[OpenSIPS-Users] Hairpin routing and Loop avoidance

[Steve Ames]

On Wed, Jun 24, 2009 at 06:01:52PM -0400, I?aki Baz Castillo wrote:
> El Mi?rcoles, 24 de Junio de 2009, Steven E. Ames escribi?:
> > 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.

Does that translate as I'm out of luck pending asterisk people fixing this, or
can someone point me at a workaround or the right phrase to google up some
additional info on the asterisk bug?

-steve