This is a solution they proposed, using GSM gateways, but it wont let me handle 1000 simultaneous calls, the other option was using an E1 but the cost would be too much to deploy 35 E1s to support that many calls. There might be a better way of doing it.
On Wed, Apr 14, 2010 at 11:08 AM, William Stillwell (Lists) < [email protected]> wrote: > http://www.voip-info.org/wiki/view/Setup+MV-370+GSM+Gateway+with+Asterisk > > > > > > > > *From:* [email protected] [mailto: > [email protected]] *On Behalf Of *Pascal Bruno > *Sent:* Wednesday, April 14, 2010 10:52 AM > *To:* [email protected] > *Subject:* [asterisk-users] Converting GSM calls to SIP > > > > I have asked a GSM operator in my country if he can route a number or a > short code to my asterisk server via SIP (since they dont give DIDs in my > country) the operator said they do not support SIP, they have no way of > converting GSM calls to SIP to then send them to me. I would like to know > what is needed from the operator side to do this, what kind of material is > needed, or what can be done from their side to send SIP calls to my server. > > Thank you > > -- > _____________________________________________________________________ > -- 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 >
-- _____________________________________________________________________ -- 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
