Re: [asterisk-users] GSM Gateway behind SIP ATA?
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.? > > thanks, > Belal > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- A human being should be able to change a diaper, plan an invasion, butcher a hog, conn a ship, design a building, write a sonnet, balance accounts, build a wall, set a bone, comfort the dying, take orders, give orders, cooperate, act alone, solve equations, analyze a new problem, pitch manure, program a computer, cook a tasty meal, fight efficiently, die gallantly. Specialization is for insects. ---Heinlein -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GSM Gateway behind SIP ATA?
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 http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GSM Gateway behind SIP ATA?
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" <[EMAIL PROTECTED]> wrote in message news:[EMAIL PROTECTED]
>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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>
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> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Re: [asterisk-users] GSM Gateway behind SIP ATA?
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 quality. > > However, given the cost of that unit i would have to be calling mobile > phones 24 hours per day for at least the next 10 years of my life to > earn the investment back, so definatively economically unviable. > > But thanks for the tip :) Remco, If you ever want dialplan/sip.conf hints for the voiceblue in the netherlands let me know. We have this unit in production and really like it. It's stable as hell and the quality is very good. Just make sure you dont put too much pressure on the antenna connection, that one breaks easily and forces you to buy a new antenna. Without the antenna the unit is close to a brick. -- Michiel van Baak [EMAIL PROTECTED] 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?" ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GSM Gateway behind SIP ATA?
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 have to be calling mobile phones 24 hours per day for at least the next 10 years of my life to earn the investment back, so definatively economically unviable. But thanks for the tip :) ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GSM Gateway behind SIP ATA?
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,Dial(SIP/gsm_gateway,45,D(${EXTEN})
>
>
> It worked
>
> Here is how i did it in FreePBX :
>
> 1) Setup a SIP extension for the ATA device, in my case i give it
> extension number 298. Edit the extension after creating it set DISALLOW to
> all and set ALLOW to alaw to make sure DTMF sending will work.
>
> 2) Create a custom trunk, and set as Custom Dial String :
> Local/[EMAIL PROTECTED]
>
> 3) add to extensions_custom.conf :
> [custom-gsmvoip-out]
> exten => _.,1,Dial(SIP/298,,D(ww0${EXTEN}))
>
> Note that i put a leading zero there, because for my fallback outbound
> routes i needed to strip the leading zero so i added it again here.
>
> 4) Insert the custom trunk in outbound routes
>
> That's it
>
> Hope this will save somebody else 2 days of frustration :)))
>
> Cheers!
>
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>
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> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Re: [asterisk-users] GSM Gateway behind SIP ATA?
>
> You can use the D option with the Dial command.
> Something like this should work:
> exten => _06,1,Dial(SIP/gsm_gateway,45,D(${EXTEN})
It worked
Here is how i did it in FreePBX :
1) Setup a SIP extension for the ATA device, in my case i give it
extension number 298. Edit the extension after creating it set DISALLOW to
all and set ALLOW to alaw to make sure DTMF sending will work.
2) Create a custom trunk, and set as Custom Dial String :
Local/[EMAIL PROTECTED]
3) add to extensions_custom.conf :
[custom-gsmvoip-out]
exten => _.,1,Dial(SIP/298,,D(ww0${EXTEN}))
Note that i put a leading zero there, because for my fallback outbound
routes i needed to strip the leading zero so i added it again here.
4) Insert the custom trunk in outbound routes
That's it
Hope this will save somebody else 2 days of frustration :)))
Cheers!
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Re: [asterisk-users] GSM Gateway behind SIP ATA?
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, 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. Yep. I've found a gsm gateway that does "...calls from VoIP to GSM and GSM to VoIP and uses the SIP protocols so is ideal for use as a SIP Trunk on many SIP based VoIP PBX Phone Systems..." Sorry, didn't know such a thing exists. I don't think it matters dialing DTMF or not a simple dialplan trick should do. >From home (Europe) I do: [gsm-out] exten => _0N.,1,Dial(SIP/gsm_gateway) exten => _0N.,2,Hangup Means all calls starting with zero and have digits from 2-9 afterwards go here. The mobile numbers start with 088 or 089. Otherwise I dial 01 for US and 011 for International. These are just ideas. You could figure out something else that fits your needs. Boyko ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GSM Gateway behind SIP ATA?
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 => _06,1,Dial(SIP/gsm_gateway,45,D(${EXTEN})
--
Michiel van Baak
[EMAIL PROTECTED]
http://michiel.vanbaak.eu
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x71C946BD
"Why is it drug addicts and computer afficionados are both called users?"
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Re: [asterisk-users] GSM Gateway behind SIP ATA?
Remco Barendse wrote:
> I have an analog GSM Gateway that is connected to a normal SIP ATA device.
>
> Basically what it does is this : when you call the extension nr. of the
> SIP ATA port, the GSM Gateway will pick up the phone and presents a (new)
> dial tone, and then dials whichever DTMF tones it received. The SIP ATA ia
> a Grandstream HT286.
>
> I would like to use the GSM Gateway to route my outbound cellular calls,
> how do i do this in Asterisk? Basically Asterisk should dial the extension
> number and then send required number as DTMF tones to the Gateway through
> the ATA.
>
> I am using FreePBX, which allows me to create a custom trunk for the
> outgoing calls. Hope this could work :)
This should work:
context out-gateway {
_X. {
Dial(SIP/gateway,30,M(dial-gateway^${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
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Re: [asterisk-users] GSM Gateway behind SIP ATA?
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!! ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GSM Gateway behind SIP ATA?
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 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] GSM Gateway behind SIP ATA?
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 :) ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
