SIP transfer OK

The dtmf's was sent. Bad

sflphone-gtk/src/actions.c

@@ -55,6 +55,28 @@ sflphone_notify_voice_mail (guint count)
 		status_bar_message_add(message,  __MSG_VOICE_MAILS);
 		g_free(message);
 	}
+	// TODO: add ifdef
+	if( account_list_get_size() > 0 )
+	{
+	  account_t* acc = account_list_get_by_state( ACCOUNT_STATE_REGISTERED );
+	  if( acc == NULL )
+	  {
+	    // Notify that no account is registered
+	    //notify_no_account_registered();
+	  }
+	  else
+	  {
+	    if( account_list_get_default() == NULL ){
+	      // Notify that the first registered account has count voice mails
+	      notify_voice_mails( count , acc );	
+	    }
+	    else
+	    {
+	      // Notify that the default registered account has count voice mails
+	      notify_voice_mails( count , account_list_get_by_id(account_list_get_default()) );
+	    } 
+	  }     
+	}
 }
 
 void
@@ -345,6 +367,7 @@ sflphone_unset_transfert()
 	}
 	toolbar_update_buttons();
 }
+
 	void
 sflphone_incoming_call (call_t * c) 
 {
@@ -357,7 +380,7 @@ sflphone_incoming_call (call_t * c)
 void process_dialing(call_t * c, guint keyval, gchar * key)
 {
 	// We stop the tone
-	if(strlen(c->to) == 0){
+	if(strlen(c->to) == 0 && c->state != CALL_STATE_TRANSFERT){
 	  dbus_start_tone( FALSE , 0 );
 	  dbus_play_dtmf( key );
 	}
@@ -421,10 +444,10 @@ void process_dialing(call_t * c, guint keyval, gchar * key)
 
 call_t * sflphone_new_call()
 {
+	// Play a tone when creating a new call
 	if( call_list_get_size() == 0 )
-	{
 	  dbus_start_tone( TRUE , ( voice_mails > 0 )? TONE_WITH_MESSAGE : TONE_WITHOUT_MESSAGE) ;
-	}
+
 	call_t * c = g_new0 (call_t, 1);
 	c->state = CALL_STATE_DIALING;
 	c->from = g_strconcat("\"\" <>", NULL);

src/managerimpl.cpp

@@ -377,6 +377,7 @@ ManagerImpl::transferCall(const CallID& id, const std::string& to)
   if (_dbus) _dbus->getCallManager()->callStateChanged(id, "HUNGUP");
   switchCall("");
 
+
   return returnValue;
 }
 

src/sipvoiplink.cpp

@@ -840,6 +840,7 @@ SIPVoIPLink::transfer(const CallID& id, const std::string& to)
   }
   eXosip_unlock();
 
+  _audiortp.closeRtpSession();
   // shall we delete the call?
   //removeCall(id);
   return true;