[Asterisk-Users] SIP Context from domain?

[Tristan 'Minty' Colgate]

Yep,

  I did a checkout after posting that and noticed the SIPDOMAIN var, it is
pretty much ideal for my needs (just pass u${EXTEN}@${SIPDOMAIN} to the
voicemail app I hope.

  My only concern is in using the CVS code, but in this case I think it is
worth it, I'm using a fairly minimal set of functionatlity from *, I'm just
using it for voicemail and PSTN from a SER proxy.

On Tue, Nov 18, 2003 at 02:56:32PM -0500, John Todd wrote:
> At 8:00 PM +0000 11/18/03, Tristan 'Minty' Colgate wrote:
> >From: "Tristan 'Minty' Colgate" <minty at deadweb.net>
> >To: asterisk-users at lists.digium.com
> >Subject: [Asterisk-Users] SIP Context from domain?
> >Reply-To: asterisk-users at lists.digium.com
> >Date: Tue, 18 Nov 2003 20:00:55 +0000
> >
> >Hi,
> >
> >  Is it possible to pick the context of a call from chan_sip based on the
> >domain of the To: header of the INVUTE? I've had a quick look throught he 
> >code
> >and can't see anything, I want to use the voicemail virtual hosting with
> >chan_sip. Can the sip domain be picked out with a global in 
> >extensions.conf?
> >This woud also solve my problem.
> >
> >  If not is there any specifc reason/restriction that I am missing? If it 
> >  is
> >not already supported and there aren't any specific objections then I don't
> >mind putting together a patch for it.
> >
> >  I'm working with the last stable release and haven;t checked out CVS yet.
> >
> >--
> >Tristan 'Minty' Colgate
> ><minty at deadweb.net> | ICQ #154577755
> >-----------
> >  "I don't mean to sound bitter, cold, or cruel, but
> > I am, so that's how it comes out"
> >				- Bill Hicks
> 
> 
> With some appropriate thought, and a basic understanding of how 
> Asterisk handles call routing, this recent CVS note should point you 
> in the right direction.
> 
> JT
> 
> 
> 
> >From: markster at lists.digium.com
> >To: asterisk-cvs at lists.digium.com
> >Subject: [Asterisk-cvs] asterisk README.variables,1.9,1.10
> >Date: Wed, 12 Nov 2003 17:28:02 -0600 (CST)
> >
> >Update of /usr/cvsroot/asterisk
> >In directory mongoose.digium.com:/tmp/cvs-serv6345
> >
> >Modified Files:
> >	README.variables
> >Log Message:
> >Improve documentation of ${SIPDOMAIN}
> >
> >
> >Index: README.variables
> >===================================================================
> >RCS file: /usr/cvsroot/asterisk/README.variables,v
> >retrieving revision 1.9
> >retrieving revision 1.10
> >diff -u -d -r1.9 -r1.10
> >--- README.variables	11 Nov 2003 20:46:41 -0000	1.9
> >+++ README.variables	12 Nov 2003 23:54:16 -0000	1.10
> >@@ -44,7 +44,7 @@
> > ${DNID}         Dialed Number Identifier
> > ${RDNIS}        Redirected Dial Number ID Service
> > ${HANGUPCAUSE}	Hangup cause on last PRI hangup
> >-${SIPDOMAIN}    SIP domain (if appropriate)
> >+${SIPDOMAIN}    SIP destination domain of an inbound call (if appropriate)
> >
> > There are two reference modes - reference by value and reference by name.
> > To refer to a variable with its name (as an argument to a function that
> >
> >_______________________________________________
> >Asterisk-Cvs mailing list
> >Asterisk-Cvs at lists.digium.com
> >http://lists.digium.com/mailman/listinfo/asterisk-cvs
> 
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users

-- 
Tristan 'Minty' Colgate
<minty at deadweb.net> | ICQ #154577755
-----------
  "I don't mean to sound bitter, cold, or cruel, but
 I am, so that's how it comes out"
				- Bill Hicks