[Asterisk-Users] Extension Problems

[Rich Adamson]

> Thanks, but those extensions are listed in that list!
> 
> I'm stumped
> 
> Richard Adamson wrote:
> 
> >Better read up on why a sip phone should register with asterisk. Do a 'sip show peers' and 
that will be the list of phones that can "receive" calls.
> >-------------------
> >
> >    I've double checked this.  Everything is logging in fine, because I can 
> >    make calls, check my voicemail, everything except recieve calls on the 
> >    SIP devices.
> >    
> >    David Phelan wrote:

Okay, let's go through basic sip stuff then...

In basic asterisk configurations, each sip phone is expected to
register with asterisk. That register process essentially informs
asterisk which IP address is associated with that phone (or extension).
The register process will happen at the time the phone boots
up AND about every 3600 seconds thereafter. (The exact time might
be different then 3600 seconds and is either sip phone vendor
dependent or configuable on some sip phones.)

After the register process is complete, executing a 'sip show peers'
should display something like:
3000/3000   211.222.191.73  D  255.255.255.255  5060 Unmonitored

If there is an IP address shown, the registration was successful.
If there is no IP, then the registration either expired or it failed.

If an IP is displayed, it only indicates the registration process
was successful at "some previous time", and that could have been
ten minutes ago or 30 minutes ago. It does NOT indicate there is
still contact with that phone. (Eg, unplug the sip phone ethernet
cable and you'll still see that same display an hour later.)

If you want to "ensure" the 'sip show peers' is always up to date
and accurate, then include "qualify=2000" in that sip phone's
definition within sip.conf.  That will force asterisk to essentially
test to see if that IP address is still reachable every 2000 milli-
seconds (or every two seconds).  Add that statement into your
sip.conf definition and do your tests again. If that statement
resolves your problem, then one of two things are occurring:
 1. your loosing contact between the sip phone and asterisk, which
 is most likely related to nat issues (since I don't recall you
 stating that phones and asterisk are on the same wire).
 2. you have another problem that none of us can even guess at
 since you've not provided any real technical information about
 your system.

If the above does not help you resolve the issue, then you will
have to post more technical info before anyone can actually help
you. That should include:
 - your exact sip.conf entry for the sip phones in question,
 - your exact extensions.conf entries (including the 'context')
   associated with the problem,
 - output from 'sip show peers'
 - any command line debug statements shown during an attempted
   call to that extension.