[Asterisk-Users] RE: Asterisk stops responding

[Andres]

It happened once again here.  This time I called an IVR (SIP to SIP) and upon 
sending the 1st DTMF tone, * bombed out.  The console got filled with these 
messages (and they wouldn't stop):

DEBUG[4101]: File chan_sip.c, Line 5001 (sipsock_read): Failed to grab lock, 
trying again...
DEBUG[4101]: File chan_sip.c, Line 5001 (sipsock_read): Failed to grab lock, 
trying again...
DEBUG[4101]: File chan_sip.c, Line 5001 (sipsock_read): Failed to grab lock, 
trying again...
DEBUG[4101]: File chan_sip.c, Line 5001 (sipsock_read): Failed to grab lock, 
trying again...
DEBUG[4101]: File chan_sip.c, Line 5001 (sipsock_read): Failed to grab lock, 
trying again..

* stopped responding and I had to kill the process manually.
*CLI> show version
Asterisk CVS-08/22/03-22:24:05 built by root at maui on a i686 running Linux

Has anybody else seen this message?
Regards,
Andres


On Thursday 28 August 2003 13:37, Andres wrote:
> We run Iptel's SER as our SIP Server.  All subs register with our SIP
> Server, but if anyone needs to call the PSTN then the call gets forwared to
> *.
>
> The "Request to schedule in the past"  messages have to do with MOH and I
> was told it was due to a slow PC.  I don't think it is related with
> Asterisk hanging up.
>
> Regards,
> Andres
>
> On Thursday 28 August 2003 13:27, David Harris wrote:
> > >Gazing at the console I was able to determine the exact time Asterisk
> > >froze.
> > >Even with DEBGUG on it did not show anything important.   The moment it
> > >freezes is when a call from Phone1 tries to connect to a SIP Provider
> >
> > like
> >
> > >Iconnect:
> >
> > I have not been able to pin point exactly what event causes the
> > freeze-up but I have been on the console when it has happened.  It
> > didn't print out anything interesting.  The call I was on cut off.
> >
> > >Phone1----Our SIP Server-------Our Asterisk--------SIP Provider
> > >
> > >
> > >It was by no means 100% reproducible.  Maybe 1 out of 10 calls caused
> >
> > the
> >
> > >trouble.
> >
> > Same here except I would say more like 1 out of 100 calls.
> >
> > > A bad symptom would be that the command "show sip channels"
> > >would show several calls, even though they had hungup a long time ago.
> >
> > I definitely have this problem.
> >
> > >Troubleshooting revealed that the BYE message was not being sent by our
> >
> > SIP
> >
> > >Server to the Asterisk server upon hangup.  We rectified this and we no
> > >longer see those phantom SIP Channels and Aterisk has not froze for
> >
> > about a >week.
> >
> > What is your "SIP Server" what does it do?  Maybe I have the same issue
> > with my Cisco Voice Gateway not sending the BYE message sometimes.  But
> > would this cause asterisk to freeze?
> >
> >
> > Other "symptoms" I have are these errors in the asterisk messages log
> > file
> >
> > Aug 27 09:21:00 NOTICE[1081364]: File sched.c, Line 209 (sched_settime):
> > Request to schedule in the past?!?!
> > Aug 27 09:21:24 NOTICE[1081364]: File sched.c, Line 209 (sched_settime):
> > Request to schedule in the past?!?!
> > Aug 27 09:21:29 NOTICE[1081364]: File sched.c, Line 209 (sched_settime):
> > Request to schedule in the past?!?!
> > Aug 27 09:21:35 NOTICE[1081364]: File sched.c, Line 209 (sched_settime):
> > Request to schedule in the past?!?!
> > Aug 27 09:22:05 NOTICE[1081364]: File sched.c, Line 209 (sched_settime):
> > Request to schedule in the past?!?!
> >
> > Thanks,
> > David Harris
> >
> >
> >
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