[asterisk-users] GSM Gateway behind SIP ATA?

[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 <asterisk at barendse.to> wrote:

> >
> > You can use the D option with the Dial command.
> > Something like this should work:
> > exten => _06XXXXXXXX,1,Dial(SIP/gsm_gateway,45,D(${EXTEN})
>
>
> It worked!!!!
>
> Here is how i did it in FreePBX :
>
> 1) Setup a SIP extension for the ATA device, in my case i give it
> extension number 298. Edit the extension after creating it set DISALLOW to
> all and set ALLOW to alaw to make sure DTMF sending will work.
>
> 2) Create a custom trunk, and set as Custom Dial String :
> Local/$OUTNUM$@custom-gsmvoip-out
>
> 3) add to extensions_custom.conf :
> [custom-gsmvoip-out]
> exten => _.,1,Dial(SIP/298,,D(wwwwww0${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!
>
> _______________________________________________
> --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
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080104/a7fc4477/attachment.htm