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