diff options
| author | Terry Wilson <twilson@digium.com> | 2012-01-16 20:15:24 +0000 |
|---|---|---|
| committer | Terry Wilson <twilson@digium.com> | 2012-01-16 20:15:24 +0000 |
| commit | fb5924ebe8104a9a5d95d6d1f2f6df41874ab93b (patch) | |
| tree | adacfa3e53976219e4885bd2b4e1d9b3aee8f8aa | |
| parent | c60d15222c8c414b8f4a09a287d52916d70e27dd (diff) | |
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)
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Merged revisions 351080 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 351081 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351082 65c4cc65-6c06-0410-ace0-fbb531ad65f3
| -rw-r--r-- | channels/chan_sip.c | 2 |
1 files changed, 2 insertions, 0 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c index 1d720adad..f26d7d5e0 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -26632,6 +26632,8 @@ 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; } } |