<html xmlns:v="urn:schemas-microsoft-com:vml" xmlns:o="urn:schemas-microsoft-com:office:office" xmlns:w="urn:schemas-microsoft-com:office:word" xmlns:m="http://schemas.microsoft.com/office/2004/12/omml" xmlns="http://www.w3.org/TR/REC-html40"><head><meta http-equiv=Content-Type content="text/html; charset=us-ascii"><meta name=Generator content="Microsoft Word 12 (filtered medium)"><style><!--
/* Font Definitions */
@font-face
{font-family:Helvetica;
panose-1:2 11 6 4 2 2 2 2 2 4;}
@font-face
{font-family:Wingdings;
panose-1:5 0 0 0 0 0 0 0 0 0;}
@font-face
{font-family:"Cambria Math";
panose-1:2 4 5 3 5 4 6 3 2 4;}
@font-face
{font-family:Calibri;
panose-1:2 15 5 2 2 2 4 3 2 4;}
@font-face
{font-family:Tahoma;
panose-1:2 11 6 4 3 5 4 4 2 4;}
@font-face
{font-family:"Lucida Grande";}
/* Style Definitions */
p.MsoNormal, li.MsoNormal, div.MsoNormal
{margin:0in;
margin-bottom:.0001pt;
font-size:12.0pt;
font-family:"Times New Roman","serif";}
a:link, span.MsoHyperlink
{mso-style-priority:99;
color:blue;
text-decoration:underline;}
a:visited, span.MsoHyperlinkFollowed
{mso-style-priority:99;
color:purple;
text-decoration:underline;}
p
{mso-style-priority:99;
mso-margin-top-alt:auto;
margin-right:0in;
mso-margin-bottom-alt:auto;
margin-left:0in;
font-size:12.0pt;
font-family:"Times New Roman","serif";}
p.MsoAcetate, li.MsoAcetate, div.MsoAcetate
{mso-style-priority:99;
mso-style-link:"Balloon Text Char";
margin:0in;
margin-bottom:.0001pt;
font-size:8.0pt;
font-family:"Tahoma","sans-serif";}
p.MsoListParagraph, li.MsoListParagraph, div.MsoListParagraph
{mso-style-priority:34;
margin-top:0in;
margin-right:0in;
margin-bottom:0in;
margin-left:.5in;
margin-bottom:.0001pt;
font-size:12.0pt;
font-family:"Times New Roman","serif";}
span.EmailStyle18
{mso-style-type:personal-reply;
font-family:"Calibri","sans-serif";
color:#1F497D;}
span.BalloonTextChar
{mso-style-name:"Balloon Text Char";
mso-style-priority:99;
mso-style-link:"Balloon Text";
font-family:"Tahoma","sans-serif";}
.MsoChpDefault
{mso-style-type:export-only;}
@page WordSection1
{size:8.5in 11.0in;
margin:1.0in 1.0in 1.0in 1.0in;}
div.WordSection1
{page:WordSection1;}
/* List Definitions */
@list l0
{mso-list-id:1066103320;
mso-list-type:hybrid;
mso-list-template-ids:-2001805200 67698689 67698691 67698693 67698689 67698691 67698693 67698689 67698691 67698693;}
@list l0:level1
{mso-level-number-format:bullet;
mso-level-text:\F0B7;
mso-level-tab-stop:none;
mso-level-number-position:left;
text-indent:-.25in;
font-family:Symbol;}
ol
{margin-bottom:0in;}
ul
{margin-bottom:0in;}
--></style><!--[if gte mso 9]><xml>
<o:shapedefaults v:ext="edit" spidmax="1026" />
</xml><![endif]--><!--[if gte mso 9]><xml>
<o:shapelayout v:ext="edit">
<o:idmap v:ext="edit" data="1" />
</o:shapelayout></xml><![endif]--></head><body lang=EN-US link=blue vlink=purple><div class=WordSection1><p class=MsoNormal><span 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 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 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 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'>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 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:-.25in;mso-list:l0 level1 lfo1'><![if !supportLists]><span 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 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:-.25in;mso-list:l0 level1 lfo1'><![if !supportLists]><span 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 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:-.25in;mso-list:l0 level1 lfo1'><![if !supportLists]><span 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 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 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 0in 0in 0in'><p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span 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><o:p> </o:p></p><div><div><p class=MsoNormal>I have deployed SIP between a 6.1 cluster and an 8.6 cluster for migration purposes.<o:p></o:p></p></div><div><p class=MsoNormal> <o:p></o:p></p></div><div><p class=MsoNormal>It worked perfect once I configured it correctly. <o:p></o:p></p></div><div><p class=MsoNormal> <o:p></o:p></p></div><div><p class=MsoNormal>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></p></div></div><div><p class=MsoNormal style='margin-bottom:12.0pt'><o:p> </o:p></p><div><p class=MsoNormal>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></p><div><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-IE 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><span lang=EN-IE><o:p></o:p></span></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-IE style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'> </span><span lang=EN-IE><o:p></o:p></span></p><p><span lang=EN-IE style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>1)</span><span lang=EN-IE style='font-size:7.0pt;color:#1F497D'> </span><span lang=EN-IE 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><span lang=EN-IE><o:p></o:p></span></p><p><span lang=EN-IE style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>2)</span><span lang=EN-IE style='font-size:7.0pt;color:#1F497D'> </span><span lang=EN-IE 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><span lang=EN-IE><o:p></o:p></span></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-IE style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'> </span><span lang=EN-IE><o:p></o:p></span></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-IE style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Cheers</span><span lang=EN-IE><o:p></o:p></span></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-IE style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Russell</span><span lang=EN-IE><o:p></o:p></span></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-IE style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'> </span><span lang=EN-IE><o:p></o:p></span></p><div><div style='border:none;border-top:solid windowtext 1.0pt;padding:3.0pt 0in 0in 0in;border-color:currentColor currentColor'><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span 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><span lang=EN-IE><o:p></o:p></span></p></div></div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-IE> <o:p></o:p></span></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-IE>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></span></p><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-IE><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></span></p></div><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-IE> <o:p></o:p></span></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 lang=EN-IE 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><span lang=EN-IE><o:p></o:p></span></p></blockquote><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-IE> <o:p></o:p></span></p></div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-IE>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></span></p><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-IE> <o:p></o:p></span></p><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-IE style='font-family:"Helvetica","sans-serif"'>-Ryan</span><span lang=EN-IE> <o:p></o:p></span></p></div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-IE> <o:p></o:p></span></p><div><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-IE>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></span></p></div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-IE> <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'>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-IE><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'>multicast, not building rtp streams correctly. Dropped calls DTMF.</span><span lang=EN-IE><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'> </span><span lang=EN-IE><o:p></o:p></span></p></div><div style='border:none;border-top:solid windowtext 1.0pt;padding:3.0pt 0in 0in 0in;border-color:currentColor currentColor'><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span 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><span lang=EN-IE><o:p></o:p></span></p></div></div><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <span lang=EN-IE><o:p></o:p></span></p></div><div><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>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 lang=EN-IE><o:p></o:p></span></p></div></div><div><p class=MsoNormal style='mso-margin-top-alt:auto;margin-bottom:12.0pt'> <span lang=EN-IE><o:p></o:p></span></p><div><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>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 lang=EN-IE><o:p></o:p></span></p></div><div><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>Has anyone done a sip trunk between CUCM clusters? I am running 8.6 in production and 8.0 in the lab.<span lang=EN-IE><o:p></o:p></span></p></div><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>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 lang=EN-IE><o:p></o:p></span></p></div><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>Or can you point me in the right direction.<span lang=EN-IE><o:p></o:p></span></p></div><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <span lang=EN-IE><o:p></o:p></span></p></div><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>Thanks.<span lang=EN-IE><o:p></o:p></span></p></div><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <span lang=EN-IE><o:p></o:p></span></p></div></div><p class=MsoNormal style='mso-margin-top-alt:auto;margin-bottom:12.0pt'><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 lang=EN-IE><o:p></o:p></span></p></div><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <span lang=EN-IE><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:13.5pt;font-family:"Lucida Grande"'>_______________________________________________<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><span lang=EN-IE><o:p></o:p></span></p></div></div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-IE> <o:p></o:p></span></p></div></div></div></div><p class=MsoNormal style='margin-bottom:12.0pt'><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></p></div><p class=MsoNormal><o:p> </o:p></p></div></div></body></html>