[asterisk-bugs] [JIRA] (ASTERISK-21387) Asterisk Deadlocks at least once during the day

Duane Larson (JIRA) noreply at issues.asterisk.org
Thu Apr 11 14:16:01 CDT 2013


    [ https://issues.asterisk.org/jira/browse/ASTERISK-21387?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=205229#comment-205229 ] 

Duane Larson commented on ASTERISK-21387:
-----------------------------------------

Ok.  So I recompiled with BETTER_BACKTRACES, DONT_OPTIMIZE and DEBUG_THREADS and set up my init.d file so that when Asterisk is started it adds the "-g" arg.

root at MediaServer01:/tmp# ps -C asterisk u
USER       PID %CPU %MEM    VSZ   RSS TTY      STAT START   TIME COMMAND
asterisk  8503  0.5 64.7 846564 165656 ?       Ssl  Apr10   5:21 /home/asterisk/asterisk-bin/sbin/asterisk -U asterisk -G asterisk -g

I had two 11.3 servers fail today and I can't find any core file.  The only file in /tmp is refs and there is no core file in /home/asterisk/asterisk-bin/sbin.

I will try to run asterisk directly from the /home/asterisk/asterisk-bin/sbin directory as root and see if that will produce a core file.

On a side note I just wanted to say that I spun up a new server and installed Asterisk 1.8.21 and this version does not appear to give me the bug I am seeing.


                
> Asterisk Deadlocks at least once during the day
> -----------------------------------------------
>
>                 Key: ASTERISK-21387
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-21387
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/General
>    Affects Versions: 11.0.1, 11.2.1, 11.3.0
>         Environment: Debian 6.0.4
>            Reporter: Duane Larson
>            Assignee: Duane Larson
>
> I have tested on three different versions of Asterisk and I still have the same issue.  It appears to happen once or twice a day.  It appears to happen when a VoIP phone trys to make a call (SIP Invite).  I have two Asterisk servers that are both behind an OpenSIPS server that load balances between the two servers.  Sometimes a SIP INVITE will make only one server have a deadlock and the call can go out the second server, but most of the time both Asterisk servers will deadlock and calls cannot be made.  When I am notified that there is an issue I get on the box and execute "netstat -nap |grep 5060" add see that "Recv-Q" column value is building up.
> I've posted on the Mailing list and was told that version 11.3 fixed a lot of deadlock issues but the upgrade did not help.
> See Mailing List email here 
> http://lists.digium.com/pipermail/asterisk-users/2013-April/278436.html
> 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"));
> I've seen the following debug logs
> 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
> Here is a gdb debug I grabbed when the issue occurred with version 11.3.0
> http://pastebin.com/gd291Bqz
> I am thinking this is a bug since it happens randomly.  Any help is appreciated.

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