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?
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${EXTEN},1)
;I have a list of all of our company's cell phone numbers. (We get free Cell to
Cell)
[outrt-006-CellGateway]
include = outrt-006-CellGateway-custom
Have you looked into
http://www.2n.cz/products/gsm_gateways/voip_gsm_gateway/voiceblue_voip_gsm_gateway.html
-E
On Jan 4, 2008 8:43 AM, Remco Barendse [EMAIL PROTECTED] wrote:
You can use the D option with the Dial command.
Something like this should work:
exten =
You can use the D option with the Dial command.
Something like this should work:
exten = _06,1,Dial(SIP/gsm_gateway,45,D(${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
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 would
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 outstanding
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 Gateway.
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
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
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,
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