[asterisk-users] RPID

[Kristian Kielhofner]

Freddi Hansen wrote:
> Hi,
> Here is how I have it working:
> 
> If  Alice calls Bob and Bob's phone diverts the call to Carol.
> You want Bob to pay for the call and the callerid  shown to Carol to be 
> 'Alice'
> 
> On Bob's server
> exten _X.,1,sipaddheader(Divertion:<Bob at anonynous.invalid;user=phone>
> 
> The proxy that routes bob's call to Carol  will then charge Bob for the 
> call and the From: field will be Alice
> 
> If you are an ITSP using Asterisk then you must look for the 'Divertion' 
> header in incoming SIP invite's yourselves with a
> 
> sipheader(Divertion) command
> 
> I have this working in a few different scenarios
> 
> I think that the right thing todo would be setting the RDNIS if the 
> 'Divertion' is present on inbound side but I am not sure about this so I 
> am using a private variable and doing this outside the SIP channel
> 
> b.r.
> Freddi

Freddi,

	Diversion as specified in draft-levy-sip-diversion-08.txt probably 
won't ever be a standard.  Cisco adds a header called "CC-Diversion", 
for instance.  Asterisk doesn't fully support it yet, although it looks 
like it will get added to app_transfer eventually:

http://bugs.digium.com/view.php?id=5484

	Then there is History-Info:, which is also expired:

http://www.softarmor.com/wgdb/docs/draft-ietf-sip-history-info-06.txt

	Back to the OP - in your situation, if you have a complete SIP 
environment, app_transfer should work with 302 redirects.

--
Kristian Kielhofner