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 :)
On Thursday 03 January 2008 15:28:15 Remco Barendse wrote:> 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.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? Boyko
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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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!!
Remco Barendse wrote:> 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 :)This should work: context out-gateway { _X. { Dial(SIP/gateway,30,M(dial-gateway^${EXTEN})); } } macro dial-gateway(number) { Wait(1); SendDTMF(${number}); } You dial to gateway, and execute macro upon answer (if i remember correctly, it should be executed within dialed channel), so macro sends the number you need to dial on GSM gateway in DTMF, and after that bridges the call. You might try removing the Wait(1), but your GSM gateway could expect some idle time before receiving digits so i put it there. Regards, Atis
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}) -- Michiel van Baak michiel at vanbaak.eu http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x71C946BD "Why is it drug addicts and computer afficionados are both called users?"