Assuming you are using SIP phones and IIRC, you can hint at the codec to be used by setting the SIP_CODEC variable in the dialplan; before Dial()'ing, of course ! :-)
I think this is still an area where asterisk needs improvement... Dynamic codec (re) negotiation. Anyone care to correct me ? Cheers, -- exvito _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
