On Tue, 15 Dec 2009, Ben Schorr wrote: > Asterisk 1.4 - FreePBX - Polycom 330 and 501 phones. > > > > I've got G.729 loaded in the modules on the Asterisk server and on the > Polycom phones I've set G.729 to be the first preference of codec, but > still when I go SIP SHOW CHANNELS during active calls it still shows > "(ULAW)" (G.711) as the codec in use. > > > > I'm a newbie at Asterisk, can anybody suggest what I might be > overlooking? >
In the sip.conf entry for your peer you need to specify the codec negotiation order. Though you put g.729 first on the phone, asterisk probably has ulaw first, and is taking precedence. In the sip.conf entry put this: disallow=all allow=g729 j _______________________________________________ -- 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