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 [EMAIL PROTECTED] 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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