[asterisk-dev] SIP invite to localhost/loopback

[Tilghman Lesher]

On Monday 11 January 2010 20:28:09 Geoffrey Mina wrote:
> Is it a limitation of asterisk or the SIP protocol which prevents
> asterisk from being able to Dial(SIP/exten at localhost)?
>
> It seems as though checking of the direction of the message combined
> with the totag and fromtag of the SIP INVITE, Asterisk should be able
> to deal with this scenario without raising a "482 Loop Detected".
>
> I messed around with chan_sip for a bit today, but was unable to
> convince asterisk it was OK to accept the inbound call.
>
> Are there any patches to chan_sip that might enable this, or is it
> just impossible inside the asterisk architecture?

Um, why not just do Dial(Local/exten at context) ?  Involving the SIP protocol
seems to be non-optimal, at best.

-- 
Tilghman Lesher
Digium, Inc. | Senior Software Developer
twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
Check us out at: www.digium.com & www.asterisk.org