[OpenSIPS-Users] Hairpin routing and Loop avoidance

[Iñaki Baz Castillo]

El Miércoles, 24 de Junio de 2009, Steven E. Ames escribió:
> Hi. I have an OpenSIPS and Asterisk setup. Incoming calls come from VoIP
> Carrier to OpenSIPS. OpenSIPS does some dbaliases because sometimes
> multiple numbers are assigned to same asterisk extension. This all works
> great.
>
> However, when asterisk makes a call outward for a number that is actually
> local then I get a Loop and asterisk is unhappy.
>
> Example: I have 2 incoming DIDs (111-222-3333 and 111-222-3334). On
> OpenSIPS in dbaliases I translated 111-222-3333 to 111-222-3334 and send it
> to asterisk. All is fine. On asterisk it knows about 3334 but not 3333. So
> if another extension on asterisk dials 111-222-3333 it gets to OpenSIPS.
> OpenSIPS does know about 3333 and knows how to handle it. It converts it to
> 3334 and sends it back to asterisk. Voila. Loop. Now the actual behavior is
> what I want but I want to modify the SIP INVITE such that asterisk will
> accept it and not gripe about the loop.
>
> Any pointers?

This is a kwnown bug of Asterisk.


-- 
Iñaki Baz Castillo <ibc at aliax.net>