asterisk users - Sep 2003 - echo cancelation

HI all,

Having a mental block today - can someone confirm which direction the echo
cancelation applies to for the Zap PRI channels?

ie. is it removing traces of the transmitted data to the PSTN from the
received data,

*or* is it removing traces of the data transmitted to Asterisk from the data
received back from Asterisk?

Got a configuration that is based on a call made from the PSTN to a SIP
user:

PSTN -> Telco Switch -> PRI -> Asterisk -> SIP -> X-Lite

PSTN caller hears no echo
X-Lite user hears significant (1 second or so) echo

If we setup a call so that the 'telco switch' runs an IVR process
instead of
linking to the PSTN, keeping the call entirely in the digital domain so no
signal reflection the X-Lite user does not hear echo.

Echocancel=yes etc. is enabled in the zapata.conf

Which party is at fault?

Linus

Check the debug log to see if echo cancel is really kicking in.
Also, you have to have the echocancel=yes before the individual channel
statements
----- Original Message -----
From: "Linus Surguy" <linus@magrathea-telecom.co.uk>
To: <asterisk-users@lists.digium.com>
Sent: Monday, September 15, 2003 11:19 AM
Subject: [Asterisk-Users] echo cancelation

> HI all,
>
> Having a mental block today - can someone confirm which direction the echo
> cancelation applies to for the Zap PRI channels?
>
> ie. is it removing traces of the transmitted data to the PSTN from the
> received data,
>
> *or* is it removing traces of the data transmitted to Asterisk from the
data
> received back from Asterisk?
>
> Got a configuration that is based on a call made from the PSTN to a SIP
> user:
>
> PSTN -> Telco Switch -> PRI -> Asterisk -> SIP -> X-Lite
>
> PSTN caller hears no echo
> X-Lite user hears significant (1 second or so) echo
>
> If we setup a call so that the 'telco switch' runs an IVR process
instead
of
> linking to the PSTN, keeping the call entirely in the digital domain so no
> signal reflection the X-Lite user does not hear echo.
>
> Echocancel=yes etc. is enabled in the zapata.conf
>
> Which party is at fault?
>
> Linus
>
>
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users

i've browsed the the voip site... but cant find is there a way the asterisk
server
to cut the ECHO, or it is only possible with hardware solution ?
I'm connecting 2 voip phones. 

tia