Fixed second call audio codec name broadcasting to client

po/es.po

@@ -114,6 +114,7 @@ msgid "%d voice mail"
 msgstr "%d mensaje de voz"
 
 #: ../sflphone-gtk/src/actions.c:80
+#, c-format
 msgid "%s account- %s"
 msgstr "%s cuenta: %s"
 
@@ -304,6 +305,7 @@ msgid "Ringtones"
 msgstr "_Permitir los ringtones"
 
 #: ../sflphone-gtk/src/calltree.c:209
+#, fuzzy, c-format
 msgid "\"Voicemail\" <%s>"
 msgstr "_Numero de Caja de Voz"
 
@@ -625,6 +627,7 @@ msgid "_Show main window"
 msgstr "Mostrar ventana principal"
 
 #: ../sflphone-gtk/src/statusicon.c:124
+#, c-format
 msgid "SFLphone - %i accounts registered"
 msgstr "SFLphone - %i cuenta registrada"
 

src/audio/audiortp.cpp

@@ -182,15 +182,17 @@ AudioRtpRTX::initBuffers()
   void
 AudioRtpRTX::initAudioRtpSession (void) 
 {
+  _debug("AudioRtpRTX::initAudioRtpSession: call %s \n",_ca->getCallId().c_str());
   try {
     if (_ca == 0) { return; }
+    
     _audiocodec = Manager::instance().getCodecDescriptorMap().getCodec( _ca->getAudioCodec() );
-    _codecSampleRate = _audiocodec->getClockRate();	
+    
+    _codecSampleRate = _audiocodec->getClockRate();
 
-    _debug("Init audio RTP session\n");
     ost::InetHostAddress remote_ip(_ca->getRemoteIp().c_str());
+    
     if (!remote_ip) {
-      _debug("! ARTP Thread Error: Target IP address [%s] is not correct!\n", _ca->getRemoteIp().data());
       return;
     }
 
@@ -249,6 +251,7 @@ AudioRtpRTX::initAudioRtpSession (void)
     _debugException("! ARTP Failure: initialisation failed");
     throw;
   }
+  _debug("AudioRtpRTX::initAudioRtpSession: Init audio RTP sessionend of method\n");
 }
 
   void
@@ -442,7 +445,7 @@ AudioRtpRTX::run () {
 
     audiolayer->startStream();
     _start.post();
-    _debug("- ARTP Action: Start\n");
+    _debug("- ARTP Action: Start call %s\n",_ca->getCallId().c_str());
     while (!testCancel()) {
       
       // printf("AudioRtpRTX::run() _session->getFirstTimestamp() %i \n",_session->getFirstTimestamp());
@@ -452,12 +455,14 @@ AudioRtpRTX::run () {
       ////////////////////////////
       // Send session
       ////////////////////////////
-      sendSessionFromMic(timestamp);
+      sendSessionFromMic(timestamp); 
       timestamp += step;
+      
       ////////////////////////////
       // Recv session
       ////////////////////////////
       receiveSessionForSpkr(countTime);
+      
       // Let's wait for the next transmit cycle
 
       if(_session->isWaiting())
@@ -465,13 +470,14 @@ AudioRtpRTX::run () {
       else
         _ca->recAudio.recData(micData,_nSamplesMic);
 
-      Thread::sleep(TimerPort::getTimer()); 
+      Thread::sleep(TimerPort::getTimer());
       TimerPort::incTimer(_layerFrameSize); // 'frameSize' ms
+      
     }
-
+    
     // _debug("stop stream for audiortp loop\n");
     audiolayer->stopStream();
-    _debug("- ARTP Action: Stop\n");
+    _debug("- ARTP Action: Stop call %s\n",_ca->getCallId().c_str());
   //} catch(std::exception &e) {
     //_start.post();
     //_debug("! ARTP: Stop %s\n", e.what());

src/call.cpp

@@ -122,7 +122,6 @@ Call::getRemoteIp()
 AudioCodecType 
 Call::getAudioCodec()
 {
-  ost::MutexLock m(_callMutex);  
   return _audioCodec;  
 }
 

src/iaxvoiplink.cpp

@@ -43,6 +43,7 @@
     IAXVoIPLink::IAXVoIPLink(const AccountID& accountID)
 : VoIPLink(accountID)
 {
+    // _debug("IAXVoIPLink::IAXVoIPLink : creating eventhread \n ");
     _evThread = new EventThread(this);
     _regSession = NULL;
     _nextRefreshStamp = 0;
@@ -88,6 +89,7 @@ IAXVoIPLink::init()
     if (initDone())
         return false;
 
+    
     bool returnValue = false;
     // _localAddress = "127.0.0.1";
     // port 0 is default

src/main.cpp

@@ -110,6 +110,7 @@ main (int argc, char **argv) {
     }
     bool initOK = false;
     try {
+      _debug("Initialize manager in main \n");
       Manager::instance().initConfigFile();
       Manager::instance().init();
       initOK = true;

src/managerimpl.cpp

@@ -221,6 +221,7 @@ ManagerImpl::outgoingCall(const std::string& accountid, const CallID& id, const
   bool
 ManagerImpl::answerCall(const CallID& id)
 {
+
   stopTone(false); 
   _debug("Try to answer call: %s\n", id.data());
   AccountID accountid = getAccountFromCall( id );
@@ -240,14 +241,15 @@ ManagerImpl::answerCall(const CallID& id)
     return false;
   }
 
-  std::string codecName = getCurrentCodecName(id);
-  _debug("ManagerImpl::hangupCall(): broadcast codec name %s \n",codecName.c_str());
-  if (_dbus) _dbus->getCallManager()->currentSelectedCodec(id,codecName.c_str());
-  
   // if it was waiting, it's waiting no more
   if (_dbus) _dbus->getCallManager()->callStateChanged(id, "CURRENT");
   removeWaitingCall(id);
   switchCall(id);
+ 
+  std::string codecName = getCurrentCodecName(id);
+  _debug("ManagerImpl::hangupCall(): broadcast codec name %s \n",codecName.c_str());
+  if (_dbus) _dbus->getCallManager()->currentSelectedCodec(id,codecName.c_str());
+
   return true;
 }
 
@@ -367,6 +369,10 @@ ManagerImpl::offHoldCall(const CallID& id)
   }
   switchCall(id);
 
+  std::string codecName = getCurrentCodecName(id);
+  _debug("ManagerImpl::hangupCall(): broadcast codec name %s \n",codecName.c_str());
+  if (_dbus) _dbus->getCallManager()->currentSelectedCodec(id,codecName.c_str());
+
   return returnValue;
 }
 
@@ -748,7 +754,7 @@ bool ManagerImpl::playATone(Tone::TONEID toneId)
         return false;
     
     audiolayer = getAudioDriver();
-
+   
     if (_telephoneTone != 0) {
         _toneMutex.enterMutex();
         _telephoneTone->setCurrentTone(toneId);

src/sipvoiplink.cpp

@@ -446,7 +446,7 @@ SIPVoIPLink::answer(const CallID& id)
     pj_status_t status;
     pjsip_tx_data *tdata;
 
-    _debug("- SIP Action: start answering\n");
+    _debug("SIPVoIPLink::answer: start answering \n");
 
     call = getSIPCall(id);
 
@@ -458,22 +458,23 @@ SIPVoIPLink::answer(const CallID& id)
     // User answered the incoming call, tell peer this news
     if (call->startNegociation(_pool)) {
         // Create and send a 200(OK) response
-        _debug("UserAgent: Negociation success!\n");
+        _debug("SIPVoIPLink::answer:UserAgent: Negociation success! : call %s \n", call->getCallId().c_str());
         status = pjsip_inv_answer(call->getInvSession(), PJSIP_SC_OK, NULL, NULL, &tdata);
         PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);
         status = pjsip_inv_send_msg(call->getInvSession(), tdata);
         PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);
 
-        _debug("* SIP Info: Starting AudioRTP when answering\n");
+        _debug("SIPVoIPLink::answer: Starting AudioRTP when answering : call %s \n", call->getCallId().c_str());
         if (_audiortp->createNewSession(call) >= 0) {
             call->setAudioStart(true);
             call->setConnectionState(Call::Connected);
             call->setState(Call::Active);
             return true;
         } else {
-            _debug("! SIP Failure: Unable to start sound when answering %s/%d\n", __FILE__, __LINE__);
+            _debug("SIPVoIPLink::answer: Unable to start sound when answering %s/%d\n", __FILE__, __LINE__);
         }
     }
+    _debug("SIPVoIPLink::answer: fail terminate call %s \n",call->getCallId().c_str());
     terminateOneCall(call->getCallId());
     removeCall(call->getCallId());
     return false;
@@ -804,10 +805,16 @@ std::string
 SIPVoIPLink::getCurrentCodecName()
 {
 
-  // _debug("SIPVoIPLink::getCurrentCodecName : \n");
+  _debug("SIPVoIPLink::getCurrentCodecName Ok 1 : \n");
+ 
+  printf("AAAAAAAAAAAAAHHHHHHH!!!!!!:: %s \n",Manager::instance().getCurrentCallId().c_str());
 
   SIPCall *call = getSIPCall(Manager::instance().getCurrentCallId());
   
+  _debug("SIPVoIPLink::getCurrentCodecName Ok 2 : \n");
+  
+  printf("OOOOOOOUUUUUUPPPPPPPPPPSSSSSSSSSSS!!!!!!:: %s \n",call->getCallId().c_str());  
+
   AudioCodec *ac = call->getCodecMap().getCodec(call->getAudioCodec());
 
   return ac->getCodecName();