[asterisk-dev] PJSIP transport problem

[Joshua Colp]

On 14-02-28 02:58 PM, Steve Murphy wrote:
> ​
> Many Thanks, Josh...
> 
> I removed the NAT options (external_signalling/media_*) set on transport,
> and used the above suggested options on the endpoint,
> and got it so phones can call other phones.
> 
> Two issues:
> 
> 1.I notice that I can't see the registration status
> of the various endpoints; is this something on the to-do
> list of things to be developed, or am I missing seomthing?
> I can get a list of outgoing registrations, tho.

You can see current associated contacts by using "pjsip list contacts".
Unlike chan_sip where a peer has one reachable address chan_pjsip
follows a much more SIP approach where contacts are bound to an AOR.

> 2. I'm seeing nat table expirations drop out from under dumber
> phones, where I can neither shorten the registration times
> nor send options. Is there a way to get pjsip to send out
> keepalives (OPTIONS)? It's kind of a bummer to have phones
> that think they are still registered, but you can't reach them.
> Looked thru the literature, haven't seen anything "juicy" about
> this.

OPTIONS functionality is controlled by specifying qualify_frequency on
the AOR.

-- 
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at:  www.digium.com  & www.asterisk.org