[asterisk-users] GSM Gateway behind SIP ATA?

[Remco Barendse]

On Thu, 3 Jan 2008, Benchev wrote:

> Basically Grandstream HT286 is a single port FXS ATA.
> In order to interconnect GSM gateway one would need FXO.
> Are you sure it gives you "new" dialing tone or this is the * itself
> you hear?

Yes, i am positive that i get a new dialtone from the GSM Gateway.

If i dial DTMF codes from a SIP phone connected to Asterisk, i can see the 
digits appear in the display of the GSM Gateway. But it is a bit 
incovenient to call an internal extension, wait for the dialtone and then 
punch in all the numbers of the cell phone i need to call.

I would prefer Asterisk to decide where / how to route the call and send 
the DTMF inband to the ATA device.

Thanks!!