> >Define that per user. > > > Of course... The user part is not the problem. If I force a user in its extensions > to use G729 only, he actually talks G729 to Asterisk, but asterisk still talks ULAW to the PSTN gateway, doing the transcoding. This is driving me crazy... >
"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. _______________________________________________ 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
