[asterisk-users] Handling SIP 482 condition

[Chris Miller]

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