[OpenSIPS-Users] Input on my loadbalancer configuration

[geoffreymina at gmail.com]

I invoked the dispatcher, but it doesn't appear anything was set. My flow  
is currently:


ds_select_dst("1","4");
sl_send_reply("301","redirected");



INVITE sip:+15552021000 at 10.2.14.100 SIP/2.0
Via: SIP/2.0/UDP 10.2.252.190:5060;branch=z9hG4bK62554e20;rport
From: "Geoff" <sip:6789050671 at connectfirst.com>;tag=as1d87dcd5
To: <sip:+15552021000 at 10.2.14.100>
Contact: <sip:6789050671 at 10.2.252.190>
Call-ID: 7e842b5a2fc02f6336a8552e4809fa67 at connectfirst.com
CSeq: 102 INVITE
User-Agent: G-Tel v1.0
Max-Forwards: 70
Date: Tue, 11 Nov 2008 15:03:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 240

SIP/2.0 301 redirected
Via: SIP/2.0/UDP 10.2.252.190:5060;branch=z9hG4bK62554e20;rport=5060
From: "Geoff" <sip:6789050671 at connectfirst.com>;tag=as1d87dcd5
To: <sip:+15552021000 at 10.2.14.100>;tag=b7b83ebb8b164c9706593e1cbef4dcee.af60
Call-ID: 7e842b5a2fc02f6336a8552e4809fa67 at connectfirst.com
CSeq: 102 INVITE
Server: OpenSIPS (1.4.2-notls (x86_64/linux))
Content-Length: 0



On Nov 11, 2008 9:59am, Bogdan-Andrei Iancu <bogdan at voice-system.ro> wrote:
> Well, after the new RURI was set with dispatcher, do  
sl_send_reply("301","redirected");
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> Automatically, the RURI will be placed as contact in 3xx reply.
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> Regards,
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> Bogdan
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> Jeff Pyle wrote:
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> Figures you would ask that. Ha! I'm not sure. I'm quite new to
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> Opensips, and haven't made it to the dispatcher module yet.
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> Anyone? Buehler?
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> - Jeff
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> -----Original Message-----
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> From: Geoffrey Mina [mailto:geoffreymina at gmail.com] Sent: Tuesday,  
November 11, 2008 9:36 AM
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> To: Jeff Pyle
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> Cc: users at lists.opensips.org
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> Subject: Re: [OpenSIPS-Users] Input on my loadbalancer configuration
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> well.. that would just be too simple!! How would I integrate sending a
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> 302 with the dispatcher module?
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> thanks for the out-of-the-box idea!
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> -Geoff
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> On Tue, Nov 11, 2008 at 9:13 AM, Jeff Pyle jpyle at fidelityvoice.com>
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> What if instead of routing the INVITE, Opensips returned a 302 to the  
originator, redirecting them to the Asterisk server(s) of choice for that  
call?
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> - Jeff
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