Looks like version 11.3 did not fix my issue. http://pastebin.com/gd291Bqz
On Thu, Apr 4, 2013 at 1:23 PM, Duane Larson <duane.lar...@gmail.com> wrote: > 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 > *--*--*--*--*--* > -- > -- -- *--*--*--*--*--* Duane *--*--*--*--*--* --
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