[asterisk-users] how to check Asterisk SIP registration

[Vieri]

I appreciate everyone's feedback.

I did not post the "sip show peers" output because I did not have time to save it but I'm fairly sure that qualify was "OK" and that IP addresses did show up.
NAT/firewall is not an issue because Asterisk and the sip devices are on the same network (open LAN).

Anyway, regardless of the "sip show peers" output, the fact that the SIP devices registered fine and communication was re-established after killing asterisk and starting it, demonstrates that the root cause is not the "network" but the Asterisk's SIP service.

I am using an alias IP address on the SIP server. Usually it works fine but maybe this time something went wrong. At the time I had my issue, I checked that the alias IP address was defined. Maybe Asterisk's SIP service was not correctly bound/listening to that alias IP address... 
Maybe removing and adding the alias IP address would have magically solved the issue but I did not try that.

Can the SIP service be restarted without affecting the rest of Asterisk? (I don't think "sip reload" does this)

Thanks,

Vieri

--- On Sat, 12/26/09, Olle E. Johansson <oej at edvina.net> wrote:

> You've unfortunately gotten a lot of
> confused answers. To try to clear this up:
> 
> 1. Only type=peer objects accept registrations. "sip show
> users" or "sip show registry" has nothing to do with peers.
> A peer might be part of a type=friend
> 2. If you see IP addresses when you run "sip show peers"
> then those objects have an active registration, Asterisk
> knows where to reach them.
> 3. Nat's or firewalls between the device and Asterisk might
> cause issues with Asterisk sending messages to them or
> devices sending messages to Asterisk
> 4. Your output below indicates that Asterisk doesn't know
> how to reach the device, that Asterisk has no IP and port
> address to send messages to, thus the device is not
> registered at all.
> 5. Turning "qualify" on can help with keeping a NAT binding
> open. 
> 
> To summarize, start with looking for IP address in "sip
> show peers". If we have an IP address, check the result of
> the Qualify option in the same output. If there's an IP, the
> device could reach Asterisk. If the status is "unreachable"
> Asterisk could not reach the device on the IP address.
> Then go hunting in your network to find the issue.
> 
> Best regards,
> /Olle
> 
> 
> 24 dec 2009 kl. 17.39 skrev 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