Hi! thank you for your good answers. Another related question: I tried using Page() and it works perfectly, but I need to implement a slightly different behavior, and I'm looking into ways of implementing it.
When a user picks up the phone and chooses to page the speakers, the call should start (so that it's ready for talking), but in muted status. When the user pushes a push-to-talk button, then a bell sound needs to be played through all the speakers, then she can start talking freely. Everytime the PTT button is released, the mic needs to mute, but that's something I can work out in the softphone. How can I implement it? I am thinking of using some call parking method and some DTMF code to pass to the next state, but I am open to advice, since I'm quite new to Asterisk. Could I also create a macro to do the same thing Page is doing, but with ConfBridge? Last question: is there a way of reinviting periodically remotes to the conference, so that they can recover after e.g. a reboot? Thank you in advance, Matteo Il 22/09/2010 21:51, Philipp von Klitzing ha scritto: > Hi! > > >> I need the system to be resilient to any network partition, so that >> anyone can send announces from any mic to all the reachable clients. >> I'd need also to page a subset of all the speakers. >> > Most of the major phone vendors (that are employed by the users of this > list) have support for multi-cast of some sort. In recent firmware > release notes I have read that SNOM has now also added a feature to feed > multicast directly from a phone (and not just play multicast audio on the > speaker as long as the phone is not in use). > > >> I'm currently using some software I wrote which sends voice over >> multicast RTP and coordinates all the sites with multicast messages. >> > app_page has been around for quite some in Asterisk, and the new Asterisk > 1.8 now also adds the channel driver "MulticastRTP". > > >> Is there a way asterisk could be of use, or would I need to bend it >> too much? >> > Look here: > http://www.voip-info.org/wiki/view/Asterisk+cmd+Page > http://www.voip-info.org/wiki/view/Asterisk+Paging+and+Intercom > > I have made good experience with MAST for multicasting SNOM phones: > http://www.aelius.com/njh/mast/ > > Philipp > > > -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users