Thanks Jim. Searched through the change log for "deadlock" but nothing really stuck out. I'll upgrade to 11.3 and see if that makes a difference.
On Thu, Apr 4, 2013 at 10:59 AM, Jim Lucas <li...@cmsws.com> wrote: > On 04/03/2013 08:15 PM, Duane Larson wrote: > >> So it just happened again on both machines at the same time and I was >> running debug on both servers. I am running OpenSIPS and load balancing >> between both servers so I am guessing when the invite was sent to the >> first >> server it was frozen for some reason and then OpenSIPS sent the invite to >> the second server and that server was also frozen/deadlocked because of >> the >> SIP message. I noticed on both servers the last log that was posted with >> Asterisk deadlocked was the following >> >> >> Asterisk version 11.0.1 >> [Apr 3 21:39:42] DEBUG[12984] res_timing_timerfd.c: Expected to >> acknowledge 1 ticks but got 11805 instead >> >> Asterisk version 11.2.1 >> [Apr 3 21:39:50] DEBUG[1854] res_timing_timerfd.c: Expected to >> acknowledge >> 1 ticks but got 12423 instead >> >> >> In my last email I posted the debug from the Asterisk server with 11.0.1 >> version of code. Here is a post of the debug for the Asterisk server with >> version 11.2.1 >> >> http://pastebin.com/mbjSSAWM >> >> >> This has to be a bug right? I am thinking of opening an issue on the >> Asterisk JIRA system >> >> > A number of deadlocks were fixed in the current release of 11.3. Please > read the change log to see if any fit your issue. > > http://downloads.asterisk.org/**pub/telephony/asterisk/** > ChangeLog-11-current<http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11-current> > > > >> >> On Wed, Apr 3, 2013 at 4:45 PM, Duane Larson <duane.lar...@gmail.com> >> wrote: >> >> It just happened again on the 11.0.1 box and I was able to grab a debug. >>> I am hoping someone can tell me if this is a bug or something wrong >>> with >>> my config. >>> >>> gdb asterisk-bin/sbin/asterisk 29048 >>> >>> Go here for the debug output >>> http://pastebin.com/DGXx0BSk >>> >>> >>> On Tue, Apr 2, 2013 at 7:42 PM, Duane Larson <duane.lar...@gmail.com >>> >wrote: >>> >>> I am currently running two different versions of Asterisk >>>> >>>> 11.0.1 >>>> 11.2.1 >>>> >>>> I have noticed the bug occur on both servers. >>>> >>>> The issue is that when I try to dial a phone number sometimes the call >>>> will never go out. I will check the Asterisk server with NGREP and see >>>> that the SIP messages are making it to Asterisk but Asterisk isn't >>>> responding. >>>> >>>> I do the following command "netstat -nap |grep 5060" and see that >>>> Asterisk has a lot under the "Recv-Q" column. >>>> >>>> It usually takes about 10 minutes before Asterisk becomes responsive >>>> again or else before 10 minutes is up I could restart Asterisk and >>>> everything will be back to normal. >>>> >>>> I see in the message logs the following errors >>>> >>>> On the 11.0.1 Asterisk server >>>> WARNING[23723][C-00000010] chan_sip.c: Unable to cancel schedule ID >>>> 11473. This is probably a bug (chan_sip.c: >>>> update_provisional_keepalive, >>>> line 4406). >>>> >>>> On the 11.2.1 Asterisk server >>>> WARNING[3493][C-0000001f] chan_sip.c: Unable to cancel schedule ID >>>> 30810. >>>> This is probably a bug (chan_sip.c: update_provisional_keepalive, line >>>> 4683). >>>> >>>> >>>> When I look in chan_sip.c on both servers I see that they are the same >>>> line of code >>>> >>>> AST_SCHED_DEL_UNREF(sched, pvt->provisional_keepalive_**sched_id, >>>> dialog_unref(pvt, "when you delete the provisional_keepalive_sched_**id, >>>> you >>>> should dec the refcount for the stored dialog ptr")); >>>> >>>> >>>> >>>> What could be causing this because it seems to happen at least once a >>>> day. >>>> >>>> >>> >>> >>> -- >>> -- >>> *--*--*--*--*--* >>> Duane >>> *--*--*--*--*--* >>> -- >>> >>> >> >> >> >> >> -- >> ______________________________**______________________________**_________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> >> http://lists.digium.com/**mailman/listinfo/asterisk-**users<http://lists.digium.com/mailman/listinfo/asterisk-users> >> >> > > -- > Jim Lucas > > http://www.cmsws.com/ > http://www.cmsws.com/examples/ > -- -- *--*--*--*--*--* Duane *--*--*--*--*--* --
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users