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

Reply via email to