[asterisk-dev] SIP invite to localhost/loopback

[Geoffrey Mina]

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?

thanks.