>From memory, it's really down to making the right selections in sip.conf
We did a large installation, with phones at the Head Office using g711 and phones at remote sites using g729. Asterisk happily transcoded for us. Which was great. PalH On Wed, 2006-02-15 at 15:37 +1100, Lisa Wolf wrote: > I've got a situation here that I thought was trivial. I have two > phones, and an asterisk box. > > The first phone knows about g723, alaw and g729, as does the second phone. > sip.conf has allows for those codecs. > > Now, within the dialplan I make a determination which codec will be used > (this is a simulation, obviously. Think 'within the office', 'to > another office' and 'to a device that only knows g723'). > > I do this with setvar(SIP_CODEC=[appropriatecodecname]). Then I can > either do an immediate answer, or wait for the other end to pick up to > ensure the answer. > > However, what I'm finding is that the originating phone decides on the > codec specified in SIP_CODEC, but the destination phone replies with its > preferred codec (whichever I specified first) out of the three. If this > isn't what I put in the SIP_CODEC, then asterisk is unable to connect > the channels. > > My question is this: is it possible for the codec chosen with something > like the SIP_CODEC to be 'passed' to the second channel as the only > choice? Or another way to get asterisk to determine the codec that > should be used based on destination? > > Thanks, > Lisa > > _______________________________________________ > --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 _______________________________________________ --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
