These are also for Outbound.... I need to match on an INBOUND To field.<br><br><div class="gmail_quote">On Mon, Jul 2, 2012 at 8:25 AM, Jason Burns <span dir="ltr"><<a href="mailto:burns.jason@gmail.com" target="_blank">burns.jason@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">IOS does have the ability to match on more than just the calling and called numbers. Unfortunately the Cisco dial-peer matching document hasn't been updated and I haven't found good official documentation, but look at this link<br>
<br><a href="https://supportforums.cisco.com/docs/DOC-25219" target="_blank">https://supportforums.cisco.com/docs/DOC-25219</a><br><br>You can match on many parts of the SIP headers to route SIP calls. Even more importantly, these header matches come before the usual calling and called number type matches in terms of matching order. That doc should get you started. It's for matching the incoming dial-peer, but the outbound dial-peer configuration is similar.<br>
<br>Actually - here is something that is a bit closer, but still not perfect. At Cisco Live I found some great slides describing this. Let me see if I can find those slides and come back with a better answer.<br><br><a href="http://www.cisco.com/en/US/docs/ios/voice/ivr/configuration/guide/gt_url.html#wp1064328" target="_blank">http://www.cisco.com/en/US/docs/ios/voice/ivr/configuration/guide/gt_url.html#wp1064328</a><span class="HOEnZb"><font color="#888888"><br>
<br>-Jason</font></span><div class="HOEnZb"><div class="h5"><br><br><div class="gmail_quote">On Mon, Jul 2, 2012 at 1:28 AM, Divin John <span dir="ltr"><<a href="mailto:dijohn@cisco.com" target="_blank">dijohn@cisco.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div style="font-size:14px;font-family:Calibri,sans-serif;word-wrap:break-word"><div>Try SIP Profiles.</div><div><a href="http://www.cisco.com/en/US/docs/ios-xml/ios/voice/cube_sip/configuration/15-2mt/voi-condl-header.html#GUID-980E7543-A7EC-41F4-800A-0ECAFAD8899F" target="_blank">http://www.cisco.com/en/US/docs/ios-xml/ios/voice/cube_sip/configuration/15-2mt/voi-condl-header.html#GUID-980E7543-A7EC-41F4-800A-0ECAFAD8899F</a></div>
<div><br></div><span><div style="border-right:medium none;padding-right:0in;padding-left:0in;padding-top:3pt;text-align:left;font-size:11pt;border-bottom:medium none;font-family:Calibri;border-top:#b5c4df 1pt solid;padding-bottom:0in;border-left:medium none">
<span style="font-weight:bold">From: </span> Jonathan Charles <<a href="mailto:jonvoip@gmail.com" target="_blank">jonvoip@gmail.com</a>><br><span style="font-weight:bold">Date: </span> Monday 2 July 2012 10:32 AM<br>
<span style="font-weight:bold">To: </span> <<a href="mailto:cisco-voip@puck.nether.net" target="_blank">cisco-voip@puck.nether.net</a>><br><span style="font-weight:bold">Subject: </span> [cisco-voip] Need SIP invite to route on TO field instead of URI<br>
</div><div><div><div><br></div>I have a CUBE running 15.1(4)M4 that is SIP to the provider and H.323 to CUCM 8.6.2<div><br></div><div>Inbound calls are all showing the billing number on the request URI:</div><div>
<br></div><div><div><br></div><div>Jul 1 09:48:43.709 CDT: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:</div><div>Received: </div><div>INVITE <a href="http://sip:1000@10.50.1.7:5060" target="_blank">sip:1000@10.50.1.7:5060</a> SIP/2.0</div>
<div>Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK-b09eefee3692d5c02b969912a045c958;rport</div><div>From: "DOE, JOHN" <<a href="mailto:sip%3A16305551414@call.message-alert.com" target="_blank">sip:16305551414@call.message-alert.com</a>>;tag=1813914414</div>
<div>To: <<a href="mailto:sip%3A16305551212@1.1.1.1" target="_blank">sip:16305551212@1.1.1.1</a>></div><div>Call-ID: 47e9ec2b@pbx</div><div>CSeq: 22580 INVITE</div><div>Max-Forwards: 70</div><div>Contact: <sip:<a href="mailto:1000@1.1.1.1" target="_blank">1000@1.1.1.1</a>:5060;transport=udp></div>
<div>Supported: 100rel, replaces, norefersub</div><div>Allow-Events: refer</div><div>Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE</div><div>Accept: application/sdp</div><div>User-Agent: talkingplatforms/2.1.15.2503</div>
<div>Alert-Info: <Bellcore-dr3></div><div>Content-Type: application/sdp</div><div>Content-Length: 323</div><div><br></div><div>v=0</div><div>o=- 1303493202 1303493202 IN IP4 66.159.89.13</div><div>s=-</div><div>c=IN IP4 1.1.1.1</div>
<div>t=0 0</div><div>m=audio 57094 RTP/AVP 0 9 18 2 3 101</div><div>a=rtpmap:0 pcmu/8000</div><div>a=rtpmap:9 g722/8000</div><div>a=rtpmap:18 g729/8000</div><div>a=fmtp:18 annexb=no</div><div>a=rtpmap:2 g726-32/8000</div>
<div>a=rtpmap:3 gsm/8000</div><div>a=rtpmap:101 telephone-event/8000</div><div>a=fmtp:101 0-16</div><div>a=sendrecv</div></div><div><br></div><div><br></div><div><br></div><div><br></div><div>As you can see the INVITE is to <a href="http://sip:1000@10.50.1.7:5060" target="_blank">sip:1000@10.50.1.7:5060</a>, which is our billing code.</div>
<div><br></div><div>The TO field has the actual DID to route to.</div><div><br></div><div><br></div><div>Per the RFC, 3261, section <a href="http://8.1.1.1" target="_blank">8.1.1.1</a>:</div><div><br></div><div>"The initial Request-URI of the message SHOULD be set to the value of the URI
in the To field." </div><div><br></div><div><br></div><div>This provider doesn't do that and says everyone but Cisco supports this, which goes back to my theory that SIP is not a protocol, but an idea of things you might want to do, but can really do whatever you want (hence the effect that every SIP provider seems to be doing their own thing)...</div>
<div><br></div><div><br></div><div>Anyway, here is what I need to do:</div><div><br></div><div>I need to have the call route on the TO field instead of the INVITE field.</div><div><br></div><div>How?</div><div><br></div>
<div>
<br></div><div><br></div><div>Jonathan</div></div></div>
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