> "If" I understood your initial objective correctly (and I may not have), > the user's phones are negotiating the codec to be used for each rtp session. > > Asterisk parameters can be used to dictate rtp sessions between the sip > phone and asterisk, but that won't influence the next step in which the sip > phone negotiates a new rtp session directly with the gateway. > > The gateway and the phone will negotiate a common codec based on > whatever logic those two devices have been programmed with by their > respective manufacturers; asterisk isn't involved. > > So, it sounds like the issue is understanding the codec selection logic > that has been programmed into the gateway and the phone.
I think you're getting my point, at least I think so (I'm getting more and more confused myself about this...) The problem is that the phone negotiates a codec with asterisk when placing the call (remember I have all reinvite's set to "no", so the gateway and the phone won't talk directly to each other!). This negotiation actually works correctly, because I force the phone's codec using "disallow=all; allow=g729" in the SIP phone's peer configuration. The negotiation which doesn't work the way I want is the asterisk-to-gateway part. The gateway can talk either ULAW or G729, whatever I tell it, if I "force" it using the disallow/allow method in sip.conf. The problem is that I need asterisk to talk to the gateway sometimes with ULAW, sometimes with G729, depending on the SIP phone who placed the call in the first place. What I need is some sort of command which says "OK, now Dial(... @gateway), but force G729" (which works *if* I tell asterisk that the gateway supports G729 *only* in sip.conf, but we want it to support both codecs, right?). Apparently I can only force the codec on incoming channels, not on outgoing channels. Is this really an asterisk limitation? -Manuel ___________________________________________________ Ticinocom SA - Via Stazione 5 - 6600 Muralto Tel 0844 007070 - Fax 0844 007071 http://www.ticinocom.com _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
