[asterisk-users] how to check Asterisk SIP registration

[Vieri]

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 <danny at 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-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com]
> On Behalf Of Vieri
> Sent: Wednesday, December 23, 2009 1:09 PM
> To: asterisk-users at 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