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 => _06XXXXXXXX,1,Dial(SIP/gsm_gateway,45,D(${EXTEN}) > > > It worked!!!! > > Here is how i did it in FreePBX : > > 1) Setup a SIP extension for the ATA device, in my case i give it > extension number 298. Edit the extension after creating it set DISALLOW to > all and set ALLOW to alaw to make sure DTMF sending will work. > > 2) Create a custom trunk, and set as Custom Dial String : > Local/[EMAIL PROTECTED] > > 3) add to extensions_custom.conf : > [custom-gsmvoip-out] > exten => _.,1,Dial(SIP/298,,D(wwwwww0${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 >
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