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 -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
