I invoked the dispatcher, but it doesn't appear anything was set. My flow is currently:
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<br /> ds_select_dst("1","4");
<br /> sl_send_reply("301","redirected");
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<br />INVITE sip:+15552021000@10.2.14.100 SIP/2.0
<br />Via: SIP/2.0/UDP 10.2.252.190:5060;branch=z9hG4bK62554e20;rport
<br />From: "Geoff" <sip:6789050671@connectfirst.com>;tag=as1d87dcd5
<br />To: <sip:+15552021000@10.2.14.100>
<br />Contact: <sip:6789050671@10.2.252.190>
<br />Call-ID: 7e842b5a2fc02f6336a8552e4809fa67@connectfirst.com
<br />CSeq: 102 INVITE
<br />User-Agent: G-Tel v1.0
<br />Max-Forwards: 70
<br />Date: Tue, 11 Nov 2008 15:03:34 GMT
<br />Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
<br />Supported: replaces
<br />Content-Type: application/sdp
<br />Content-Length: 240
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<br />SIP/2.0 301 redirected
<br />Via: SIP/2.0/UDP 10.2.252.190:5060;branch=z9hG4bK62554e20;rport=5060
<br />From: "Geoff" <sip:6789050671@connectfirst.com>;tag=as1d87dcd5
<br />To: <sip:+15552021000@10.2.14.100>;tag=b7b83ebb8b164c9706593e1cbef4dcee.af60
<br />Call-ID: 7e842b5a2fc02f6336a8552e4809fa67@connectfirst.com
<br />CSeq: 102 INVITE
<br />Server: OpenSIPS (1.4.2-notls (x86_64/linux))
<br />Content-Length: 0
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<br />On Nov 11, 2008 9:59am, Bogdan-Andrei Iancu <bogdan@voice-system.ro> wrote:
<br />> Well, after the new RURI was set with dispatcher, do sl_send_reply("301","redirected");
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<br />> Automatically, the RURI will be placed as contact in 3xx reply.
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<br />> Regards,
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<br />> Bogdan
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<br />> Jeff Pyle wrote:
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<br />> Figures you would ask that. Ha! I'm not sure. I'm quite new to
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<br />> Anyone? Buehler?
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<br />> - Jeff
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<br />> -----Original Message-----
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<br />> From: Geoffrey Mina [mailto:geoffreymina@gmail.com] Sent: Tuesday, November 11, 2008 9:36 AM
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<br />> To: Jeff Pyle
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<br />> Cc: users@lists.opensips.org
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<br />> Subject: Re: [OpenSIPS-Users] Input on my loadbalancer configuration
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<br />> well.. that would just be too simple!! How would I integrate sending a
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<br />> 302 with the dispatcher module?
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<br />> thanks for the out-of-the-box idea!
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<br />> -Geoff
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<br />> On Tue, Nov 11, 2008 at 9:13 AM, Jeff Pyle jpyle@fidelityvoice.com>
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<br />> 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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<br />> - Jeff
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