Adding new method to iaxvoiplink (answer, refuse, transfer, senddtmf)

TODO

@@ -17,7 +17,7 @@ For sflphone-qt:
 ---------------
 Add samplerate combobox if sample rate is compiled
 Save account status if modified in configuration
-
+Bug when moving sflphone and clicking inside the lcd
 
 
 From FIXME:

src/audio/audiortp.cpp

@@ -54,8 +54,9 @@ AudioRtp::createNewSession (SIPCall *ca) {
 
   // something should stop the thread before...
   if ( _RTXThread != 0 ) { 
-    _debug("Try to create a new audio rtp thread...\n");
-    return -1; 
+    _debug("AudioRTP Failure: Thread already exists..., stopping it\n");
+    delete _RTXThread; _RTXThread = 0;
+    //return -1; 
   }
 
   // Start RTP Send/Receive threads
@@ -64,6 +65,7 @@ AudioRtp::createNewSession (SIPCall *ca) {
 
   try {
     if (_RTXThread->start() != 0) {
+      _debug("AudioRTP Failure: unable to start RTX Thread\n");
       return -1;
     }
   } catch(...) {

src/iaxvoiplink.cpp

@@ -292,6 +292,19 @@ IAXVoIPLink::newOutgoingCall(const CallID& id, const std::string& toUrl)
   return call;
 }
 
+bool 
+IAXVoIPLink::answer(const CallID& id) 
+{
+  IAXCall* call = getIAXCall(id);
+  if (call==0) { _debug("Call doesn't exists\n"); return false; }
+  _mutexIAX.enterMutex();
+  iax_answer(call->getSession());
+  _mutexIAX.leaveMutex();
+  call->setState(Call::Active);
+  call->setConnectionState(Call::Connected);
+  return true;
+}
+
 bool 
 IAXVoIPLink::hangup(const CallID& id)
 {
@@ -335,6 +348,44 @@ IAXVoIPLink::offhold(const CallID& id)
   return true;
 }
 
+bool 
+IAXVoIPLink::transfer(const CallID& id, const std::string& to)
+{
+  IAXCall* call = getIAXCall(id);
+  if (call==0) { _debug("Call doesn't exists\n"); return false; }
+
+  char callto[to.length()+1];
+  strcpy(callto, to.c_str());
+  
+  _mutexIAX.enterMutex();
+  iax_transfer(call->getSession(), callto); 
+  _mutexIAX.leaveMutex();
+
+  // should we remove it?
+  // removeCall(id);
+}
+
+bool 
+IAXVoIPLink::refuse(const CallID& id)
+{
+  IAXCall* call = getIAXCall(id);
+  if (call==0) { _debug("Call doesn't exists\n"); return false; }
+  _mutexIAX.enterMutex();
+  iax_reject(call->getSession(), "Call rejected manually.");
+  _mutexIAX.leaveMutex();
+  removeCall(id);
+}
+
+bool
+IAXVoIPLink::carryingDTMFdigits(const CallID& id, char code)
+{
+  IAXCall* call = getIAXCall(id);
+  if (call==0) { _debug("Call doesn't exists\n"); return false; }
+  _mutexIAX.enterMutex();
+  iax_send_dtmf(call->getSession(), code);
+  _mutexIAX.leaveMutex();
+}
+
 bool
 IAXVoIPLink::iaxOutgoingInvite(IAXCall* call) 
 {

src/iaxvoiplink.h

@@ -51,15 +51,15 @@ public:
   bool setUnregister (void);
 
   Call* newOutgoingCall(const CallID& id, const std::string& toUrl);
-  bool answer(const CallID& id) {return false;}
+  bool answer(const CallID& id);
 
   bool hangup(const CallID& id);
   bool cancel(const CallID& id) { return false; }
   bool onhold(const CallID& id);
   bool offhold(const CallID& id);
-  bool transfer(const CallID& id, const std::string& to) { return false; }
-  bool refuse (const CallID& id) { return false; }
-  bool carryingDTMFdigits(const CallID& id, char code) { return false; }
+  bool transfer(const CallID& id, const std::string& to);
+  bool refuse (const CallID& id);
+  bool carryingDTMFdigits(const CallID& id, char code);
   bool sendMessage(const std::string& to, const std::string& body) { return false; }
 
 public: // iaxvoiplink only

src/sipvoiplink.cpp

@@ -613,6 +613,13 @@ SIPVoIPLink::onhold(const CallID& id)
   SIPCall* call = getSIPCall(id);
   if (call==0) { _debug("Call doesn't exist\n"); return false; }  
 
+  // Stop sound
+  call->setAudioStart(false);
+  call->setState(Call::Hold);
+  _debug("SIP: Stopping AudioRTP when onhold\n");
+  _audiortp.closeRtpSession();
+
+
   int did = call->getDid();
 
   eXosip_lock ();
@@ -661,12 +668,6 @@ SIPVoIPLink::onhold(const CallID& id)
     osip_message_set_content_type (invite, "application/sdp");
   }
   
-  // Stop sound
-  call->setAudioStart(false);
-  call->setState(Call::Hold);
-  _debug("SIP: Stopping AudioRTP when onhold\n");
-  _audiortp.closeRtpSession();
-
   // send request
   _debug("< Send on hold request\n");
   eXosip_lock ();

tools/sflphone-bash/call

@@ -30,7 +30,7 @@ if [ -z `$pidof 'sflphoned'` ]; then
   echo -en "\n";
 fi
 
-callstring="call someone with $callid $telephonenumber\n"
+callstring="call someone SIP0 $callid $telephonenumber\n"
 echo "Calling $telephonenumber..."
 
 echo -en $callstring | $netcat