Merged revisions 234526 via svnmerge from

https://origsvn.digium.com/svn/asterisk/trunk

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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

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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: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@234533 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 38130dd925f6d998d93e85025813836104a49d83..06417a83ee9ba3546dbf1b86036ea116efb79387 100644 (file)
--- 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