[OpenSIPS-Users] Number portability
brett at nemeroff.com
Fri Jul 10 21:06:15 CEST 2009
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
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
> > > 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>
> Users mailing list
> Users at lists.opensips.org
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