Asterisk SVN-branch-1.2-r48484
I get a SIP Response 482 (loop detected) back from my SIP provider whenever I dial from/to DIDs on the same server. The call is assumed "from an unknown peer", then gets routed to Local/<DID>@from-sip-external which fails. No SIP headers/messages are generated because the SIP channel is gone. It all makes sense, but how can I go about telling Asterisk not to dial out of a trunk when the number is local?
I could list the DIDs under from-sip-external, but that would potentially allow anyone to connect to the server by spoofing the DID. Seems like there ought to be an easy way get Asterisk to consult it's own inbound DID routes before selecting an outbound trunk, and without populating the dialplan with a parallel list of DIDs. I can't imagine I'm the only one to have run into this, but there's nothing on the lists about this scenario.
Chris _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
