asterisk users - Dec 2005 - SIP and echo cancel

I known that sip channel should be free from echo. I am find this is not the
case for me.

The setup here is
Sipura 3000 connected to vonage
extensions are SIPURA 841 or SIPURA 2002 ATA.

I am getting echos on some of the outbound calls. I would like to be able to
have one
of the software echo cancel working on SIP channel (RTP stream)

Before I start hacking this into asterisk 1.2.1 I would like to known if
others are running
into this kind of problem ?

Is anyone working on this problem ?

If I make this work what is needed to get it put into a standard release ?
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> 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