diff --git a/autogen.sh b/autogen.sh
index 06b6a97c95e0c7f4db9c8bd21a358a14784e9187..ac85bd137c9dcf2f99eef6992751d1a92876cdcd 100755
--- a/autogen.sh
+++ b/autogen.sh
@@ -4,6 +4,6 @@
aclocal -I m4
libtoolize --force
autoheader
-autoconf -f
+autoconf -v -f
automake -a
./configure $@
diff --git a/build-package.sh b/build-package.sh
index a2c775ac077e8d1f9aedf5cc837277e5f99aca2b..78aa9774b07f8bf2ccc14c2cdb902a8f35adf96c 100755
--- a/build-package.sh
+++ b/build-package.sh
@@ -2,16 +2,48 @@
#
# @author: Yun Liu <yun.liu@savoirfairelinux.com>
#
-# Build sflphone debian packages for Ubuntu 8.04
+# Build sflphone rpm packages for Fedora 10 and openSUSE 11
# 1 - The SFLphone package must be build with a specific GnuPG key. Please contact us to have more information about that (<sflphoneteam@savoirfairelinux.com>)
# 2. The source code can be teched through anonymous http access. So no need of special access.
-# 3. After having all the prerequisites, you can run "build-package.sh" to build debian packages for sflphone.
-# All the source packages and binary packages will be generated in the current directory.
+# 3. After having all the prerequisites, you can run "build-rpm-package.sh" to build rpm packages for sflphone.
+#
# Refer to http://www.sflphone.org for futher information
+# Analyze parameters
+if [ "$1" == "--help" ] || [ "$1" == "" ];then
+ echo -e '\E[34mThis script is used to build sflphone rpm packages on ubuntu series(8.04,8,10,9), Fedora 10 and SUSE 11 platform.'
+ echo -e '\E[34mYou can add --fedora, --suse or --ubuntu to start packaging.'
+ echo
+ echo "The SFLphone package must be build with a specific GnuPG key. Please contact us to have more information about that (<sflphoneteam@savoirfairelinux.com>)"
+ echo
+ echo "For fedora and SUSE, you also need to add the following lines to $HOME/.rpmmacros:"
+ echo -e '\E[32m%_gpg_path /home/yun/.gnupg'
+ echo -e '\E[32m%_gpg_name Savoir-Faire Linux Inc. (Génération des paquets pour SFLphone) <sflphoneteam@savoirfairelinux.com>'
+ echo -e '\E[32m%_gpgbin /usr/bin/gpg'
+ echo
+ echo -e '\E[34mAfter all these preparations done, you can run ./build-package.sh --platform-name'
+ echo
+ echo -e '\E[36mHave fun!'
+
+ tput sgr0 # Reset colors to "normal."
+ echo
+ exit 1
+elif [ $1 == "--fedora" ];then
+ BUILDDIR=$HOME/rpmbuild
+ platform="fedora"
+elif [ $1 == "--suse" ];then
+ BUILDDIR=/usr/src/packages
+ platform="suse"
+elif [ $1 == "--ubuntu" ];then
+ platform="ubuntu"
+else
+ echo "This script can only be used for Ubuntu series, Fedora 10 and SUSE 11 platform. Use --help to get more information."
+ exit 1
+fi
+
if [ -d "sflphone" ]; then
echo "Directory sflphone already exists. Please remove it first."
- exit 1
+ exit 1
fi
# Anonymous git http access
@@ -22,33 +54,54 @@ git checkout origin/release -b release
# Get system parameters
arch_flag=`getconf -a|grep LONG_BIT | sed -e 's/LONG_BIT\s*//'`
os_version=`lsb_release -d -s -c | sed -e '1d'`
+ver=0.9.2
+
+if [ $platform == "ubuntu" ];then
+ # Generate the changelog, according to the distribution and the git commit messages
+ cp debian/changelog.$os_version debian/changelog
+ git-dch --debian-branch=release --release
+fi
-# Generate the changelog, according to the distribution and the git commit messages
-cp debian/changelog.$os_version debian/changelog
-git-dch --debian-branch=release --release
cd ..
# Remove useless git directory
rm sflphone/.git/ -rf
-# Copy the appropriate control file based on different archtecture
-cp sflphone/debian/control.$os_version sflphone/debian/control
+# Get the public gpg key to sign the packages
+wget -q http://www.sflphone.org/downloads/gpg/sflphone.gpg.asc -O- | gpg --import -
-echo "Building sflphone package on Ubuntu $os_version $arch_flag bit architecture...."
+if [ $platform == "ubuntu" ];then
+ # Copy the appropriate control file based on different archtecture
+ cp sflphone/debian/control.$os_version sflphone/debian/control
-# Provide prerequisite directories used by debuild
-cp sflphone sflphone-0.9.2 -r
-cp sflphone sflphone-0.9.2.orig -r
+ echo "Building sflphone package on Ubuntu $os_version $arch_flag bit architecture...."
+ # Provide prerequisite directories used by debuild
+ cp sflphone sflphone-$ver -r
+ cp sflphone sflphone-$ver.orig -r
+
+ # Build packages
+ cd sflphone-$ver/debian; debuild -k'Savoir-Faire Linux Inc.'
-# Get the public gpg key to sign the packages
-wget -q http://www.sflphone.org/downloads/gpg/sflphone.gpg.asc -O- | gpg --import -
+ # Post clean
+ cd ..
+ rm sflphone-$ver sflphone -rf
+ echo "Done! All the source packages and binary packages are generated in the current directory"
-# Build packages
-cd sflphone-0.9.2/debian; debuild -k'Savoir-Faire Linux Inc.'
+else
+ # Prepare for packaging
+ mv sflphone sflphone-$ver
-# Clean
-cd ../..
-rm sflphone-0.9.2/ -rf
-rm sflphone/ -rf
+ cp sflphone-$ver/platform/$platform.spec $BUILDDIR/SPECS/
+ cp sflphone-$ver/libs/pjproject-1.0/libpj-sfl.pc $BUILDDIR/SOURCES
+ tar zcvf sflphone-$ver.tar.gz sflphone-$ver
-echo "Building package finished successullly!"
+ rm sflphone-$ver -rf
+ mv sflphone-$ver.tar.gz $BUILDDIR/SOURCES
+ echo "Building sflphone package on $platform $arch_flag bit architecture...."
+
+ # Build packages
+ cd $BUILDDIR/SPECS/
+ rpmbuild -ba --sign sflphone.spec
+
+ echo "Done! All source rpms and binary rpms are stored in $BUILDDIR/SRPMS and $BUILDDIR/RPMS"
+fi
diff --git a/configure.ac b/configure.ac
index d9b4ed6dd946e89a73809b67d9bf1b53b14214aa..100f5d33d603f38acaefc85422c0899c844034cb 100644
--- a/configure.ac
+++ b/configure.ac
@@ -2,8 +2,8 @@ dnl SFLPhone - configure.ac for automake 1.9 and autoconf 2.59
dnl
dnl Process this file with autoconf to produce a configure script.
AC_PREREQ(2.59)
-AC_INIT([SFLPhone],[0.7],[sflphoneteam@savoirfairelinux.com],[sflphone])
-AC_COPYRIGHT([[Copyright (c) Savoir-Faire Linux 2004-2008]])
+AC_INIT([SFLPhone],[0.9.2-7],[sflphoneteam@savoirfairelinux.com],[sflphone])
+AC_COPYRIGHT([[Copyright (c) Savoir-Faire Linux 2004-2009]])
AC_REVISION([$Revision$])
dnl Compute canonical system name
@@ -13,7 +13,7 @@ dnl Rewrite name of programs to be installed
AC_ARG_PROGRAM
dnl Automake options
-AM_INIT_AUTOMAKE([-Wall -Wno-portability])
+AM_INIT_AUTOMAKE([-Wno-portability])
dnl Improve make variable MAKE
AC_PROG_MAKE_SET
@@ -37,7 +37,8 @@ AC_CONFIG_FILES([src/Makefile \
src/audio/codecs/ilbc/Makefile \
src/config/Makefile \
src/dbus/Makefile \
- src/zeroconf/Makefile])
+ src/zeroconf/Makefile \
+ src/plug-in/Makefile])
dnl Unitary test section
AC_CONFIG_FILES([test/Makefile])
@@ -95,7 +96,7 @@ dnl Define C++ as default language
AC_LANG(C++)
dnl Setup C++ compiler flags
-CXXFLAGS="$CXXFLAGS $USE_EXCEPTIONS $USER_INCLUDES -O0 -g -Wall -Wextra"
+CXXFLAGS="$CXXFLAGS $USE_EXCEPTIONS $USER_INCLUDES -O0 -g -Wno-return-type"
dnl Check for header files
AC_FUNC_ALLOCA
diff --git a/debian/changelog b/debian/changelog
index 645b87bab9f9fcf59117b4c64fee8730ee87b5fb..93faa51b23cf7bd923cc0d115ff12fca96d0a49e 100644
--- a/debian/changelog
+++ b/debian/changelog
@@ -1,3 +1,53 @@
+sflphone (0.9.2-2ubuntu6) intrepid; urgency=low
+
+ [ Emmanuel Milou ]
+ * Migrate STUN configuration to the main config window
+ * Update french translation
+ * Other tiny memory leaks
+ * Fix memory leak in sampleconverter.cpp
+ * Generate packages from the release branch
+ * update the build package script
+ * modify the control files with architecture=any
+ * Remove valgring uninitialized value
+ * IAX and SIP use the same global variables to set account
+ configuration ; fix broken code
+
+ [ Maxime Chambreuil ]
+ * Update spanish translation
+
+ [ Hussein Abdallah ]
+ * Update russian translation
+
+ [ Yun Liu ]
+ * Update translation files
+ * Fix the bug when user uncheck the account which fails in the
+ previous registration
+ * Add stun error status
+ * Fix bug ticket #143
+ * Script for auto-install dependencies
+ * Fix bug ticket #140
+ * Fix bug ticket 141
+ * Fix the reregister process when user change the details of an
+ account
+
+ -- Emmanuel Milou <manu@sulfur.inside.savoirfairelinux.net> Fri, 16 Jan 2009 18:19:05 -0500
+
+sflphone (0.9.2-2ubuntu5) intrepid; urgency=low
+
+ * Fix memory leak in the pulseaudio callback
+ * Update debian package generation script
+ * Warnings removal in GTK+ client
+ * Clean adjust volume method in alsalayer
+ * Plug the sflphone playback volume control to the pulseaudio volume
+ manager
+ * Display the date in history according to the current locale
+ * Generate the changelog according to the git commit messages
+ * Complete header in chinese translation file
+ * Use the right gpg key to sign the packages
+ * add debian jaunty jackalope support
+
+ -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Wed, 14 Jan 2009 21:17:20 -0500
+
sflphone (0.9.2-2ubuntu4) intrepid; urgency=low
[ Emmanuel Milou ]
diff --git a/debian/changelog.hardy b/debian/changelog.hardy
index 1eb567db468442e164f2ce7bca174775c1fd8a6a..93faa51b23cf7bd923cc0d115ff12fca96d0a49e 100644
--- a/debian/changelog.hardy
+++ b/debian/changelog.hardy
@@ -1,32 +1,82 @@
-sflphone (0.9.2-0ubuntu4) hardy; urgency=low
+sflphone (0.9.2-2ubuntu6) intrepid; urgency=low
[ Emmanuel Milou ]
- * add german translation
+ * 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 ]
- * Fix GUI crash in Ubuntu8.10 64bit system
+ * Update translation files
+ * Fix the bug when user uncheck the account which fails in the
+ previous registration
+ * Add stun error status
+ * Fix bug ticket #143
+ * Script for auto-install dependencies
+ * Fix bug ticket #140
+ * Fix bug ticket 141
+ * Fix the reregister process when user change the details of an
+ account
+
+ -- Emmanuel Milou <manu@sulfur.inside.savoirfairelinux.net> Fri, 16 Jan 2009 18:19:05 -0500
+
+sflphone (0.9.2-2ubuntu5) intrepid; urgency=low
+
+ * Fix memory leak in the pulseaudio callback
+ * Update debian package generation script
+ * Warnings removal in GTK+ client
+ * Clean adjust volume method in alsalayer
+ * Plug the sflphone playback volume control to the pulseaudio volume
+ manager
+ * Display the date in history according to the current locale
+ * Generate the changelog according to the git commit messages
+ * Complete header in chinese translation file
+ * Use the right gpg key to sign the packages
+ * add debian jaunty jackalope support
+
+ -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Wed, 14 Jan 2009 21:17:20 -0500
+
+sflphone (0.9.2-2ubuntu4) intrepid; urgency=low
+ [ Emmanuel Milou ]
+ * add german translation
+
+ [ Yun Liu ]
+ * Fix GUI crash in Ubuntu8.10 64bit system
+
-- Yun Liu <yun.liu@savoirfairelinux.com> Thu, 08 Jan 2009 13:08:51 -0500
-sflphone (0.9.2-0ubuntu3) hardy; urgency=low
+sflphone (0.9.2-2ubuntu3) intrepid; urgency=low
[ Emmanuel Milou ]
* The main thread synchronizes the ringtone thread
* disable custom ringtone for the ALSA layer
* Fix the Makefile.am in man directory, add a SEE ALSO section
-
+
[ Yun Liu ]
* Fix daemon crash caused by the previous patch ( for bug ticket #129)
-
+
-- Yun Liu <yun.liu@savoirfairelinux.com> Tue, 06 Jan 2009 16:18:38 -0500
-sflphone (0.9.2-0ubuntu2) hardy; urgency=low
+sflphone (0.9.2-2ubuntu2) intrepid; urgency=low
* Fix bug ticket #129
-- Yun Liu <yun.liu@savoirfairelinux.com> Wed, 5 Jan 2009 15:54:53 -0500
-sflphone (0.9.2-0ubuntu1) hardy; urgency=low
+sflphone (0.9.2-2ubuntu1) intrepid; urgency=low
* Migrate from eXosip library to pjsip
* Add multiple SIP accounts support
diff --git a/debian/changelog.intrepid b/debian/changelog.intrepid
index 645b87bab9f9fcf59117b4c64fee8730ee87b5fb..93faa51b23cf7bd923cc0d115ff12fca96d0a49e 100644
--- a/debian/changelog.intrepid
+++ b/debian/changelog.intrepid
@@ -1,3 +1,53 @@
+sflphone (0.9.2-2ubuntu6) intrepid; urgency=low
+
+ [ Emmanuel Milou ]
+ * Migrate STUN configuration to the main config window
+ * Update french translation
+ * Other tiny memory leaks
+ * Fix memory leak in sampleconverter.cpp
+ * Generate packages from the release branch
+ * update the build package script
+ * modify the control files with architecture=any
+ * Remove valgring uninitialized value
+ * IAX and SIP use the same global variables to set account
+ configuration ; fix broken code
+
+ [ Maxime Chambreuil ]
+ * Update spanish translation
+
+ [ Hussein Abdallah ]
+ * Update russian translation
+
+ [ Yun Liu ]
+ * Update translation files
+ * Fix the bug when user uncheck the account which fails in the
+ previous registration
+ * Add stun error status
+ * Fix bug ticket #143
+ * Script for auto-install dependencies
+ * Fix bug ticket #140
+ * Fix bug ticket 141
+ * Fix the reregister process when user change the details of an
+ account
+
+ -- Emmanuel Milou <manu@sulfur.inside.savoirfairelinux.net> Fri, 16 Jan 2009 18:19:05 -0500
+
+sflphone (0.9.2-2ubuntu5) intrepid; urgency=low
+
+ * Fix memory leak in the pulseaudio callback
+ * Update debian package generation script
+ * Warnings removal in GTK+ client
+ * Clean adjust volume method in alsalayer
+ * Plug the sflphone playback volume control to the pulseaudio volume
+ manager
+ * Display the date in history according to the current locale
+ * Generate the changelog according to the git commit messages
+ * Complete header in chinese translation file
+ * Use the right gpg key to sign the packages
+ * add debian jaunty jackalope support
+
+ -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Wed, 14 Jan 2009 21:17:20 -0500
+
sflphone (0.9.2-2ubuntu4) intrepid; urgency=low
[ Emmanuel Milou ]
diff --git a/debian/changelog.jaunty b/debian/changelog.jaunty
index 99db3887ffc3de8da7f5c2c9093807d59bba73f1..93faa51b23cf7bd923cc0d115ff12fca96d0a49e 100644
--- a/debian/changelog.jaunty
+++ b/debian/changelog.jaunty
@@ -1,4 +1,54 @@
-sflphone (0.9.2-2ubuntu4) jaunty; urgency=low
+sflphone (0.9.2-2ubuntu6) intrepid; urgency=low
+
+ [ Emmanuel Milou ]
+ * Migrate STUN configuration to the main config window
+ * Update french translation
+ * Other tiny memory leaks
+ * Fix memory leak in sampleconverter.cpp
+ * Generate packages from the release branch
+ * update the build package script
+ * modify the control files with architecture=any
+ * Remove valgring uninitialized value
+ * IAX and SIP use the same global variables to set account
+ configuration ; fix broken code
+
+ [ Maxime Chambreuil ]
+ * Update spanish translation
+
+ [ Hussein Abdallah ]
+ * Update russian translation
+
+ [ Yun Liu ]
+ * Update translation files
+ * Fix the bug when user uncheck the account which fails in the
+ previous registration
+ * Add stun error status
+ * Fix bug ticket #143
+ * Script for auto-install dependencies
+ * Fix bug ticket #140
+ * Fix bug ticket 141
+ * Fix the reregister process when user change the details of an
+ account
+
+ -- Emmanuel Milou <manu@sulfur.inside.savoirfairelinux.net> Fri, 16 Jan 2009 18:19:05 -0500
+
+sflphone (0.9.2-2ubuntu5) intrepid; urgency=low
+
+ * Fix memory leak in the pulseaudio callback
+ * Update debian package generation script
+ * Warnings removal in GTK+ client
+ * Clean adjust volume method in alsalayer
+ * Plug the sflphone playback volume control to the pulseaudio volume
+ manager
+ * Display the date in history according to the current locale
+ * Generate the changelog according to the git commit messages
+ * Complete header in chinese translation file
+ * Use the right gpg key to sign the packages
+ * add debian jaunty jackalope support
+
+ -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Wed, 14 Jan 2009 21:17:20 -0500
+
+sflphone (0.9.2-2ubuntu4) intrepid; urgency=low
[ Emmanuel Milou ]
* add german translation
@@ -8,7 +58,7 @@ sflphone (0.9.2-2ubuntu4) jaunty; urgency=low
-- Yun Liu <yun.liu@savoirfairelinux.com> Thu, 08 Jan 2009 13:08:51 -0500
-sflphone (0.9.2-2ubuntu3) jaunty; urgency=low
+sflphone (0.9.2-2ubuntu3) intrepid; urgency=low
[ Emmanuel Milou ]
* The main thread synchronizes the ringtone thread
@@ -20,13 +70,13 @@ sflphone (0.9.2-2ubuntu3) jaunty; urgency=low
-- Yun Liu <yun.liu@savoirfairelinux.com> Tue, 06 Jan 2009 16:18:38 -0500
-sflphone (0.9.2-2ubuntu2) jaunty; urgency=low
+sflphone (0.9.2-2ubuntu2) intrepid; urgency=low
* Fix bug ticket #129
-- Yun Liu <yun.liu@savoirfairelinux.com> Wed, 5 Jan 2009 15:54:53 -0500
-sflphone (0.9.2-2ubuntu1) jaunty; urgency=low
+sflphone (0.9.2-2ubuntu1) intrepid; urgency=low
* Migrate from eXosip library to pjsip
* Add multiple SIP accounts support
diff --git a/globals.mak b/globals.mak
index c03985035c4058b00b378f1b0809017241339313..7a0b085b9efbe0f7deefb226deed286cbba7982a 100644
--- a/globals.mak
+++ b/globals.mak
@@ -1,6 +1,7 @@
# Global variables
src=$(top_srcdir)
sflcodecdir=$(libdir)/sflphone/codecs
+sflplugindir=$(libdir)/sflphone/plugins
PJSIP_LIBS = -lpjnath-sfl -lpjsua-sfl -lpjsip-sfl -lpjmedia-sfl -lpjsip-simple-sfl -lpjsip-ua-sfl -lpjmedia-codec-sfl -lpjlib-util-sfl -lpj-sfl
@@ -16,5 +17,6 @@ AM_CPPFLAGS = \
@SIP_CFLAGS@ \
@DBUSCPP_CFLAGS@ \
-DCODECS_DIR=\""$(sflcodecdir)"\" \
+ -DPLUGINS_DIR=\""$(sflplugindir)"\" \
-DENABLE_TRACE
diff --git a/sflphone-gtk/configure.ac b/sflphone-gtk/configure.ac
index b90af938de2786df192558b41aa3befd45ee1f23..fc3a5108d538a149843449ba7d88325bb88129fc 100644
--- a/sflphone-gtk/configure.ac
+++ b/sflphone-gtk/configure.ac
@@ -1,8 +1,8 @@
-AC_INIT([SFLphone],[0.9.2],[sflphoneteam@savoirfairelinux.com],[sflphone])
+AC_INIT([SFLphone],[0.9.2-7],[sflphoneteam@savoirfairelinux.com],[sflphone])
AM_CONFIG_HEADER(config.h)
PACKAGE=SFLphone
-VERSION=0.9.2
+VERSION=0.9.2-7
AM_INIT_AUTOMAKE($PACKAGE,$VERSION)
diff --git a/sflphone-gtk/src/actions.c b/sflphone-gtk/src/actions.c
index 19b024322b132ca6795e49cc34c007ebb390ce66..6efb5cbcc125d58bc629f2cddcf0b1107a4e7471 100644
--- a/sflphone-gtk/src/actions.c
+++ b/sflphone-gtk/src/actions.c
@@ -164,6 +164,8 @@ sflphone_fill_account_list(gboolean toolbarInitialized)
{
account_t * a = account_list_get_nth (i);
GHashTable * details = (GHashTable *) dbus_account_details(a->accountID);
+ if( details == NULL )
+ break;
a->properties = details;
gchar * status = g_hash_table_lookup(details, "Status");
@@ -215,15 +217,8 @@ sflphone_fill_account_list(gboolean toolbarInitialized)
toolbar_update_buttons();
}
-gboolean
-sflphone_init()
+gboolean sflphone_init()
{
- current_calls = calltab_init();
- history = calltab_init();
- sleep(0.5);
- if(SHOW_SEARCHBAR) histfilter = create_filter(GTK_TREE_MODEL(history->store));
- account_list_init ();
- codec_list_init();
if(!dbus_connect ()){
main_window_error_message(_("Unable to connect to the SFLphone server.\nMake sure the daemon is running."));
@@ -232,6 +227,11 @@ sflphone_init()
else
{
dbus_register(getpid(), "Gtk+ Client");
+ current_calls = calltab_init();
+ history = calltab_init();
+ if(SHOW_SEARCHBAR) histfilter = create_filter(GTK_TREE_MODEL(history->store));
+ account_list_init ();
+ codec_list_init();
sflphone_fill_account_list(FALSE);
sflphone_fill_codec_list();
sflphone_set_current_account();
@@ -643,60 +643,63 @@ sflphone_keypad( guint keyval, gchar * key)
void
sflphone_place_call ( call_t * c )
{
- if(c->state == CALL_STATE_DIALING)
- {
- if( account_list_get_size() == 0 )
- {
- notify_no_accounts();
- sflphone_fail(c);
- }
- else if( account_list_get_by_state( ACCOUNT_STATE_REGISTERED ) == NULL )
- {
- notify_no_registered_accounts();
- sflphone_fail(c);
- }
- else
+
+ if(c->state == CALL_STATE_DIALING && strcmp(c->to, "") != 0)
{
- account_t * current = account_list_get_current();
- if( current )
- {
- if(g_strcasecmp(g_hash_table_lookup( current->properties, "Status"),"REGISTERED")==0)
- {
- // OK, everything alright - the call is made with the current account
- c -> accountID = current -> accountID;
- status_bar_display_account(c);
- dbus_place_call(c);
- }
- else
- {
- // Current account is not registered
- // So we place a call with the first registered account
- // And we switch the current account
- current = account_list_get_by_state( ACCOUNT_STATE_REGISTERED );
- c -> accountID = current -> accountID;
- dbus_place_call(c);
- notify_current_account( current );
- status_bar_display_account(c);
- account_list_set_current_id( c-> accountID );
- }
- }
- else
- {
- // No current accounts have been setup.
- // So we place a call with the first registered account
- // and we change the current account
- current = account_list_get_by_state( ACCOUNT_STATE_REGISTERED );
- c -> accountID = current -> accountID;
- dbus_place_call(c);
- notify_current_account( current );
- status_bar_display_account(c);
- account_list_set_current_id( c-> accountID );
- }
+ if( account_list_get_size() == 0 )
+ {
+ notify_no_accounts();
+ sflphone_fail(c);
+ }
+
+ else if( account_list_get_by_state( ACCOUNT_STATE_REGISTERED ) == NULL )
+ {
+ notify_no_registered_accounts();
+ sflphone_fail(c);
+ }
+
+ else
+ {
+ account_t * current = account_list_get_current();
+ if( current )
+ {
+ if(g_strcasecmp(g_hash_table_lookup( current->properties, "Status"),"REGISTERED")==0)
+ {
+ // OK, everything alright - the call is made with the current account
+ c -> accountID = current -> accountID;
+ status_bar_display_account(c);
+ dbus_place_call(c);
+ }
+ else
+ {
+ // Current account is not registered
+ // So we place a call with the first registered account
+ // And we switch the current account
+ current = account_list_get_by_state( ACCOUNT_STATE_REGISTERED );
+ c -> accountID = current -> accountID;
+ dbus_place_call(c);
+ notify_current_account( current );
+ status_bar_display_account(c);
+ account_list_set_current_id( c-> accountID );
+ }
+ }
+ else
+ {
+ // No current accounts have been setup.
+ // So we place a call with the first registered account
+ // and we change the current account
+ current = account_list_get_by_state( ACCOUNT_STATE_REGISTERED );
+ c -> accountID = current -> accountID;
+ dbus_place_call(c);
+ notify_current_account( current );
+ status_bar_display_account(c);
+ account_list_set_current_id( c-> accountID );
+ }
+ }
+ // Update history
+ c->history_state = OUTGOING;
+ call_list_add(history, c);
}
- // Update history
- c->history_state = OUTGOING;
- call_list_add(history, c);
- }
}
/* Internal to action - set the __CURRENT_ACCOUNT variable */
diff --git a/sflphone-gtk/src/configwindow.c b/sflphone-gtk/src/configwindow.c
index f67dd87b750a81dd9323fdd81406801a1cba54f1..be137c73cf4bc7c7e1f9257351c5e5c3eacc656e 100644
--- a/sflphone-gtk/src/configwindow.c
+++ b/sflphone-gtk/src/configwindow.c
@@ -184,7 +184,7 @@ clean_history( void )
select_account(GtkTreeSelection *selection, GtkTreeModel *model)
{
GtkTreeIter iter;
- GValue val;
+ GValue val = {0};
if (!gtk_tree_selection_get_selected(selection, &model, &iter))
{
@@ -194,7 +194,7 @@ select_account(GtkTreeSelection *selection, GtkTreeModel *model)
return;
}
- val.g_type = G_TYPE_POINTER;
+ // The Gvalue will be initialized in the following function
gtk_tree_model_get_value(model, &iter, COLUMN_ACCOUNT_DATA, &val);
selectedAccount = (account_t*)g_value_get_pointer(&val);
diff --git a/sflphone-gtk/src/dbus.c b/sflphone-gtk/src/dbus.c
index 8795c1ef8baededd09bc6c910dc1215262c45923..68e0e2d423367cc834a682c13e40b233726aea64 100644
--- a/sflphone-gtk/src/dbus.c
+++ b/sflphone-gtk/src/dbus.c
@@ -39,6 +39,7 @@ DBusGConnection * connection;
DBusGProxy * callManagerProxy;
DBusGProxy * configurationManagerProxy;
DBusGProxy * instanceProxy;
+DBusGProxy * nameOwnerProxy;
static void
incoming_call_cb (DBusGProxy *proxy UNUSED,
@@ -172,17 +173,32 @@ error_alert(DBusGProxy *proxy UNUSED,
sflphone_throw_exception( errCode );
}
+
+static void nameOwnerChanged(DBusGProxy *proxy, char *name, char *old_owner, char *new_owner, gpointer data )
+{
+
+ g_print("******************************************************************\n");
+ g_print("Owner name of the service %s changed from %s to %s\n", name, old_owner, new_owner);
+ g_print("******************************************************************\n");
+
+ if (strcmp(name, "org.sflphone.SFLphone")!=0) return;
+
+}
+
gboolean
dbus_connect ()
{
+
GError *error = NULL;
+ connection = NULL;
+ instanceProxy = NULL;
+ nameOwnerProxy = NULL;
g_type_init ();
- error = NULL;
- connection = dbus_g_bus_get (DBUS_BUS_SESSION,
- &error);
- if (connection == NULL)
+ connection = dbus_g_bus_get (DBUS_BUS_SESSION, &error);
+
+ if (error)
{
g_printerr ("Failed to open connection to bus: %s\n",
error->message);
@@ -190,13 +206,33 @@ dbus_connect ()
return FALSE;
}
+
+ nameOwnerProxy = dbus_g_proxy_new_for_name_owner( connection,
+ DBUS_SERVICE_DBUS,
+ DBUS_PATH_DBUS,
+ DBUS_INTERFACE_DBUS,
+ &error);
+
+ if( nameOwnerProxy==NULL)
+ {
+ g_printerr ("Failed to get proxy to NameOwner\n");
+ return FALSE;
+ }
+
+ dbus_g_proxy_add_signal( nameOwnerProxy, "NameOwnerChanged",
+ G_TYPE_STRING, G_TYPE_STRING, G_TYPE_STRING, G_TYPE_INVALID);
+ dbus_g_proxy_connect_signal (nameOwnerProxy, "NameOwnerChanged",
+ G_CALLBACK (nameOwnerChanged), NULL, NULL);
+
+
/* Create a proxy object for the "bus driver" (name "org.freedesktop.DBus") */
- instanceProxy = dbus_g_proxy_new_for_name (connection,
+ instanceProxy = dbus_g_proxy_new_for_name_owner (connection,
"org.sflphone.SFLphone",
"/org/sflphone/SFLphone/Instance",
- "org.sflphone.SFLphone.Instance");
- if (!instanceProxy)
+ "org.sflphone.SFLphone.Instance",
+ &error);
+ if (instanceProxy==NULL)
{
g_printerr ("Failed to get proxy to Instance\n");
return FALSE;
@@ -205,11 +241,12 @@ dbus_connect ()
g_print ("DBus connected to Instance\n");
- callManagerProxy = dbus_g_proxy_new_for_name (connection,
+ callManagerProxy = dbus_g_proxy_new_for_name_owner (connection,
"org.sflphone.SFLphone",
"/org/sflphone/SFLphone/CallManager",
- "org.sflphone.SFLphone.CallManager");
- if (!callManagerProxy)
+ "org.sflphone.SFLphone.CallManager",
+ &error);
+ if (callManagerProxy==NULL)
{
g_printerr ("Failed to get proxy to CallManagers\n");
return FALSE;
@@ -251,10 +288,11 @@ dbus_connect ()
dbus_g_proxy_connect_signal (callManagerProxy,
"volumeChanged", G_CALLBACK(volume_changed_cb), NULL, NULL);
- configurationManagerProxy = dbus_g_proxy_new_for_name (connection,
+ configurationManagerProxy = dbus_g_proxy_new_for_name_owner (connection,
"org.sflphone.SFLphone",
"/org/sflphone/SFLphone/ConfigurationManager",
- "org.sflphone.SFLphone.ConfigurationManager");
+ "org.sflphone.SFLphone.ConfigurationManager",
+ &error);
if (!configurationManagerProxy)
{
g_printerr ("Failed to get proxy to ConfigurationManager\n");
@@ -280,6 +318,7 @@ dbus_clean ()
{
g_object_unref (callManagerProxy);
g_object_unref (configurationManagerProxy);
+ g_object_unref (instanceProxy);
}
@@ -1026,7 +1065,7 @@ gchar*
dbus_get_current_audio_output_plugin()
{
g_print("Before get audio plugin");
- gchar* plugin;
+ gchar* plugin="";
GError* error = NULL;
org_sflphone_SFLphone_ConfigurationManager_get_current_audio_output_plugin(
configurationManagerProxy,
@@ -1174,18 +1213,20 @@ dbus_get_searchbar()
{
int state;
GError* error = NULL;
- org_sflphone_SFLphone_ConfigurationManager_get_searchbar(
- configurationManagerProxy,
- &state,
- &error);
- g_print("After");
- if(error)
- {
- g_error_free(error);
- }
+ if(!org_sflphone_SFLphone_ConfigurationManager_get_searchbar( configurationManagerProxy, &state, &error))
+ {
+ if(error->domain == DBUS_GERROR && error->code == DBUS_GERROR_REMOTE_EXCEPTION)
+ g_printerr ("Caught remote method (get_searchbar) exception %s: %s\n", dbus_g_error_get_name(error), error->message);
+ else
+ g_printerr("Error while calling get_searchbar: %s\n", error->message);
+ g_error_free (error);
+ return -1;
+ }
else
+ {
g_print("DBus called get_searchbar on ConfigurationManager\n");
- return state;
+ return state;
+ }
}
void
@@ -1354,24 +1395,24 @@ dbus_set_notify( void )
guint
dbus_get_notify( void )
{
- g_print("Before dbus_get_notif_level()\n");
gint level;
GError* error = NULL;
- org_sflphone_SFLphone_ConfigurationManager_get_notify(
- configurationManagerProxy,
- &level,
- &error);
- if(error)
- {
- g_print("Error calling dbus_get_notif_level\n");
- g_error_free(error);
- }
- else
- g_print("Called dbus_get_notif_level\n");
-
- return (guint)level;
+ if( !org_sflphone_SFLphone_ConfigurationManager_get_notify( configurationManagerProxy,&level, &error) )
+ {
+ if(error->domain == DBUS_GERROR && error->code == DBUS_GERROR_REMOTE_EXCEPTION)
+ g_printerr ("Caught remote method (get_notify) exception %s: %s\n", dbus_g_error_get_name(error), error->message);
+ else
+ g_printerr("Error while calling get_notify: %s\n", error->message);
+ g_error_free (error);
+ return 0;
+ }
+ else{
+ g_print ("DBus called get_notify() on ConfigurationManager\n");
+ return (guint)level;
+ }
}
+
void
dbus_set_mail_notify( void )
{
@@ -1390,7 +1431,7 @@ dbus_set_mail_notify( void )
guint
dbus_get_mail_notify( void )
{
- g_print("Before dbus_get_mail_notif_level()\n");
+ g_print("Before dbus_get_mail_notif_level()\n");
gint level;
GError* error = NULL;
org_sflphone_SFLphone_ConfigurationManager_get_mail_notify(
diff --git a/sflphone-gtk/src/menus.c b/sflphone-gtk/src/menus.c
index bf9fbe93d0e6941edffa4ec8c9f19d14502502df..2f7a36f156e6520a97137cae06b0b1c7f952df82 100644
--- a/sflphone-gtk/src/menus.c
+++ b/sflphone-gtk/src/menus.c
@@ -128,7 +128,7 @@ help_about ( void * foo UNUSED)
"artists", artists,
"authors", authors,
"comments", _("SFLphone is a VoIP client compatible with SIP and IAX2 protocols."),
- "copyright", "Copyright © 2004-2008 Savoir-faire Linux Inc.",
+ "copyright", "Copyright © 2004-2009 Savoir-faire Linux Inc.",
"name", PACKAGE,
"title", _("About SFLphone"),
"version", VERSION,
diff --git a/src/Makefile.am b/src/Makefile.am
index 632a388a91672784901a393daa6d5d102a33c3ba..16725b63fb2d28f416c88bb7fd4f6453d0cb56d0 100644
--- a/src/Makefile.am
+++ b/src/Makefile.am
@@ -23,7 +23,7 @@ IAXSOURCES =
IAXHEADERS =
endif
-SUBDIRS = audio config dbus $(ZEROCONFDIR)
+SUBDIRS = audio config dbus $(ZEROCONFDIR) plug-in
# Add here the cpp files to be build with sflphone
sflphoned_SOURCES = \
@@ -63,6 +63,8 @@ sflphoned_LDADD = \
-luuid \
@SAMPLERATE_LIBS@
+#sflphoned_LDFLAGS=-pg
+
noinst_LTLIBRARIES = libsflphone.la
noinst_HEADERS = \
@@ -88,6 +90,7 @@ libsflphone_la_LIBADD = \
./audio/libaudio.la \
./dbus/libdbus.la \
./config/libconfig.la \
+ ./plug-in/libplugin.la \
$(IAX_LIBS)
libsflphone_la_SOURCES =
diff --git a/src/audio/alsalayer.cpp b/src/audio/alsalayer.cpp
index acd933c9d071b4d82e1a02c7206f07ab069a279d..c72378229e9a7b69f83a5560312ffa7724326414 100644
--- a/src/audio/alsalayer.cpp
+++ b/src/audio/alsalayer.cpp
@@ -54,7 +54,7 @@ AlsaLayer::~AlsaLayer (void)
deviceClosed = true;
ringtone_thread_is_running = false;
- pthread_join(ringtone_thread, NULL);
+ //pthread_join(ringtone_thread, NULL);
}
void
@@ -104,7 +104,7 @@ AlsaLayer::openDevice (int indexIn, int indexOut, int sampleRate, int frameSize,
_debugAlsa(" : nb channel in=%2d, out=%2d\n", _inChannel, _outChannel);
_debugAlsa(" : sample rate=%5d, format=%s\n", _sampleRate, SFLDataFormatString);
- ost::MutexLock lock( _mutex );
+ //ost::MutexLock lock( _mutex );
/*void **hint;
@@ -163,6 +163,7 @@ void* ringtoneThreadEntry( void *ptr )
( ( AlsaLayer *) ptr) -> playTones();
//sleep(0.1);
}
+
/*
pthread_mutex_lock(&mut);
while( ((AlsaLayer*)ptr)->_manager->getTelephoneTone() == NULL )
@@ -171,6 +172,7 @@ void* ringtoneThreadEntry( void *ptr )
}
( AlsaLayer *) ptr -> playTones();
pthread_mutex_unlock(&mut);*/
+
return 0;
}
@@ -206,7 +208,7 @@ AlsaLayer::fillHWBuffer( void)
bool
AlsaLayer::isStreamActive (void)
{
- ost::MutexLock lock( _mutex );
+ //ost::MutexLock lock( _mutex );
return (isPlaybackActive() && isCaptureActive());
}
@@ -279,7 +281,7 @@ AlsaLayer::getMic(void *buffer, int toCopy)
bool
AlsaLayer::isStreamStopped (void)
{
- ost::MutexLock lock( _mutex );
+ //ost::MutexLock lock( _mutex );
return !(isStreamActive());
}
@@ -291,20 +293,18 @@ void AlsaLayer::restorePulseAppsVolume( void ){}
///////////////// ALSA PRIVATE FUNCTIONS ////////////////////////////////////////////////
////////////////////////////////////////////////////////////////////////////////////////////
-
-
void
AlsaLayer::playTones( void )
{
int frames;
int maxBytes;
- pthread_mutex_lock(&mut);
- while(!_manager-> getTelephoneTone() && !_manager->getTelephoneFile())
- {
+ //pthread_mutex_lock(&mut);
+ //while(!_manager-> getTelephoneTone() && !_manager->getTelephoneFile())
+ //{
_debug("Make the ringtone thread wait\n");
pthread_cond_wait(&cond, &mut);
- }
+ //}
//frames = _periodSize ;
frames = 940 ;
@@ -324,13 +324,13 @@ AlsaLayer::playTones( void )
}
// free the temporary data buffer
free( out ); out = 0;
- pthread_mutex_unlock(&mut);
+ //pthread_mutex_unlock(&mut);
}
bool
AlsaLayer::isPlaybackActive(void) {
- ost::MutexLock guard( _mutex );
+ //ost::MutexLock guard( _mutex );
if( _PlaybackHandle )
return (snd_pcm_state(_PlaybackHandle) == SND_PCM_STATE_RUNNING ? true : false);
else
@@ -339,7 +339,7 @@ AlsaLayer::isPlaybackActive(void) {
bool
AlsaLayer::isCaptureActive(void) {
- ost::MutexLock guard( _mutex );
+ //ost::MutexLock guard( _mutex );
if( _CaptureHandle )
return (snd_pcm_state( _CaptureHandle) == SND_PCM_STATE_RUNNING ? true : false);
else
@@ -452,11 +452,11 @@ bool AlsaLayer::alsa_set_params( snd_pcm_t *pcm_handle, int type, int rate ){
// So the loop could start when the ringtone thread entry function is reached
ringtone_thread_is_running = true;
- if( pthread_create(&ringtone_thread, NULL, ringtoneThreadEntry, this) != 0 )
+ /*if( pthread_create(&ringtone_thread, NULL, ringtoneThreadEntry, this) != 0 )
{
_debug("Unable to start the ringtone posix thread\n");
return false;
- }
+ }*/
}
snd_pcm_sw_params_free( swparams );
diff --git a/src/audio/audiortp.cpp b/src/audio/audiortp.cpp
index c2f3ec87673b1d73b58029c8215bd0cfc3a24b43..85ec5fea464f198be3e410b68d8cd130dc562bb9 100644
--- a/src/audio/audiortp.cpp
+++ b/src/audio/audiortp.cpp
@@ -52,7 +52,7 @@ AudioRtp::~AudioRtp (void) {
int
AudioRtp::createNewSession (SIPCall *ca) {
- ost::MutexLock m(_threadMutex);
+ //ost::MutexLock m(_threadMutex);
// something should stop the thread before...
if ( _RTXThread != 0 ) {
@@ -66,7 +66,7 @@ AudioRtp::createNewSession (SIPCall *ca) {
_RTXThread = new AudioRtpRTX (ca, _symmetric);
try {
if (_RTXThread->start() != 0) {
- _debug("! ARTP Failure: unable to start RTX Thread\n");
+ _debug("! ARTP Failure: unable to start RTX Thread\n");
return -1;
}
} catch(...) {
@@ -79,7 +79,7 @@ AudioRtp::createNewSession (SIPCall *ca) {
void
AudioRtp::closeRtpSession () {
- ost::MutexLock m(_threadMutex);
+ //ost::MutexLock m(_threadMutex);
// This will make RTP threads finish.
// _debug("Stopping AudioRTP\n");
try {
diff --git a/src/audio/audiostream.cpp b/src/audio/audiostream.cpp
index 63654ce0768347134a292e829eddf722e5379e1a..5e4d295871dc2f614f2bf0b1266aac706f3e044b 100644
--- a/src/audio/audiostream.cpp
+++ b/src/audio/audiostream.cpp
@@ -30,7 +30,7 @@ AudioStream::AudioStream( pa_context* context, int type, std::string desc, doubl
sample_spec.rate = 44100;
sample_spec.channels = 1;
channel_map.channels = 1;
- pa_cvolume_set( &_volume , 1 , PA_VOLUME_MUTED ) ; // * vol / 100 ;
+ pa_cvolume_set( &_volume , 1 , PA_VOLUME_NORM ) ; // * vol / 100 ;
_audiostream = createStream( context );
}
diff --git a/src/audio/pulselayer.cpp b/src/audio/pulselayer.cpp
index 98a6cd6aa54fcf9e3a25a0ae50eb1e263a9f53c5..b1e6f6f725ea9c061cbbbcf8c04d811ff181643f 100644
--- a/src/audio/pulselayer.cpp
+++ b/src/audio/pulselayer.cpp
@@ -89,7 +89,7 @@ PulseLayer::connectPulseAudioServer( void )
}
pa_threaded_mainloop_unlock( m );
- serverinfo();
+ //serverinfo();
//muteAudioApps(99);
_debug("Context creation done\n");
}
diff --git a/src/dbus/callmanager.cpp b/src/dbus/callmanager.cpp
index 31929f2983df4d2e8af68239fd678c0e4f63e143..2fa7a779edcfad17266412642454f1eb177a218a 100644
--- a/src/dbus/callmanager.cpp
+++ b/src/dbus/callmanager.cpp
@@ -34,7 +34,9 @@ CallManager::placeCall( const std::string& accountID,
const std::string& to )
{
_debug("CallManager::placeCall received\n");
- Manager::instance().outgoingCall(accountID, callID, to);
+ // Check if a destination number is available
+ if( to == "") _debug("No number entered - Call stopped\n");
+ else Manager::instance().outgoingCall(accountID, callID, to);
}
void
diff --git a/src/managerimpl.cpp b/src/managerimpl.cpp
index c6177f35fc920738250824a767cd0f07635b32f2..626aca506c6114558d6a1db42fae6259e55a019b 100644
--- a/src/managerimpl.cpp
+++ b/src/managerimpl.cpp
@@ -64,20 +64,20 @@ ManagerImpl::ManagerImpl (void)
: _hasTriedToRegister(false)
, _config()
, _currentCallId2()
- , _currentCallMutex()
+ //, _currentCallMutex()
, _codecBuilder(NULL)
, _audiodriver(NULL)
, _dtmfKey(NULL)
, _codecDescriptorMap()
- , _toneMutex()
+ //, _toneMutex()
, _telephoneTone(NULL)
, _audiofile()
, _spkr_volume(0)
, _mic_volume(0)
- , _mutex()
+ //, _mutex()
, _dbus(NULL)
, _waitingCall()
- , _waitingCallMutex()
+ //, _waitingCallMutex()
, _nbIncomingWaitingCall(0)
, _path("")
, _exist(0)
@@ -86,7 +86,7 @@ ManagerImpl::ManagerImpl (void)
, _firewallAddr("")
, _hasZeroconf(false)
, _callAccountMap()
- , _callAccountMapMutex()
+ //, _callAccountMapMutex()
, _accountMap()
, _userAgent(NULL)
, _userAgentInitlized(false)
@@ -203,13 +203,13 @@ void ManagerImpl::terminate()
bool
ManagerImpl::isCurrentCall(const CallID& callId) {
- ost::MutexLock m(_currentCallMutex);
+ //ost::MutexLock m(_currentCallMutex);
return (_currentCallId2 == callId ? true : false);
}
bool
ManagerImpl::hasCurrentCall() {
- ost::MutexLock m(_currentCallMutex);
+ //ost::MutexLock m(_currentCallMutex);
_debug("Current call ID = %s\n", _currentCallId2.c_str());
if ( _currentCallId2 != "") {
return true;
@@ -219,13 +219,13 @@ ManagerImpl::hasCurrentCall() {
const CallID&
ManagerImpl::getCurrentCallId() {
- ost::MutexLock m(_currentCallMutex);
+ //ost::MutexLock m(_currentCallMutex);
return _currentCallId2;
}
void
ManagerImpl::switchCall(const CallID& id ) {
- ost::MutexLock m(_currentCallMutex);
+ //ost::MutexLock m(_currentCallMutex);
_currentCallId2 = id;
}
@@ -295,7 +295,9 @@ ManagerImpl::answerCall(const CallID& id)
ManagerImpl::hangupCall(const CallID& id)
{
stopTone(true);
+
if (_dbus) _dbus->getCallManager()->callStateChanged(id, "HUNGUP");
+
AccountID accountid = getAccountFromCall( id );
if (accountid == AccountNULL) {
/** @todo We should tell the GUI that the call doesn't exist, so
@@ -305,6 +307,7 @@ ManagerImpl::hangupCall(const CallID& id)
}
bool returnValue = getAccountLink(accountid)->hangup(id);
+
_debug("After voip link hungup!\n");
removeCallAccount(id);
switchCall("");
@@ -561,20 +564,20 @@ ManagerImpl::playDtmf(char code, bool isTalking)
// Multi-thread
bool
ManagerImpl::incomingCallWaiting() {
- ost::MutexLock m(_waitingCallMutex);
+ //ost::MutexLock m(_waitingCallMutex);
return (_nbIncomingWaitingCall > 0) ? true : false;
}
void
ManagerImpl::addWaitingCall(const CallID& id) {
- ost::MutexLock m(_waitingCallMutex);
+ //ost::MutexLock m(_waitingCallMutex);
_waitingCall.insert(id);
_nbIncomingWaitingCall++;
}
void
ManagerImpl::removeWaitingCall(const CallID& id) {
- ost::MutexLock m(_waitingCallMutex);
+ //ost::MutexLock m(_waitingCallMutex);
// should return more than 1 if it erase a call
if (_waitingCall.erase(id)) {
_nbIncomingWaitingCall--;
@@ -583,7 +586,7 @@ ManagerImpl::removeWaitingCall(const CallID& id) {
bool
ManagerImpl::isWaitingCall(const CallID& id) {
- ost::MutexLock m(_waitingCallMutex);
+ //ost::MutexLock m(_waitingCallMutex);
CallIDSet::iterator iter = _waitingCall.find(id);
if (iter != _waitingCall.end()) {
return false;
@@ -735,9 +738,9 @@ ManagerImpl::playATone(Tone::TONEID toneId) {
if (!hasToPlayTone) return false;
if (_telephoneTone != 0) {
- _toneMutex.enterMutex();
+ //_toneMutex.enterMutex();
_telephoneTone->setCurrentTone(toneId);
- _toneMutex.leaveMutex();
+ //_toneMutex.leaveMutex();
AudioLoop* audioloop = getTelephoneTone();
unsigned int nbSampling = audioloop->getSize();
@@ -774,16 +777,16 @@ ManagerImpl::stopTone(bool stopAudio=true) {
}
- _toneMutex.enterMutex();
+ //_toneMutex.enterMutex();
if (_telephoneTone != 0) {
_telephoneTone->setCurrentTone(Tone::TONE_NULL);
}
- _toneMutex.leaveMutex();
+ //_toneMutex.leaveMutex();
// for ringing tone..
- _toneMutex.enterMutex();
+ //_toneMutex.enterMutex();
_audiofile.stop();
- _toneMutex.leaveMutex();
+ //_toneMutex.leaveMutex();
}
/**
@@ -847,13 +850,13 @@ ManagerImpl::ringtone()
int sampleRate = audiolayer->getSampleRate();
AudioCodec* codecForTone = _codecDescriptorMap.getFirstCodecAvailable();
- _toneMutex.enterMutex();
+ //_toneMutex.enterMutex();
bool loadFile = _audiofile.loadFile(ringchoice, codecForTone , sampleRate);
- _toneMutex.leaveMutex();
+ //_toneMutex.leaveMutex();
if (loadFile) {
- _toneMutex.enterMutex();
+ //_toneMutex.enterMutex();
_audiofile.start();
- _toneMutex.leaveMutex();
+ //_toneMutex.leaveMutex();
if(CHECK_INTERFACE( layer, ALSA )){
/*int size = _audiofile.getSize();
SFLDataFormat output[ size ];
@@ -880,7 +883,7 @@ ManagerImpl::ringtone()
ManagerImpl::getTelephoneTone()
{
if(_telephoneTone != 0) {
- ost::MutexLock m(_toneMutex);
+ //ost::MutexLock m(_toneMutex);
return _telephoneTone->getCurrentTone();
}
else {
@@ -891,7 +894,7 @@ ManagerImpl::getTelephoneTone()
AudioLoop*
ManagerImpl::getTelephoneFile()
{
- ost::MutexLock m(_toneMutex);
+ //ost::MutexLock m(_toneMutex);
if(_audiofile.isStarted()) {
return &_audiofile;
} else {
@@ -1837,7 +1840,7 @@ ManagerImpl::detachZeroconfEvents(Pattern::Observer& observer)
ManagerImpl::getCallStatus(const std::string& sequenceId UNUSED)
{
if (!_dbus) { return false; }
- ost::MutexLock m(_callAccountMapMutex);
+ //ost::MutexLock m(_callAccountMapMutex);
CallAccountMap::iterator iter = _callAccountMap.begin();
TokenList tk;
std::string code;
@@ -2139,7 +2142,7 @@ ManagerImpl::associateCallToAccount(const CallID& callID, const AccountID& accou
{
if (getAccountFromCall(callID) == AccountNULL) { // nothing with the same ID
if ( accountExists(accountID) ) { // account id exist in AccountMap
- ost::MutexLock m(_callAccountMapMutex);
+ //ost::MutexLock m(_callAccountMapMutex);
_callAccountMap[callID] = accountID;
_debug("Associate Call %s with Account %s\n", callID.data(), accountID.data());
return true;
@@ -2154,7 +2157,7 @@ ManagerImpl::associateCallToAccount(const CallID& callID, const AccountID& accou
AccountID
ManagerImpl::getAccountFromCall(const CallID& callID)
{
- ost::MutexLock m(_callAccountMapMutex);
+ //ost::MutexLock m(_callAccountMapMutex);
CallAccountMap::iterator iter = _callAccountMap.find(callID);
if ( iter == _callAccountMap.end()) {
return AccountNULL;
@@ -2166,7 +2169,7 @@ ManagerImpl::getAccountFromCall(const CallID& callID)
bool
ManagerImpl::removeCallAccount(const CallID& callID)
{
- ost::MutexLock m(_callAccountMapMutex);
+ //ost::MutexLock m(_callAccountMapMutex);
if ( _callAccountMap.erase(callID) ) {
return true;
}
diff --git a/src/managerimpl.h b/src/managerimpl.h
index a76987d758b48188f60e853759be06a51d5b75de..b13fc911a88d79580e739ac8ea8b5439d5ad7424 100644
--- a/src/managerimpl.h
+++ b/src/managerimpl.h
@@ -868,7 +868,7 @@ class ManagerImpl {
CallID _currentCallId2;
/** Protected current call access */
- ost::Mutex _currentCallMutex;
+ //ost::Mutex _currentCallMutex;
/** Vector of CodecDescriptor */
CodecDescriptor* _codecBuilder;
@@ -886,7 +886,7 @@ class ManagerImpl {
/////////////////////
// Protected by Mutex
/////////////////////
- ost::Mutex _toneMutex;
+ //ost::Mutex _toneMutex;
TelephoneTone* _telephoneTone;
AudioFile _audiofile;
@@ -899,7 +899,7 @@ class ManagerImpl {
// Multithread variable (protected by _mutex)
//
/** Mutex to protect access to code section */
- ost::Mutex _mutex;
+ //ost::Mutex _mutex;
// Multithread variable (non protected)
DBusManagerImpl * _dbus;
@@ -908,7 +908,7 @@ class ManagerImpl {
CallIDSet _waitingCall;
/** Protect waiting call list, access by many voip/audio threads */
- ost::Mutex _waitingCallMutex;
+ //ost::Mutex _waitingCallMutex;
/** Number of waiting call, synchronize with waitingcall callidvector */
unsigned int _nbIncomingWaitingCall;
@@ -956,7 +956,7 @@ class ManagerImpl {
CallAccountMap _callAccountMap;
/** Mutex to lock the call account map (main thread + voiplink thread) */
- ost::Mutex _callAccountMapMutex;
+ //ost::Mutex _callAccountMapMutex;
/** Associate a new CallID to a AccountID
* Protected by mutex
diff --git a/src/plug-in/Makefile.am b/src/plug-in/Makefile.am
new file mode 100644
index 0000000000000000000000000000000000000000..53bf6b050cba69da07a0f8fd60b2f4b52c8f41d6
--- /dev/null
+++ b/src/plug-in/Makefile.am
@@ -0,0 +1,8 @@
+include ../../globals.mak
+
+noinst_LTLIBRARIES = libplugin.la
+
+libplugin_la_SOURCES = \
+ pluginmanager.cpp \
+ plugin.h
+
diff --git a/src/plug-in/plugin.cpp b/src/plug-in/plugin.cpp
new file mode 100644
index 0000000000000000000000000000000000000000..495437076549dee7c9bafbdc5ca6782076400d97
--- /dev/null
+++ b/src/plug-in/plugin.cpp
@@ -0,0 +1,27 @@
+#include "plugin.h"
+
+::sflphone::Plugin::Plugin( const std::string &filename UNUSED )
+{
+ //TODO IMPLEMENT
+}
+
+::sflphone::Plugin::Plugin( const Plugin &plugin UNUSED )
+{
+ //TODO IMPLEMENT
+}
+
+::sflphone::Plugin::~Plugin()
+{
+ //TODO IMPLEMENT
+}
+
+int ::sflphone::Plugin::getCoreVersion( void ) const
+{
+ //TODO IMPLEMENT
+ return 1;
+}
+
+void ::sflphone::Plugin::registerPlugin( PluginManager & )
+{
+ //TODO IMPLEMENT
+}
diff --git a/src/plug-in/plugin.h b/src/plug-in/plugin.h
new file mode 100644
index 0000000000000000000000000000000000000000..100d6ea5b40bc618e3e6cae9c4435b9b192af676
--- /dev/null
+++ b/src/plug-in/plugin.h
@@ -0,0 +1,43 @@
+#ifndef PLUGIN_H
+#define PLUGIN_H
+
+#include <string>
+
+#include "global.h"
+
+/*
+ * @file plugin.h
+ * @brief Define a plugin object
+ */
+
+namespace sflphone {
+
+class PluginManager;
+
+ class Plugin {
+
+ public:
+ Plugin( const std::string &name );
+ //Plugin( const Plugin &plugin );
+ virtual ~Plugin() {}
+
+ public:
+ /**
+ * Return the minimal core version required so that the plugin could work
+ * @return int The version required
+ */
+ virtual int getCoreVersion() const = 0;
+
+ /**
+ * Register the plugin to the plugin manager
+ */
+ virtual void registerPlugin( PluginManager & ) = 0;
+
+ private:
+ Plugin &operator =(const Plugin &plugin);
+
+ };
+}
+
+#endif //PLUGIN_H
+
diff --git a/src/plug-in/pluginmanager.cpp b/src/plug-in/pluginmanager.cpp
new file mode 100644
index 0000000000000000000000000000000000000000..6012a59bde292396b2f70fdce8d0a1d458ef8aa7
--- /dev/null
+++ b/src/plug-in/pluginmanager.cpp
@@ -0,0 +1,98 @@
+#include <dirent.h>
+#include <dlfcn.h>
+
+#include "pluginmanager.h"
+
+::sflphone::PluginManager::PluginManager():_loadedPlugins()
+{
+ _instance = this;
+}
+
+::sflphone::PluginManager::~PluginManager()
+{
+ _instance = 0;
+}
+
+int ::sflphone::PluginManager::loadPlugins( const std::string &path )
+{
+ std::string pluginDir, current;
+ ::sflphone::Plugin *plugin;
+ DIR *dir;
+ dirent *dirStruct;
+ int result=0;
+
+ const std::string pDir = "..";
+ const std::string cDir = ".";
+
+ /* The directory in which plugins are dropped. Default: /usr/lib/sflphone/plugins/ */
+ ( path == "" )? pluginDir = std::string(PLUGINS_DIR).append("/"):pluginDir = path;
+ _debug("Loading plugins from %s...\n", pluginDir.c_str());
+
+ dir = opendir( pluginDir.c_str() );
+ /* Test if the directory exists or is readable */
+ if( dir ){
+ /* Read the directory */
+ while( (dirStruct=readdir(dir)) ){
+ /* Get the name of the current item in the directory */
+ current = dirStruct->d_name;
+ /* Test if the current item is not the parent or the current directory */
+ if( current != pDir && current != cDir ){
+ loadDynamicLibrary( current );
+ result++;
+ }
+ }
+ }
+ /* Close the directory */
+ closedir( dir );
+
+ return result;
+}
+
+::sflphone::Plugin* ::sflphone::PluginManager::isPluginLoaded( const std::string &name )
+{
+ if(_loadedPlugins.empty()) return NULL;
+
+ /* Use an iterator on the loaded plugins map */
+ pluginMap::iterator iter;
+
+ iter = _loadedPlugins.begin();
+ while( iter != _loadedPlugins.end() ) {
+ if ( iter->first == name ) {
+ /* Return the associated plugin */
+ return iter->second;
+ }
+ iter++;
+ }
+
+ /* If we are here, it means that the plugin we were looking for has not been loaded */
+ return NULL;
+}
+
+void* ::sflphone::PluginManager::loadDynamicLibrary( const std::string& filename ) {
+
+ void *pluginHandlePtr = NULL;
+ const char *error;
+
+ _debug("Loading dynamic library %s\n", filename.c_str());
+
+#if defined(Q_OS_UNIX)
+ /* Load the library */
+ pluginHandlePtr = dlopen( filename.c_str(), RTLD_LAZY );
+ if( !pluginHandlePtr ) {
+ error = dlerror();
+ _debug("Error while opening plug-in: %s\n", error);
+ }
+ dlerror();
+#endif
+
+ return pluginHandlePtr;
+}
+
+void ::sflphone::PluginManager::unloadDynamicLibrary( void * pluginHandlePtr ) {
+
+ dlclose( pluginHandlePtr );
+ dlerror();
+}
+
+::sflphone::PluginManager* ::sflphone::PluginManager::_instance = 0;
+
diff --git a/src/plug-in/pluginmanager.h b/src/plug-in/pluginmanager.h
new file mode 100644
index 0000000000000000000000000000000000000000..651a0902f8e0db872052a2b0c234c66866ee92f1
--- /dev/null
+++ b/src/plug-in/pluginmanager.h
@@ -0,0 +1,72 @@
+#ifndef PLUGIN_MANAGER_H
+#define PLUGIN_MANAGER_H
+
+/*
+ * @file pluginmanager.h
+ * @brief Base class of the plugin manager
+ */
+
+#include "plugin.h"
+#include "global.h"
+
+#include <map>
+#include <string>
+
+namespace sflphone {
+
+ class PluginManager {
+
+ public:
+ /**
+ * Default constructor
+ */
+ PluginManager();
+
+ /**
+ * Destructor
+ */
+ ~PluginManager();
+
+ /**
+ * Returns the unique instance of the plugin manager
+ */
+ static PluginManager* instance();
+
+ /**
+ * Load all the plugins found in a specific directory
+ * @param path The absolute path to the directory
+ * @return int The number of items loaded
+ */
+ int loadPlugins( const std::string &path = "" );
+
+ /**
+ * Check if a plugin has been already loaded
+ * @param name The name of the plugin looked for
+ * @return Plugin* The pointer on the plugin or NULL if not found
+ */
+ Plugin* isPluginLoaded( const std::string &name );
+
+ private:
+ /**
+ * Load a unix dynamic/shared library
+ * @param filename The path to the dynamic/shared library
+ * @return void* A pointer on it
+ */
+ void * loadDynamicLibrary( const std::string &filename );
+
+ /**
+ * Unload a unix dynamic/shared library
+ * @param pluginHandleptr The pointer on the loaded plugin
+ */
+ void unloadDynamicLibrary( void * pluginHandlePtr );
+
+ /* Map of plugins associated by their string name */
+ typedef std::map<std::string, ::sflphone::Plugin*> pluginMap;
+ pluginMap _loadedPlugins;
+
+ /* The unique static instance */
+ static PluginManager* _instance;
+ };
+}
+
+#endif //PLUGIN_MANAGER_H
diff --git a/src/sipvoiplink.cpp b/src/sipvoiplink.cpp
index cc7e147a8b12f7549173727a6fa01d02d1f934ea..c71fd1da9eb1392f4074c80f29d88a3c6ea40327 100644
--- a/src/sipvoiplink.cpp
+++ b/src/sipvoiplink.cpp
@@ -81,7 +81,7 @@ void
SIPVoIPLink::terminateSIPCall()
{
- ost::MutexLock m(_callMapMutex);
+ //ost::MutexLock m(_callMapMutex);
CallMap::iterator iter = _callMap.begin();
SIPCall *call;
while( iter != _callMap.end() ) {
@@ -219,18 +219,21 @@ SIPVoIPLink::answer(const CallID& id)
bool
SIPVoIPLink::hangup(const CallID& id)
{
- SIPCall* call = getSIPCall(id);
- if (call==0) { _debug("! SIP Error: Call doesn't exist\n"); return false; }
+ SIPCall* call = getSIPCall(id);
+ if (call==0) { _debug("! SIP Error: Call doesn't exist\n"); return false; }
- Manager::instance().getUserAgent()->hangup(call);
+ if(!Manager::instance().getUserAgent()->hangup(call))
+ return false;
- // Release RTP thread
- if (Manager::instance().isCurrentCall(id)) {
- _debug("* SIP Info: Stopping AudioRTP for hangup\n");
- _audiortp->closeRtpSession();
- }
- removeCall(id);
- return true;
+ // Release RTP thread
+ if (Manager::instance().isCurrentCall(id)) {
+ _debug("* SIP Info: Stopping AudioRTP for hangup\n");
+ _audiortp->closeRtpSession();
+ }
+
+ removeCall(id);
+
+ return true;
}
bool
@@ -259,49 +262,59 @@ SIPVoIPLink::onhold(const CallID& id)
_debug("* SIP Info: Stopping AudioRTP for onhold action\n");
_audiortp->closeRtpSession();
- Manager::instance().getUserAgent()->onhold(call);
-
- return true;
+ return Manager::instance().getUserAgent()->onhold(call);
}
bool
SIPVoIPLink::offhold(const CallID& id)
{
- SIPCall* call = getSIPCall(id);
- if (call==0) { _debug("! SIP Error: Call doesn't exist\n"); return false; }
+ SIPCall *call;
- Manager::instance().getUserAgent()->offhold(call);
+ call = getSIPCall(id);
+ if (call==0) {
+ _debug("! SIP Error: Call doesn't exist\n");
+ return false;
+ }
- // Enable audio
- _debug("* SIP Info: Starting AudioRTP when offhold\n");
- call->setState(Call::Active);
- // it's sure that this is the current call id...
- if (_audiortp->createNewSession(call) < 0) {
- _debug("! SIP Failure: Unable to start sound (%s:%d)\n", __FILE__, __LINE__);
- return false;
- }
- return true;
+ if(!Manager::instance().getUserAgent()->offhold(call))
+ return false;
+
+ // Enable audio
+ _debug("* SIP Info: Starting AudioRTP when offhold\n");
+ call->setState(Call::Active);
+ // it's sure that this is the current call id...
+ if (_audiortp->createNewSession(call) < 0) {
+ _debug("! SIP Failure: Unable to start sound (%s:%d)\n", __FILE__, __LINE__);
+ return false;
+ }
+
+ return true;
}
bool
SIPVoIPLink::transfer(const CallID& id, const std::string& to)
{
- SIPCall* call = getSIPCall(id);
- if (call==0) { _debug("! SIP Failure: Call doesn't exist\n"); return false; }
+ SIPCall *call;
+ std::string tmp_to;
- std::string tmp_to = SIPToHeader(to);
- if (tmp_to.find("@") == std::string::npos) {
- tmp_to = tmp_to + "@" + getHostname();
- }
+ call = getSIPCall(id);
+ if (call==0) {
+ _debug("! SIP Failure: Call doesn't exist\n");
+ return false;
+ }
+
+ tmp_to = SIPToHeader(to);
+ if (tmp_to.find("@") == std::string::npos) {
+ tmp_to = tmp_to + "@" + getHostname();
+ }
- _debug("In transfer, tmp_to is %s\n", tmp_to.data());
+ _debug("In transfer, tmp_to is %s\n", tmp_to.data());
- Manager::instance().getUserAgent()->transfer(call, tmp_to);
+ return Manager::instance().getUserAgent()->transfer(call, tmp_to);
//_audiortp->closeRtpSession();
// shall we delete the call?
//removeCall(id);
- return true;
}
bool SIPVoIPLink::transferStep2()
@@ -313,34 +326,45 @@ bool SIPVoIPLink::transferStep2()
bool
SIPVoIPLink::refuse (const CallID& id)
{
- SIPCall* call = getSIPCall(id);
+ SIPCall *call;
- if (call==0) { _debug("Call doesn't exist\n"); return false; }
+ call = getSIPCall(id);
- // can't refuse outgoing call or connected
- if (!call->isIncoming() || call->getConnectionState() == Call::Connected) {
- _debug("It's not an incoming call, or it's already answered\n");
- return false;
- }
+ if (call==0) {
+ _debug("Call doesn't exist\n");
+ return false;
+ }
- Manager::instance().getUserAgent()->refuse(call);
-
- return true;
+ // can't refuse outgoing call or connected
+ if (!call->isIncoming() || call->getConnectionState() == Call::Connected) {
+ _debug("It's not an incoming call, or it's already answered\n");
+ return false;
+ }
+
+ return Manager::instance().getUserAgent()->refuse(call);
}
bool
-SIPVoIPLink::carryingDTMFdigits(const CallID& id, char code UNUSED)
+SIPVoIPLink::carryingDTMFdigits(const CallID& id, char code)
{
- SIPCall* call = getSIPCall(id);
- if (call==0) { _debug("Call doesn't exist\n"); return false; }
- int duration = Manager::instance().getConfigInt(SIGNALISATION, PULSE_LENGTH);
- const int body_len = 1000;
- char *dtmf_body = new char[body_len];
+ SIPCall *call;
+ int duration;
+ const int body_len = 1000;
+ char *dtmf_body;
+
+ call = getSIPCall(id);
+ if (call==0) {
+ _debug("Call doesn't exist\n");
+ return false;
+ }
+
+ duration = Manager::instance().getConfigInt(SIGNALISATION, PULSE_LENGTH);
+ dtmf_body = new char[body_len];
- snprintf(dtmf_body, body_len - 1, "Signal=%c\r\nDuration=%d\r\n", code, duration);
+ snprintf(dtmf_body, body_len - 1, "Signal=%c\r\nDuration=%d\r\n", code, duration);
- return Manager::instance().getUserAgent()->carryingDTMFdigits(call, dtmf_body);
+ return Manager::instance().getUserAgent()->carryingDTMFdigits(call, dtmf_body);
}
bool
@@ -447,9 +471,9 @@ SIPVoIPLink::SIPCallServerFailure(SIPCall *call)
//SIPCall* call = findSIPCallWithCid(event->cid);
if (call != 0) {
_debug("Server error!\n");
- CallID id = call->getCallId();
- Manager::instance().callFailure(id);
- removeCall(id);
+ CallID id = call->getCallId();
+ Manager::instance().callFailure(id);
+ removeCall(id);
}
//break;
//}
@@ -522,45 +546,6 @@ SIPVoIPLink::SIPCallAnswered(SIPCall *call, pjsip_rx_data *rdata)
}
}
-SIPCall*
-SIPVoIPLink::findSIPCallWithCid(int cid)
-{
- if (cid < 1) {
- _debug("! SIP Error: Not enough information for this event\n");
- return NULL;
- }
- ost::MutexLock m(_callMapMutex);
- SIPCall* call = 0;
- CallMap::iterator iter = _callMap.begin();
- while(iter != _callMap.end()) {
- call = dynamic_cast<SIPCall*>(iter->second);
- if (call && call->getCid() == cid) {
- return call;
- }
- iter++;
- }
- return NULL;
-}
-
-SIPCall*
-SIPVoIPLink::findSIPCallWithCidDid(int cid, int did)
-{
- if (cid < 1 && did < -1) {
- _debug("! SIP Error: Not enough information for this event\n");
- return NULL;
- }
- ost::MutexLock m(_callMapMutex);
- SIPCall* call = 0;
- CallMap::iterator iter = _callMap.begin();
- while(iter != _callMap.end()) {
- call = dynamic_cast<SIPCall*>(iter->second);
- if (call && call->getCid() == cid && call->getDid() == did) {
- return call;
- }
- iter++;
- }
- return NULL;
-}
SIPCall*
SIPVoIPLink::getSIPCall(const CallID& id)
@@ -571,61 +556,7 @@ SIPVoIPLink::getSIPCall(const CallID& id)
}
return NULL;
}
-/*
-bool
-SIPVoIPLink::handleDtmfRelay(eXosip_event_t* event) {
-
- SIPCall* call = findSIPCallWithCidDid(event->cid, event->did);
- if (call==0) { return false; }
-
-
- bool returnValue = false;
- osip_body_t *body = NULL;
- // Get the message body
- if (0 == osip_message_get_body(event->request, 0, &body) && body->body != 0 ) {
- _debug("* SIP Info: Text body: %s\n", body->body);
- std::string dtmfBody(body->body);
- std::string::size_type posStart = 0;
- std::string::size_type posEnd = 0;
- std::string signal;
- std::string duration;
- // search for signal=and duration=
- posStart = dtmfBody.find("Signal=");
- if (posStart != std::string::npos) {
- posStart += strlen("Signal=");
- posEnd = dtmfBody.find("\n", posStart);
- if (posEnd == std::string::npos) {
- posEnd = dtmfBody.length();
- }
- signal = dtmfBody.substr(posStart, posEnd-posStart+1);
- _debug("* SIP Info: Signal value: %s\n", signal.c_str());
-
- if (!signal.empty()) {
- if (Manager::instance().isCurrentCall(call->getCallId())) {
- Manager::instance().playDtmf(signal[0], true);
- returnValue = true;
- }
- }
-
- // we receive the duration, but we use our configuration...
-
- posStart = dtmfBody.find("Duration=");
- if (posStart != std::string::npos) {
- posStart += strlen("Duration=");
- posEnd = dtmfBody.find("\n", posStart);
- if (posEnd == std::string::npos) {
- posEnd = dtmfBody.length();
- }
- duration = dtmfBody.substr(posStart, posEnd-posStart+1);
- _debug("Duration value: %s\n", duration.c_str());
- returnValue = true;
- }
- }
- }
- return returnValue;
-}
-*/
///////////////////////////////////////////////////////////////////////////////
// Private functions
///////////////////////////////////////////////////////////////////////////////
diff --git a/src/sipvoiplink.h b/src/sipvoiplink.h
index fb863c491ab20b61e5034cff70460240a1835503..865703e2151eb4345c40e178b5fa2187aa06230a 100644
--- a/src/sipvoiplink.h
+++ b/src/sipvoiplink.h
@@ -268,22 +268,6 @@ class SIPVoIPLink : public VoIPLink
*/
void SIPCallReleased(SIPCall *call);
- /**
- * Find a SIPCall with cid
- * Explication there is no DID when the dialog is not establish...
- * @param cid call ID
- * @return SIPCall* SIPCall pointer or 0
- */
- SIPCall* findSIPCallWithCid(int cid);
-
- /**
- * Find a SIPCall with cid and did
- * @param cid call ID
- * @param did domain ID
- * @return SIPCall* SIPCall pointer or 0
- */
- SIPCall* findSIPCallWithCidDid(int cid, int did);
-
/**
* SIPCall accessor
* @param id The call identifier
diff --git a/src/useragent.cpp b/src/useragent.cpp
index cb39635f05d2593c5090cd43e6c501a683aa9db1..6e0e104cec4ada90606d980fbf15212e2870e450 100644
--- a/src/useragent.cpp
+++ b/src/useragent.cpp
@@ -1126,42 +1126,65 @@ void UserAgent::call_on_state_changed(pjsip_inv_session *inv, pjsip_event *e) {
}
bool UserAgent::onhold(SIPCall *call) {
- _debug("UserAgent: Before onhold pjsip_inv_reinite begins!\n");
+
pj_status_t status;
pjsip_tx_data *tdata;
pjmedia_sdp_attr *attr;
+ pjmedia_sdp_session* local_sdp;
+ local_sdp = call->getLocalSDPSession();
+ if( local_sdp == NULL ){
+ _debug("! SIP Failure: unable to find local_sdp\n");
+ return false;
+ }
/* Create re-INVITE with new offer */
- pjmedia_sdp_media_remove_all_attr(call->getLocalSDPSession()->media[0], "sendrecv");
+ // Remove all the attributes with the specified name
+ pjmedia_sdp_media_remove_all_attr(local_sdp->media[0], "sendrecv");
attr = pjmedia_sdp_attr_create(_pool, "sendonly", NULL);
- pjmedia_sdp_media_add_attr(call->getLocalSDPSession()->media[0], attr);
+ pjmedia_sdp_media_add_attr(local_sdp->media[0], attr);
- status = pjsip_inv_reinvite( call->getInvSession(), NULL, call->getLocalSDPSession(), &tdata);
+ status = pjsip_inv_reinvite( call->getInvSession(), NULL, local_sdp, &tdata);
+ if( status != PJ_SUCCESS )
+ {
+ _debug("On hold: creation of the Re-invite request failed\n");
+ return false;
+ }
/* Send the request */
status = pjsip_inv_send_msg( call->getInvSession(), tdata);
- _debug("UserAgent: After pjsip_inv_reinite begins!\n");
return (status == PJ_SUCCESS);
}
bool UserAgent::offhold(SIPCall *call) {
- _debug("UserAgent: Before offhold pjsip_inv_reinite begins!\n");
+
pj_status_t status;
pjsip_tx_data *tdata;
pjmedia_sdp_attr *attr;
+ pjmedia_sdp_session* local_sdp;
+ local_sdp = call->getLocalSDPSession();
+ if( local_sdp == NULL ){
+ _debug("! SIP Failure: unable to find local_sdp\n");
+ return false;
+ }
/* Create re-INVITE with new offer */
- pjmedia_sdp_media_remove_all_attr(call->getLocalSDPSession()->media[0], "sendonly");
+ // Remove all the attributes with the specified name
+ pjmedia_sdp_media_remove_all_attr(local_sdp->media[0], "sendonly");
attr = pjmedia_sdp_attr_create(_pool, "sendrecv", NULL);
- pjmedia_sdp_media_add_attr(call->getLocalSDPSession()->media[0], attr);
+ pjmedia_sdp_media_add_attr(local_sdp->media[0], attr);
+
+ status = pjsip_inv_reinvite( call->getInvSession(), NULL, local_sdp , &tdata);
+ if( status != PJ_SUCCESS )
+ {
+ _debug("Off hold: creation of the Re-invite request failed\n");
+ return false;
+ }
- status = pjsip_inv_reinvite( call->getInvSession(), NULL, call->getLocalSDPSession(), &tdata);
/* Send the request */
status = pjsip_inv_send_msg( call->getInvSession(), tdata);
- _debug("UserAgent: After pjsip_inv_reinite begins!\n");
return (status == PJ_SUCCESS);
}
@@ -1172,16 +1195,16 @@ bool UserAgent::hangup(SIPCall* call) {
// User hangup current call. Notify peer
status = pjsip_inv_end_session(call->getInvSession(), 404, NULL, &tdata);
if(status != PJ_SUCCESS)
- return false;
+ return false;
_debug("UserAgent: Before send msg!\n");
if(tdata == NULL)
- return true;
+ return true;
status = pjsip_inv_send_msg(call->getInvSession(), tdata);
if(status != PJ_SUCCESS)
- return false;
+ return false;
call->getInvSession()->mod_data[getInstance()->getModId()] = NULL;
return true;
@@ -1194,10 +1217,12 @@ bool UserAgent::refuse(SIPCall* call)
// User refuse current call. Notify peer
status = pjsip_inv_end_session(call->getInvSession(), PJSIP_SC_DECLINE, NULL, &tdata); //603
- PJ_ASSERT_RETURN(status == PJ_SUCCESS, false);
+ if(status != PJ_SUCCESS)
+ return false;
status = pjsip_inv_send_msg(call->getInvSession(), tdata);
- PJ_ASSERT_RETURN(status == PJ_SUCCESS, false);
+ if(status != PJ_SUCCESS)
+ return false;
call->getInvSession()->mod_data[getInstance()->getModId()] = NULL;
return true;
@@ -1253,9 +1278,6 @@ bool UserAgent::transfer(SIPCall *call, const std::string& to)
{
pjsip_evsub *sub;
pjsip_tx_data *tdata;
- //pjsip_dialog *dlg;
- //pjsip_generic_string_hdr *gs_hdr;
- //const pj_str_t str_ref_by = { (char*)"Referred-By", 11 };
struct pjsip_evsub_user xfer_cb;
pj_status_t status;
pj_str_t dest;
diff --git a/test/Makefile.am b/test/Makefile.am
index 14411aa140bbc3264308b27f0cd418287544821c..86e70dbc38e76a6fda92b43be969afd51e260781 100644
--- a/test/Makefile.am
+++ b/test/Makefile.am
@@ -1,6 +1,6 @@
include ../globals.mak
-bin_PROGRAMS = configurationTester
+bin_PROGRAMS = configurationTester pluginmanagerTester
OBJECT_FILES= \
../src/sflphoned-managerimpl.o \
@@ -16,6 +16,7 @@ OBJECT_FILES= \
../src/sflphoned-iaxaccount.o \
../src/sflphoned-eventthread.o \
../src/sflphoned-useragent.o \
+ ../src/plug-in/pluginmanager.o \
../src/sflphoned-samplerateconverter.o
configurationTester_SOURCES = \
@@ -23,6 +24,11 @@ configurationTester_SOURCES = \
configurationTest.h \
TestMain.cpp
+pluginmanagerTester_SOURCES = \
+ pluginmanagerTest.h \
+ pluginmanagerTest.cpp \
+ TestMain.cpp
+
configurationTester_LDADD = \
../src/libsflphone.la \
$(SFLPHONE_LIBS) $(ZEROCONFLIB) $(LIB_DNSSD) $(IAX_LIBS) \
@@ -36,4 +42,18 @@ configurationTester_LDADD = \
@SAMPLERATE_LIBS@ \
$(PJSIP_LIBS) \
$(OBJECT_FILES)
-
+
+pluginmanagerTester_LDADD = \
+ ../src/libsflphone.la \
+ $(SFLPHONE_LIBS) $(ZEROCONFLIB) $(LIB_DNSSD) $(IAX_LIBS) \
+ @ALSA_LIBS@ \
+ @PULSEAUDIO_LIBS@ \
+ @CPPUNIT_LIBS@ \
+ @CCEXT2_LIBS@ \
+ @CCGNU2_LIBS@ \
+ @CCRTP_LIBS@ \
+ @DBUSCPP_LIBS@ \
+ @SAMPLERATE_LIBS@ \
+ $(PJSIP_LIBS) \
+ $(OBJECT_FILES)
+
diff --git a/test/pluginmanagerTest.cpp b/test/pluginmanagerTest.cpp
new file mode 100644
index 0000000000000000000000000000000000000000..b9a3d413a400d422a7510963d36124ba1d026e14
--- /dev/null
+++ b/test/pluginmanagerTest.cpp
@@ -0,0 +1,44 @@
+/*
+ * Copyright (C) 2009 Savoir-Faire Linux inc.
+ * Author: Emmanuel Milou <emmanuel.milou@savoirfairelinux.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 3 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ */
+
+#include <stdio.h>
+#include <sstream>
+
+#include "pluginmanagerTest.h"
+
+using std::cout;
+using std::endl;
+
+void PluginManagerTest::setUp(){
+ // Instanciate the plugin manager object
+ _pm = new ::sflphone::PluginManager();
+}
+
+void PluginManagerTest::testLoadPluginDirectory(){
+ _pm->loadPlugins();
+}
+
+void PluginManagerTest::testNonloadedPlugin(){
+ CPPUNIT_ASSERT( _pm->isPluginLoaded("test") == NULL );
+}
+
+void PluginManagerTest::tearDown(){
+ // Delete the plugin manager object
+ delete _pm; _pm=NULL;
+}
diff --git a/test/pluginmanagerTest.h b/test/pluginmanagerTest.h
new file mode 100644
index 0000000000000000000000000000000000000000..e43ba8f7ab09262243fddd3edc21130ddca5dd80
--- /dev/null
+++ b/test/pluginmanagerTest.h
@@ -0,0 +1,75 @@
+/*
+ * Copyright (C) 2009 Savoir-Faire Linux inc.
+ * Author: Emmanuel Milou <emmanuel.milou@savoirfairelinux.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 3 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ */
+
+// Cppunit import
+#include <cppunit/extensions/HelperMacros.h>
+#include <cppunit/TestCaller.h>
+#include <cppunit/TestCase.h>
+#include <cppunit/TestSuite.h>
+
+#include <assert.h>
+
+// Application import
+#include "plug-in/pluginmanager.h"
+
+/*
+ * @file pluginManagerTest.cpp
+ * @brief Regroups unitary tests related to the plugin manager.
+ */
+
+#ifndef _PLUGINMANAGER_TEST_
+#define _PLUGINMANAGER_TEST_
+
+class PluginManagerTest : public CppUnit::TestCase {
+
+ /*
+ * Use cppunit library macros to add unit test the factory
+ */
+ CPPUNIT_TEST_SUITE( PluginManagerTest );
+ CPPUNIT_TEST( testLoadPluginDirectory );
+ CPPUNIT_TEST( testNonloadedPlugin );
+ CPPUNIT_TEST_SUITE_END();
+
+ public:
+ PluginManagerTest() : CppUnit::TestCase("Plugin Manager Tests") {}
+
+ /*
+ * Code factoring - Common resources can be initialized here.
+ * This method is called by unitcpp before each test
+ */
+ void setUp();
+
+ /*
+ * Code factoring - Common resources can be released here.
+ * This method is called by unitcpp after each test
+ */
+ inline void tearDown();
+
+ void testLoadPluginDirectory();
+
+ void testNonloadedPlugin();
+
+ private:
+ ::sflphone::PluginManager *_pm;
+};
+
+/* Register our test module */
+CPPUNIT_TEST_SUITE_REGISTRATION( PluginManagerTest );
+
+#endif