From: Terry Wilson <twilson@digium.com>
Date: Mon, 16 Jan 2012 20:13:55 +0000 (+0000)
Subject: Don't prematurely stop SIP session timer
X-Git-Tag: 10.2.0-rc1~55
X-Git-Url: http://git.ipfire.org/cgi-bin/gitweb.cgi?a=commitdiff_plain;h=75f8a7940a5626e7138bd3c697de4c6c199c2326;p=thirdparty%2Fasterisk.git

Don't prematurely stop SIP session timer

When Asterisk is the UAS (incoming call, endpoint is re-inviting) the SIP session timer expires after half the time the sip endpoint indicates in the Session-expires header in proc_session_timer(). The session timer was being stopped totally and being handled as an error case instead of running again until the second expiry. This patch treats the half-time expiry as a non-error case and continues the timer until the true expiry.

(closes issue ASTERISK-18996)
Reported by: Thomas Arimont
Tested by: Thomas Arimont
Patches: session_timer_fix.diff by Terry Wilson (License #5357)
  based on session_timer.patch by Thomas Arimont (License #5525)
........

Merged revisions 351080 from http://svn.asterisk.org/svn/asterisk/branches/1.8


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@351081 65c4cc65-6c06-0410-ace0-fbb531ad65f3
---

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 1e95f5e418..091dacb84c 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -26529,6 +26529,8 @@ static int proc_session_timer(const void *vp)
 			ast_softhangup_nolock(p->owner, AST_SOFTHANGUP_DEV);
 			ast_channel_unlock(p->owner);
 			sip_pvt_unlock(p);
+		} else {
+			res = 1;
 		}
 	}