[asterisk-commits] oej: branch 1.6.0 r128952 - in /branches/1.6.0: ./ channels/chan_sip.c

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Tue Jul 8 05:06:33 CDT 2008


Author: oej
Date: Tue Jul  8 05:06:32 2008
New Revision: 128952

URL: http://svn.digium.com/view/asterisk?view=rev&rev=128952
Log:
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

........
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?
*/

........

................

Modified:
    branches/1.6.0/   (props changed)
    branches/1.6.0/channels/chan_sip.c

Propchange: branches/1.6.0/
------------------------------------------------------------------------------
Binary property 'trunk-merged' - no diff available.

Modified: branches/1.6.0/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/branches/1.6.0/channels/chan_sip.c?view=diff&rev=128952&r1=128951&r2=128952
==============================================================================
--- branches/1.6.0/channels/chan_sip.c (original)
+++ branches/1.6.0/channels/chan_sip.c Tue Jul  8 05:06:32 2008
@@ -14719,21 +14719,17 @@
 		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 */




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