[Asterisk-Users] SIP Context from domain?

John Todd jtodd at loligo.com
Tue Nov 18 12:56:32 MST 2003


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
>
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>Asterisk-Cvs at lists.digium.com
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