[Asterisk-Dev] ${RDNIS} from SIP-GW

[Adam Roach]

The "Diversion" header hasn't actually received much traction
within any IETF working group, and has (IMHO) virtually no
chance of ever being published as an RFC. I would not recommend
implementing it in asterisk, simply because it will likely
never be anything remotely approaching standard. However,
if you must add support (due to your gateway vendor's implementation),
the format is described in:
<http://www.ietf.org/internet-drafts/draft-levy-sip-diversion-05.txt>

There is a little more support for a broader effort that covers
some of the same ground:
<http://www.ietf.org/internet-drafts/draft-barnes-sipping-history-info-02.tx
t>
although I'll note that this hasn't been picked up by any
IETF working group yet as a working group item. In fact,
it's still considered very controversial. It does
have a lot more chance of being eventually published as an
RFC than the "Diversion" header draft, though -- simply
because it does things in a much more general way.

I would discuss this with your gateway vendor, and see if
you can get them to support the "History-Info" header
instead of "Diversion".

In either case, I strongly suspect (although I haven't
checked) that you'll need to add support to whichever
header your gateway sends into Asterisk. Note that
I'm not talking about turning on some configuration
option; you'd need to write a patch to the code.

The truth, though, is that neither approach is particularly
close to any sort of standards approval, which means
that at any given time, the document that describes how
to do it is six months away from being rather hard to find.
(The IETF removes internet-drafts automatically after 6
months).

/a


> -----Original Message-----
> From: Christoph Frei [mailto:asterisk at digitalcity.ch]
> Sent: Monday, May 12, 2003 15:30
> To: asterisk-dev at lists.digium.com
> Subject: [Asterisk-Dev] ${RDNIS} from SIP-GW
> 
> 
> Hello,
> 
> i'm trying to get a virtual voicemail done by using RDNIS. I 
> can see with
> sip debug that the gateway sends a header with the correct number:
> 
> Diversion: <sip:012345678 at 192.168.1.1>;reason=unconditional
> 
> Is this simply not yet supported by asterisk, or do i have to 
> activate 
> the RDNIS
> feature on SIP somehow? ${RDNIS} is always empty...
> 
> 
> Greetings,
> 
> Christoph
> 
> 
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