-c++ from freedesktop
- * [#1430] Bring account changed signal/callback back to normal
- * [#1060] Update german translation - Sven Werlen
- * [#1430] Add marshaller one string define
- * [#1430] Send account change signal broadcast using account id
- * [#1430] Remove condition on setRegistrationState, cause stun to
- crash
- * [#1317] Centralized version handling
- * [#1317] Fixed version number on sfl-git-dch
- * [#1317] Refactoring for new distributions
- * [#1215] Fix account order at startup if latency
- * [#1088] Restore sip dns srv
- * [#1214] Add squeleton for history manager
- * [#1430] Add accout id to accout changed method
- * [#1430] No connectionStatusNotification (account changed) if no
- changes
- * [#1538] Add COPYING file
- * [#1430] Add audio rtp thread tests
- * [#1317] Changed version detection
- * [#1538] Document license in libs/stund
- * [#1317] Added version files
- * [#1538] Apply François patches - debian packages
- * [#1317] Updated spec files
- * add files
- * [#1538] Apply François patches - debian packages
- * [#1535] Change program file structure (directory src...)
- * [#1317] Updated build system scripts
- * [#1317] Cleaning
- * [#1317] Copied introspect files to gnome client
- * [#1317] Added opensuse to build-system : first-shot
- * [#1317] Remove spec files from configure
- * [#1317] Added missing prefix
- * removed debug for daemon account fix
- * [#1430] Add a connection reference which most likely belong to
- libdbus
- * [#1430] Use shared connection instead of private
- * make daemon find the account, added userMatch
- * Clean code, add comments...
- * [#1317] Fixed packaging rules
- * [#1317] Updated autogen
- * Updated autogen.sh for pjsip
- * [#1526] Set accounts order
- * [#1317] Fixed pjsip lib dirs
- * [#1317] Updated debian packaging for new pjsip configuration script
- * [#1317] Switch to autogenerated guess and sub files
- * [#1317] Updated pjsip inclusion in build system
- * [#1317] Replaced pjsip guess and sub files
- * [#1317] Fixed compilation issues on opensuse 11
- * [#1505] account list seem to crash the application when clicking
- Apply very fast...
- * [#1456] Add a flag to be replaced in the control files
- * [#1456] Added version dependancy handling
- * put account alias in AccountWidgetItem rather than in the item with
- " " before.
- * [#1034] The KDE client should start sflphoned if it is not started
- * [#1500] Handle options for notifications and display on incoming
- call.
- * [#1443] Client should not crash when receive an unexpected
- stateChanged signal
- * [#1403] Do not stop the notification anymore
- * [#1456] Added version dependancy handling
- * [#1426] Daemon crashes when get alsa plugin
- * [#1422] Improved error messages
- * commit for merge
- * [#1424] Change logo in tray icon and put a different one when
- incoming call
- * [#1425] first part done, window title...
- * [#1413] add manpages creating and installing in build system
- * [#1417] The client should start the account creation wizard if
- started for the first time (if config file doesn't exist)
- * [#1421] Make volume bars horizontal when dialpad is hidden.
- * Changed main window title and fixed a mistake in sflphone_const.h
- * [#1412] make debian package building work
- * changelog changed.
- * Changed addAccount method in gnome client.
- * Debian and man folders added.
- * [#1388] Change project name from sflphone_kde to sflphone-client-kde
- * Better handle of kabc check.
- * [#1351] Automatic generation of dbus interfaces in makefile
- generated by cmake
- * [#1307] Implement "edit before call" in history and address book.
- * [#1344] change action_call label in call history from "call" to
- "call back".
- * [#1308] Implement Hook feature in kde client
- * Improved build system.
- * #1219 : Add address book configuration page
- * Better handling of registration to the daemon.
- * #1039 : Add tray icon in kde.
- * Issue no 1216 : Double click on item in history or address book
- causes call.
- * display peer name in call list and call history when called from
- address book.
- * Address book functionnal with photo displayed.
- * Help menu kde available but actions disappeared. All fonctions in
- view.
- * Address book functionnal but ugly and making its own sort in the
- complete address book.
- * Account choice on right click, clean out includes, page address
- book, fixed bugs...
- * Wizard, double click, context menu...
- * Removed sflphone_kde.kdevelop.filelist
- * Added account creation wizard and translated interface in english.
- * Transfer functionnal but ugly.
- * transfer not functionnal
- * Bug fixed : unholding (UNHOLD_CURRENT, UNHOLD_RECORD)
- * Commit functional for push. With install.sh
- * Before merge.
- * Problem with enable accounts. Account display increased.
- * Functional with codec order working , playDTMF.
- * Commit functional.
- * sflphone_kde/build added in .gitignore.
- * complete commit for checkout previous.
- * Commit before checkout previous version to check the display
- bug(little font everywhere...)
- * Functionnal client. Rest : history icons, config icons and
- functionalities
- * commit before merge asavard for isRecording.
- * Call and Automate fusion done and seems to work.
- * Commiting before putting Automate class in Call class.
- * Functionnal main window without recording, history, voicemail, kio
- widgets.
- * client kde avec kdevelop.
- * Config Dialog almost finished.
- * Base of QT client
-
- -- SFLphone Automatic Build System <team@sflphone.org> Tue, 23 Jun 2009 11:13:42 -0400
-
-sflphone (0.9.5-SYSTEM) SYSTEM; urgency=low
-
- ** 0.9.5 release **
-
- * [#1060] FIx bug in chinese translation
- * [#1313] git add rtpTest.cpp rtpTest.h
- * [#1313] Add init/close rtp tests
- * [#1313] Basic instanciation of the rtp layer
- * [#1449] Gtk-Critical concerning history filters and new calls
- * [#1400] Make the match with the hostname instead of username
- * [#1324] Change status bar label for "Using %s (%s)"
- * [#1403] Icon size: 60x60 px
- * [#1403] Do not remove notification, improve icon quality
- * [#1403] Add smaller icon for gnome notifications
- * [#1403] Prevent crash when hangup && no notification
- * [#1403] Remove all actions on notifications; code refactoring
- * [#1451] Use stun.sflphone.org as default STUN server
- * [#1060] New po files - need to be translated
- * [#1060] Update french translation - Rebuild template file
- * [#1456] Add a flag to be replaced in the control files
- * [#1454] Make cppunit optional; remove from build deps in control
- files
- * [#1401] Add libexpat1-dev dependency in control files
- * [#1448] Take off these ugly debug messages
- * [#1448] fixed getTelephoneTone and getTelephoneFile() called
- repeatedly
- * [#1406] add liblog4c-dev in build-depends
- * [#1409] Restore .desktop icon
-
- -- SFLphone Automatic Build System <team@sflphone.org> Mon, 25 May 2009 11:34:48 -0400
-
-sflphone (0.9.5-SYSTEM~rc2) SYSTEM; urgency=low
-
- ** 0.9.5 rc2 **
-
- * [#1422] Improved error message
- * [#1402] Fix pjsip build
- * [#1404] Clear GTK-Critical Bug at client startup
- * [#1422] Added automatic VM shutdown when building on more than one
- VM
- * [#1422] Fixed some issues with new changelog generation script
- * [#1422] Moved distribution update to specific file
- * [#1422] Dropped git-dch, replace by home made implementation
- * [#1402] Fix pjsip build
- * [#1404] Clear GTK-Critical Bug at client startup
- * Changes for name based dbus connection
- * Clean changelogs
- * [#1343] Gnome: Implement a callback system to handle focus on
- different widgets
- * Debus Session
- * Refactoring Python code, PEP8
- * [#1430] Get back dbus_g_proxy_new_for_name
- * [#1430] Get back DBUS_BUS_SESSION type
- * [#1430] Dbus fixed owner message binding
- * Second test with DBUS owner
- * [#1404] Gnome -> Preferences -> Hooks
- * [#1404] Gnome -> Preferences -> Recordings
- * [#1404] Call History
- * [#1404] Gnome -> Preferences -> Address Book
- * [#1404] IF the first notification option disable the second
- notification
- * Dbus with fixed owner does not automatically start the deamon
- * Add codec debug tests in pysflphone
- * [#1407] Some print info
- * [#1407] Add a scenario to pick_up action
- * Test client dbus connection to a fixed owner
- * Add python dbus test suite
- * [#1161] Modified version handling in build system
- * [#1314] Test pulse audio and audio streams connect and disconnect
- * [#1402] Add info message after configure
- * [#1402] Build the daemon with the local pjsip library (vs the
- installed one)
- * [#1009] Fix Codec Sampling Rate set to zeros
- * [#1314] Add mutex to pulse layer audio streams
- * [#1314] Refactoring pulseaudio stream to test connect disconnect
- * [#1314] Refactoring of pulselayer to test conect/disconnect
- * Add debug messages in debus calls concerning account
- * [#1314] Add some return values to audio init functions
- * [#1406] add liblog4c-dev in build-depends
- * [#1409] Restore .desktop icon
- * Bug #1405: Fix strings as requested.
- * Bug #1404: Fix strings in preferences panel.
-
- -- SFLphone Automatic Build System <team@sflphone.org> Tue, 19 May 2009 12:08:18 -0400
-
-sflphone (0.9.5-0ubuntu1~rc1) SYSTEM; urgency=low
-
- [ SFLphone Project ]
- * [#1262] Updated changelogs for version 0.9.5-0ubuntu1 Snapshot 2009-
- 05-05
-
- [ Emmanuel Milou ]
- * Add some python CLI client code; not really functional
- * [#1108] Fix peerHungup method for IP to IP call
-
- [ Alexandre Savard ]
- * [#1108] Correct setting of SIP contact for direct IP call
- * [#1108] SIP user agent handles incoming REFER
-
- [ Emmanuel Milou ]
- * Remove website from repository
- * Update translation
-
- [ Alexandre Savard ]
- * Sflphone icon's tooltip changed for "configured" instead of
- "registered"
-
- [ Emmanuel Milou ]
- * Update translation
-
- [ Sflphone Project ]
-
- -- Sflphone Project <sflphone@mtl.savoirfairelinux.net> Tue, 05 May 2009 19:16:13 -0400
-
-sflphone (0.9.5-0ubuntu1~beta) SYSTEM; urgency=low
-
- [ Julien Bonjean ]
- * Updated Eclipse stuff
- * Improved addressbook config window
- * Added sflphone Eclipse stuff
- * Implemented addressbook list server side
- * Moved dbus stuff in dbus directory
- * Updated addressbook configuration
-
- [ Emmanuel Milou ]
- * Remove unuseful installation scripts. Use apt-get build-dep sflphone
- instead
- * fix bug #1090
-
- [ Alexandre Savard ]
- * defining speex 16khz
-
- [ Emmanuel Milou ]
- * Remove unuseful file from build system
- * Start dns srv resolver
-
- [ Alexandre Savard ]
- * Basic ogg/vorbis initialization
-
- [ Emmanuel Milou ]
- * Handle incoming IP-to-IP invite correctly
-
- [ Alexandre Savard ]
- * speex wideband 16000
-
- [ Emmanuel Milou ]
- * Better handling of incoming IP to IP call
- * DNS SRV resolution functional
- * Implement IAX2 incoming URL
- * Allow user to make IP call without any accounts configured
- * Add a contextual menu to edit a number from the contacts tab
- * Add comments, tooltip and new button to the contextual menu
- * add delete event, migrate to GTK 2.16 for sexy icons
- * Resolve ticket #1118
- * Update suse spec file
- * Add phone number cleanup functions, unit tests and panel
- configuration
- * Add pertinent test that fails
- * fix dependencies for suse package
- * Add contextual edit menu in history - #1120
-
- [ Alexandre Savard ]
- * Temporary comit: make speex wideband (16 khz)
- * Temporary: shared object for speex narrow band
- * Temporary: speex narrowband and wideband coexist
-
- [ Julien Bonjean ]
- * Fixed bug when no book selected
- * Fixed addressbook related compilation warnings
- * Fixed GTK client remaining compilation warnings
- * Fixed segfault when book removed since last sflphone run
- * Fixed bug when book is unreachable (ldap error)
-
- [ Alexandre Savard ]
- * Fix codec list in audio config window
- * Active/inactive speex codec by payload
-
- [ Julien Bonjean ]
- * Updated gitignore
- * Added some comments
-
- [ Emmanuel Milou ]
- * Add callto: handler script for browsers and al.
- * Integrate test compilation in the daemon build-system
-
- [ Julien Bonjean ]
- * Fixed g_object_unref warning for pixbuf
- * Cleaned too verbose output
- * Fixed toolbar update warning
- * Added support for asynchornous books open (first shot)
-
- [ Emmanuel Milou ]
- * Add a DBus call to fetch the call details from a call ID - Ticket
- #928
-
- [ Julien Bonjean ]
- * Improved async open books
- * Fixed bug #1139
-
- [ Emmanuel Milou ]
- * Add a way to save account order
- * commit missing files
-
- [ Julien Bonjean ]
- * Introduced log4c (ticket #1162)
-
- [ Emmanuel Milou ]
- * Load/save account order functionnal - ticket #813
-
- [ Alexandre Savard ]
- * Add CELT codec (#1143)
- * Make celt frame size 256 (*1143)
-
- [ Julien Bonjean ]
- * Switched everything to log4c (ticket #1162)
- * Updated eclipse settings
-
- [ Emmanuel Milou ]
- * Restore adding account - ticket #1172
- * Add liblog4c dependecy - ticket #1179
-
- [ Alexandre Savard ]
- * Double maxAvailByte for frame size in rtp (#1143)
-
- [ Emmanuel Milou ]
- * Add User-Agent SIP header - Ticket #1173
-
- [ Julien Bonjean ]
- * Fixed autoresize issue (#708)
-
- [ Emmanuel Milou ]
- * Remove libcppuint dependency for the debian packages
- * Look for libsexy only if gtk version < 2.16 - Ticket #1116
- * Remove libsexy dependency for jaunty. ticket #1116
-
- [ Julien Bonjean ]
- * Introduced unit tests (#1146)
- * Updated gitignore
- * Fixed Makefile (#1146)
-
- [ Emmanuel Milou ]
- * [TICKET #1112] Add a test on the voice buffer to send through iax
- packets
- * Remove doublon in dependencies
- * Remove warnings from the client test framework
- * Update version number to 0.9.5~beta
- * Update build-package script
- * Add check dependency in build-deps control file field
- * Create debian files for the new sflphone
- * [TICKET #1212] Add Replaces field in control files
- * [TICKET #1212] Fix manpages installation path
- * [TICKET #1212] Add maintainer scripts to create alternatives
- * [#1212] Update the manpages generation - edit preinst maintainer
- script
- * [#1212] Fix reference error in manpage
- * [#1212] Add missing files on the client side
- * [#1212] Fix debian docs files - no TODO file
- * [1212] Fix manpage creation problem
- * [#1220] Generate client-side glue files and marshaller at
- compilation time
- * [#1220] Generate server-side glue files at compilation time
- * [#1212] Change binary name to sflphone
- * [#1212] Update .gitignore to fit the new working tree
- * [#1220] Explicitly generate glue files before building the library
- * [#1220] Compile dbus directory before audio
- * [#1212] Create sflphone-common at the root of the repository
- * [#1212] Re-add pjproject
- * [#1212] Remove Makefile from repo
- * [#1220] Fix Makefile.am
- * [#1212] New working directory functional
- * [#1212] Update .gitignore
- * [#1212] Hack to make pjsip compile..
- * [#1220] Use non-installed binary for dbusxx-xml2cpp
- * [#1212] Add descriptive files, remove unuseful scripts from tools/
-
- [ Alexandre Savard ]
- * Restore speex codecs
- * add frame size for celt (#1143)
- * add framesize to codec, independant from audiolayer (#1143)
- * use codec frame size in rtp (#1143)
- * compute fixed_codec_framesize (#1143)
- * do not resample if not required (#1143)
- * add condition on resampling for decoder (#1143)
- * add a condition on bytesAvail == 0 from mic data
- * no maximum in rtp decode (#1143)
- * compute maximum for decoding (#1143)
-
- [ Emmanuel Milou ]
- * [#1146] Implement unitary tests on the client-side
-
- [ Alexandre Savard ]
- * use float instead of int to compute max nb of sample (#1143)
- * add nbSampleMax for unresampled data (#1143)
- * make thread sleep during 5 ms insead of 20 (#1143)
- * use unix usleep (#1143)
- * 50 usecond thread!!!!! (#1143)
- * try with the smallest compression (#1143)
- * use timer set at framesize (#1143)
-
- [ Emmanuel Milou ]
- * [#1161] Restore changelog version
-
- [ Alexandre Savard ]
- * Remove celt stuff
-
- [ Emmanuel Milou ]
- * [#1161] Update changelog
- * [#1220] Add Conflicts: sflphone in debian control files
- * [#1179] Add liblog4c3 runtime dependency
- * [#1212] FIx typo error in dependency list for itnrepid
- * [#1212] FIx .desktop file to point on the right exec
- * [#1212] Modify changelog replacing tag
-
- [ Sflphone Project ]
- * "[#1262] Updated changelogs for version 0.9.5-0ubuntu1~beta"
-
- [ Emmanuel Milou ]
- * [#1212] restore changelogs
-
- [ Sflphone Project ]
- * [#1262] Updated changelogs for version 0.9.5-0ubuntu1 Snapshot 2009-
- 04-27
-
- [ Emmanuel Milou ]
- * [#1212] restore changelogs
-
- [ Sflphone Project ]
- * [#1262] Updated changelogs for version 0.9.5-0ubuntu1~beta
-
- [ Emmanuel Milou ]
- * [#1212] restore changelogs
-
- [ Sflphone Project ]
-
- -- Sflphone Project <sflphone@mtl.savoirfairelinux.net> Mon, 27 Apr 2009 17:00:03 -0400
-
-sflphone (0.9.4-0ubuntu2) SYSTEM; urgency=low
-
- [ Alexandre Savard ]
- * Restore speex and GSM detection
-
- [ Emmanuel Milou ]
- * Fix bug #1090
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Wed, 8 Apr 2009 11:29:15 -0500
-
-sflphone (0.9.4-0ubuntu1) SYSTEM; urgency=low
-
- [ Emmanuel Milou ]
- * Integrate DBus-c++ and libiax2 in the main build system
- * Clean up in the working repository
- * Reorder hooks configuration panel
- * Protect case when no codecs are active
- * Fix some return values
- * Add unitary tests for the hook manager (premisces)
-
- [Yun Liu]
- * Update chinese translation
-
- [Sven Werlen]
- * Update german translation
-
- [Hussein Abdallah]
- * Update russian translation
-
- [Maxime Chambreuil]
- * Update spanish translation
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Fri, 3 Apr 2009 18:29:15 -0500
-
-
-sflphone (0.9.4-rc1) SYSTEM; urgency=low
-
- [ Emmanuel Milou ]
- * Fix bug while trying to hold/unhold several simultaneous call
- * Improve address book build system
- * Implement SIP url popup on incoming call
- * Improve GTK+ panel configuration
- [ Julien Bonjean ]
- * GTK+ client refactoring
- * GTK+ clean up
- * Address book improvment
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Fri, 27 Mar 2009 18:29:15 -0500
-
-sflphone (0.9.4-0beta1) SYSTEM; urgency=low
-
- [ Alexandre Savard ]
- * Display codec used during conversation on the GUI
- * Enable/disable STUN parameters at runtime
- * Refactor search bar use
- [ Emmanuel Milou ]
- * Build system fixes
- * Implement SIP re-invite
- * Implement IP to IP call
- [ Julien Bonjean ]
- * Integrate GNOME address book based on evolution data server
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Fri, 20 Mar 2009 18:29:15 -0500
-
-
-sflphone (0.9.3-0ubuntu3) SYSTEM; urgency=low
-
- [ Alexandre Savard ]
- * Both playback and record streams in PA_STREAM_CORKED (pulseaudio)
- * Use PLUGHW device for ALSA capture
- * Functional IAX and SIP recording for voicemail
- * Use the less CPU-consuming interpolator algorithm for resampling
- * Display in GTK GUI the codec used in conversation
- * GTK GUI use ASCII instread of utf-8
- * Add record menus in GTK GUI
- * Put on hold when dialing a new number
- * AccountID's are saved in the history
-
- [ Emmanuel Milou ]
- * Integrate DBUS C++, libiax2 in the git repository
- * Update website
- * Use libspeexdsp only if available on the system
- * Updated .gitignore file
-
- [Cyrille Béraud]
- * Account assistant manager improvment
- * Add an email request when creating a new account to receive voicemails
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Sat, 14 Feb 2009 13:29:15 -0500
-
-sflphone (0.9.3-0ubuntu2) SYSTEM; urgency=low
-
- [ Emmanuel Milou ]
- * Add compilation note in README
- * Use default ALSA plugin for capture
- * Fix the ALSA capture problem one more time
- * Clean up debug messages in dbus.c
- * Add libspeexdsp dependency
- * Remove implicit declaration compilation warnings
- * Fix links in the website, add release note
- * Change capture for the website front page
- * Add alsa devel dependency in build-depends control file field
- * Clean up, indentation, try to handle latency problems in iax/pulseaudio
- * Remove pjsip generated files from the repo
- * Use the previous declared curAlias function in accountwindow
- * Fix bug in history call duration when the call fails
- * Remove runtime warning in the GTK+ client
- * Add librsvg2-common dependency to load SVG under KDE
- * Refresh .gitignore
- * Update locales files + french translation
- * Add configuration panel for future noise reduction
- * Add configuration panel for audio record module
- * Daemon less verbose; accounts don't try to access STUn options anymore
- * Fix typo in configwindow
- * Add content in the official website
- * use a GTK_STOCK icon for the record button
- * Complete description text in the assistant manager
- * Add libtool flags in client configure.ac
- * Remove unuseful dependency (snd)
- * Fix SIP transfer problems
- * Remove previous version of PJSIP from the repo
- * Upgrade PJSIP to version 1.0.1
- * Add the new website source in the repository
- * Use libspeexdsp for silence detection only if available
-
- [ Loïc Faure-Lacroix ]
- * Ajout du logo gpl3
- * Ajout des images
- * Ajout de la section screenshot pour le site
- * Ajout du favicon dans le header
- * Modification des cartes
-
- [ Alexandre Savard ]
- * Clean up <speex/libspeexdsp>
- * Small cleanup
- * Save Wave fixed
- * Fix new call button when recording
- * libspeexdsp added
- * Recording: default home folder at startup
- * Minor changes to config window
- * IAX recording fixed
- * Set / get recording path, still need some GTK for client
- * AudioRecord file name format
- * Now recording in HOME folder
-
- [ Cyrille Béraud ]
- * Fix bug in reqaccount.c
-
- [ Maxime Chambreuil ]
- * Update spanish translation
-
- [Yun Liu ]
- * Update chinese translation
-
- [ Hussein Abdallah ]
- * Update russian translation
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Sat, 14 Feb 2009 13:29:15 -0500
-
-sflphone (0.9.3-0ubuntu1) SYSTEM; urgency=low
-
- * Remove debug
- * Join thread before leaving
- * Fix implicit declaration in reqaccount
- * Add REST code to build the request to server
- * Fix GValue initialization warnings
- * Update version number, fix implicit declaration, fix GTK markup
- warnings
- * Apply patch to create custom SIP account from our own server
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Fri, 06 Feb 2009 19:17:32 -0500
-
-sflphone (0.9.2-2ubuntu9) SYSTEM; urgency=low
-
- [ Alexandre Savard ]
- * Speex audio codec preprocessing initialization
- * peer hung up segmentation fault solved
- * Stop recording when transfering
- * Terminate only one call
- * Add isRecording() function
- * Fix call_icon GTK client
- * Fix SIPCallClose() function, recorded file now close properly
- * Function terminateSIPCall added in sipvoiplink and managerimpl
- * Fix thread destructor
- * setRecordingOption function implement in audiorecord
- * Record now implemented in Call class
- * Record interface complete (on hold erase previous recording)
- * Added recButton in client
- * Added: record button related icons
- * Record button added
- * Overload AudioRecord::recData to get mic and speaker data mixed
- * Recording now in audiortp::run() method
- * Audio recording working in AudioRTP: receiveSessionForSpeaker
- * Open/close a wave file when pulse audio stream start/stop
-
- [ Emmanuel Milou ]
- * Fix path for GTK+ icons; clean up
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Thu, 05 Feb 2009 18:27:53 -0500
-
-sflphone (0.9.2-2ubuntu8) SYSTEM; urgency=low
-
- [ Emmanuel Milou ]
- * Update changelogs
- * Fix bug in merge and in Makefile.am
- * Terminate only one call
- * Disable PJsip shutdown when changing STUN parameters
- * Function terminateSIPCall added in sipvoiplink and managerimpl
- * Add a timer to the alsa thread to not jam the CPU load
- * Fix bug in sipvoiplink.cpp
- * Clean shutdown of pulseaudio on quiting
- * Fix DTMF at first start with Pulseaudio
- * Remove zeroconf from the build system
- * Add a library manager + exception handling
- * Clean up in the working directory
- * Better handling of capture XRUNs
- * Restore mic adjust volume on ALSA layer
- * Protect device ALSA operation if not opened
- * Fix the switching layer bug
- * Use dynamic_cast<> to use audiolayer-specific methods
- * Open the audio devices only once at startup
- * Refactoring of the ALSA part
- * Functional plug-in manager
- * Use a C++ thread to handle tones and DTMF in ALSA
- * Restore IAXVoIPLink, restore Mutex
- * Make the plugins registering against the plugin manager
- * Migrate to 1->N relationship between voiplink and accounts
- * API plugin for registration
- * Use C++ thread in SIP, move everything in sipvoiplink
- * Complete singleton pattern for the plugin manager
- * Add -Wno-return-type compilation flag to remove warnings; Update
- version number in configure.ac
- * Add the dynamic loading for the plugin framework; integate unittest
-
- [ Yun Liu ]
- * Update rpm spec file
- * modify build package script and spec file for suse
-
- [ Alexandre Savard ]
- * Add audiorecorder plugin and testaudiorecorder
- * Add audio Recording class, edit global.h
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Wed, 04 Feb 2009 14:00:30 -0500
-
-sflphone (0.9.2-2ubuntu7) SYSTEM; urgency=low
-
- [ Emmanuel Milou ]
- * Update changelog to 0.9.2-6
- * Fix some dbus-glib implementation details on the client side
- * Init history after dbus initialization
- * Add error checking in useragent; Clean sipvoiplink
- * Prevent crash when trying to call an empty number
- * Set the volume of the playback stream to PA_VOLUME_NORM at startup
- * Fix GTK+ generic value double initialization
- * Fix jaunty control file dependency problems
- * Fix jaunty control file dependency problems
-
- [ Yun Liu ]
- * Fix bug ticket # 137
- * Tolerant to gsm library of OpenSuse 11
-
- [ Sven Werlen ]
- * Update german translation
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Fri, 23 Jan 2009 17:48:13 -0500
-
-sflphone (0.9.2-2ubuntu6) SYSTEM; urgency=low
-
- [ Emmanuel Milou ]
- * Migrate STUN configuration to the main config window
- * Update french translation
- * Other tiny memory leaks
- * Fix memory leak in sampleconverter.cpp
- * Generate packages from the release branch
- * update the build package script
- * modify the control files with architecture=any
- * Remove valgring uninitialized value
- * IAX and SIP use the same global variables to set account
- configuration ; fix broken code
-
- [ Maxime Chambreuil ]
- * Update spanish translation
-
- [ Hussein Abdallah ]
- * Update russian translation
-
- [ Yun Liu ]
- * Update translation files
- * Fix the bug when user uncheck the account which fails in the
- previous registration
- * Add stun error status
- * Fix bug ticket #143
- * Script for auto-install dependencies
- * Fix bug ticket #140
- * Fix bug ticket 141
- * Fix the reregister process when user change the details of an
- account
-
- -- Emmanuel Milou <manu@sulfur.inside.savoirfairelinux.net> Fri, 16 Jan 2009 18:19:05 -0500
-
-sflphone (0.9.2-2ubuntu5) SYSTEM; urgency=low
-
- * Fix memory leak in the pulseaudio callback
- * Update debian package generation script
- * Warnings removal in GTK+ client
- * Clean adjust volume method in alsalayer
- * Plug the sflphone playback volume control to the pulseaudio volume
- manager
- * Display the date in history according to the current locale
- * Generate the changelog according to the git commit messages
- * Complete header in chinese translation file
- * Use the right gpg key to sign the packages
- * add debian jaunty jackalope support
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Wed, 14 Jan 2009 21:17:20 -0500
-
-sflphone (0.9.2-2ubuntu4) SYSTEM; urgency=low
-
- [ Emmanuel Milou ]
- * add german translation
-
- [ Yun Liu ]
- * Fix GUI crash in Ubuntu8.10 64bit system
-
- -- Yun Liu <yun.liu@savoirfairelinux.com> Thu, 08 Jan 2009 13:08:51 -0500
-
-sflphone (0.9.2-2ubuntu3) SYSTEM; urgency=low
-
- [ Emmanuel Milou ]
- * The main thread synchronizes the ringtone thread
- * disable custom ringtone for the ALSA layer
- * Fix the Makefile.am in man directory, add a SEE ALSO section
-
- [ Yun Liu ]
- * Fix daemon crash caused by the previous patch ( for bug ticket #129)
-
- -- Yun Liu <yun.liu@savoirfairelinux.com> Tue, 06 Jan 2009 16:18:38 -0500
-
-sflphone (0.9.2-2ubuntu2) SYSTEM; urgency=low
-
- * Fix bug ticket #129
-
- -- Yun Liu <yun.liu@savoirfairelinux.com> Wed, 5 Jan 2009 15:54:53 -0500
-
-sflphone (0.9.2-2ubuntu1) SYSTEM; urgency=low
-
- * Migrate from eXosip library to pjsip
- * Add multiple SIP accounts support
- * Fix ringtones problems
- * Add a pulseaudio support
- * Improve audio quality with ALSA
- * Add chinese translation
- * Improve spanish translation
- * Migrate to a maintained C++ DBus bindings
- * Clean and improve the build system
- * Add build-dependency on Perl because we need pod2man to generate manpages
-
- -- Yun Liu <yun.liu@savoirfairelinux.com> Wed, 26 Nov 2008 09:47:53 -0500
-
-sflphone (0.9.1) unstable; urgency=low
- * Add a search tool in the history
- * Migrate some gtk_entry_new to sexy_icon_entry_new
- * Bug fix (Ticket #78): The voicemail password isn't displayed anymore in
- the history tab
- * Add the SIP registration expire value in the user file.
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Thu, 22 May 2008 11:14:25 -0500
-
-sflphone (0.9.0) unstable; urgency=low
- * Add history features
- * Call date
- * Call duration
- * Mouse events in the history tab
- * Smooth switch from the history tab to the calls tab
- * Remove most of GTK-Critical warnings
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Wed, 13 May 2008 16:58:25 -0500
-
-sflphone (0.9-2008-06-06) unstable; urgency=low
- * Audio bug correction: capture stopped after a few minutes of conversation
- with USB Plantronics sound card
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Tue, 06 May 2008 16:58:25 -0500
-
-sflphone (0.9-2008-05-06) unstable; urgency=low
- * Bug correction: account creation with the assistant
- * GTK+ warnings removal
- * libnotify warnings removal
- * Remove aliasing on the SFLphone logo
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Mon, 05 May 2008 16:58:25 -0500
-
-sflphone (0.9) unstable; urgency=low
- * Clean dependencies ( removal of libboost )
- * Several GTK improvement and updates
- -account window
- -configuration window
- * Migrate from GtkCheckMenuItem to GtkImageMenuItem
- * ALSA standard I/O transfers: MMAP instead of R/W
- * Fix speex audio quality
- * IAX2 protocol
- -Fix hold/unhold situation
- -Add on hold music
- * SIP protocol
- -Ringtone on incoming call
- -Fix transfer situation
- * Add desktop notification ( libnotify )
- * Improve the system tray icon behaviour
- * Improve registration error handling
- * Register/unregister from the account window takes effect without starting back SFLphone
- * Compilation warnings removal
- * Call history
- * Add an account configuration wizard
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Wed, 30 Apr 2008 16:58:25 -0500
-
-sflphone (0.8.2) unstable; urgency=low
- * Internationalization of the GTK GUI
- * English / French
- * STUN support
- * Slight modifications of the graphical interface ( tooltips, dialpad, ...)
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Fri, 21 Mar 2008 11:37:53 -0500
diff --git a/tools/build-system/launchpad/sflphone/debian/control b/tools/build-system/launchpad/sflphone/debian/control
deleted file mode 100644
index 050440b8b5b28df8b200a5efc523807f5ff394ec..0000000000000000000000000000000000000000
--- a/tools/build-system/launchpad/sflphone/debian/control
+++ /dev/null
@@ -1,12 +0,0 @@
-Package: sflphone
-Section: gnome
-Priority: optional
-Architecture: all
-Depends: sflphone-gnome, sflphone-gnome
-Maintainer: Savoir-faire Linux Inc <emmanuel.milou@savoirfairelinux.com>
-Description: metapackage providing the GNOME client for SFLphone
- Provide a GNOME client for SFLphone.
- SFLphone is meant to be a robust enterprise-class desktop phone.
- SFLphone is released under the GNU General Public License.
- SFLphone is being developed by the global community, and maintained by
- Savoir-faire Linux, a Montreal, Quebec, Canada-based Linux consulting company.
diff --git a/tools/build-system/make-telify-package.sh b/tools/build-system/make-telify-package.sh
deleted file mode 100644
index f7a4d3cef8661b78f963ab9fff356ebeea9f0fae..0000000000000000000000000000000000000000
--- a/tools/build-system/make-telify-package.sh
+++ /dev/null
@@ -1,45 +0,0 @@
-#!/bin/bash
-#####################################################
-# File Name: make-telify-package.sh
-#
-# Purpose :
-#
-# Author: Julien Bonjean (julien@bonjean.info)
-#
-# Creation Date: 2009-12-15
-# Last Modified: 2009-12-15 18:16:47 -0500
-#####################################################
-
-#set -x
-
-. `dirname $0`/setenv.sh
-
-# change to working directory
-cd ${LAUNCHPAD_DIR}
-
-if [ "$?" -ne "0" ]; then
- echo " !! Cannot cd to launchpad directory"
- exit -1
-fi
-
-cd ${REFERENCE_REPOSITORY}
-
-for LAUNCHPAD_DISTRIBUTION in ${LAUNCHPAD_DISTRIBUTIONS[*]}
-do
- LOCAL_VERSION="${SOFTWARE_VERSION}~ppa${VERSION_INDEX}~${LAUNCHPAD_DISTRIBUTION}"
-
- cp ${DEBIAN_DIR}/control ${DEBIAN_DIR}/control
- cp ${DEBIAN_DIR}/changelog.generic ${DEBIAN_DIR}/changelog
-
- sed -i "s/SYSTEM/${LAUNCHPAD_DISTRIBUTION}/g" ${DEBIAN_DIR}/changelog
-
- cd ${LAUNCHPAD_DIR}/${LAUNCHPAD_PACKAGE}
- ./autogen.sh
- debuild -S -sa -kFDFE4451
- cd ${LAUNCHPAD_DIR}
-
- if [ ${DO_UPLOAD} ] ; then
- dput -f -c ${LAUNCHPAD_DIR}/dput.conf ${LAUNCHPAD_CONF_PREFIX}-${LAUNCHPAD_DISTRIBUTION} ${LAUNCHPAD_PACKAGE}_${LOCAL_VERSION}_source.changes
- fi
-done
-
diff --git a/tools/build-system/rpm/sflphone.spec b/tools/build-system/rpm/sflphone.spec
deleted file mode 100644
index 69a4df19636e07918594726ef591c7f660204366..0000000000000000000000000000000000000000
--- a/tools/build-system/rpm/sflphone.spec
+++ /dev/null
@@ -1,372 +0,0 @@
-%bcond_with video
-Name: sflphone
-Version: 1.4.2
-%if 0%{?nightly}
-%define rel rc%{nightly}
-%define tarball %{name}-%{version}-rc%{nightly}
-%else
-%define rel 1
-%define tarball %{name}-%{version}
-%endif
-Release: %{rel}%{?dist}
-Summary: SIP/IAX2 compatible enterprise-class software phone
-Group: Applications/Internet
-License: GPLv3
-URL: http://sflphone.org/
-Source0: https://projects.savoirfairelinux.com/attachments/download/6423/%{tarball}.tar.gz
-BuildRoot: %{_tmppath}/%{name}-%{version}-%{release}-root-%(%{__id_u} -n)
-BuildRequires: gettext gnutls-devel desktop-file-utils perl libuuid-devel
-BuildRequires: yaml-cpp-devel alsa-lib-devel pulseaudio-libs-devel
-BuildRequires: ccrtp-devel libzrtpcpp-devel dbus-c++-devel pcre-devel
-BuildRequires: gsm-devel opus-devel speex-devel expat-devel libsamplerate-devel
-BuildRequires: gnome-doc-utils libtool libsexy-devel intltool yelp-tools
-BuildRequires: libnotify-devel check-devel rarian-compat ilbc-devel
-BuildRequires: evolution-data-server-devel gnome-common libsndfile-devel
-BuildRequires: pjproject-devel libsrtp-devel
-# KDE requires
-BuildRequires: cmake kdepimlibs-devel
-BuildRequires: perl-podlators
-%if %{with video} && 0%{?fedora} < 18
-BuildRequires: libudev-devel
-%endif
-%if %{with video} && 0%{?fedora} >= 18
-BuildRequires: systemd-devel
-%endif
-
-%description
-SFLphone is a robust standards-compliant enterprise software phone,
-for desktop and embedded systems. It is designed to handle
-several hundreds of calls a day. It supports both SIP and IAX2
-protocols.
-
-%prep
-%setup -q -n %{tarball}
-
-%build
-# Build some dependencies with contrib since no Fedora packages exist (yet)
-mkdir -p daemon/contrib/native
-pushd daemon/contrib/native
-../bootstrap
-make .iax
-make .dht
-popd
-# Compile the daemon
-pushd daemon
-./autogen.sh
-%if %{with video}
-%configure --enable-video
-%else
-%configure
-%endif
-make %{?_smp_mflags}
-make doc
-popd
-pushd plugins
-./autogen.sh
-%configure
-make %{?_smp_mflags}
-popd
-# Compile kde client (only without video)
-pushd kde
-sed -i '/^[^#]add_subdirectory.*test/s/^[^#]/#/' src/CMakeLists.txt
-./config.sh --prefix=%{_prefix}
-cd build
-make %{?_smp_mflags}
-popd
-# Compile gnome client
-pushd gnome
-./autogen.sh
-%if %{with video}
-%configure --enable-video
-%else
-%configure
-%endif
-make %{?_smp_mflags}
-popd
-
-
-%if %{with video}
-%package gnome-video
-Summary: SIP/IAX2 compatible enterprise-class software phone
-Group: Applications/Internet
-Requires: %{name}-common-video
-Conflicts: sflphone-gnome sflphone
-BuildRequires: ffmpeg-devel clutter-gtk-devel
-%description gnome-video
-SFLphone is a robust standards-compliant enterprise software phone,
-for desktop and embedded systems. It is designed to handle
-several hundreds of calls a day. It supports both SIP and IAX2
-protocols.
-
-This package includes the Gnome client with videoconferencing ability
-
-%package common-video
-Summary: SIP/IAX2 compatible enterprise-class software phone
-Group: Applications/Internet
-Conflicts: sflphone sflphone-daemon sflphone-common
-%description common-video
-SFLphone is a robust standards-compliant enterprise software phone,
-for desktop and embedded systems. It is designed to handle
-several hundreds of calls a day. It supports both SIP and IAX2
-protocols.
-
-This package includes the SFLPhone daemon with videoconferencing enabled
-%else
-%package common
-Summary: SIP/IAX2 compatible enterprise-class software phone
-Group: Applications/Internet
-Conflicts: sflphone sflphone-daemon-video
-%description common
-SFLphone is a robust standards-compliant enterprise software phone,
-for desktop and embedded systems. It is designed to handle
-several hundreds of calls a day. It supports both SIP and IAX2
-protocols.
-
-This package includes the SFLPhone common
-
-%package gnome
-Summary: Gnome interface for SFLphone
-Group: Applications/Internet
-%if %{with video}
-Requires: %{name}-common-video = %{version}
-%else
-Requires: %{name}-common = %{version}
-%endif
-Obsoletes: sflphone < 1.2.2-2
-Conflicts: sflphone-video
-%description gnome
-SFLphone is a robust standards-compliant enterprise software phone,
-for desktop and embedded systems. It is designed to handle
-several hundreds of calls a day. It supports both SIP and IAX2
-protocols.
-
-This package includes the Gnome client
-
-%endif
-
-%package kde-video
-Summary: KDE interface for SFLphone
-Group: Applications/Internet
-%if %{with video}
-Requires: %{name}-common-video = %{version}
-%else
-Requires: %{name}-common = %{version}
-%endif
-%description kde-video
-SFLphone is a robust standards-compliant enterprise software phone,
-for desktop and embedded systems. It is designed to handle
-several hundreds of calls a day. It supports both SIP and IAX2
-protocols.
-
-This package includes the KDE client
-
-%package plugins
-Summary: Plugins (address book) for SFLphone
-Group: Applications/Internet
-%if %{with video}
-Requires: %{name}-common-video = %{version}
-%else
-Requires: %{name}-common = %{version}
-%endif
-%description plugins
-SFLphone is a robust standards-compliant enterprise software phone,
-for desktop and embedded systems. It is designed to handle
-several hundreds of calls a day. It supports both SIP and IAX2
-protocols.
-
-This package includes the address book plugin.
-
-%install
-rm -rf %{buildroot}
-pushd daemon
-make install DESTDIR=$RPM_BUILD_ROOT
-popd
-# Gnome install
-pushd gnome
-make install DESTDIR=$RPM_BUILD_ROOT
-# Find Lang files
-popd
-# Plugins install
-pushd plugins
-make install DESTDIR=$RPM_BUILD_ROOT
-popd
-%find_lang sflphone --with-gnome
-# Handling desktop file
-desktop-file-validate %{buildroot}%{_datadir}/applications/%{name}.desktop
-# KDE install
-pushd kde/build
-make install DESTDIR=$RPM_BUILD_ROOT
-popd
-%find_lang sflphone-client-kde --with-kde -f sflphone-client-kde
-%find_lang sflphone-kde --with-kde -f sflphone-kde
-
-%if %{with video}
-%pre gnome-video
-if [ "$1" -gt 1 ] ; then
- glib-compile-schemas %{_datadir}/glib-2.0/schemas &>/dev/null
-fi
-
-%post gnome-video
- glib-compile-schemas %{_datadir}/glib-2.0/schemas &>/dev/null
-
-%preun gnome-video
-if [ "$1" -eq 0 ] ; then
- glib-compile-schemas %{_datadir}/glib-2.0/schemas &>/dev/null
-fi
-%else
-%pre gnome
-if [ "$1" -gt 1 ] ; then
- glib-compile-schemas %{_datadir}/glib-2.0/schemas &>/dev/null
-fi
-
-%post gnome
- glib-compile-schemas %{_datadir}/glib-2.0/schemas &>/dev/null
-
-%preun gnome
-if [ "$1" -eq 0 ] ; then
- glib-compile-schemas %{_datadir}/glib-2.0/schemas &>/dev/null
-fi
-%endif
-
-%post kde-video -p /usr/sbin/ldconfig
-%postun kde-video -p /usr/sbin/ldconfig
-
-%if %{with video}
-%files common-video
-%else
-%files common
-%endif
-%defattr(-,root,root,-)
-%doc daemon/AUTHORS COPYING NEWS README
-%{_libdir}/%{name}/*
-%{_libexecdir}/sflphoned
-%{_datadir}/dbus-1/services/org.%{name}.SFLphone.service
-%{_mandir}/man1/sflphoned.1.gz*
-%{_datadir}/pixmaps/%{name}.svg
-%{_datadir}/%{name}/*
-
-%if %{with video}
-%files -f sflphone.lang gnome-video
-%else
-%files -f sflphone.lang gnome
-%endif
-%defattr(-,root,root,-)
-%{_bindir}/sflphone
-%{_bindir}/sflphone-client-gnome
-%exclude %{_libdir}/libsflphone.a
-%exclude %{_libdir}/libsflphone.la
-%{_datadir}/glib-2.0/schemas/org.sflphone.SFLphone.gschema.xml
-%{_datadir}/applications/%{name}.desktop
-%{_mandir}/man1/sflphone.1.gz
-%{_mandir}/man1/sflphone-client-gnome.1.gz
-%{_datadir}/pixmaps/%{name}.svg
-%{_datadir}/%{name}/*
-
-%files plugins
-%{_libdir}/sflphone/plugins/libevladdrbook.so
-
-%files kde-video -f sflphone-kde -f sflphone-client-kde
-%{_bindir}/sflphone-client-kde
-%{_datadir}/kde4/apps/sflphone-client-kde
-%{_datadir}/config.kcfg/sflphone-client-kde.kcfg
-%{_datadir}/applications/kde4
-%doc %{_mandir}/man1/*kde*
-%{_datadir}/icons/hicolor
-%{_libdir}/libksflphone.so*
-%{_libdir}/libqtsflphone.so*
-%exclude %{_includedir}/kde4/ksflphone/*.h
-%exclude %{_includedir}/qtsflphone/*.h
-
-%changelog
-* Tue Nov 25 2014 Tristan Matthews <tristan.matthews@savoirfairelinux.com> - 1.4.2-6
-- Build dht from contrib
-
-* Mon Nov 24 2014 Tristan Matthews <tristan.matthews@savoirfairelinux.com> - 1.4.2-5
-- drop .h and .so that are no longer built
-
-* Wed Nov 19 2014 Simon Piette <simon.piette@savoirfairelinux.com> - 1.4.2-4
-- add libstrp build require
-
-* Fri Nov 14 2014 Simon Piette <simon.piette@savoirfairelinux.com> - 1.4.2-3
-- Changed sflphoned path
-
-* Tue Nov 4 2014 Tristan Matthews <tristan.matthews@savoirfairelinux.com> - 1.4.2-2
-- Use Fedora's pjproject package
-
-* Thu Sep 25 2014 Tristan Matthews <tristan.matthews@savoirfairelinux.com> - 1.4.2-1
-- Bump version after release
-
-* Fri Sep 12 2014 Tristan Matthews <tristan.matthews@savoirfairelinux.com> - 1.4.1-5
-- Enable opus
-
-* Thu Sep 4 2014 Tristan Matthews <tristan.matthews@savoirfairelinux.com> - 1.4.1-4
-- Depend on yaml-cpp-devel instead of libyaml-devel
-
-* Mon Aug 25 2014 Tristan Matthews <tristan.matthews@savoirfairelinux.com> - 1.4.1-3
-- Build iax and pjproject with contrib
-
-* Wed Jul 23 2014 Simon Piette <simon.piette@savoirfairelinux.com> - 1.4.1-2
-- Always build kde package
-
-* Tue Jul 15 2014 Tristan Matthews <tristan.matthews@savoirfairelinux.com> - 1.4.1-1
-- Start development of 1.4.1
-
-* Tue Jul 15 2014 Tristan Matthews <tristan.matthews@savoirfairelinux.com> - 1.4.0-1
-- Update to 1.4.0
-
-* Wed Jul 9 2014 Tristan Matthews <tristan.matthews@savoirfairelinux.com> - 1.3.0-5
-- Drop uuid dependency
-
-* Mon Jul 07 2014 Simon Piette <simon.piette@savoirfairelinux.com> - 1.3.0-n
-- Support both nightly and release
-
-* Thu May 15 2014 Simon Piette <simon.piette@savoirfairelinux.com> - 1.3.0-rc%{nightly}
-- Adapt for nightly builds
-
-* Tue Jan 21 2014 Simon Piette <simon.piette@savoirfairelinux.com> - 1.3.0-2
-- Fix "Fix KDE paths"
-
-* Mon Jan 13 2014 Tristan Matthews <tristan.matthews@savoirfairelinux.com> - 1.3.0-1
-- Update to 1.3.0
-- Fix KDE paths (tested on f20)
-- Added libuuid dependency for pjsip
-
-* Wed Jun 19 2013 Simon Piette <simonp@fedoraproject.org> - 1.2.3-1
-- Update to 1.2.3
-- Enable ilbc
-
-* Mon Feb 18 2013 Simon Piette <simonp@fedoraproject.org> - 1.2.2-6
-- Add sflphone-plugins
-
-* Mon Feb 18 2013 Simon Piette <simonp@fedoraproject.org> - 1.2.2-5
-- Renamed daemon to config
-
-* Mon Feb 18 2013 Simon Piette <simonp@fedoraproject.org> - 1.2.2-4
-- Video variant for gnome
-
-* Wed Feb 13 2013 Simon Piette <simonp@fedoraproject.org> - 1.2.2-3
-- split daemon and gnome packages
-
-* Wed Feb 13 2013 Simon Piette <simonp@fedoraproject.org> - 1.2.2-2
-- creates a kde client package
-
-* Tue Jan 15 2013 Simon Piette <simonp@fedoraproject.org> - 1.2.2-1
-- upgraded to 1.2.2
-- updated BuildRequires
-- disabled ilbc
-- replaced gconf with gsettings
-
-* Tue Sep 11 2012 Simon Piette <simonp@fedoraproject.org> - 1.2.0-1
-- upgraded to 1.2.0 (tested on f16)
-- updated BuildRequires
-
-* Wed Apr 20 2011 Prabin Kumar Datta <prabindatta@fedoraproject.org> - 0.9.13-1
-- avoiding compling with Celt codec support to resolve build problem
-- removed clean section since not required
-- upgraded to 0.9.13
-
-* Mon Apr 18 2011 Prabin Kumar Datta <prabindatta@fedoraproject.org> - 0.9.12-2
-- Fixed schema registration problem
-
-* Fri Mar 25 2011 Prabin Kumar Datta <prabindatta@fedoraproject.org> - 0.9.12-1
-- Initial build
diff --git a/tools/build-system/scripts/run_package_test.sh b/tools/build-system/scripts/run_package_test.sh
deleted file mode 100755
index bda3f9ed4b88aeda590417bca5d42e169a77a785..0000000000000000000000000000000000000000
--- a/tools/build-system/scripts/run_package_test.sh
+++ /dev/null
@@ -1,291 +0,0 @@
-#!/bin/bash
-
-# package_test.sh is free software: you can redistribute it and/or modify
-# it under the terms of the GNU General Public License as published by
-# the Free Software Foundation, either version 3 of the License, or
-# (at your option) any later version.
-#
-# package_test.sh is distributed in the hope that it will be useful,
-# but WITHOUT ANY WARRANTY; without even the implied warranty of
-# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
-# GNU General Public License for more details.
-#
-# You should have received a copy of the GNU General Public License
-# along with package_test.sh If not, see <http://www.gnu.org/licenses/>.
-#
-# @author: Emmanuel Lepage Vallee <elv1313@gmail.com>
-# @copyright: Savoir-faire Linux 2014
-#
-# Description:
-# This script is used to quickly test packages from a CI system such as Jenkins
-# by hand. It was developed in response to numerous issues with the sflphone
-# ppa over the years and to quickly reproduce environment from similar to those
-# used by users whose packages crashed and were reported to errors.ubuntu.com
-
-# Settings
-DEBIAN_VERSION="trusty"
-IMAGEPATH="/var/chroot/"
-ARCH="amd64"
-IMAGENAME=${DEBIAN_VERSION}_integration.img
-MASTERMOUNTPOINT=/tmp/ppa_testing/master
-SNAPSHOTMOUNTPOINT=/tmp/ppa_testing/mountpoint
-SNAPSHOTNAME=snapshot_$(date '+%d.%m.%y')
-IMAGE_SIZE=8 # In Gigabyte
-PACKAGE_CACHE_SIZE=4 # In Gigabyte
-MIRROR="http://ftp.ussg.iu.edu/linux/ubuntu/"
-PACKAGES=()
-REPOSITORIES=()
-POST_COMMANDS=()
-PRE_COMMANDS=()
-PRE_UMOUNT_COMMANDS=()
-
-# Parse arguments
-OLD_IFS=$IFS
-IFS=`echo -en "\n\b"`
-while getopts ":a:d:p:r:s:c:b:u:m:h" opt; do
- case $opt in
- d)
- echo "Using alternate Debian derivative $OPTARG"
- DEBIAN_VERSION=$OPTARG
- ;;
- a)
- echo "Using alternate architecture $OPTARG"
- ARCH=$OPTARG
- ;;
- r)
- echo "Use alt repository (or PPA) $OPTARG"
- PPA=$OPTARG
- REPOSITORIES+=($OPTARG)
- ;;
- p)
- echo "Add package $OPTARG"
- PACKAGES+=($OPTARG)
- ;;
- p)
- MIRROR=$OPTARG
- ;;
- c)
- POST_COMMANDS+=($OPTARG)
- ;;
- b)
- echo -e "\n\n\n\nADDING " $OPTARG
- PRE_COMMANDS+=($OPTARG)
- ;;
- u)
- PRE_UMOUNT_COMMANDS+=($OPTARG)
- ;;
- s)
- echo "Open a shell before deleting the snapshot"
- OPEN_SHELL=true
- ;;
- :|h)
- echo "./create_jail.sh [-arch trusty] [-h] [-ppa link]"
- echo "a: architechture (i386, amd64, armhf)"
- echo "r: Add an apt repository to /etc/apt/sources.list (multiple allowed)"
- echo "p: Package to install (multiple allowed)"
- echo "d: alternative distribution (trusty, precise, wheezy, etc)"
- echo "c: Bash command to execute before exiting (multiple allowed)"
- echo "b: Bash command to execute before installing packages (multiple allowed)"
- echo "u: LOCAL (as ROOT on your *REAL* Linux) commands to execute before"\
- "unmounting the jail (PWD in the jail /), be careful!"
- echo "m: Ubuntu/Debian repository mirror"
- echo "shell: Open a shell before exiting"
- echo
- echo "Example usage:"
- echo "./create_jail.sh"
- exit 0
- ;;
- \?)
- echo "Invalid option: -$OPTARG", use -h for helo >&2
- exit 1
- ;;
- esac
-done
-IFS=$OLD_IFS
-
-# Unmount a jail
-function unmountjail() {
- CHROOT_PATH=`pwd`
- if [ "$1" != "" ]; then
- CHROOT_PATH=$1
- fi
- echo Unmounting jail on $CHROOT_PATH
- umount -l $CHROOT_PATH/dev 2> /dev/null
- umount -l $CHROOT_PATH/dev/pts 2> /dev/null
- umount -l $CHROOT_PATH/sys 2> /dev/null
- umount -l $CHROOT_PATH/proc 2> /dev/null
- umount -l $CHROOT_PATH/var/cache/apt/archives/ 2> /dev/null
-}
-
-# Mount the special APT packet cache to avoid redundant downloads
-function mountcache() {
- MOUNT_PATH=$1
- CACHE_PATH=$IMAGEPATH/cache_${IMAGENAME}
- #If the cache doesn't exist, create it
- if [ ! -f $CACHE_PATH ]; then
- echo "Creating a package cache, this may take a while"
- dd if=/dev/zero of=$CACHE_PATH bs=1M count=${PACKAGE_CACHE_SIZE}000
- mkfs.btrfs $CACHE_PATH
- fi
-
- # Mount the cache
- mount -o loop $CACHE_PATH $MOUNT_PATH
-
- # Check if the cache is full, clear it
- PERCENT_USE=`df /$MOUNT_PATH | egrep "([0-9.]+)%" -o | egrep "([0-9.]+)" -o`
- if [ $PERCENT_USE -gt 75 ]; then
- echo "The cache is full, forcing a cleanup (${PERCENT_USE} used of ${PACKAGE_CACHE_SIZE}Gb)"
- rm $MOUNT_PATH/*.deb
- fi
-}
-
-# Mount a chroot jail in the current PWD or $1
-function mountjail() {
- CHROOT_PATH=`pwd`
- if [ "$1" != "" ]; then
- CHROOT_PATH=$1
- fi
- unmountjail $CHROOT_PATH
- echo Mounting jail on $CHROOT_PATH
- mount -o bind /dev $CHROOT_PATH/dev
- mount -o bind /dev/pts $CHROOT_PATH/dev/pts
- mount -o bind /sys $CHROOT_PATH/sys
- mount -o bind /proc $CHROOT_PATH/proc
- mountcache $CHROOT_PATH/var/cache/apt/archives/
-}
-
-function clearmountpoints() {
- # Close the jails
- unmountjail $SNAPSHOTMOUNTPOINT
-
- # Delete the snapshot
- umount -l $SNAPSHOTMOUNTPOINT 2> /dev/null
- btrfs subvolume delete ./${SNAPSHOTNAME} 2> /dev/null
-
- # Unmount the master
- umount -l $MASTERMOUNTPOINT 2> /dev/null
-}
-
-# Check the dependencies
-if ! command -v debootstrap ; then
- echo Please install debootstrap
- exit 1
-fi
-if ! command -v btrfs ; then
- echo Please install btrfs-tools
- exit 1
-fi
-
-# Check the script can be executed
-if [ "$(whoami)" != "root" ]; then
- echo This script needs to be executed as root
- exit 1
-fi
-
-# Make sure the mount points exists
-mkdir $MASTERMOUNTPOINT $SNAPSHOTMOUNTPOINT $IMAGEPATH -p
-
-cd $IMAGEPATH
-
-# Create the container image if it doesn't already exist
-if [ ! -f $IMAGENAME ]; then
- echo "Creating a disk image (use space now), this may take a while"
- dd if=/dev/zero of=$IMAGEPATH/$IMAGENAME bs=1M count=${IMAGE_SIZE}000
- mkfs.btrfs $IMAGENAME
-fi
-
-# Mount the image master snapshot
-clearmountpoints
-mount -o loop $IMAGEPATH/$IMAGENAME $MASTERMOUNTPOINT
-cd $MASTERMOUNTPOINT
-
-# Create the chroot if empty
-if [ "$(ls)" == "" ]; then
- debootstrap --variant=buildd --arch $ARCH $DEBIAN_VERSION ./ $MIRROR #http://archive.ubuntu.com/ubuntu
-
- # We need universe packages
- sed -i 's/main/main universe restricted multiverse/' ./etc/apt/sources.list
-
- # Add the deb-src repository
- cat ./etc/apt/sources.list | sed "s/deb /deb-src /" >> ./etc/apt/sources.list
-fi
-
-
-# Apply updates
-mountjail
-chroot ./ apt-get update > /dev/null
-chroot ./ apt-get upgrade -y --force-yes > /dev/null
-unmountjail
-
-# Create
-btrfs subvolume snapshot . ./${SNAPSHOTNAME}
-
-# Mount the subvolume
-mount -t btrfs -o loop,subvol=${SNAPSHOTNAME} $IMAGEPATH/$IMAGENAME $SNAPSHOTMOUNTPOINT
-
-
-
-###################################################
-# Begin testing #
-###################################################
-
-mountjail $SNAPSHOTMOUNTPOINT
-
-# Execute all PRE commands
-OLD_IFS=$IFS
-IFS=`echo -en "\n\b"`
-for COMMAND in ${PRE_COMMANDS[@]}; do
- echo EXEC $COMMAND
- chroot $SNAPSHOTMOUNTPOINT bash -c "$COMMAND"
-done
-IFS=$OLD_IFS
-
-# Add the PPA/repositories to the clear/vanilla snapshot
-for REPOSITORY in ${REPOSITORIES[@]}; do
- echo deb $REPOSITORY $DEBIAN_VERSION main \
- >> $SNAPSHOTMOUNTPOINT/etc/apt/sources.list
- echo deb-src $REPOSITORY $DEBIAN_VERSION main \
- >> $SNAPSHOTMOUNTPOINT/etc/apt/sources.list
-done
-
-# Fetch/Update the repositories
-chroot $SNAPSHOTMOUNTPOINT apt-get update > /dev/null 2> /dev/null
-
-# Install each package individually
-for PACKAGE in ${PACKAGES[@]}; do
- echo -e "\n\n===========Installing ${PACKAGE} ==============\n"
- chroot $SNAPSHOTMOUNTPOINT apt-get install $PACKAGE -y --force-yes
- RET=$?
- if [ "$RET" != "0" ]; then
- echo -e "\n\n\nInstall PPA to vanilla Ubuntu completed with $RET \n"
- clearmountpoints
- exit $RET
- else
- echo -e "\n\n${PACKAGE} successfully installed"
- chroot $SNAPSHOTMOUNTPOINT dpkg -s $PACKAGE
- fi
-done
-
-# Execute the post install bash commands
-OLD_IFS=$IFS
-IFS=`echo -en "\n\b"`
-for COMMAND in ${POST_COMMANDS[@]}; do
- chroot $SNAPSHOTMOUNTPOINT /bin/bash -c "$COMMAND"
-done
-IFS=$OLD_IFS
-
-# Execute commands on the *REAL OS*, as root
-OLD_IFS=$IFS
-IFS=`echo -en "\n\b"`
-for COMMAND in ${PRE_UMOUNT_COMMANDS[@]}; do
- /bin/bash -c "cd $SNAPSHOTMOUNTPOINT;$COMMAND"
-done
-IFS=$OLD_IFS
-
-# Open an interactive shell
-if [ "$OPEN_SHELL" == "true" ]; then
- chroot $SNAPSHOTMOUNTPOINT /bin/bash
-fi
-
-clearmountpoints
-echo "Completed successfully!"
diff --git a/tools/build-system/scripts/sflphone_integration.sh b/tools/build-system/scripts/sflphone_integration.sh
deleted file mode 100755
index 14dd2422ac26ee61861a2997e47697bb19c06594..0000000000000000000000000000000000000000
--- a/tools/build-system/scripts/sflphone_integration.sh
+++ /dev/null
@@ -1,90 +0,0 @@
-#!/bin/bash
-
-NIGHTLY_PPA="http://ppa.launchpad.net/savoirfairelinux/sflphone-nightly/ubuntu"
-FAILED=0
-
-# Print an error if a test failed
-function() checkResult() {
- RET=$?
- if [ "$RET" != "0" ]; then
- echo !! " [FAILED]"
- let FAILED=$FAILED+1
- fi
-}
-
-#
-# Install the PPA packages
-#
-
-# Install the gnome client from the PPA
-sudo ./run_package_test.sh -r $NIGHTLY_PPA -p sflphone-gnome
-checkResult
-
-# Install the KDE client from the PPA
-sudo ./run_package_test.sh -r $NIGHTLY_PPA -p sflphone-kde
-checkResult
-
-# Install both clients from the PPA
-sudo ./run_package_test.sh -r $NIGHTLY_PPA -p sflphone-gnome-video sflphone-kde
-checkResult
-
-
-
-
-#
-# Upgrade stock Ubuntu sflphone packages to our PPA
-#
-
-# Install the stock gnome client, then upgrade to the PPA
-sudo ./run_package_test.sh -r $NIGHTLY_PPA \
- -b "apt-get install sflphone-gnome" -p sflphone-gnome
-checkResult
-
-# Install the stock KDE client, then upgrade to the PPA
-sudo ./run_package_test.sh -r $NIGHTLY_PPA \
- -b "apt-get install sflphone-kde" -p sflphone-kde
-checkResult
-
-
-
-
-#
-# Toggle the PPA gnome client video support
-#
-
-# Upgrade from non-video Gnome client to Video Gnome client
-sudo ./run_package_test.sh -r $NIGHTLY_PPA \
- -b "apt-get install sflphone-gnome" -p sflphone-gnome-video
-checkResult
-
-# Downgrade from non-video Gnome client to Video Gnome client
-sudo ./run_package_test.sh -r $NIGHTLY_PPA \
- -b "apt-get install sflphone-gnome-video" -p sflphone-gnome
-checkResult
-
-
-
-
-#
-# List build dependencies
-#
-
-
-# List Gnome client build-dep versus PPA
-sudo ./run_package_test.sh -r $NIGHTLY_PPA \
- -b "apt-get build-dep sflphone-gnome -s" -c "apt-get build-dep sflphone-gnome -s" \
- -c "debtree -b --arch=all --no-recommends --no-conflicts sflphone-gnome > /tmp/graph" \
- -p debtree -u "dotty tmp/graph"
-checkResult
-
-# List Gnome client (+video) build-dep versus PPA
-sudo ./run_package_test.sh -r $NIGHTLY_PPA \
- -b "apt-get build-dep sflphone-gnome -s" -c "apt-get build-dep sflphone-gnome-video -s" \
- -c "debtree -b --arch=all --no-recommends --no-conflicts sflphone-gnome-video > /tmp/graph"\
- -p debtree -u "dotty tmp/graph"
-checkResult
-
-# Exit with error 1 if one or more test failed
-if [ $FAILED ~= "0" ]; then
- exit 1
-fi
diff --git a/tools/build-system/setenv.sh b/tools/build-system/setenv.sh
deleted file mode 100755
index e65c18496d8e47199efdcb22e76cbf55fcdfc220..0000000000000000000000000000000000000000
--- a/tools/build-system/setenv.sh
+++ /dev/null
@@ -1,19 +0,0 @@
-#!/bin/bash
-# Export environment variables for launch-build-machine-jenkins.sh script.
-
-# home directory
-export ROOT_DIR=${HOME}
-
-# In case the script is executed manually, replace the variables set by Jenkins
-export WORKSPACE=${WORKSPACE:=.}
-
-# gpg passphrase file
-export GPG_FILE="${WORKSPACE}/.gpg-sflphone"
-
-export EDITOR="echo"
-
-export REFERENCE_REPOSITORY="${WORKSPACE}"
-
-export WORKING_DIR="${WORKSPACE}/tools/build-system"
-export LAUNCHPAD_DIR="${WORKING_DIR}/launchpad"
-export LAUNCHPAD_DISTRIBUTIONS=("trusty utopic")
diff --git a/tools/build-system/sfl-git-dch-2.sh b/tools/build-system/sfl-git-dch-2.sh
deleted file mode 100755
index cde69a6a2c07bf2c1e4a1f9ac7e556c8ec2ddd4e..0000000000000000000000000000000000000000
--- a/tools/build-system/sfl-git-dch-2.sh
+++ /dev/null
@@ -1,94 +0,0 @@
-#!/bin/bash
-#####################################################
-# File Name: sfl-git-dch.sh
-#
-# Purpose :
-#
-# Author: Julien Bonjean (julien@bonjean.info)
-#
-# Creation Date: 2009-10-21
-# Last Modified: 2009-10-21 14:58:22 -0400
-#####################################################
-
-#set -x
-
-. $1
-
-echo "********************************************************************************"
-echo "Software: ${SOFTWARE}"
-echo "Version: ${VERSION}"
-echo "Distribution: ${DISTRIBUTION}"
-echo "Generating changelog (from commit ${COMMIT_HASH_BEGIN} to ${COMMIT_HASH_END}) in file ${CHANGELOG_FILE}"
-if [ ${IS_RELEASE} ] ; then
- echo "Release mode"
-else
- echo "Snapshot mode"
-fi
-
-cd ${WORKING_DIR}
-
-# use git log to retrieve changelog content
-CHANGELOG_CONTENT=`git log --no-merges --pretty=format:"%s" ${COMMIT_HASH_BEGIN}..${COMMIT_HASH_END} $2 | grep -v "\[\#1262\]"`
-
-if [ "$?" -eq "1" ]; then
- echo " !! No new commit since last release"
- CHANGELOG_CONTENT="No new commit"
-fi
-
-if [ "$?" -ne "0" ]; then
- echo " !! Error when retrieving changelog content"
- exit -1
-fi
-
-rm -f ${CHANGELOG_FILE}.dch >/dev/null 2>&1
-
-IS_FIRST=1
-echo "${CHANGELOG_CONTENT}" | while read line
-do
- if [ ${IS_FIRST} ]; then
-
- yes | dch --changelog ${CHANGELOG_FILE} -b --allow-lower-version --no-auto-nmu --distribution ${DISTRIBUTION} --newversion ${VERSION} "$line" >/dev/null 2>&1
-
- if [ "$?" -ne "0" ]; then
- echo
- echo " !! Error with new version"
- exit -1
- fi
-
- IS_FIRST=
-
- else
- dch --changelog ${CHANGELOG_FILE} --no-auto-nmu "$line"
- if [ "$?" -ne "0" ]; then
- echo
- echo " !! Error when adding changelog entry"
- exit -1
- fi
- fi
- echo -n .
-done
-
-# add snapshot or release flag if needed
-echo
-if [ ${IS_RELEASE} ]; then
- sed -i "3i\ ** ${VERSION} **\n" ${CHANGELOG_FILE}
- if [ "$?" -ne "0" ]; then
- echo " !! Error when adding snapshot flag"
- exit -1
- fi
-else
- sed -i "3i\ ** SNAPSHOT ${VERSION} **\n" ${CHANGELOG_FILE}
- if [ "$?" -ne "0" ]; then
- echo " !! Error when adding snapshot flag"
- exit -1
- fi
-fi
-
-echo
-echo "All done !"
-echo "********************************************************************************"
-
-cd -
-
-exit 0
-