[Asterisk-Users] RE: Asterisk stops responding

[Paul Cheng]

Update to latest CVS and check the bug report that I filed re:DTMF. 
Your problem could be related. Latest CVS seems to fix the blocking 
problem for me.

On Friday, September 5, 2003, at 01:15  AM, Andres wrote:

> 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
>>>
>>>
>>>
>>> _______________________________________________
>>> Asterisk-Users mailing list
>>> Asterisk-Users at lists.digium.com
>>> http://lists.digium.com/mailman/listinfo/asterisk-users
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