I wrote a script to check clients and restart asterisk if registrations died (external IAX)...but you could modify for your needs. Check it out on www.generationd.com
-----Original Message----- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vieri Sent: Thursday, December 24, 2009 12:06 PM To: Asterisk Users List Subject: Re: [asterisk-users] how to check Asterisk SIP registration Thanks but "sip show registry" yields nothing. --- On Thu, 12/24/09, Danny Nicholas <da...@debsinc.com> wrote: > "sip show registry" might be more > helpful. > > -----Original Message----- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] > On Behalf Of Vieri > Sent: Thursday, December 24, 2009 10:39 AM > To: asterisk-users@lists.digium.com > Subject: Re: [asterisk-users] how to check Asterisk SIP registration > > Unfortunately, "sip show peers" did not "work" in my case. > The sip peers > were apparently "online" and "OK" (I use qualify=yes) but they > weren't... > The SIP clients could NOT register, so they were offline but "sip show > peers" stated that they were OK. > > I would prefer to perform an "automated" SIP registration (via cron > script). > If it fails then I can spawn a "rescue" script. > Surely, a "real" sip registration is more reliable then "sip show > peers". > > Any ideas? > > Vieri > > > --- On Wed, 12/23/09, Danny Nicholas <da...@debsinc.com> > wrote: > > > "Sip show users" or "sip show peers" > > should do the trick, but I'm not sure about 1.2; all of my > > experience is in the 1.4 branch. > > > > -----Original Message----- > > From: asterisk-users-boun...@lists.digium.com > > [mailto:asterisk-users-boun...@lists.digium.com] > > On Behalf Of Vieri > > Sent: Wednesday, December 23, 2009 1:09 PM > > To: asterisk-users@lists.digium.com > > Subject: [asterisk-users] how to check Asterisk SIP registration > > > > Hi, > > > > This is the first time I experience this problem with > > Asterisk: > > all of a sudden SIP registrations stopped working. > Active > > calls kept working > > but new calls could not be established (I did NOT > perform a > > "graceful > > restart"). > > > > Besides, would a "restart gracefully" actually deny > SIP > > registration? > > > > I did not have a network issue because killing > asterisk and > > starting it > > again solved the problem. > > > > How can I diagnose what happened to the SIP service > (I > > checked the log but > > am quite lost)? > > > > Also, how can I do a simple command-line "check" to > see > > that SIP > > registrations are OK? I would like to use a SIP > client > > (like sipsak) to > > perform a simple registration from a custom bash > script so > > I can quickly > > detect if this problem occurs again and > "auto-kill+restart" > > the asterisk > > process. I know this sounds ugly but on my production server, it's > > better to bring the whole system down and back up in as little > time > > as possible. > > > > Any suggestions? > > > > Asterisk is 1.2.31.1 > > > > Some log lines: > > > > Dec 23 13:13:16 DEBUG[11482] channel.c: Avoiding > initial > > deadlock for > > 'SIP/4053-b4520e98' > > Dec 23 13:13:16 WARNING[11482] channel.c: Avoided > initial > > deadlock for > > '0xb4302278', 9 retries! > > > > Dec 23 13:13:43 VERBOSE[18837] logger.c: > > -- Executing > > Dial("SIP/6174-b456d828", > "SIP/4062|20|tTwWM(auto-blkvm)") > > in new stack > > Dec 23 13:13:43 NOTICE[18837] app_dial.c: Unable to > create > > channel of type > > 'SIP' (cause 3 - No route to destination) Dec 23 13:13:43 > > VERBOSE[18837] > > logger.c: == Everyone is busy/congested at this time (1:0/0/1) Dec > > 23 13:13:43 DEBUG[18837] app_dial.c: Exiting with > > DIALSTATUS=CHANUNAVAIL. > > > > Thanks, > > > > Vieri > > > > > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users