[Asterisk-Users] SIP and ECHO

[Don Pobanz]

Ever since updating through cvs I have also noticed echo on the pstn 
lines using a fxo channel bank. I had no echo prior to the cvs update. 
I have backed off the gain for the FXO channels also (from 2.0 to 0.0) 
and this helped but there is still sometimes (but not all the time) a 
very distinct echo. It is like sometimes the echo canceling works and 
sometimes it does not.

I am now running asterisk cvs-08/26/03-14:04:16. I also did a cvs 
update for Zaptel and libpri on August 26th since * would not compile 
without it. Previously I was running the cvs version that I downloaded 
on April 16, 2003.

I am not using a sip phone but an ADSI analog set so it appears that 
this is not sip related.

If someone could verify this, a bug could be filed in bugtracker.

Don Pobanz

On Thursday, August 28, 2003 10:34 AM, Dan [SMTP:dtoma at fx.ro] 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