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
