[OpenSIPS-Users] Number portability

[Brett Nemeroff]

Just throwing this out.. Not all equipment can handle SIP Spiral properly.
<cough> asterisk <cough> (although I know there was work done on
Asterisk+SIP Sprial, I don't know where that ended up)
so be careful before you spend a lot of time on that.  I'd love to hear how
all of that works for you. I've got plans to do something similar in the LNP
space..
-Brett

On Fri, Jul 10, 2009 at 2:02 PM, Iñaki Baz Castillo <ibc at aliax.net> wrote:

> El Viernes, 10 de Julio de 2009, Alex Balashov escribió:
> > > 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 <sip%3A%2B12345678 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).
> >
> > How does that change the other logical attributes of a call leg, i.e.
> > Call ID GUID, From tag, CSeq, etc?
>
> If the RURI changes, then it's *not* a loop, but a spiral. Re-read the
> appropiate section in RFC 3261 :)
>
>
> --
> Iñaki Baz Castillo <ibc at aliax.net>
>
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