Merge branch 'master' into m_savard

sflphone-gtk/src/actions.c

@@ -373,7 +373,6 @@ sflphone_off_hold ()
         g_print("Currently recording! \n");
     else
         g_print("Not recording currently \n");
-    
 }
 
 
@@ -701,11 +700,13 @@ sflphone_place_call ( call_t * c )
         {
             account_t * current;
 
-            if(c->accountID != 0)
+            if(g_strcasecmp(c->accountID, "") != 0) {
+                g_print ("account_list_get_by_id : %s\n", c->accountID);
                 current = account_list_get_by_id(c->accountID);
-            else
+            } else {
+                g_print ("account_list_get_current\n");
                 current = account_list_get_current();
-
+            }
             // printf("sflphone_place_call :: c->accountID : %i \n",c->accountID);
 
             // account_t * current = c->accountID;

src/sdp.cpp

@@ -379,7 +379,6 @@ void Sdp::set_local_media_capabilities () {
         }
     } 
     _local_media_cap.push_back (audio);
-    _debug ("%s\n", audio->to_string ().c_str());
 }
 
 void Sdp::attribute_port_to_all_media (int port) {
@@ -408,7 +407,6 @@ void Sdp::fetch_remote_ip_from_sdp (pjmedia_sdp_session *r_sdp) {
     std::string remote_ip;
 
     remote_ip = r_sdp->conn->addr.ptr;
-    _debug("**************************************************            Remote Audio IP: %s\n", remote_ip.c_str());
     this->set_remote_ip(remote_ip);
 }
 

src/sipvoiplink.cpp

@@ -1698,14 +1698,12 @@ std::string SIPVoIPLink::getSipTo(const std::string& to_url, std::string hostnam
 
             // The call is ringing - We need to handle this case only on outgoing call
             if (inv->state == PJSIP_INV_STATE_EARLY && e->body.tsx_state.tsx->role == PJSIP_ROLE_UAC){
-                _debug ("*************************** PJSIP_INV_STATE_EARLY - PEER RINGING ***********************************\n");
                 call->setConnectionState(Call::Ringing);
                 Manager::instance().peerRingingCall(call->getCallId());
             }
 
             // We receive a ACK - The connection is established
             else if( inv->state == PJSIP_INV_STATE_CONFIRMED ){
-                _debug ("*************************** PJSIP_INV_STATE_CONFIRMED ***********************************\n");
                     
                 /* If the call is a direct IP-to-IP call */
                 if (call->getCallConfiguration () == Call::IPtoIP) {
@@ -1721,7 +1719,6 @@ std::string SIPVoIPLink::getSipTo(const std::string& to_url, std::string hostnam
             }
 
             else if( inv->state == PJSIP_INV_STATE_DISCONNECTED ){
-                _debug ("*************************** PJSIP_INV_STATE_DISCONNECTED  %i***********************************\n", inv->cause);
                 switch( inv->cause )
                 {
                     /* The call terminates normally - BYE / CANCEL */
@@ -2443,8 +2440,6 @@ void call_on_tsx_changed(pjsip_inv_session *inv, pjsip_transaction *tsx, pjsip_e
 
     void on_rx_offer( pjsip_inv_session *inv, const pjmedia_sdp_session *offer ){
 
-        _debug ( "********************************* REINVITE RECEIVED *******************************\n" );
-
 #ifdef CAN_REINVITE
         _debug ("reinvite                                                  SIP\n");