[svn-commits] oej: branch 1.6.0 r126791 - in /branches/1.6.0: ./ channels/chan_sip.c

SVN commits to the Digium repositories svn-commits at lists.digium.com
Tue Jul 1 07:02:41 CDT 2008


Author: oej
Date: Tue Jul  1 07:02:41 2008
New Revision: 126791

URL: http://svn.digium.com/view/asterisk?view=rev&rev=126791
Log:
Merged revisions 126790 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/trunk

................
r126790 | oej | 2008-07-01 13:58:17 +0200 (Tis, 01 Jul 2008) | 14 lines

Merged revisions 126789 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r126789 | oej | 2008-07-01 13:51:38 +0200 (Tis, 01 Jul 2008) | 6 lines

Report 200 OK to all in-dialog OPTIONs requests (to confirm that the dialog
exist). Don't bother checking the request URI.

(closes issue #11264)
Reported by: ibc

........

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

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=126791&r1=126790&r2=126791
==============================================================================
--- branches/1.6.0/channels/chan_sip.c (original)
+++ branches/1.6.0/channels/chan_sip.c Tue Jul  1 07:02:41 2008
@@ -16036,6 +16036,13 @@
 		\todo Fix handle_request_options device handling with optional authentication
 			(this needs to be fixed in 1.4 as well)
 	*/
+
+	if (p->lastinvite) {
+		/* if this is a request in an active dialog, just confirm that the dialog exists. */
+		transmit_response_with_allow(p, "200 OK", req, 0);
+		return 0;
+	}
+
 	res = get_destination(p, req);
 	build_contact(p);
 
@@ -16051,8 +16058,7 @@
 
 	/* Destroy if this OPTIONS was the opening request, but not if
 	   it's in the middle of a normal call flow. */
-	if (!p->lastinvite)
-		sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
+	sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
 
 	return res;
 }




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