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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070814/66d4a5b9/attachment.htm
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
On Tue, 14 Aug 2007, 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.You can do this by having a default widlcard extension to point to the co-lo box. Simple (crude?) but effective... So in the diaplan in the office asterisk box: exten => _X.,1,Dial(IAX2/co-lo/${EXTEN}) so anything not matched locally will get punted through to the co-lo box. This assumes 'co-lo' has an entry in iax.conf. (and corresponding authentication on the co-lo side) You can narrow it down if you know the extension numbering scheme - so if you want to point all numbers starting with 0 and all 3-digit extensions in the range 200 through 399, to the co-lo, then: exten => _0.,1,Dial(IAX2/co-lo/${EXTEN}) exten => _2XX,1,Dial(IAX2/co-lo/${EXTEN}) exten => _3XX,1,Dial(IAX2/co-lo/${EXTEN}) and so on. Although the last 2 might be combined with: exten => _[23]XX,1,Dial(IAX2/co-lo/${EXTEN}) etc. If you explicitly wanted to dial 9 for an outside line, then: exten => _9.,1,Dial(IAX2/co-lo/${EXTEN:1})> 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.With so many ways to do something, who defines "proper" :) Gordon