Douglas Garstang wrote: > Realtime, as stated by Digium, does not work with sip users. This isn't > related to sip subscriptions. > Realtime certainly works with SIP users.
And yes, SIP subscriptions are lost in the current code. In 1.0, the support for subscriptions was very poor so it wasn't used. In 1.2, a few of us *non-Digum* developers enhanced it quite a lot. A community effort, not related to Digium at all. That's why SIP subscriptions is now a useful feature in Asterisk. In 1.3, because we made subscriptions very popular, we have to rewrite the whole subscription system in order to support a large quantity of subscriptions and new features users now want, like shared line apperances and call control. A lot of the work with Asterisk is done by non-Digium developers, in many cases paid for by Asterisk users that wants to enhance Asterisk to better support their business but can't or don't want to develop in house. Digium adds quite a lot of quality control, especially now when they have a team of testers working with the Asterisk Business Edition. I've also started to take Asterisk to the SIP interoperability testing events to further enhance the SIP support in Asterisk. If you want something to change, please do not go out saying "this whole product is crap, since feature yyy does not work as expected and Digium stinks by the way". We are very open to discuss bugs and missing features, as the product is being enhanced. The Asterisk.org developer community is very large, as you can see if you visit the bugtracker and check who is really contributing. Visiting the bug tracker might be a good idea anyway, that's a good place to report your bugs. Good luck with your continued work with Asterisk! Best regards, /Olle _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
