Merged revisions 128951 via svnmerge from

https://origsvn.digium.com/svn/asterisk/trunk

................
r128951 | oej | 2008-07-08 12:02:12 +0200 (Tis, 08 Jul 2008) | 19 lines

Merged revisions 128950 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r128950 | oej | 2008-07-08 11:52:21 +0200 (Tis, 08 Jul 2008) | 11 lines

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?
*/

........

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@128952 65c4cc65-6c06-0410-ace0-fbb531ad65f3

channels/chan_sip.c

@@ -14719,21 +14719,17 @@ 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 (!req->ignore)
+				if (!p->route && !req->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 */