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

[Daniel ANDRE]

Hello,

Have you succeded to use flash key to do call transfert?

Regards,

Daniel


Dave Alan Caruana a écrit:

>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
>
>
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-- 
Daniel ANDRE (mailto:dandre at iris-tech.fr)
IRIS Technologies - http://www.iris-tech.com
Serveur kwartz - http://www.kwartz.com