[asterisk-users] how to check Asterisk SIP registration

Michelle Dupuis support at ocg.ca
Thu Dec 24 15:56:39 CST 2009


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-bounces at lists.digium.com
[mailto:asterisk-users-bounces at 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 <danny at debsinc.com> wrote:

> "sip show registry" might be more
> helpful.
> 
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com]
> On Behalf Of Vieri
> Sent: Thursday, December 24, 2009 10:39 AM
> To: asterisk-users at 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 <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
> 
> 
> 
>       
> 
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