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</o:shapelayout></xml><![endif]--></head><body lang=EN-IE link=blue vlink=purple><div class=WordSection1><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Excellent tips. Thanks guys<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>When you say SIP to H323 back to SIP are you talking about endpoints? My other concern is that one of these CUCM clusters already has a H323 ICT to a 3<sup>rd</sup> 8.X cluster. Calls need to be able to traverse through all of them but presence only shared between the 1<sup>st</sup> two and there are MGCP QSIG and H323 ISDN gateways hanging off all of them – I suspect I will run into more problems that will out way any possible benefits like presence sharing.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><div><div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm 0cm 0cm'><p class=MsoNormal><b><span lang=EN-US style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span lang=EN-US style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'> Carlo Calabrese [mailto:carlo_calabrese2006@yahoo.com] <br><b>Sent:</b> 12 July 2013 17:11<br><b>To:</b> 'Erick Wellnitz'; Russell Chaseling<br><b>Cc:</b> 'Ryan Ratliff'; 'Cisco-voip'<br><b>Subject:</b> RE: [cisco-voip] SIP between CUCM clusters<o:p></o:p></span></p></div></div><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>I am running 8.6 and just got it working In the lab. One gotcha is make sure you use the IP address of the servers in the System->Cisco Unified CM.<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>“Cisco Unified Communications Manager Name”<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>It will use that as the name in the SIP request. I had the publisher set up with the name and it wouldn’t work. One I fixed that, it works great.<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>I haven’t test the BLF but will try and work on that in the next few weeks.<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>I think it is worth it, when you go between SIP to H323 back to SIP you run into all sorts of tiny problems including <o:p></o:p></span></p><p class=MsoListParagraph style='text-indent:-18.0pt;mso-list:l0 level1 lfo2'><![if !supportLists]><span lang=EN-US style='font-size:11.0pt;font-family:Symbol;color:#1F497D'><span style='mso-list:Ignore'>·<span style='font:7.0pt "Times New Roman"'> </span></span></span><![endif]><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>no audio because the rtp paths don’t get built correctly. ( found that one by putting the call on hold and unhold and the RTP streams rebuilt and worked fine.<o:p></o:p></span></p><p class=MsoListParagraph style='text-indent:-18.0pt;mso-list:l0 level1 lfo2'><![if !supportLists]><span lang=EN-US style='font-size:11.0pt;font-family:Symbol;color:#1F497D'><span style='mso-list:Ignore'>·<span style='font:7.0pt "Times New Roman"'> </span></span></span><![endif]><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Multicast setups and break downs.<o:p></o:p></span></p><p class=MsoListParagraph style='text-indent:-18.0pt;mso-list:l0 level1 lfo2'><![if !supportLists]><span lang=EN-US style='font-size:11.0pt;font-family:Symbol;color:#1F497D'><span style='mso-list:Ignore'>·<span style='font:7.0pt "Times New Roman"'> </span></span></span><![endif]><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>DTMF fails once in a while.<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm 0cm 0cm'><p class=MsoNormal><b><span lang=EN-US style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span lang=EN-US style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'> Erick Wellnitz [mailto:ewellnitzvoip@gmail.com] <br><b>Sent:</b> Friday, July 12, 2013 8:08 AM<br><b>To:</b> Russell Chaseling<br><b>Cc:</b> Ryan Ratliff; Carlo Calabrese; Cisco-voip<br><b>Subject:</b> Re: [cisco-voip] SIP between CUCM clusters<o:p></o:p></span></p></div><p class=MsoNormal><span lang=EN-US><o:p> </o:p></span></p><div><div><p class=MsoNormal><span lang=EN-US>I have deployed SIP between a 6.1 cluster and an 8.6 cluster for migration purposes.<o:p></o:p></span></p></div><div><p class=MsoNormal><span lang=EN-US> <o:p></o:p></span></p></div><div><p class=MsoNormal><span lang=EN-US>It worked perfect once I configured it correctly. <o:p></o:p></span></p></div><div><p class=MsoNormal><span lang=EN-US> <o:p></o:p></span></p></div><div><p class=MsoNormal><span lang=EN-US>Make sure each side only points to CUCM nodes in the CM group of the trunk's device pool...if that makes sense. Or, I believe you could check the 'run on all nodes' box in the trunk configuration for both/all clusters.<o:p></o:p></span></p></div></div><div><p class=MsoNormal style='margin-bottom:12.0pt'><span lang=EN-US><o:p> </o:p></span></p><div><p class=MsoNormal><span lang=EN-US>On Fri, Jul 12, 2013 at 8:08 AM, Russell Chaseling <<a href="mailto:rchaseling@plannet21.ie" target="_blank">rchaseling@plannet21.ie</a>> wrote:<o:p></o:p></span></p><div><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>I’m also looking at deploying SIP trunks between two CUCM 8.6 clusters for IP calling for the simple reason that I believe that presence status can be shared between the two ie IP Phone BLF and CUEAC BLF (particularly CUEAC BLF)</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'> </span><o:p></o:p></p><p><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>1)</span><span style='font-size:7.0pt;color:#1F497D'> </span><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>First questions is have I heard this right? Will BLF work between 2 x CUCM clusters? </span><o:p></o:p></p><p><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>2)</span><span style='font-size:7.0pt;color:#1F497D'> </span><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Has anyone deployed SIP trunks between clusters in production? If so is it worth the pain?</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'> </span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Cheers</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Russell</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'> </span><o:p></o:p></p><div><div style='border:none;border-top:solid windowtext 1.0pt;padding:3.0pt 0cm 0cm 0cm;border-color:currentColor currentColor'><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><b><span lang=EN-US style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span lang=EN-US style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'> cisco-voip [mailto:<a href="mailto:cisco-voip-bounces@puck.nether.net" target="_blank">cisco-voip-bounces@puck.nether.net</a>] <b>On Behalf Of </b>Ryan Ratliff<br><b>Sent:</b> 05 June 2013 15:24<br><b>To:</b> Carlo Calabrese<br><b>Cc:</b> 'Cisco-voip'<br><b>Subject:</b> Re: [cisco-voip] SIP between CUCM clusters</span><o:p></o:p></p></div></div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>You can use SIP trunks between CUCM clusters in lieu of ICTs. The 8.x SRND specifically states that QSIG-tunneled SIP trunks have feature parity with H.323 ICTs.<o:p></o:p></p><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><a href="http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/8x/trunks.html#wp1122456" target="_blank">http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/8x/trunks.html#wp1122456</a><o:p></o:p></p></div><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p></div><div><blockquote style='margin-top:5.0pt;margin-bottom:5.0pt'><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-size:9.0pt;font-family:"Arial","sans-serif";background:white'>The SIP trunk features available in the current release of Unified CM make SIP the preferred choice for new and existing trunk connections. The QSIG over SIP feature provides parity with H.323 intercluster trunks and can also be used to provide QSIG over SIP trunk connections to Cisco IOS gateways (and on to QSIG-based TDM PBXs). The ability to run on all Unified CM nodes and to handle up to 16 destination IP addresses improves outbound call distribution from Unified CM clusters and reduces the number of SIP trunks required between clusters and devices. SIP OPTIONS ping provides dynamic reachability detection for SIP trunk destinations, rather than per-call reachability determination. SIP Early Offer support for voice and video calls (insert MTP if needed) can reduce or eliminate the need to use MTPs and allows voice, video, and encrypted calls to be made over SIP Early Offer trunks.</span><o:p></o:p></p></blockquote><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p></div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>In general anything you can do to reduce the complexity of your call flows makes your life that much easier both in configuration and troubleshooting. <o:p></o:p></p><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-family:"Helvetica","sans-serif"'>-Ryan</span> <o:p></o:p></p></div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p><div><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>On Jun 5, 2013, at 8:32 AM, Carlo Calabrese <<a href="mailto:carlo_calabrese2006@yahoo.com" target="_blank">carlo_calabrese2006@yahoo.com</a>> wrote:<o:p></o:p></p></div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p><div><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>This is the problem I am having. Intercluster turnks are H323 based not sip. I have them built now, but it breaks all sorts of stuff.</span><o:p></o:p></p></div><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>multicast, not building rtp streams correctly. Dropped calls DTMF.</span><o:p></o:p></p></div><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'> </span><o:p></o:p></p></div><div style='border:none;border-top:solid windowtext 1.0pt;padding:3.0pt 0cm 0cm 0cm;border-color:currentColor currentColor'><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><b><span lang=EN-US style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span lang=EN-US style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'> Kenneth Hayes [mailto:<a href="mailto:kennethwhayes@" target="_blank">kennethwhayes@</a><a href="http://gmail.com/" target="_blank"><span style='color:purple'>gmail.com</span></a>] <br><b>Sent:</b> Wednesday, June 05, 2013 3:26 AM<br><b>To:</b> Carlo Calabrese<br><b>Subject:</b> Re: [cisco-voip] SIP between CUCM clusters</span><o:p></o:p></p></div></div><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US> </span><o:p></o:p></p></div><div><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>That would be call a Intercluster trunk. Build it like you would a regular SIP trunk but instead of selecting SIP trunk go to Trunk->Add New->Trunk type should be InterCluster Trunk Non-Gatekeeper Controlled" put in the required information and I'm not sure if MTP is needed or not it's been awhile since I've done a ICT but that should get you going. Also you will have to make sure you have the correct PT's and CSS on both ends and dial-plan modifications.</span><o:p></o:p></p></div></div><div><p class=MsoNormal style='mso-margin-top-alt:auto;margin-bottom:12.0pt'><span lang=EN-US> </span><o:p></o:p></p><div><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>On Wed, Jun 5, 2013 at 12:19 AM, Carlo Calabrese <<a href="mailto:carlo_calabrese2006@yahoo.com" target="_blank"><span style='color:purple'>carlo_calabrese2006@yahoo.com</span></a>> wrote:</span><o:p></o:p></p></div><div><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>Has anyone done a sip trunk between CUCM clusters? I am running 8.6 in production and 8.0 in the lab.</span><o:p></o:p></p></div><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>I can make it work with intercluster trunks ok. But I keep running into odd bugs with calls going from sip to h323 back to sip</span><o:p></o:p></p></div><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>Or can you point me in the right direction.</span><o:p></o:p></p></div><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US> </span><o:p></o:p></p></div><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>Thanks.</span><o:p></o:p></p></div><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US> </span><o:p></o:p></p></div></div><p class=MsoNormal style='mso-margin-top-alt:auto;margin-bottom:12.0pt'><span lang=EN-US><br>_______________________________________________<br>cisco-voip mailing list<br><a href="mailto:cisco-voip@puck.nether.net" target="_blank"><span style='color:purple'>cisco-voip@puck.nether.net</span></a><br><a href="https://puck.nether.net/mailman/listinfo/cisco-voip" target="_blank"><span style='color:purple'>https://puck.nether.net/mailman/listinfo/cisco-voip</span></a></span><o:p></o:p></p></div><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US> </span><o:p></o:p></p></div></div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US style='font-size:13.5pt;font-family:"Lucida Grande","serif"'>_______________________________________________<br>cisco-voip mailing list<br><a href="mailto:cisco-voip@puck.nether.net" target="_blank"><span style='color:purple'>cisco-voip@puck.nether.net</span></a><br><a href="https://puck.nether.net/mailman/listinfo/cisco-voip" target="_blank"><span style='color:purple'>https://puck.nether.net/mailman/listinfo/cisco-voip</span></a></span><o:p></o:p></p></div></div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p></div></div></div></div><p class=MsoNormal style='margin-bottom:12.0pt'><span lang=EN-US><br>_______________________________________________<br>cisco-voip mailing list<br><a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a><br><a href="https://puck.nether.net/mailman/listinfo/cisco-voip" target="_blank">https://puck.nether.net/mailman/listinfo/cisco-voip</a><o:p></o:p></span></p></div><p class=MsoNormal><span lang=EN-US><o:p> </o:p></span></p></div></div></body></html>