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 - - -# 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 + $timeout)) ]; then - break - fi - sleep 15 -done - -ruby create_tarball.rb --noaccount --application sflphone-kde -rm -rf kde -rm -rf sflphone-kde-*.tar.* -rm create_tarball.rb config.ini -mv sflphone-kde-* kde diff --git a/tools/build-system/hudson-sflphone-script.sh b/tools/build-system/hudson-sflphone-script.sh deleted file mode 100755 index 0a795b41c019d8bd3eb1595cf59e9b205927ee15..0000000000000000000000000000000000000000 --- a/tools/build-system/hudson-sflphone-script.sh +++ /dev/null @@ -1,249 +0,0 @@ -#!/bin/bash -e -# -# Copyright (C) 2004-2016 Savoir-faire Linux Inc. -# -# Author: Emmanuel Milou -# -# This program is free software; you can redistribute it and/or modify -# it under the terms of the GNU General Public License as published by -# the Free Software Foundation; either version 3 of the License, or -# (at your option) any later version. -# -# This program is distributed in the hope that it will be useful, -# but WITHOUT ANY WARRANTY; without even the implied warranty of -# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the -# GNU General Public License for more details. -# -# You should have received a copy of the GNU General Public License -# along with this program; if not, write to the Free Software -# Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. -# - -# Script used by Hudson continious integration server to build SFLphone - -XML_RESULTS="cppunitresults.xml" -TEST=0 -BUILD= -CODE_ANALYSIS=0 -DOXYGEN=0 -#daemon opts -DOPTS="--prefix=/usr" -#gnome opts -GOPTS="--prefix=/usr --enable-video" - -#compiler defaults -export CC=gcc -export CXX=g++ - -CONFIGDIR=~/.config -SFLCONFDIR=${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 Fri, 21 Apr 2010 19:51:54 +0100 - -mozilla-telify-sflphone (0.4.7.3) unstable; urgency=low - - [ Julien Bonjean ] - * Package creation - - -- Julien Bonjean 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 -Uploaders: Julien Bonjean -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 -Uploaders: Julien Bonjean -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 -Uploaders: Julien Bonjean -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 -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 --- 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usr/share/telify/chrome/locale/en-US/locale.js -24604d65d67874073ad9b3df7060e221 usr/share/telify/chrome/locale/de-DE/custom_preset.js -8ea28f7144c3da1b13824c751ed045d7 usr/share/telify/chrome/locale/de-DE/lang.dtd -ded79ce41ea89b117534c1af4e95d5b5 usr/share/telify/chrome/locale/de-DE/country_locale.js -f8c99d4e0ea6514f2d4f6ad71db85ae2 usr/share/telify/chrome/locale/de-DE/lang.properties -816dac9e3dc748bc9f7e6a46dd57e029 usr/share/telify/chrome/locale/de-DE/locale.js -4200a5c7a588e6aadbd6d56dc0f72ee7 usr/share/telify/defaults/preferences/preferences.js -8bed4b9918d4f36b8c3db1f0322e31bf usr/share/telify/chrome.manifest -7a164502d511ecbe38c3ad860e833579 usr/share/telify/install.rdf -a5a82bcdce14f9439cad46169b5bc354 usr/share/doc/mozilla-telify-sflphone/changelog.gz 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 @@ - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - 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) 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 - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - 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 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 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 @@ - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - diff --git a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/telify/chrome/content/error32.png b/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/telify/chrome/content/error32.png deleted file mode 100644 index 2d5e260a5f21585e0a76c81a0a66b6dbf5be60e6..0000000000000000000000000000000000000000 Binary files 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a/tools/build-system/launchpad/mozilla-telify-sflphone/debian/mozilla-telify-sflphone/usr/share/telify/chrome/content/info32.png and /dev/null differ 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 - * 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 @@ - - - - - - - - - - - - - - - - - - - - - - - - - - - - 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= 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= 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= 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= 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= 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 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=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= 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= 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= 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 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 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= 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= 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= '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= '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"], - ["Ãœ", "Ü"], - ["ä", "ä"], - ["ö", "ö"], - ["ü", "ü"], - ["ß", "ß"], - ]; - - for (var i=0; i= 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 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(); -} - - -/* - - - - - - - - - - - - - - - - - - - - - -*/ - - -}; - 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 @@ - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - 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 @@ - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - 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 @@ - - - - {6c5f349a-ddda-49ad-bdf0-326d3fe1f938} - true - chrome://telify/content/icon32.png - 0.4.7.3 - Michael Koch - http://www.codepad.de/ - chrome://telify/content/config.xul - - - - de-DE - Telify - Erzeugt klickbare Links aus Telefonnummern - - - - - - en-US - Telify - Converts telephone numbers into clickable links - - - - - - - {ec8030f7-c20a-464f-9b0e-13a3a9e97384} - 3.0 - 3.6.* - - - - \ 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 - * * #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 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 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 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 - ZajÄ…czkowski ) - * [#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 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 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 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 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 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 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 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 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 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 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 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 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 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 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 - * 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 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 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 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 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 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 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 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 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 Tue, 06 Jan 2009 16:18:38 -0500 - -sflphone (0.9.2-2ubuntu2) SYSTEM; urgency=low - - * Fix bug ticket #129 - - -- Yun Liu 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 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 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 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 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 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 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 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 -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 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 - -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: -# * `remove' -# * `purge' -# * `upgrade' -# * `failed-upgrade' -# * `abort-install' -# * `abort-install' -# * `abort-upgrade' -# * `disappear' -# -# 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 - * * #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 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 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 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 - ZajÄ…czkowski ) - * [#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 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 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 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 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 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 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 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 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 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 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 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 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 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 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 - * 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 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 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 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 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 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 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 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 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 Tue, 06 Jan 2009 16:18:38 -0500 - -sflphone (0.9.2-2ubuntu2) SYSTEM; urgency=low - - * Fix bug ticket #129 - - -- Yun Liu 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 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 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 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 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 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 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 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 -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 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 - -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: -# * `remove' -# * `purge' -# * `upgrade' -# * `failed-upgrade' -# * `abort-install' -# * `abort-install' -# * `abort-upgrade' -# * `disappear' -# -# 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 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 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 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 - ZajÄ…czkowski ) - * [#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 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 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 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 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 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 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 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 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 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 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 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 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 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 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 - * 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 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 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 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 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 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 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 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 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 Tue, 06 Jan 2009 16:18:38 -0500 - -sflphone (0.9.2-2ubuntu2) SYSTEM; urgency=low - - * Fix bug ticket #129 - - -- Yun Liu 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 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 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 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 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 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 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 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 -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 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 - -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: -# * `remove' -# * `purge' -# * `upgrade' -# * `failed-upgrade' -# * `abort-install' -# * `abort-install' -# * `abort-upgrade' -# * `disappear' -# -# 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 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 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 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 - ZajÄ…czkowski ) - * [#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 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 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 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 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 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 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 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 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 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 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 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 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 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 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 - * 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 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 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 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 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 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 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 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 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 Tue, 06 Jan 2009 16:18:38 -0500 - -sflphone (0.9.2-2ubuntu2) SYSTEM; urgency=low - - * Fix bug ticket #129 - - -- Yun Liu 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 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 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 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 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 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 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 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 -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 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 - -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: -# * `remove' -# * `purge' -# * `upgrade' -# * `failed-upgrade' -# * `abort-install' -# * `abort-install' -# * `abort-upgrade' -# * `disappear' -# -# 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 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 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 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 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 -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 - -SFLPhone KDE: - - Copyright (C) 2008-2009 Savoir-faire Linux - Copyright (C) 2009-2012 Savoir-faire Linux 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 can be included directly" error - * #8449: Update version 1.0.2 - * * #8542: removed trailing whitespace from tree - - -- Julien Bonjean 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 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 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