Function terminateSIPCall added in sipvoiplink and managerimpl

src/audio/audiortp.cpp

@@ -142,7 +142,7 @@ AudioRtpRTX::~AudioRtpRTX () {
     _debugException("! ARTP: Thread destructor didn't terminate correctly");
     throw;
   }
-  _debug("terminate audiortprtx ended...\n");
+  _debug("AudioRtpRTX::~AudioRtpRTX() :: terminate audiortprtx ended...\n");
   _ca = 0;
   if (!_sym) {
     delete _sessionRecv; _sessionRecv = NULL;
@@ -161,8 +161,6 @@ AudioRtpRTX::~AudioRtpRTX () {
   delete time; time = NULL;
 
   delete converter; converter = NULL;
-
-  _debug("AudioRtpRTX::~AudioRtpRTX () : Deleting the call instance \n");
   
 }
 

src/managerimpl.cpp

@@ -249,6 +249,7 @@ ManagerImpl::answerCall(const CallID& id)
   bool
 ManagerImpl::hangupCall(const CallID& id)
 {
+  _debug("ManagerImpl::hangupCall(): This function is called when user hangup \n");
     PulseLayer *pulselayer;
     AccountID accountid;
     bool returnValue;
@@ -370,6 +371,7 @@ ManagerImpl::transferCall(const CallID& id, const std::string& to)
   bool
 ManagerImpl::refuseCall (const CallID& id)
 {
+  _debug("ManagerImpl::refuseCall(): method called");
   stopTone(true);
   AccountID accountid = getAccountFromCall( id );
   if (accountid == AccountNULL) {
@@ -631,8 +633,10 @@ ManagerImpl::peerRingingCall(const CallID& id)
   void
 ManagerImpl::peerHungupCall(const CallID& id)
 {
+    _debug("ManagerImpl::peerHungupCall():this function is called when peer hangup \n");
     PulseLayer *pulselayer;
     AccountID accountid;
+    bool returnValue;
 
     accountid = getAccountFromCall( id );
     if (accountid == AccountNULL) {
@@ -648,6 +652,8 @@ ManagerImpl::peerHungupCall(const CallID& id)
         switchCall("");
     }
 
+    returnValue = getAccountLink(accountid)->hangup(id);
+
     removeWaitingCall(id);
     removeCallAccount(id);
   

src/sipvoiplink.cpp

@@ -149,6 +149,8 @@ SIPVoIPLink* SIPVoIPLink::_instance = NULL;
     , _useStun(false)
     , _clients(0)
 {
+    _debug("SIPVoIPLink::~SIPVoIPLink(): sipvoiplink constructor called \n");    
+
     // to get random number for RANDOM_PORT
     srand (time(NULL));
 
@@ -161,6 +163,7 @@ SIPVoIPLink* SIPVoIPLink::_instance = NULL;
 
 SIPVoIPLink::~SIPVoIPLink()
 {
+    _debug("SIPVoIPLink::~SIPVoIPLink(): sipvoiplink destructor called \n");
     terminate();
 }
 
@@ -210,7 +213,7 @@ SIPVoIPLink::terminate()
     void
 SIPVoIPLink::terminateSIPCall()
 {
-    _debug("SIPVoIPLink::terminateSIPCall(): function called");
+    _debug("SIPVoIPLink::terminateSIPCall(): function called \n");
     ost::MutexLock m(_callMapMutex);
     CallMap::iterator iter = _callMap.begin();
     SIPCall *call;
@@ -218,7 +221,7 @@ SIPVoIPLink::terminateSIPCall()
         call = dynamic_cast<SIPCall*>(iter->second);
         if (call) {
             // terminate the sip call
-      	    _debug("SIPVOIP::the call is deleted, should close recording file \n");
+      	    _debug("SIPVoIPLink::terminateSIPCall()::the call is deleted, should close recording file \n");
             delete call; call = 0;
         }
         iter++;
@@ -485,6 +488,8 @@ SIPVoIPLink::hangup(const CallID& id)
         _debug("* SIP Info: Stopping AudioRTP for hangup\n");
         _audiortp->closeRtpSession();
     }
+ 
+    terminateSIPCall();
 
     removeCall(id);
 
@@ -686,7 +691,7 @@ SIPVoIPLink::refuse (const CallID& id)
     pj_status_t status;
     pjsip_tx_data *tdata;
 
-
+    _debug("SIPVoIPLink::refuse() : teh call is refused \n");
     call = getSIPCall(id);
 
     if (call==0) { 
@@ -710,6 +715,8 @@ SIPVoIPLink::refuse (const CallID& id)
         return false;
 
     call->getInvSession()->mod_data[getModId()] = NULL;
+
+    terminateSIPCall();
     return true;
 }
 
@@ -929,6 +936,8 @@ SIPVoIPLink::SIPCallServerFailure(SIPCall *call)
 void
 SIPVoIPLink::SIPCallClosed(SIPCall *call) 
 {
+
+  _debug("SIPVoIPLink::SIPCallClosed():: function called when peer hangup");
     // it was without did before
     //SIPCall* call = findSIPCallWithCid(event->cid);
     if (!call) { return; }