Bug fix

sflphone-gtk/src/actions.c

@@ -207,6 +207,7 @@ sflphone_pick_up()
         			printf("accountID=%s\n",selectedCall->accountID);
 				break;
 			case CALL_STATE_INCOMING:
+				printf("CALL ID = %s\n", selectedCall->callID);
 				dbus_accept (selectedCall);
 				break;
 			case CALL_STATE_HOLD:

sflphone-gtk/src/configwindow.c

@@ -199,8 +199,8 @@ select_codec( GtkComboBox* wid)
 	guint item = gtk_combo_box_get_active(wid);
 	/* now we want this selected codec to be used as the preferred codec */
 	/* ie first in the list in the user config */
-	codec_set_prefered_order(item);
-	dbus_set_prefered_codec(codec_list_get_nth(0)->name);  
+	//codec_set_prefered_order(item);
+	//dbus_set_prefered_codec(codec_list_get_nth(0)->name);  
 }
 
 void

src/audio/audiortp.cpp

@@ -66,7 +66,7 @@ AudioRtp::createNewSession (SIPCall *ca) {
 	// Start RTP Send/Receive threads
 	_symmetric = Manager::instance().getConfigInt(SIGNALISATION,SYMMETRIC) ? true : false;
 	_RTXThread = new AudioRtpRTX (ca, _symmetric);
-
+	printf("_sym=%i\n\n", _symmetric);
 	try {
 		if (_RTXThread->start() != 0) {
 			_debug("! ARTP Failure: unable to start RTX Thread\n");
@@ -111,7 +111,6 @@ AudioRtpRTX::AudioRtpRTX (SIPCall *sipcall, bool sym)
 	// TODO: this should be the local ip not the external (router) IP
 	std::string localipConfig = _ca->getLocalIp(); // _ca->getLocalIp();
 	ost::InetHostAddress local_ip(localipConfig.c_str());
-
 	if (!_sym) {
 		_sessionRecv = new ost::RTPSession(local_ip, _ca->getLocalAudioPort());
 		_sessionSend = new ost::RTPSession(local_ip, _ca->getLocalAudioPort());
@@ -182,7 +181,7 @@ AudioRtpRTX::initAudioRtpSession (void)
 	try {
 		if (_ca == 0) { return; }
 
-		//_debug("Init audio RTP session\n");
+		_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());
@@ -224,7 +223,7 @@ AudioRtpRTX::initAudioRtpSession (void)
 			_sessionSend->setMark(true);
 		} else {
 
-			//_debug("AudioRTP Thread: Added session destination %s:%d\n", remote_ip.getHostname(), (unsigned short) _ca->getRemoteSdpAudioPort());
+			//_debug("AudioRTP Thread: Added session destination %s\n", remote_ip.getHostname() );
 
 			if (!_session->addDestination (remote_ip, (unsigned short) _ca->getRemoteAudioPort())) {
 				return;
@@ -330,13 +329,14 @@ AudioRtpRTX::receiveSessionForSpkr (int& countTime)
 			adu = _session->getData(_session->getFirstTimestamp());
 		}
 		if (adu == NULL) {
+			//_debug("No RTP audio stream\n");
 			return;
 		}
 
 		int payload = adu->getType(); // codec type
 		unsigned char* data  = (unsigned char*)adu->getData(); // data in char
 		unsigned int size    = adu->getSize(); // size in char
-		  
+
 
 		// Decode data with relevant codec
 		AudioCodec* audiocodec = _ca->getCodecMap().getCodec((CodecType)payload);
@@ -359,7 +359,7 @@ AudioRtpRTX::receiveSessionForSpkr (int& countTime)
 				_debug("We have decoded an RTP packet larger than expected: %s VS %s. Cropping.\n", nbInt16, max);
 				nbInt16=max;
 			}
-
+		
 			SFLDataFormat* toAudioLayer;
 			int nbSample = nbInt16;
 
@@ -501,7 +501,6 @@ AudioRtpRTX::run () {
 			// Recv session
 			////////////////////////////
 			receiveSessionForSpkr(countTime);
-
 			// Let's wait for the next transmit cycle
 			Thread::sleep(TimerPort::getTimer());
 			TimerPort::incTimer(_layerFrameSize); // 'frameSize' ms

src/managerimpl.cpp

@@ -166,6 +166,7 @@ ManagerImpl::isCurrentCall(const CallID& callId) {
 bool
 ManagerImpl::hasCurrentCall() {
   ost::MutexLock m(_currentCallMutex);
+  _debug("Current call ID = %s\n", _currentCallId2.c_str());
   if ( _currentCallId2 != "") {
     return true;
   }
@@ -220,10 +221,10 @@ bool
 ManagerImpl::answerCall(const CallID& id)
 {
   stopTone(false); 
-  if (hasCurrentCall()) 
+  /*if (hasCurrentCall()) 
   { 
     onHoldCall(getCurrentCallId());
-  }
+  }*/
   AccountID accountid = getAccountFromCall( id );
   if (accountid == AccountNULL) {
     _debug("Answering Call: Call doesn't exists\n");
@@ -611,6 +612,7 @@ ManagerImpl::incomingCall(Call* call, const AccountID& accountId)
   associateCallToAccount(call->getCallId(), accountId);
 
   if ( !hasCurrentCall() ) {
+    _debug("INCOMING CALL!!!!!!\n");
     call->setConnectionState(Call::Ringing);
     ringtone();
     switchCall(call->getCallId());
@@ -990,7 +992,7 @@ ManagerImpl::behindNat(const std::string& svr, int port)
 {
   StunAddress4 stunSvrAddr;
   stunSvrAddr.addr = 0;
-  
+   
   // Convert char* to StunAddress4 structure
   bool ret = stunParseServerName ((char*)svr.data(), stunSvrAddr);
   if (!ret) {
@@ -1085,9 +1087,12 @@ void
 ManagerImpl::initAudioCodec (void)
 {
   _debugInit("Active Codecs");
-  _codecDescriptorMap.setActive(getConfigString("Audio", "Codecs.codec1"));
+  //_codecDescriptorMap.setActive(getConfigString("Audio", "Codecs.codec1"));
   //_codecDescriptorMap.setActive(getConfigString("Audio", "Codec.codec2"));
   //_codecDescriptorMap.setActive(getConfigString("Audio", "Codec.codec3"));
+  _codecDescriptorMap.setActive("G711a");
+  _codecDescriptorMap.setActive("G711u");
+  _codecDescriptorMap.setActive("GSM");
 }
 
 void
@@ -1102,8 +1107,10 @@ ManagerImpl::setPreferedCodec(const ::DBus::String& codec_name)
 	list[0] = list[i];
 	list[i] = tmp; 
 	_codecDescriptorMap.setActive(list[0]);
-	_codecDescriptorMap.setInactive(list[1]);
-	_codecDescriptorMap.setInactive(list[2]);
+	//_codecDescriptorMap.setInactive(list[1]);
+	//_codecDescriptorMap.setInactive(list[2]);
+	_codecDescriptorMap.setActive(list[1]);
+	_codecDescriptorMap.setActive(list[2]);
         setConfig("Audio", "Codecs.codec1", list[0]);	
 	setConfig("Audio", "Codecs.codec2", list[1]);
 	setConfig("Audio", "Codecs.codec3", list[2]);

src/sipaccount.cpp

@@ -20,7 +20,7 @@
 #include "sipaccount.h"
 #include "sipvoiplink.h"
 #include "manager.h"
-
+#include "user_cfg.h"
 
 SIPAccount::SIPAccount(const AccountID& accountID)
  : Account(accountID)
@@ -43,8 +43,7 @@ SIPAccount::registerVoIPLink()
   int useStun = Manager::instance().getConfigInt(_accountID,SIP_USE_STUN);
   
   SIPVoIPLink* thislink = dynamic_cast<SIPVoIPLink*> (_link);
-
-  thislink->setStunServer(Manager::instance().getConfigString(_accountID,SIP_STUN_SERVER));
+  thislink->setStunServer(Manager::instance().getConfigString(_accountID,DFT_STUN_SERVER));
   thislink->setUseStun( useStun!=0 ? true : false);
 
   _link->init();

src/sipvoiplink.cpp

@@ -951,7 +951,9 @@ SIPVoIPLink::SIPStartCall(SIPCall* call, const std::string& subject)
   CodecMap::iterator iter = map.begin();
 
   while(iter != map.end()) {
+    printf("codec = %s\n",iter->second->getCodecName().data());
     if (iter->second!=0 && iter->second->isActive()) {
+    printf("codec = %s\n",iter->second->getCodecName().data());
       payload = iter->first;
       // add each payload in the list of payload
       media_audio << payload << " ";

src/user_cfg.h

@@ -56,8 +56,8 @@
 #define PLAY_TONES      "DTMF.playTones" 
 #define PULSE_LENGTH	"DTMF.pulseLength"
 #define SEND_DTMF_AS	"DTMF.sendDTMFas"
-//#define STUN_SERVER		"STUN.STUNserver"
-//#define USE_STUN		"STUN.useStun"
+#define STUN_SERVER		"STUN.STUNserver"
+#define USE_STUN		"STUN.useStun"
 #define DRIVER_NAME		"Drivers.driverName"
 #define DRIVER_NAME_IN		"Drivers.driverNameIn"
 #define DRIVER_NAME_OUT		"Drivers.driverNameOut"