"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
