> Before I start hacking this into asterisk 1.2.1 I would like to known > if others are running into this kind of problem ?
Asterisk doesn't do any echo cancellation in the setup you describe; it just passes the audio data, and transcodes if necessary. The endpoints (the 841 phone and the 2002 and 3000 ATAs) are responsible for cancelling echo. The Sipura ATA's generally do a good job cancelling echo. You may want to play with the gain settings in the admin web config for the Sipura ATA. As far as the 841 is concerned, if the handset volume is too loud I noticed you may be getting acoustic echo. Hasn't been a problem for me for PSTN calls or SIP to SIP calls though. If you really want to patch asterisk to apply echo cancellation on the RTP stream on pure VoIP calls, that would be interesting to see how well it works. --Luki _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
