[Asterisk-Users] SIP and ECHO

[Dan]

----- Original Message ----- 
From: "Brian J. Schrock" <brians at anistonetech.com>
To: <asterisk-users at lists.digium.com>
Sent: Thursday, August 28, 2003 6:16 PM
Subject: [Asterisk-Users] SIP and ECHO


> Hello,
>
> I have read the information on echo and SIP in the FAQ and I have
> scoured the mailing list for possible solutions, but as yet I have not
> been able to get rid of this echo.
>
> I have a Sip phone, Digium 4 Port FXS, and 3 Digium FXO cards installed
> into an asterisk server. If I call between the Sip Phone
> (Budgettone-100) and the 4 FXS ports everything sounds great. If I call
> out to the PSTN through the FXO cards I get horrible echo, I have even
> been able when talking loud enough to get a horrible feedback loop
> going. I have tried 4 different echo cancellers in the Makefile for the
> Zap drivers and nonoe of them changed the situation.
>
> I have echocancel = (Any where from 1 - 256, I have tried alot of
> different values), and I have echocanelwhenbridged = yes.I only hear the
> echo start when the call gets bridged onto the outgoing PSTN lines.
>
> Is there anything I can do?
>
> Brian J. Schrock
>


Hi,

For me:

rxgain=0.8
txgain=0.8

in zapata conf do the trick.
Now the echo is allmost inexistant. Maybe the sound is not very strong but
the quality is very good.
I have the default echo canceller (no modification in the source files).

Tested with a lot of SIP phones (ATA (G.711), X-Lite(GSM), SJ_phone(G.711),
Cisco 79x0) and one X100P card.

BR,
Dan