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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'>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)<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=MsoListParagraph style='text-indent:-18.0pt;mso-list:l0 level1 lfo1'><![if !supportLists]><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><span style='mso-list:Ignore'>1)<span style='font:7.0pt "Times New Roman"'> </span></span></span><![endif]><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? <o:p></o:p></span></p><p class=MsoListParagraph style='text-indent:-18.0pt;mso-list:l0 level1 lfo1'><![if !supportLists]><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><span style='mso-list:Ignore'>2)<span style='font:7.0pt "Times New Roman"'> </span></span></span><![endif]><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?<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'>Cheers<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Russell<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"'> cisco-voip [mailto:cisco-voip-bounces@puck.nether.net] <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<o:p></o:p></span></p></div></div><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>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><a href="http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/8x/trunks.html#wp1122456">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><o:p> </o:p></p></div><div><blockquote style='margin-top:5.0pt;margin-bottom:5.0pt'><p class=MsoNormal><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><o:p> </o:p></p></div><p class=MsoNormal>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><o:p> </o:p></p><div><p class=MsoNormal><span class=apple-style-span><span style='font-family:"Helvetica","sans-serif"'>-Ryan</span></span> <o:p></o:p></p></div><p class=MsoNormal><o:p> </o:p></p><div><div><p class=MsoNormal>On Jun 5, 2013, at 8:32 AM, Carlo Calabrese <<a href="mailto:carlo_calabrese2006@yahoo.com">carlo_calabrese2006@yahoo.com</a>> wrote:<o:p></o:p></p></div><p class=MsoNormal><o:p> </o:p></p><div><div><p class=MsoNormal><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><span lang=EN-US><o:p></o:p></span></p></div><div><p class=MsoNormal><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><span lang=EN-US><o:p></o:p></span></p></div><div><p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'> </span><span lang=EN-US><o:p></o:p></span></p></div><div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm 0cm 0cm'><div><p class=MsoNormal><b><span lang=EN-US style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span class=apple-converted-space><span lang=EN-US style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'> </span></span><span lang=EN-US style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>Kenneth Hayes [mailto:kennethwhayes@<a href="http://gmail.com/"><span style='color:purple'>gmail.com</span></a>]<span class=apple-converted-space> </span><br><b>Sent:</b><span class=apple-converted-space> </span>Wednesday, June 05, 2013 3:26 AM<br><b>To:</b><span class=apple-converted-space> </span>Carlo Calabrese<br><b>Subject:</b><span class=apple-converted-space> </span>Re: [cisco-voip] SIP between CUCM clusters</span><span lang=EN-US><o:p></o:p></span></p></div></div><div><p class=MsoNormal><span lang=EN-US> <o:p></o:p></span></p></div><div><div><p class=MsoNormal><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.<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><div><p class=MsoNormal><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:<o:p></o:p></span></p></div><div><div><p class=MsoNormal><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.<o:p></o:p></span></p></div><div><p class=MsoNormal><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<o:p></o:p></span></p></div><div><p class=MsoNormal><span lang=EN-US>Or can you point me in the right direction.<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>Thanks.<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 style='margin-bottom:12.0pt'><span lang=EN-US><br>_______________________________________________<br>cisco-voip mailing list<br><a href="mailto:cisco-voip@puck.nether.net"><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><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 style='font-size:13.5pt;font-family:"Lucida Grande","serif"'>_______________________________________________<br>cisco-voip mailing list<br><a href="mailto:cisco-voip@puck.nether.net"><span style='color:purple'>cisco-voip@puck.nether.net</span></a><br><a href="https://puck.nether.net/mailman/listinfo/cisco-voip"><span style='color:purple'>https://puck.nether.net/mailman/listinfo/cisco-voip</span></a><o:p></o:p></span></p></div></div><p class=MsoNormal><o:p> </o:p></p></div></div></div></body></html>