[Asterisk-Users] SIP and ECHO

[Brian J. Schrock]

I can minimize doing those tricks, but I cannot seem to get it to go
away.

On Thu, 2003-08-28 at 11:33, Dan wrote:
> ----- 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
> 
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