[asterisk-dev] IAX support for SIP URI as called number

Matthew Jordan mjordan at digium.com
Thu Aug 30 08:05:39 CDT 2012



----- Original Message -----
> From: "Alec Davis" <sivad.a at paradise.net.nz>
> To: "Asterisk Developers Mailing List" <asterisk-dev at lists.digium.com>
> Sent: Wednesday, August 29, 2012 11:25:57 PM
> Subject: [asterisk-dev] IAX support for SIP URI as called number
> 
> Just implemented distributed call queues using jabber, and
> sucessfully
> distributing device state.
> With the queue making calls to the remote site over an IAX trunk, but
> having
> to use the remote extension number.
> 
> queues.conf:
> ...
> member => IAX2/server2/8819,0,Test User,SIP/XYZ9000
> 
> But the IAX protocol should be able to handle a SIP URI in the
> CALLED_NUMBER
> 
> http://www.rfc-editor.org/rfc/rfc5456.txt
> 8.6.1.  CALLED NUMBER
>    The purpose of the CALLED NUMBER information element ...
>    ...It is possible for a number or extension to include non-numeric
> characters.  The CALLED
>    NUMBER IE MAY contain a SIP URI, [RFC3261] or a URI in any other
>    format.  The ability to serve a CALLED NUMBER is server dependent.
> 
> So we should be able to have
> queues.conf:
> ...
> member => IAX2/server2/sip:XYZ9000,0,Test User,SIP/XYZ9000
> 
> The first and easy change is in parse_dial_string to look for a ':'in
> the
> exten.
> The other side yet to determine.
> 
> The question is. Do we want it?
> Am I barking up the wrong tree.
> 

Well, it'd certainly be interesting.  If you're going to make chan_iax2
able to parse a SIP URI, you'll probably want to look at using the
parsing routines in reqresp_parser, rather than rolling anything new.
Since those are already exposed by reqresp_parser.h, that should keep
the cross-pollination between chan_iax2 and chan_sip to a minimum - but
it does feel a little odd to have chan_iax2 including things from
chan_sip.

That being said, it sounds like there's no reason why the CALLED NUMBER
can't be any dial string ("URI in any other format" is rather vague) - so
this may not have to be SIP specific.

--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org



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