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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

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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@234526 f38db490-d61c-443f-a65b-d21fe96a405b
This commit is contained in:
oej 2009-12-14 10:46:20 +00:00
parent 9a6e6a6b16
commit 0641077477
1 changed files with 1 additions and 0 deletions
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1
channels/chan_sip.c
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@ -6398,6 +6398,7 @@ static int sip_hangup(struct ast_channel *ast)
}
} 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