[Asterisk-Users] SIP + Grandstream 100 + TDM100P = lots of local echo, & questions about call transfers

[Dave Alan Caruana]

could you send me the exact syntax for rxgain / txgain?
I think that might help towards my problem
becuase i'm having to turn the handset volume all the
way up ..

thanks
Dave

----- Original Message -----
From: "WipeOut ." <wipeout at linuxmail.org>
To: <asterisk-users at lists.digium.com>
Sent: Tuesday, August 05, 2003 9:45 AM
Subject: Re: [Asterisk-Users] SIP + Grandstream 100 + TDM100P = lots of
local echo, & questions about call transfers


> > my error .. the cards are X100P which is why I wrote FXO.
> >
> > The Grandstream phones are on a LAN, the * server connects to the
phonelines
> > via the X100P cards. When I call from the Grandstream phones onto the
PSTN
> > there is a VERY big amount of echo, ie. I can hear myself in the
earpiece.
> >
> > cheers
> > Dave
> >
>
> An echo at the begining of a call is normal as the * and phone "trains"
themselves but this should dissappear after about 30 seconds to 1 min..
>
> So my only suggesttions are..
>
> First make sure you have echocancel=yes and echocancelwhenbridged=yes in
your zapata.conf..
>
> If that doesn't help try lowering the volume on the sip handset and play
with the rxgain= and txgain= in zapata.conf for the X100P's..
>
> Other than that I don't really know what else you can try..
>
> Later..
> --
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