[asterisk-users] Re: Re: ECHO Cancellation in SIP Calls

[Steven]

If you are calling from a SIP phone through asterisk and through a Digium card, one could argue that the Digium card IS farside of 
the SIP phone.

SIP -- asterisk -- Digium -- T1/PRI/analog -- PSTN -- PSTN -- PSTN -- Destination.

I would argue that the Digium card IS on the farside of asterisk as far as the SIP phone is concerned.

We just switched from Legacy PBX to Asterisk and we get occasional echo.
Everything past the Digium card is the same as the old PBX.
We never got echo on the old PBX.



-- 
-- 
Steven

http://www.glimasoutheast.org



"Tzafrir Cohen" <tzafrir.cohen at xorcom.com> wrote in message news:20061027095405.GK4594 at xorcom.com...
> On Thu, Oct 26, 2006 at 04:17:16PM -0700, Alyed Tzompa wrote:
>>
>> Echo is generated by the analog end to where you place the call, not the IP side of it.
>>
>> As far as I know the echo cancelation in the Asterisk can only be tweaked in the zapata.conf (since IP calls don't generate it)
>>
>> I'm afraid there is little you can do to here.
>
> A digital zaptel card (PRI/BRI) does not generate them either.
>
> If the echo would be generated on your side, you wouldn't be the one to
> hear it.
>
> -- 
>               Tzafrir Cohen
> icq#16849755                    jabber:tzafrir at jabber.org
> +972-50-7952406           mailto:tzafrir.cohen at xorcom.com
> http://www.xorcom.com  iax:guest at local.xorcom.com/tzafrir
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