Re: [asterisk-users] GSM Gateway behind SIP ATA?

2016-01-26 Thread Belal
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 -- _ -- Bandwidth and Colocation Provided by

Re: [asterisk-users] GSM Gateway behind SIP ATA?

2016-01-26 Thread John Kiniston
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?

Re: [asterisk-users] GSM Gateway behind SIP ATA?

2008-01-07 Thread 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

Re: [asterisk-users] GSM Gateway behind SIP ATA?

2008-01-04 Thread EdPimentl
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 =

Re: [asterisk-users] GSM Gateway behind SIP ATA?

2008-01-04 Thread Remco Barendse
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

Re: [asterisk-users] GSM Gateway behind SIP ATA?

2008-01-04 Thread Remco Barendse
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

Re: [asterisk-users] GSM Gateway behind SIP ATA?

2008-01-04 Thread Michiel van Baak
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

[asterisk-users] GSM Gateway behind SIP ATA?

2008-01-03 Thread Remco Barendse
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

Re: [asterisk-users] GSM Gateway behind SIP ATA?

2008-01-03 Thread Benchev
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,

Re: [asterisk-users] GSM Gateway behind SIP ATA?

2008-01-03 Thread Remco Barendse
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.

Re: [asterisk-users] GSM Gateway behind SIP ATA?

2008-01-03 Thread Atis Lezdins
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

Re: [asterisk-users] GSM Gateway behind SIP ATA?

2008-01-03 Thread Michiel van Baak
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

Re: [asterisk-users] GSM Gateway behind SIP ATA?

2008-01-03 Thread 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,