[asterisk-users] GSM Gateway behind SIP ATA?

[Steven]

I am using freePBX, so my dialplan uses macros and such, but here is what I do.

exten => 91248640ABCD,1,Goto(outrt-006-CellGateway,394${EXTEN},1)
;I have a list of all of our company's cell phone numbers. (We get free Cell to Cell)

[outrt-006-CellGateway]
include => outrt-006-CellGateway-custom
exten => _3949.,1,Macro(dialout-trunk,12,${EXTEN:4},,)
exten => _3949.,n,Macro(dialout-trunk,11,${EXTEN:4},,)
exten => _3949.,n,Macro(dialout-trunk,1,${EXTEN:4},,)
exten => _3949.,n,Macro(outisbusy,)
; end of [outrt-006-CellGateway]

;I have a two port SIP-GSM Gateway.
;Trunk 12 is port2
:Trunk 11 is port1
;Trunk 1 is my PRI, in case the other two port are busy.


-- 
-- 
Steven

http://www.connectech.org/



"Remco Barendse" <asterisk at barendse.to> wrote in message news:Pine.LNX.4.64.0801031421460.19715 at raveon.vaag.nu...
>I have an analog GSM Gateway that is connected to a normal SIP ATA device.
>
> Basically what it does is this : when you call the extension nr. of the
> SIP ATA port, the GSM Gateway will pick up the phone and presents a (new)
> dial tone, and then dials whichever DTMF tones it received. The SIP ATA ia
> a Grandstream HT286.
>
> I would like to use the GSM Gateway to route my outbound cellular calls,
> how do i do this in Asterisk? Basically Asterisk should dial the extension
> number and then send required number as DTMF tones to the Gateway through
> the ATA.
>
> I am using FreePBX, which allows me to create a custom trunk for the
> outgoing calls. Hope this could work :)
>
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