[OpenSIPS-Users] Number portability

[Alex Balashov]

Iñaki Baz Castillo wrote:
> 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).

How does that change the other logical attributes of a call leg, i.e. 
Call ID GUID, From tag, CSeq, etc?


-- 
Alex Balashov
Evariste Systems
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