[Freeswitch-users] Modify sip_invite_req_uri, sip_invite_to_uri

Brian Foster bdfoster at endigotech.com
Fri Jan 20 19:21:43 MSK 2012


I'm sending it to a local extension. Nothing fancy. I'm asking where I
should put it in a dialplan.

-BDF

On Fri, Jan 20, 2012 at 10:34 AM, Brian West <brian at freeswitch.org> wrote:

> OK how are you sending calls? I'll take dialplan for 300 Alex!
>
> /b
>
> On Jan 20, 2012, at 8:55 AM, Brian Foster wrote:
>
> In this example:
>
> send 1149 bytes to udp/[<extension_ip>]:5060 at 14:47:46.972424:
>   ------------------------------------------------------------------------
>   INVITE sip:<extension>@<extension_ip> SIP/2.0
>   Via: SIP/2.0/UDP 204.93.201.164;rport;branch=z9hG4bKDZKU0eQBtr3Be
>   Max-Forwards: 64
>   From: "UNKNOWN" <sip:<caller_id_number>@<FS_ip>>;tag=FFr3B9gSB0KZK
>   To: <sip:<extension>@<extension_ip>:5060>
>   ...
>
> I need to change sip_invite_req_uri and sip_invite_to_uri so that the
> switch downstream can route the call based on the DID they have with me.
> I've tried setting this several different ways but no avail. I'd like it to
> be global if possible. Is there a good spot to set these variables?
>
>
> --
> Brian West
> FreeSWITCH Solutions, LLC
> Phone: +1 (918) 420-9266
> Fax:   +1 (918) 420-9267
> brian at freeswitch.org
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
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>
> 
> 
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>


-- 
Brian D. Foster
Endigo Computer LLC
Email: bdfoster at endigotech.com
Phone: 317-429-1069
Indianapolis, Indiana, USA

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