[Asterisk-Users] SIP and ECHO

[Dave Alan Caruana]

I tried specifying rxgain & txgain,
copied the format some some message on asterisk-users
Result was asterisk bombed out & didn't even load
due to not being able to understand the config file ..
what's the exact syntax that works??

cheers
Dave

----- Original Message -----
From: "Fredrik Hedberg" <fredrik.hedberg at telavox.se>
To: <asterisk-users at lists.digium.com>
Sent: Tuesday, September 02, 2003 9:21 AM
Subject: Re: [Asterisk-Users] SIP and ECHO


> What have you specified as rx and txgain?
>
> Simon McAuliffe wrote:
>
> >I've been having the same problem too, except for me it only happens
> >occasionnally.
> >
> >I'm not 100% sure of this, but it seems that for very local calls (eg
across
> >the city) I never get echo.  For calls that go longer distance (say 500km
or
> >more), or through some closer call centres, I'm getting the echo.  I
don't
> >get the echo on an analogue POTS connection to the same places (it is
> >clearly only happening on our asterisk system).
> >
> >This might indicate some link between echo cancellation and delayed
audio,
> >but if so, its sensitive to very small delays.
> >
> >The echo can only be heard at our end, there is no trace of it at the
other
> >end.
> >
> >I'm using ATAs doing SIP to Asterisk and through a PRI connection to a
> >Telco.  Echo cancellation is turned on and showing as activated on the
Zap
> >channels.  Echo cancellation is also enabled on the ATAs.
> >
> >----- Original Message -----
> >From: "Brian J. Schrock" <brians at anistonetech.com>
> >To: <asterisk-users at lists.digium.com>
> >Sent: Friday, August 29, 2003 3:16 AM
> >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
> >>
> >>_______________________________________________
> >>Asterisk-Users mailing list
> >>Asterisk-Users at lists.digium.com
> >>http://lists.digium.com/mailman/listinfo/asterisk-users
> >>
> >>
> >>
> >
> >_______________________________________________
> >Asterisk-Users mailing list
> >Asterisk-Users at lists.digium.com
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> >
>
>
> --
> Fredrik Hedberg
>
> Telavox AB        Direct:  +46 46 6220013
> Lilla torg 1        Phone:   +46 46 6220000
> S-211 34 Malmo        Mobile:  +46 70 3323033
> Sweden        Web: www.telavox.se
>
>
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