On 04 Jun 2014, at 20:57, Joshua Colp <[email protected]> wrote: > Olle E. Johansson wrote: >> On 04 Jun 2014, at 20:35, Joshua Colp<[email protected]> wrote: >> >>> Olle E. Johansson wrote: >>>> On 04 Jun 2014, at 19:49, SVN commits to the Digium >>>> repositories<[email protected]> wrote: >>>> >>>>> Creating a branch for persisting subscriptions across restarts >>>>> in PJSIP. >>>> Why? You can terminate subscriptions at shutdown and the UA will >>>> re-subscribe when you're back. >>> The goal of my branch is to cover both best cases and worst cases. >>> In normal circumstances then yes, we can terminate with probation >>> or deactivated and assume the other side will re-subscribe. In >>> worst case Asterisk has terminated unexpectedly (or "core restart >>> now" is used) and we don't have a chance to do this. I would like >>> the experience for the user to be that Asterisk comes back and they >>> are subscribed, persistence achieves this in both the best and >>> worst case scenarios. >> Well, yes. Agree. >> >> Remember the hook we have so that we terminate subscriptions for >> extensions that was deleted when the dialplan was reloaded. > > Will do. > >>>> The problem with registrations is that you can not do that. >>> Can you clarify what you mean by this? >> >> You can not terminate a registration from the server. If you do >> support SUBSCRIBE for registration states you can do a bit more >> messaging to the UA, which would be a good thing to support, >> especially since you want to support multiple registrations per >> account. > > Do you know off the top of your head any endpoints which are easily available > which implement this for testing?
No, not on the top of my head. I will check. Have been discussing this a lot with Blink. It makes a lot of sense to do it. Phones will know which other phones that are connected to an account and can simplify transfer, especially if their contacts contact display names. A digium phone supporting this could have a quick transfer menu saying something like "transfer to Olles laptop Olles supervideoscreen Olles audio-only BT phone" Then I could easily transfer a call from one SIP phone to another using my account. If they all use GRUUs, IPv4/IPv6/NAT won't be a big issue either. Sorry for preaching ;-) /O -- _____________________________________________________________________ -- 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
