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authoroej <oej@f38db490-d61c-443f-a65b-d21fe96a405b>2009-12-14 11:12:45 +0000
committeroej <oej@f38db490-d61c-443f-a65b-d21fe96a405b>2009-12-14 11:12:45 +0000
commit2f8103000c1344be933509d0747df3df4447083a (patch)
tree8bcf99f053e95a0cae8aa6788e2eecbbbf6c1618
parentfe99b4acbd43591f7a1e961a33f86984f47a19fd (diff)
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 ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@234533 f38db490-d61c-443f-a65b-d21fe96a405b
-rw-r--r--channels/chan_sip.c1
1 files changed, 1 insertions, 0 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 38130dd92..06417a83e 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -5660,6 +5660,7 @@ 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