[asterisk-dev] SIP invite to localhost/loopback

[Alex Balashov]

The limitation is that loop detection is performed by proxies, not  
back-to-back user agents (B2BUAs) like Asterisk.

Loop detection relies on detecting that the same logical leg is re- 
entering the system that has previously left it.

Since Asterisk generates a completely new call leg every time you Dial 
(), how would you define "looping" formally?

Besides, there may be legitimate applications in some cases for Dial() 
ing to the local host.

-- Alex

--
Sent from mobile device

On Jan 11, 2010, at 9:28 PM, Geoffrey Mina <geoffreymina at gmail.com>  
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?
>
> thanks.
>
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