[asterisk-commits] oej: branch 1.6.0 r234528 - in /branches/1.6.0: ./ channels/chan_sip.c

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Mon Dec 14 04:55:18 CST 2009


Author: oej
Date: Mon Dec 14 04:55:13 2009
New Revision: 234528

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=234528
Log:
Merged revisions 234526 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/trunk

................
r234526 | oej | 2009-12-14 11:46:20 +0100 (Mån, 14 Dec 2009) | 16 lines

Merged revisions 234492 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r234492 | oej | 2009-12-14 11:16:00 +0100 (Mån, 14 Dec 2009) | 8 lines

Stop sending 183's after call hangup.

There where still cases where the 183 keep-alive mechanism would not stop
sending 183's even though the Asterisk server had sent a final reply to
the invite.

EDVX-28

........

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

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://svnview.digium.com/svn/asterisk/branches/1.6.0/channels/chan_sip.c?view=diff&rev=234528&r1=234527&r2=234528
==============================================================================
--- branches/1.6.0/channels/chan_sip.c (original)
+++ branches/1.6.0/channels/chan_sip.c Mon Dec 14 04:55:13 2009
@@ -5213,6 +5213,7 @@
 				}
 			} else {	/* Incoming call, not up */
 				const char *res;
+				AST_SCHED_DEL(sched, p->provisional_keepalive_sched_id);
 				if (p->hangupcause && (res = hangup_cause2sip(p->hangupcause)))
 					transmit_response_reliable(p, res, &p->initreq);
 				else 




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