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

[frank.barthe at prescom.net]

Dave Alan Caruana wrote:

> 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.
>

> ----- Original Message -----
> From: "WipeOut ." <wipeout at linuxmail.org>
> To: <asterisk-users at lists.digium.com>
> Sent: Tuesday, August 05, 2003 8:50 AM
> Subject: Re: [Asterisk-Users] SIP + Grandstream 100 + TDM100P = lots of
> local echo, & questions about call transfers
>
> > > 1st question:
> > > when i phone out
> > > or receive a call from one of the SIP phones onto the PSTN, there is
> > > a LOT of local echo in the handset .. the PSTN end of the call does not
> > > here this echo, but it's VERY annoying on the SIP end of things ..
> > > the echo seems to be about 0.3 seconds delayed to the speech ..
> > > there is no echo on incoming voice, just an echo of my own voice
> > > as I speak.

My configuration :
1 - X-TEL SIP phone
    - phone handset connected to sound blaster
    (providing no accoustic echo by itself. Tested with Netmeeting)
2 - one single FXO board
    (on the Asterisk side)
3 - remote = PSTN telephone set or GSM telephone set

The symptoms :
- broad local echo (IP side)
- tiny echo on the PSTN side

What I have tried :
1 - change the handset with an USB one (nothing change but the
    sound quality : worst !)
2 - change the echo canceller attached to the FXO board (nothing
    really noticiable)
3 - change to IAX (changing the client software) : seems to cancel
    the echo (!?)

Problem stays alive !