Andrew, Do you mind sharing how you got it working in branch 1.4? Thanks!
-----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Monday, October 01, 2007 6:21 PM To: asterisk-dev@lists.digium.com Subject: [asterisk-dev] Thanks for the G.722 codec Steve, et al., thanks for the G.722 codec in Trunk. I was just about to start getting something working and I noticed the G722 codec in trunk. I copied it back to branch 1.4 and it seems to work well. The translate time is only a "2", so it does not cost a lot of CPU time. The Polycom SoundPoint IP 650 seems to be very happy with G.722 (but the polycom is VERY LOUD by default on G.722). The Grandstream GXP-2000 is not so happy. It mostly works but has sound issues, I'll blame this on Grandstream as the Polycom seems to have no problems. G.722 sounds very lifelike! The MOH sounds much better and most of the digium recorded prompts sound like I'm talking to someone in the room. Andrew _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
