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
