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<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1F497D">We went from 15.4(3)M9 to 15.7(3)M4b<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">Noted also in this version that we previously allowed video/media to flow to the CUBE just in case we were ever to use it for routing media calls, and this version
(15.7) drops calls from Jabber, etc when we place them on hold. <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">TAC said it was something to do with the number of options in the SDP changing and it being a protocol violation. The IEC error and Q code did sort of indicate
this, but, as to why Cisco’s systems don’t want to play nice protocol wise with each other I don’t know.<o:p></o:p></span></p>
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<p class="MsoNormal"><b><span style="font-size:11.0pt;font-family:"Calibri",sans-serif">From:</span></b><span style="font-size:11.0pt;font-family:"Calibri",sans-serif"> Ryan Huff <ryanhuff@outlook.com>
<br>
<b>Sent:</b> Monday, December 2, 2019 10:25 AM<br>
<b>To:</b> Pawlowski, Adam <ajp26@buffalo.edu>; cisco-voip@puck.nether.net<br>
<b>Subject:</b> Re: Native Call Queuing on UCM 12.5 with SIP Trunks<o:p></o:p></span></p>
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<p class="MsoNormal" style="background:white"><span style="font-family:"Calibri",sans-serif;color:black">Adam,<o:p></o:p></span></p>
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<p class="MsoNormal" style="background:white"><span style="font-family:"Calibri",sans-serif;color:black">I've not encountered the streaming issue with a SIP trunk, just H.323.. interesting.<o:p></o:p></span></p>
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<p class="MsoNormal" style="background:white"><span style="font-family:"Calibri",sans-serif;color:black">Out of curiosity, what code level in the ISR G2 did you go to when you encountered this? The behavior of simplex streaming is to send a null/fake IP address
in the logic channel multimedia control message, which for anything required to check incoming RTP packets against the IP/port would fail (creating the
<i>silent</i> MoH condition) which I was under the impression was only the G3 ISRs.<o:p></o:p></span></p>
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<p class="MsoNormal"><span style="font-family:"Calibri",sans-serif;color:black">Thanks,<o:p></o:p></span></p>
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<p class="MsoNormal"><span style="font-family:"Calibri",sans-serif;color:black"><o:p> </o:p></span></p>
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<p class="MsoNormal"><span style="font-family:"Calibri",sans-serif;color:black">Ryan<o:p></o:p></span></p>
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<p class="MsoNormal"><b><span style="font-size:11.0pt;font-family:"Calibri",sans-serif;color:black">From:</span></b><span style="font-size:11.0pt;font-family:"Calibri",sans-serif;color:black"> cisco-voip <<a href="mailto:cisco-voip-bounces@puck.nether.net">cisco-voip-bounces@puck.nether.net</a>>
on behalf of Pawlowski, Adam <<a href="mailto:ajp26@buffalo.edu">ajp26@buffalo.edu</a>><br>
<b>Sent:</b> Monday, December 2, 2019 8:50 AM<br>
<b>To:</b> <a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a> <<a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>><br>
<b>Subject:</b> Re: [cisco-voip] Native Call Queuing on UCM 12.5 with SIP Trunks</span>
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<p class="xmsonormal"><span style="font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1F497D">After an upgrade on our CUBE to a later IOS (ISR G2) we needed to have the duplex streaming parameter enabled for MOH/Ringback to work properly.
</span><o:p></o:p></p>
<p class="xmsonormal"><span style="font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1F497D"> </span><o:p></o:p></p>
<p class="xmsonormal"><span style="font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1F497D">I believe there is a note about that being needed for the 4400 series gateways as well, but, I know we didn’t have it before but needed it at some point to get
media to be heard properly by the far end in a network hold kind of situation.</span><o:p></o:p></p>
<p class="xmsonormal"><span style="font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1F497D"> </span><o:p></o:p></p>
<p class="xmsonormal"><span style="font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1F497D">Adam</span><o:p></o:p></p>
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<p class="xmsonormal"><b><span style="font-size:11.0pt;font-family:"Calibri",sans-serif">From:</span></b><span style="font-size:11.0pt;font-family:"Calibri",sans-serif"> cisco-voip <<a href="mailto:cisco-voip-bounces@puck.nether.net">cisco-voip-bounces@puck.nether.net</a>>
<b>On Behalf Of </b>Mark H. Turpin<br>
<b>Sent:</b> Monday, December 2, 2019 8:20 AM<br>
<b>To:</b> Dana Tong <<a href="mailto:dana.tong@yellit.com.au">dana.tong@yellit.com.au</a>>;
<a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a><br>
<b>Subject:</b> Re: [cisco-voip] Native Call Queuing on UCM 12.5 with SIP Trunks</span><o:p></o:p></p>
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<p class="xmsonormal"> <o:p></o:p></p>
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<p class="xmsonormal"><span style="font-family:"Calibri",sans-serif;color:black">Does the problem go away if you force an MTP?</span><o:p></o:p></p>
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<p class="xmsonormal"><span style="font-family:"Calibri",sans-serif;color:black"> </span><o:p></o:p></p>
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<p class="xmsonormal"><span style="font-family:"Calibri",sans-serif;color:black">Can you provide a debug ccsip messages of an external call?</span><o:p></o:p></p>
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<p class="xmsonormal"><span style="font-family:"Calibri",sans-serif;color:black"> </span><o:p></o:p></p>
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<p class="xmsonormal"><span style="font-family:"Calibri",sans-serif;color:black">Can you share your sanitized CUBE config?</span><o:p></o:p></p>
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<p class="xmsonormal"><span style="font-family:"Calibri",sans-serif;color:black"> </span><o:p></o:p></p>
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<p class="xmsonormal"><span style="font-family:"Calibri",sans-serif;color:black">Curious if your call/media is changing after the initial announcement and your ITSP doesn't care for the way you're attempting to change it.</span><o:p></o:p></p>
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<p class="xmsonormal"><span style="font-family:"Calibri",sans-serif;color:black"> </span><o:p></o:p></p>
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<p class="xmsonormal"><b><span style="font-size:11.0pt;font-family:"Calibri",sans-serif;color:black">From:</span></b><span style="font-size:11.0pt;font-family:"Calibri",sans-serif;color:black"> cisco-voip <<a href="mailto:cisco-voip-bounces@puck.nether.net">cisco-voip-bounces@puck.nether.net</a>>
on behalf of Dana Tong <<a href="mailto:dana.tong@yellit.com.au">dana.tong@yellit.com.au</a>><br>
<b>Sent:</b> Monday, December 2, 2019 12:36 AM<br>
<b>To:</b> <a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a> <<a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>><br>
<b>Subject:</b> [cisco-voip] Native Call Queuing on UCM 12.5 with SIP Trunks</span>
<o:p></o:p></p>
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<p class="xmsonormal" style="margin-bottom:12.0pt"><span lang="EN-AU">*** EXTERNAL EMAIL - DO NOT CLICK LINKS ***
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<p class="xxmsonormal"><span lang="EN-AU" style="font-size:11.0pt;font-family:"Calibri",sans-serif">Hi all,</span><o:p></o:p></p>
<p class="xxmsonormal"><span lang="EN-AU" style="font-size:11.0pt;font-family:"Calibri",sans-serif"> </span><o:p></o:p></p>
<p class="xxmsonormal"><span lang="EN-AU" style="font-size:11.0pt;font-family:"Calibri",sans-serif">This being back on the tools is doing my head in. I think I’ve been away too long.</span><o:p></o:p></p>
<p class="xxmsonormal"><span lang="EN-AU" style="font-size:11.0pt;font-family:"Calibri",sans-serif"> </span><o:p></o:p></p>
<p class="xxmsonormal"><span lang="EN-AU" style="font-size:11.0pt;font-family:"Calibri",sans-serif">So I have configured native call queuing on UCM 12.5.x</span><o:p></o:p></p>
<p class="xxmsonormal"><span lang="EN-AU" style="font-size:11.0pt;font-family:"Calibri",sans-serif">Internal calls queue fine. The initial announcement plays. Period announcements work and the MOH is fine in between.</span><o:p></o:p></p>
<p class="xxmsonormal"><span lang="EN-AU" style="font-size:11.0pt;font-family:"Calibri",sans-serif"> </span><o:p></o:p></p>
<p class="xxmsonormal"><span lang="EN-AU" style="font-size:11.0pt;font-family:"Calibri",sans-serif">External calls hear the initial announcement.
</span><o:p></o:p></p>
<p class="xxmsonormal"><span lang="EN-AU" style="font-size:11.0pt;font-family:"Calibri",sans-serif">There is no MOH after the announcement and the external user has ring-back tone.</span><o:p></o:p></p>
<p class="xxmsonormal"><span lang="EN-AU" style="font-size:11.0pt;font-family:"Calibri",sans-serif">There is no periodic announcement.
</span><o:p></o:p></p>
<p class="xxmsonormal"><span lang="EN-AU" style="font-size:11.0pt;font-family:"Calibri",sans-serif"> </span><o:p></o:p></p>
<p class="xxmsonormal"><span lang="EN-AU" style="font-size:11.0pt;font-family:"Calibri",sans-serif">Any thoughts on why internal is okay and external is not working? Is it relating to SIP supplementary services?</span><o:p></o:p></p>
<p class="xxmsonormal"><span lang="EN-AU" style="font-size:11.0pt;font-family:"Calibri",sans-serif"> </span><o:p></o:p></p>
<p class="xxmsonormal"><span lang="EN-AU" style="font-size:11.0pt;font-family:"Calibri",sans-serif">Cheers</span><o:p></o:p></p>
<p class="xxmsonormal"><span lang="EN-AU" style="font-size:11.0pt;font-family:"Calibri",sans-serif">Dana</span><o:p></o:p></p>
<p class="xxmsonormal"><span lang="EN-AU" style="font-size:11.0pt;font-family:"Calibri",sans-serif"> </span><o:p></o:p></p>
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