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

[WipeOut .]

> 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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