Ahhh...yes, I think that may have been it. I moved G.729 to the top of that list (just below disallow) and now I have a "restart when convenient" pending. Is that sufficient or do I have to actually reboot the server for the change to take effect?
Best wishes and aloha, Ben M. Schorr Chief Executive Officer ______________________________________________ Roland Schorr & Tower www.rolandschorr.com b...@rolandschorr.com > -----Original Message----- > From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- > boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere > Sent: Tuesday, December 15, 2009 8:30 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Can't get G.729 to work... > > > 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 _______________________________________________ -- 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