Don't hangup the call if we can't resolve the Contact if there's a proxy

route set for the call.
----
This comment was added a while ago and today it hit me badly. 

/* OEJ: Possible issue that may need a check:
	If we have a proxy route between us and the device,
	should we care about resolving the contact
	or should we just send it?
*/


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@128950 65c4cc65-6c06-0410-ace0-fbb531ad65f3

channels/chan_sip.c

@@ -12256,21 +12256,17 @@ static void handle_response_invite(struct sip_pvt *p, int resp, char *rest, stru
 		if (outgoing) {
 			update_call_counter(p, DEC_CALL_RINGING);
 			parse_ok_contact(p, req);
+			/* Save Record-Route for any later requests we make on this dialogue */
+			if (!reinvite)
+				build_route(p, req, 1);
+
 			if(set_address_from_contact(p)) {
 				/* Bad contact - we don't know how to reach this device */
 				/* We need to ACK, but then send a bye */
-				/* OEJ: Possible issue that may need a check:
-					If we have a proxy route between us and the device,
-					should we care about resolving the contact
-					or should we just send it?
-				*/
-				if (!ast_test_flag(req, SIP_PKT_IGNORE))
+				if (!p->route && !ast_test_flag(req, SIP_PKT_IGNORE))
 					ast_set_flag(&p->flags[0], SIP_PENDINGBYE);	
 			} 
 
-			/* Save Record-Route for any later requests we make on this dialogue */
-			if (!reinvite)
-				build_route(p, req, 1);
 		}
 		
 		if (p->owner && (p->owner->_state == AST_STATE_UP) && (bridgepeer = ast_bridged_channel(p->owner))) { /* if this is a re-invite */