[cisco-voip] Ported Numbers to SIP call handler transfer is not working correctly.

[Ryan Huff]

Yes, RFC 2833 is the older RFC and 4733 is the newer. I believe CUCM still references 2833 because 4733 could potentially result in a non supported DTMF scenario that would appear supported if CUCM stated it supported 4733 (Ex. a SBC not supporting all the same events that CUCM would require perhaps).

I haven’t dug into the topic in depth however, so I may not be correct.

Here is an excerpt from 4734 that sums it up; “ This document provides a number of clarifications to the original document. However, it specifically differs from RFC 2833 by removing the requirement that all compliant implementations support the DTMF events. Instead, compliant implementations taking part in out-of-band negotiations of media stream content indicate what events they support. This memo adds three new procedures to the RFC 2833 framework: subdivision of long events into segments, reporting of multiple events in a single packet, and the concept and reporting of state events.“


Sent from my iPhone

On Apr 24, 2020, at 12:01, Anthony Holloway <avholloway+cisco-voip at gmail.com> wrote:


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.

On Fri, Apr 24, 2020 at 10:17 AM Ryan Huff <ryanhuff at outlook.com<mailto:ryanhuff at outlook.com>> wrote:
RCF 4733, I believe.

Sent from my iPhone

On Apr 24, 2020, at 10:58, Anthony Holloway <avholloway+cisco-voip at gmail.com<mailto:avholloway%2Bcisco-voip at gmail.com>> wrote:


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).


On Fri, Apr 24, 2020 at 9:35 AM Hamu Ebiso <hebiso2010 at hotmail.com<mailto:hebiso2010 at hotmail.com>> wrote:
Thank you Jason for your questions. how can you setup rfc2833 In CUCM trunks?

thanks

________________________________
From: Jason Aarons <scubajasona at gmail.com<mailto:scubajasona at gmail.com>>
Sent: Friday, April 24, 2020 8:24 AM
To: Hamu Ebiso <hebiso2010 at hotmail.com<mailto:hebiso2010 at hotmail.com>>
Cc: cisco-voip <cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net>>
Subject: Re: [cisco-voip] Ported Numbers to SIP call handler transfer is not working correctly.

I am doubtful porting had anything to do with it. Was it tested fully before the port?

Under dial peers is dtmf-relay rtp-nte set? In CUCM trunks is rfc2833 set? How is Unity integrated with CUCM ? SIP? CXN Version?

Without some debugs /traces I suspect you won't find much.

On Thu, Apr 23, 2020, 3:52 PM Hamu Ebiso <hebiso2010 at hotmail.com<mailto:hebiso2010 at hotmail.com>> wrote:
Hello team,

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.
What do you think might be causing this issue?

Thanks
Hamu
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