[asterisk-dev] SIP invite to localhost/loopback

[Olle E. Johansson]

12 jan 2010 kl. 03.28 skrev 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?

Why do you need it?


/O