The original called identifier is in the To header field of an initial
request.

John 

> -----Original Message-----
> From: [email protected] 
> [mailto:[email protected]] On 
> Behalf Of Alejandro Orellana
> Sent: 14 May 2009 16:29
> To: Lehto, Susan
> Cc: [email protected]
> Subject: Re: [Sip-implementors] Number delivery in diversion header
> 
> The original dialed number  is   supposed to be carried in a diversion
> header.
> for example say :
> A calls B
> B forwards to C
> C sends you an invite
> 
> you should received an invite with a diversion like
> Diversion: <[email protected]> reason=unconditional ,;  some other parameters
> Diverssion: <[email protected]>;
> 
> also you can get one Diversion header with comma separated  like
> Diversion: <[email protected]> reason=busy; more parameters   , <[email protected]> ;
> reason=unconditional; more paramertes
> 
> so the bottom most it is always the original called number 
> and topmost is
> the dialed number (current one).
> 
> take a look at  the following draft for details.
> draft-levy-sip-diversion-08.txt
> 
> Please also notice the diversion header was never moved to an 
> RFC , rather
> the History header it is the "official" .
> however Diversion it is widely implemented.
> 
> 
> --
> alejandro
> 
> 
> On Thu, May 14, 2009 at 10:43 AM, Lehto, Susan <[email protected]> wrote:
> 
> > To deliver the number originally dialed, is there an area in the SIP
> > diversion header to carry the called "telephone number" for 
> delivery to
> > a voice mail system?
> >
> > Our TDM telephone system is optioned to deliver to the 
> voice mail system
> > the number dialed, so that if the called line forwards to 
> voice mail,
> >
> >
> >
> > 1.  the voice mail system can respond with the called 
> person's greeting
> >
> > 2.  if the called line forwards to another line, which in 
> turn forwards
> > to voice mail, the voice mail system can respond with the 
> greeting of
> > the person originally dialed.
> >
> >
> >
> > Can SIP achieve the same result?
> >
> > We are testing a SIP system, and the vendor does not know 
> of a solution.
> >
> > I see that a similar question had been posed on the 
> listserve in October
> > 2000, but I have not yet found the answer
> >
> > Thanks for applying your expertise to this issue.
> >
> >
> >
> > Susan Lehto
> >
> > Sr. Analyst
> >
> > Boston University
> >
> > [email protected]
> >
> > _______________________________________________
> > Sip-implementors mailing list
> > [email protected]
> > https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors
> >
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