[Asterisk-Users] echo on the sip side

[Lee Goodman]

Did you enable echocancel and echocancelwhilebridged?
Did you put them in the correct location in the zapata.conf ? It has to be
before the channel statement (this is what threw me for a week)
If you tail -f debug in the /var/log/asterisk you can watch the call and see
if echo cancel was kicking in

Lee


----- Original Message -----
From: "John Brown" <jmbrown at chagresventures.com>
To: <asterisk-users at lists.digium.com>
Sent: Wednesday, August 20, 2003 2:23 AM
Subject: [Asterisk-Users] echo on the sip side


> so i call from a sip phone (grandstream) to
> a cell via x100p
>
>
> PSTN side hears everything nice, no echo.
>
> on the SIP side I hear myself about .1 to .2 sec
> later...
>
> any thoughts on how to resolve this.
>
> mucho thanks to everyone that has been helpful :)
>
> john
>
>
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