[Asterisk-Users] update re. Grandstream + SIP + Echo problems ..

[Dave Alan Caruana]

well .. good news :)

i've just put in
txgain=1.0
rxgain=1.0
in my zapata.conf

and upgraded the Grandstream Budgettones i'm using to version 81
of the software and all seems fine .. there is still an echo but after
the first couple of seconds of call it vanishes, as the echocancelling
kicks in .. so far my client is happy :)

now .. i have one slight problem left .. although most of my SIP
phones are on a LAN connection with the asterisk server,
there are two phones which are at a remote office bridged to
my LAN via a 128k point to point ADSL .. these do not seem
to be working well, you do hear speech but the remote person
(dialled over PSTN through an X100P) hears it low and garbled ..
I am assuming it's due to the delays in stuffing 64kbits (of g711)
over a 128k link and was thinking of switching to G729.

I already have the G729 codec installed, and configured with 1
license. Can anyone give me the correct sip.conf commands 
(or whatever I need) to get the budgettones working over G729?

many thanks
Dave