There are also settings which will turn on local echo cancellation for the handset, headset and/or speaker phone. I don't recall their names at the moment. They are off by default on the handset and headset unless you're using a very recent (3.0+) SIP app.
Tim Nelson wrote: > I'm not sure about the 3 second delay, but I've seen plenty of echo issues on > Polycom phones when the gain has been changed on the handset. Check the > voice.gain.tx and voice.gain.rx settings in your sip.cfg to make sure they're > not too high. > > You also may want to make sure there aren't any system resource constraints > such as high CPU usage or memory usage... :-) > > Tim Nelson > Systems/Network Support > Rockbochs Inc. > (218)727-4332 x105 > > ----- "c james" <[EMAIL PROTECTED]> wrote: > >> A Polycom 550 and a IAX client (Mozphone and ZoIPer were used) are >> having a conversation. Call quality is reported as good except for >> an >> echo with a 3 second delay. >> >> Most of my searches are saying echo happens only on the PSTN piece, >> but >> there isn't one here. >> >> Can someone point me in the right direction? >> >> Asterisk 1.4.21.2 >> Under 40 users >> Quad-Core AMD Opteron(tm) Processor 2352 and 4G RAM (Hey, that's what >> they wanted to use!) >> >> >> _______________________________________________ >> -- 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 > _______________________________________________ -- 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
