diff --git a/src/audio/audiortp.cpp b/src/audio/audiortp.cpp
index 222812e244459eeddea8089e3d1d77bece36d101..e9de9be9ab9aa7f5b6786f22b4300f08f23613ef 100644
--- a/src/audio/audiortp.cpp
+++ b/src/audio/audiortp.cpp
@@ -53,6 +53,7 @@ AudioRtp::~AudioRtp (void) {
 
 int 
 AudioRtp::createNewSession (SIPCall *ca) {
+   
     ost::MutexLock m(_threadMutex);
 
     // something should stop the thread before...
@@ -87,6 +88,7 @@ AudioRtp::closeRtpSession () {
   // This will make RTP threads finish.
   // _debug("Stopping AudioRTP\n");
   try {
+    
     delete _RTXThread; _RTXThread = 0;
   } catch(...) {
     _debugException("! ARTP Exception: when stopping audiortp\n");
@@ -140,7 +142,7 @@ AudioRtpRTX::~AudioRtpRTX () {
     _debugException("! ARTP: Thread destructor didn't terminate correctly");
     throw;
   }
-  //_debug("terminate audiortprtx ended...\n");
+  _debug("terminate audiortprtx ended...\n");
   _ca = 0;
   if (!_sym) {
     delete _sessionRecv; _sessionRecv = NULL;
@@ -148,7 +150,7 @@ AudioRtpRTX::~AudioRtpRTX () {
   } else {
     delete _session;     _session = NULL;
   }
-
+ 
   delete [] micData;  micData = NULL;
   delete [] micDataConverted;  micDataConverted = NULL;
   delete [] micDataEncoded;  micDataEncoded = NULL;
@@ -159,6 +161,9 @@ AudioRtpRTX::~AudioRtpRTX () {
   delete time; time = NULL;
 
   delete converter; converter = NULL;
+
+  _debug("AudioRtpRTX::~AudioRtpRTX () : Deleting the call instance \n");
+  
 }
 
   void
diff --git a/src/sipvoiplink.cpp b/src/sipvoiplink.cpp
index 356863ad8db1e767822a95ee629ead89ebfd29e7..b7b2365aa3bba3274f694f05689f76de751dd952 100644
--- a/src/sipvoiplink.cpp
+++ b/src/sipvoiplink.cpp
@@ -210,6 +210,7 @@ SIPVoIPLink::terminate()
     void
 SIPVoIPLink::terminateSIPCall()
 {
+    _debug("SIPVoIPLink::terminateSIPCall(): function called");
     ost::MutexLock m(_callMapMutex);
     CallMap::iterator iter = _callMap.begin();
     SIPCall *call;
@@ -477,7 +478,7 @@ SIPVoIPLink::hangup(const CallID& id)
         return false;
 
     call->getInvSession()->mod_data[getModId()] = NULL;
-    return true;
+    
 
     // Release RTP thread
     if (Manager::instance().isCurrentCall(id)) {
@@ -906,63 +907,63 @@ std::string SIPVoIPLink::getSipTo(const std::string& to_url, std::string hostnam
         return true;
     }
 
-    void
-        SIPVoIPLink::SIPCallServerFailure(SIPCall *call) 
-        {
-            //if (!event->response) { return; }
-            //switch(event->response->status_code) {
-            //case SIP_SERVICE_UNAVAILABLE: // 500
-            //case SIP_BUSY_EVRYWHERE:     // 600
-            //case SIP_DECLINE:             // 603
-            //SIPCall* call = findSIPCallWithCid(event->cid);
-            if (call != 0) {
-                _debug("Server error!\n");
-                CallID id = call->getCallId();
-                Manager::instance().callFailure(id);
-                removeCall(id);
-            }
-            //break;
-            //}
-        }
+void
+SIPVoIPLink::SIPCallServerFailure(SIPCall *call) 
+{
+    //if (!event->response) { return; }
+    //switch(event->response->status_code) {
+    //case SIP_SERVICE_UNAVAILABLE: // 500
+    //case SIP_BUSY_EVRYWHERE:     // 600
+    //case SIP_DECLINE:             // 603
+    //SIPCall* call = findSIPCallWithCid(event->cid);
+    if (call != 0) {
+        _debug("Server error!\n");
+        CallID id = call->getCallId();
+        Manager::instance().callFailure(id);
+        removeCall(id);
+    }
+    //break;
+    //}
+}
 
-    void
-        SIPVoIPLink::SIPCallClosed(SIPCall *call) 
-        {
-            // it was without did before
-            //SIPCall* call = findSIPCallWithCid(event->cid);
-            if (!call) { return; }
+void
+SIPVoIPLink::SIPCallClosed(SIPCall *call) 
+{
+    // it was without did before
+    //SIPCall* call = findSIPCallWithCid(event->cid);
+    if (!call) { return; }
 
-            CallID id = call->getCallId();
-            //call->setDid(event->did);
-            if (Manager::instance().isCurrentCall(id)) {
-                call->setAudioStart(false);
-                _debug("* SIP Info: Stopping AudioRTP when closing\n");
-                _audiortp->closeRtpSession();
-            }
-            _debug("After close RTP\n");
-            Manager::instance().peerHungupCall(id);
-            removeCall(id);
-            _debug("After remove call ID\n");
-        }
+    CallID id = call->getCallId();
+    //call->setDid(event->did);
+    if (Manager::instance().isCurrentCall(id)) {
+        call->setAudioStart(false);
+        _debug("* SIP Info: Stopping AudioRTP when closing\n");
+        _audiortp->closeRtpSession();
+    }
+    _debug("After close RTP\n");
+    Manager::instance().peerHungupCall(id);
+    removeCall(id);
+    _debug("After remove call ID\n");
+}
 
-    void
-        SIPVoIPLink::SIPCallReleased(SIPCall *call)
-        {
-            // do cleanup if exists
-            // only cid because did is always 0 in these case..
-            //SIPCall* call = findSIPCallWithCid(event->cid);
-            if (!call) { return; }
-
-            // if we are here.. something when wrong before...
-            _debug("SIP call release\n");
-            CallID id = call->getCallId();
-            Manager::instance().callFailure(id);
-            removeCall(id);
-        }
+void
+SIPVoIPLink::SIPCallReleased(SIPCall *call)
+{
+    // do cleanup if exists
+    // only cid because did is always 0 in these case..
+    //SIPCall* call = findSIPCallWithCid(event->cid);
+    if (!call) { return; }
+
+    // if we are here.. something when wrong before...
+    _debug("SIP call release\n");
+    CallID id = call->getCallId();
+    Manager::instance().callFailure(id);
+    removeCall(id);
+}
 
-    void
-        SIPVoIPLink::SIPCallAnswered(SIPCall *call, pjsip_rx_data *rdata)
-        {
+void
+SIPVoIPLink::SIPCallAnswered(SIPCall *call, pjsip_rx_data *rdata)
+{
             //SIPCall* call = dynamic_cast<SIPCall *>(theCall);//findSIPCallWithCid(event->cid);
             if (!call) {
                 _debug("! SIP Failure: unknown call\n");