diff --git a/configure.ac b/configure.ac
index 3fc535bcb9d9a0fe59321928b90cffa7e6a1d6b6..9d66f5b93ed208ecfc7a7e44f0ea22c781c50163 100644
--- a/configure.ac
+++ b/configure.ac
@@ -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
@@ -97,7 +97,7 @@ 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 $USER_INCLUDES -O0 -g -Wno-return-type"
dnl Check for header files
AC_FUNC_ALLOCA
diff --git a/sflphone-gtk/configure.ac b/sflphone-gtk/configure.ac
index b90af938de2786df192558b41aa3befd45ee1f23..fc3a5108d538a149843449ba7d88325bb88129fc 100644
--- a/sflphone-gtk/configure.ac
+++ b/sflphone-gtk/configure.ac
@@ -1,8 +1,8 @@
-AC_INIT([SFLphone],[0.9.2],[sflphoneteam@savoirfairelinux.com],[sflphone])
+AC_INIT([SFLphone],[0.9.2-7],[sflphoneteam@savoirfairelinux.com],[sflphone])
AM_CONFIG_HEADER(config.h)
PACKAGE=SFLphone
-VERSION=0.9.2
+VERSION=0.9.2-7
AM_INIT_AUTOMAKE($PACKAGE,$VERSION)
diff --git a/sflphone-gtk/src/dbus.c b/sflphone-gtk/src/dbus.c
index 7ed21b2118a4241b4dffd5ef73e120a12848ffe4..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,
@@ -172,18 +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 ();
connection = dbus_g_bus_get (DBUS_BUS_SESSION, &error);
- if (connection == NULL)
+ if (error)
{
g_printerr ("Failed to open connection to bus: %s\n",
error->message);
@@ -191,13 +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;
@@ -206,11 +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;
@@ -252,10 +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");
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 90953f1607ffb826cc2fe4279696b3d02f400df9..75fc40c9e8d4e1a01bb35f361c1cbd6aa4d3f02d 100644
--- a/src/Makefile.am
+++ b/src/Makefile.am
@@ -62,6 +62,8 @@ sflphoned_LDADD = \
@PULSEAUDIO_LIBS@ \
@SAMPLERATE_LIBS@
+#sflphoned_LDFLAGS=-pg
+
noinst_LTLIBRARIES = libsflphone.la
noinst_HEADERS = \
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/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/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/sipvoiplink.cpp b/src/sipvoiplink.cpp
index d70e1221216de4257fe15373b637d39446152286..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
diff --git a/src/useragent.cpp b/src/useragent.cpp
index fb527784e0a41d0754228debb329860589370d0c..6e0e104cec4ada90606d980fbf15212e2870e450 100644
--- a/src/useragent.cpp
+++ b/src/useragent.cpp
@@ -1195,16 +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;