From 434e4745fc2c1f2dbeb973ed15925a03cf59f767 Mon Sep 17 00:00:00 2001 From: Olle Johansson Date: Mon, 14 Dec 2009 10:55:13 +0000 Subject: [PATCH] Merged revisions 234526 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk MIME-Version: 1.0 Content-Type: text/plain; charset=utf8 Content-Transfer-Encoding: 8bit ................ r234526 | oej | 2009-12-14 11:46:20 +0100 (MÃ¥n, 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 (MÃ¥n, 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@234528 65c4cc65-6c06-0410-ace0-fbb531ad65f3 --- channels/chan_sip.c | 1 + 1 file changed, 1 insertion(+) diff --git a/channels/chan_sip.c b/channels/chan_sip.c index ea3955aee6..c9b6ab39e2 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -5213,6 +5213,7 @@ 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 -- 2.47.2