So, how do you do this with a Cisco CUBE?<br><br><div class="gmail_quote">On Thu, Jul 5, 2012 at 8:20 AM, Mark Holloway <span dir="ltr"><<a href="mailto:mh@markholloway.com" target="_blank">mh@markholloway.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="word-wrap:break-word">The SP is expecting you to be SIP Connect compliant. If they had an Acme Packet SBC in their core this could easily be resolved for you by using the option sip-connect-pbx-reg=rewrite-all and then the request URI and TO would match based on the TO header. There are several customer edge devices out there which will route inbound calls on the TO header, but the fact not all devices (or IP PBX's) will do it, means having the SP resolve it in the core is much better for everyone. Perhaps you could inquire with the SP? <div>
<div class="h5"><div><br></div><div><div><br><div><div>On Jul 2, 2012, at 5:51 PM, Jason Burns wrote:</div><br><blockquote type="cite">That's certainly an interesting problem.<br><br>You have an inbound SIP message, and selecting the inbound dial-peer isn't really what you're concerned with. You're more concerned that the outbound H.323 dial-peer selection is only going to look at the called number in the SIP Request URI (which is the billing number).<br>
<br>It sounds like you need to do some work on the inbound SIP dial-peer to manipulate that Request URI and stick the TO header inside of it. Then you want to have this message processed by IOS to find the right H.323 dial-peer based on your modification.<br>
<br>Another option is to change the header that IOS looks at when going from SIP to H323 for pulling out the called number, but I have no idea how to do that.<br><br>This example would copy the TO header into the Request-URI portion, but I think you're right that it's really only meant for SIP to SIP. We'd pass the modified SIP message out the far side, but I don't know if we'd make the modification on the inbound leg and then change our next hop routing based on this modification. I don't think it was intended for SIP to H.323.<br>
<br><a href="https://supportforums.cisco.com/thread/2119596" target="_blank">https://supportforums.cisco.com/thread/2119596</a><br><br><br>How about converting the CUBE to be SIP-SIP? There may be other solutions but I'm not brushed up enough on my IOS SIP modifications to come up with something better! Anyone else have ideas?<br>
<br>-Jason<br><br><br><br><div class="gmail_quote">On Mon, Jul 2, 2012 at 10:12 AM, Jonathan Charles <span dir="ltr"><<a href="mailto:jonvoip@gmail.com" target="_blank">jonvoip@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
These are also for Outbound.... I need to match on an INBOUND To field.<div><div><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><font color="#888888"><br>
<br>-Jason</font></span><div><div><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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