Function terminateSIPCall added in sipvoiplink and managerimpl

Conflicts:

	src/audio/audiortp.cpp
	src/sipvoiplink.cpp

src/audio/audiortp.cpp

@@ -126,7 +126,6 @@ AudioRtpRTX::~AudioRtpRTX () {
     _debugException("! ARTP: Thread destructor didn't terminate correctly");
     throw;
   }
-  //_debug("terminate audiortprtx ended...\n");
   _ca = 0;
   if (!_sym) {
     delete _sessionRecv; _sessionRecv = NULL;
@@ -145,6 +144,7 @@ AudioRtpRTX::~AudioRtpRTX () {
   delete time; time = NULL;
 
   delete converter; converter = NULL;
+  
 }
 
   void

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) {
@@ -632,8 +634,10 @@ 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) {
@@ -649,6 +653,8 @@ 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));
 
@@ -158,6 +160,7 @@ SIPVoIPLink* SIPVoIPLink::_instance = NULL;
 
 SIPVoIPLink::~SIPVoIPLink()
 {
+    _debug("SIPVoIPLink::~SIPVoIPLink(): sipvoiplink destructor called \n");
     terminate();
 }
 
@@ -213,7 +216,6 @@ SIPVoIPLink::terminate()
     void
 SIPVoIPLink::terminateSIPCall()
 {
-
     ost::MutexLock m(_callMapMutex);
     CallMap::iterator iter = _callMap.begin();
     SIPCall *call;
@@ -488,6 +490,8 @@ SIPVoIPLink::hangup(const CallID& id)
         _debug("* SIP Info: Stopping AudioRTP for hangup\n");
         _audiortp->closeRtpSession();
     }
+ 
+    terminateSIPCall();
 
     removeCall(id);
 
@@ -689,7 +693,7 @@ 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) { 
@@ -713,6 +717,8 @@ SIPVoIPLink::refuse (const CallID& id)
         return false;
 
     call->getInvSession()->mod_data[getModId()] = NULL;
+
+    terminateSIPCall();
     return true;
 }