Summary:	ASTERISK-17151: Deadlock - Asterisk stops processing sip calls after a few calls.
Reporter:	Claudio Villalobos (devmod)	Labels:
Date Opened:	2010-12-22 20:15:50.000-0600	Date Closed:	2013-04-08 15:40:20
Priority:	Major	Regression?	No
Status:	Closed/Complete	Components:	Channels/chan_sip/General
Versions:		Frequency of Occurrence
Related Issues:	is related to ASTERISK-17867 [patch] Random entire asterisk deadlock when use builtin_atxfer when use res_timing_timerfd module
Environment:		Attachments:	( 0) backtrace-threads.txt ( 1) core-show-locks.txt ( 2) extensions.conf ( 3) log_sip_lock.txt ( 4) sip.conf
Description:	Asterisk stops responding to sip invites after two endpoints have made a few calls to it. The AGI app that is shown on the logs is basically an adhearsion app looking at the invite and : a) If P-Conference-ID exists, doing 'exec MusicOnHold'. b) If P-Conference-ID does not exist, hanging up. The issue is easy to reproduce, however sometimes it takes more than a few calls to get asterisk to lock up.
Comments:	By: David Vossel (dvossel) 2011-11-01 16:45:18.868-0500 Related to the timerfd bug.  This should be resolved in Asterisk 1.8.7