[OpenSIPS-Users] Number portability

[Iñaki Baz Castillo]

El Viernes, 10 de Julio de 2009, Alex Balashov escribió:
> Victor Pascual Avila wrote:
> > On Fri, Jul 10, 2009 at 8:12 PM, Alex Balashov<abalashov at evaristesys.com> 
wrote:
> >> Yes, you can.
> >>
> >> Just beware that you will _have_ to use something like 302s.  If you
> >> send the INVITE request back to the switch, it will be considered a
> >> call loop.
> >
> > Unless you added ;npdi or ;rn parameters to the RURI
>
> I am not sure how adding those parameters would circumvent the
> fundamental problem.
>
>    Softswitch --> call leg 1 --> proxy --> still call leg 1 --> softswitch

npdi and rp are *userinfo* parameters (in fact they are TEL URI paremeters so 
when converting to SIP URI they become part of the userinfo part).
  http://www.tech-invite.com/Ti-sip-abnf.html#teluri

So, if the original RURI is:
  sip:+12345678 at mydomain.org

and OpenSIPS modifies it to:
  sip:+12345678;npdi=123;rn=456 at mydomain.org

then both RURI's are differents and the softsiwtch won't consider it a loop.

However, if the parameters are added as SIP URI parameters (after the 
hostpart) the it would be a loop (except if they are maddr, user, ttl).


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