[asterisk-users] GSM Gateway behind SIP ATA?

[Benchev]

On Thursday 03 January 2008 16:38:35 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.
Yep. I've found a gsm gateway that does  "...calls from VoIP to GSM and GSM to 
VoIP and uses the SIP protocols so is ideal for use as a SIP Trunk on many 
SIP based VoIP PBX Phone Systems..."
Sorry, didn't know such a thing exists.

I don't think it matters dialing DTMF or not 
a simple dialplan trick should do.
From home (Europe) I do: 
[gsm-out]
exten => _0N.,1,Dial(SIP/gsm_gateway)
exten => _0N.,2,Hangup
Means all calls starting with zero and have digits from 2-9
afterwards go here. The mobile numbers start with 088 or 089.

Otherwise I dial 01 for US and 011 for International.
These are just ideas. You could figure out something else that
fits your needs.

Boyko