I am currently testing a new Asterisk installation. The server has a T100P connected to a PRI, and about 50 Polycom IP600 phones connected via the local network. Every couple hours, Asterisk randomly stops responding to all calls, both incoming on the PRI and calls from the SIP phones. I'm not sure how or where to start debugging it. When Asterisk stops working I can still connect to it with "asterisk -r". All of the commands I've tried such as "sip show channels" and "zap show channels" show the last statuses of when asterisk stopped working. I am running asterisk at debug level 9, but nothing appears in the logs when asterisk stops responding. Calls to the PRI generate a busy signal, and calls from the SIP phones just time out. No new messages appear on the debug console. "extensions reload" did not make a difference, but a "restart now" fixes the problem for awhile. We originally were running CVS-HEAD from several weeks ago, but updated to CVS-HEAD from today and are still having the same problem. We are using Slackware 10.1, Realtime (talking to Mysql on the same machine), ldapget, app_ldap, rx and txfax, and logging CDR's to mysql. Any suggestions on how to figure out what is happening would be appreciated! I saw someone else mention using "strace -p" to figure out exactly what asterisk is doing... are there other methods? Which of the many asterisk process ID's should I use with strace? Thanks! Eric
Now I'm worried - we have exactly the same problem, but were going to upgrade to 1.2. Now it seems as if CVS-HEAD has the same issue. We have a TE405P, with 80 cisco7960 phones connected to a isdn30 pri. The same issues ocurr - Busy on inbound calls, cannot place outbound, nothing in the logs. Are you (as we are) 1) running with queues and agents 2) reloading the config (reload from the cli) 3) monitoring the system by connecting to the manager cli ? We are looking for all possible solutions to this. Julian. ewr@erols.com wrote:> I am currently testing a new Asterisk installation. The server has a > T100P connected to a PRI, and about 50 Polycom IP600 phones connected > via the local network. Every couple hours, Asterisk randomly stops > responding to all calls, both incoming on the PRI and calls from the SIP > phones. I'm not sure how or where to start debugging it. > > When Asterisk stops working I can still connect to it with "asterisk > -r". All of the commands I've tried such as "sip show channels" and "zap > show channels" show the last statuses of when asterisk stopped working. > I am running asterisk at debug level 9, but nothing appears in the logs > when asterisk stops responding. Calls to the PRI generate a busy > signal, and calls from the SIP phones just time out. No new messages > appear on the debug console. "extensions reload" did not make a > difference, but a "restart now" fixes the problem for awhile. > > We originally were running CVS-HEAD from several weeks ago, but updated > to CVS-HEAD from today and are still having the same problem. We are > using Slackware 10.1, Realtime (talking to Mysql on the same machine), > ldapget, app_ldap, rx and txfax, and logging CDR's to mysql. > > Any suggestions on how to figure out what is happening would be > appreciated! I saw someone else mention using "strace -p" to figure out > exactly what asterisk is doing... are there other methods? Which of the > many asterisk process ID's should I use with strace? > > Thanks! > > Eric > > > _______________________________________________ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > >
James Jones
2005-Aug-28 06:49 UTC
[Asterisk-Dev] Re: [Asterisk-Users] Help Solving Asterisk Lockups
If this issue exists doesn't it mean that asterisk is unstable anyway? On Sat, 2005-08-27 at 16:29 -0400, Marc Olivier Chouinard wrote:> I have repeatedly mention this issues, and I keep getting laugh at from > Mark... So I do not think donation to digium will fix the core problem. > > Digium want to sell the product like it is rightnow, and have no plan to > do masive change to fix any core problems. They think that if they > start redesign this, it will bring back asterisk to be unstable again. > > Marc O. > > James Jones wrote: > > > I know of good way to solve this problem. I have been authorize by my > > company to try to a group of people and businesses to give donations > > to get Digium to fix this issue. We will start the pot at $200. Are > > there any takers? > > > > > > On Sat, 2005-08-27 at 10:08 -0400, ewr@erols.com wrote: > > > >>> So the only thing we have in common is the remote monitoring ... > >> > >>Are you using: > >> > >>1) Realtime (and if so, with mysql, odbc, etc?) > >>2) Logging CDR records? (and if so, how) > >> > >>This post looks like it could pertain to the same problem: > >>http://lists.digium.com/pipermail/asterisk-dev/2005-August/014797.html > >>.. but I don't think it has been resolved. > >> > >>Eric > >> > >>> > >>> Julian > >>> ewr@erols.com <mailto:ewr@erols.com> wrote: > >>>>> Now I'm worried - we have exactly the same problem, but were going to > >>>>> upgrade to 1.2. Now it seems as if CVS-HEAD has the same issue. > >>>>> > >>>>> We have a TE405P, with 80 cisco7960 phones connected to a isdn30 pri. > >>>>> The same issues ocurr - Busy on inbound calls, cannot place outbound, > >>>>> nothing in the logs. > >>>>> > >>>>> Are you (as we are) > >>>>> > >>>>> 1) running with queues and agents > >>>> > >>>> > >>>> We are *not* using queues or agents. > >>>> > >>>>> 2) reloading the config (reload from the cli) > >>>> > >>>> I have used "restart now" from the cli to bring the system back when it > >>>> freezes. Honestly I'm not sure that I've tried a plain "reload." I'll > >>>> see if that brings it back next time it dies. > >>>> > >>>>> 3) monitoring the system by connecting to the manager cli ? > >>>> > >>>> We have an application (similar to the Flash Operator Panel) that > >>>> connects to the manager API (via port 5038, not the CLI) and is used by > >>>> our receptionist to monitor extensions and transfer calls. > >>>> > >>>> I intend to slowly start stripping the system down. Next time it crashes > >>>> I will change the logging from mysql to csv only. This bug makes it > >>>> sound like an mysql glitch can cause the system to hang: > >>>> http://bugs.digium.com/view.php?id=4953 > >>>> > >>>>> We are looking for all possible solutions to this. > >>>> > >>>> Me too! > >>>> > >>>> Eric > >>>> > >>>> _______________________________________________ > >>>> --Bandwidth and Colocation sponsored by Easynews.com -- > >>>> > >>>> Asterisk-Users mailing list > >>>> Asterisk-Users@lists.digium.com <mailto:Asterisk-Users@lists.digium.com> > >>>> http://lists.digium.com/mailman/listinfo/asterisk-users > >>>> To UNSUBSCRIBE or update options visit: > >>>> http://lists.digium.com/mailman/listinfo/asterisk-users > >>>> > >>>> > >>> > >>> _______________________________________________ > >>> --Bandwidth and Colocation sponsored by Easynews.com -- > >>> > >>> Asterisk-Users mailing list > >>> Asterisk-Users@lists.digium.com <mailto:Asterisk-Users@lists.digium.com> > >>> http://lists.digium.com/mailman/listinfo/asterisk-users > >>> To UNSUBSCRIBE or update options visit: > >>> http://lists.digium.com/mailman/listinfo/asterisk-users > >>> > >> > >> > >>_______________________________________________ > >>--Bandwidth and Colocation sponsored by Easynews.com -- > >> > >>Asterisk-Users mailing list > >>Asterisk-Users@lists.digium.com <mailto:Asterisk-Users@lists.digium.com> > >>http://lists.digium.com/mailman/listinfo/asterisk-users > >>To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-use <http://lists.digium.com/mailman/listinfo/asterisk-users>rs <http://lists.digium.com/mailman/listinfo/asterisk-users> > >> > >> > >------------------------------------------------------------------------ > > > >_______________________________________________ > >Asterisk-Dev mailing list > >Asterisk-Dev@lists.digium.com > >http://lists.digium.com/mailman/listinfo/asterisk-dev > >To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-dev > > > > _______________________________________________ > Asterisk-Dev mailing list > Asterisk-Dev@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-dev > To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050828/93a272ad/attachment.htm