I used the macro-stdextenion that comes with every Asterisk install, and added a new option - s-CHANUNAVAIL which then dialled the other server via IAX. Worked really well and only took a few minutes.
PaulH On Tue, 2007-08-14 at 23:51 -0700, Nicholas Blasgen wrote: > I've heard about this, but I really can't seem to find anything on it. > I've got a strange setup that exists only because of firewall issues, > and everything about it seems fine. The setup: > > SIP clients -> Asterisk (office) -> IAX -> Asterisk (colocation) -> > SIP PSTN Termination > > All the extensions I want to be able to dial are on the colocation > box. What I'd really like is for the "office" asterisk box to forward > all extension requests it doesn't know about to the colocation > Asterisk box. I think this is refered to as Trunking. I only need to > do this in a single direction, if that's any easier to setup. > > Are there any good documents on VOIP-Info or another site on setting > up something like this? The office Asterisk's job is just to act as a > SIP to IAX gateway. I've got a work-a-round that will work, but I > thought I'd learn the proper method. > > -- > /Nick > _______________________________________________ > --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 _______________________________________________ --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