[asterisk-users] Asterisk SIP deadlocks - update_provisional_keepalive

[Jim Lucas]

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


>
>
> On Wed, Apr 3, 2013 at 4:45 PM, Duane Larson <duane.larson at 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.larson at 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
>> *--*--*--*--*--*
>> --
>>
>
>
>
>
>
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-- 
Jim Lucas

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