[asterisk-users] GSM Gateway behind SIP ATA?

[Michiel van Baak]

On 15:38, Thu 03 Jan 08, Remco Barendse wrote:
> 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!!

You can use the D option with the Dial command.
Something like this should work:
exten => _06XXXXXXXX,1,Dial(SIP/gsm_gateway,45,D(${EXTEN})


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Michiel van Baak
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