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 > > > > > > > > _______________________________________________ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
