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
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
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 Gatewa
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 to
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?
>
> Y
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 hea
Have you looked into
http://www.2n.cz/products/gsm_gateways/voip_gsm_gateway/voiceblue_voip_gsm_gateway.html
-E
On Jan 4, 2008 8:43 AM, Remco Barendse <[EMAIL PROTECTED]> wrote:
> >
> > You can use the D option with the Dial command.
> > Something like this should work:
> > exten => _06,1
>
> You can use the D option with the Dial command.
> Something like this should work:
> exten => _06,1,Dial(SIP/gsm_gateway,45,D(${EXTEN})
It worked
Here is how i did it in FreePBX :
1) Setup a SIP extension for the ATA device, in my case i give it
extension number 298. Edit the e
On Fri, 4 Jan 2008, EdPimentl wrote:
> Have you looked into
> http://www.2n.cz/products/gsm_gateways/voip_gsm_gateway/voiceblue_voip_gsm_gateway.html
> -E
Yes i did, looks like an excellent product with many, many features and
of outstanding quality.
However, given the cost of that unit i woul
On 16:10, Fri 04 Jan 08, Remco Barendse wrote:
>
> On Fri, 4 Jan 2008, EdPimentl wrote:
>
> > Have you looked into
> > http://www.2n.cz/products/gsm_gateways/voip_gsm_gateway/voiceblue_voip_gsm_gateway.html
> > -E
>
> Yes i did, looks like an excellent product with many, many features and
> of
I am using freePBX, so my dialplan uses macros and such, but here is what I do.
exten => 91248640ABCD,1,Goto(outrt-006-CellGateway,394${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
Dear sir,
what about receiving call from a GSM gateway. I didn't see the caller ID?.
is it happen to you? and what is the solution,Please.?
thanks,
Belal
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Have you turned on sip debugging?
Do you see the caller ID in the invite from your Gateway to your PBX?
On Tue, Jan 26, 2016 at 2:07 AM, Belal
wrote:
> Dear sir,
>
> what about receiving call from a GSM gateway. I didn't see the caller ID?.
> is it happen to you? and what is the solution,Please
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