<div dir="ltr">Ding ding ding! Winner! I wonder why Cisco doesn't update the CUCM UI. I was looking for DTMF support in a Telepresence Admin Guide for like an SX20 or something, and I couldn't find RFC2833 mentioned anywhere, but it did mention RFC4733. Anyway, that's all the trivia I have for now.<br></div><br><div class="gmail_quote"><div dir="ltr" class="gmail_attr">On Fri, Apr 24, 2020 at 10:17 AM Ryan Huff <<a href="mailto:ryanhuff@outlook.com">ryanhuff@outlook.com</a>> wrote:<br></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">
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RCF 4733, I believe.<br>
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<div dir="ltr">Sent from my iPhone</div>
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<blockquote type="cite">On Apr 24, 2020, at 10:58, Anthony Holloway <<a href="mailto:avholloway%2Bcisco-voip@gmail.com" target="_blank">avholloway+cisco-voip@gmail.com</a>> wrote:<br>
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<div dir="ltr">Actually you don't want to set rfc2833 (pop quiz: rfc2833 is not the real RFC number. What's the real RFC number? Don't google it, but reply if you know!) on your CUCM SIP Trunk to CUBE. You want No Preference. It's a setting right on the
SIP Trunk, just scroll to the bottom of the settings page. Also, on your CUBE dial-peers you don't want solely rtp-nte either, you want both rtp-nte and at least one Out of Band (OOB) option, like sip-kpml or sip-notify (thought the latter requires a SIP
Trunk Security Profile change from default to allow unsolicited NOTIFY).
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<div dir="ltr" class="gmail_attr">On Fri, Apr 24, 2020 at 9:35 AM Hamu Ebiso <<a href="mailto:hebiso2010@hotmail.com" target="_blank">hebiso2010@hotmail.com</a>> wrote:<br>
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Thank you Jason for your questions. how can you setup rfc2833 In CUCM trunks?</div>
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thanks<br>
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<div id="gmail-m_-4378112625464179853gmail-m_-6364694289785131301divRplyFwdMsg" dir="ltr"><font style="font-size:11pt" face="Calibri, sans-serif" color="#000000"><b>From:</b> Jason Aarons <<a href="mailto:scubajasona@gmail.com" target="_blank">scubajasona@gmail.com</a>><br>
<b>Sent:</b> Friday, April 24, 2020 8:24 AM<br>
<b>To:</b> Hamu Ebiso <<a href="mailto:hebiso2010@hotmail.com" target="_blank">hebiso2010@hotmail.com</a>><br>
<b>Cc:</b> cisco-voip <<a href="mailto:cisco-voip@puck.nether.net" target="_blank">cisco-voip@puck.nether.net</a>><br>
<b>Subject:</b> Re: [cisco-voip] Ported Numbers to SIP call handler transfer is not working correctly.</font>
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<div dir="auto">I am doubtful porting had anything to do with it. Was it tested fully before the port?
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<div dir="auto">Under dial peers is dtmf-relay rtp-nte set? In CUCM trunks is rfc2833 set? How is Unity integrated with CUCM ? SIP? CXN Version?</div>
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<div dir="auto">Without some debugs /traces I suspect you won't find much. </div>
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<div dir="ltr">On Thu, Apr 23, 2020, 3:52 PM Hamu Ebiso <<a href="mailto:hebiso2010@hotmail.com" target="_blank">hebiso2010@hotmail.com</a>> wrote:<br>
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Hello team, <br>
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I hope someone have come across this issue and can help me. We ported our numbers to SIP yesterday. Now, their main menu is not transferring numbers correctly. For example, when you select classifieds, it is supposed to go to the LAC Classifieds call handler.
Selecting option 1 is not routing correctly. Calling the LAC numbers directly works.</div>
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What do you think might be causing this issue?</div>
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Thanks</div>
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Hamu<br>
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