Can you compare and contrast the location 1 call flow from the location 2
call flow in detail? What's the same? What's different?
Can you provide the CUBE sip messaging for a failed call in location 1?
On Fri, Apr 24, 2020 at 2:19 PM Hamu Ebiso wrote:
> Hello Anthony and all,
>
> I have found
Hello Anthony and all,
I have found more information about this.
3 call handlers are involved with this.
Location 1 Main call handler
Clasified department call handler
Location Main call handler
call to location 1 Main call handler >> option 3 to Classified advertising >>
option 1 to going ba
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 per
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, A
Make sure you get both sides the inbound and outbound dial-peers...
And if you have all that set, CUCM logs probably going to be the next step….
Along with some sips from the voice services voip, sip-ua and dial-peers
> On Apr 24, 2020, at 9:17 AM, Ryan Huff wrote:
>
> RCF 4733, I believe.
>
RCF 4733, I believe.
Sent from my iPhone
On Apr 24, 2020, at 10:58, Anthony Holloway
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
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
pag
Thank you Jason for your questions. how can you setup rfc2833 In CUCM trunks?
thanks
From: Jason Aarons
Sent: Friday, April 24, 2020 8:24 AM
To: Hamu Ebiso
Cc: cisco-voip
Subject: Re: [cisco-voip] Ported Numbers to SIP call handler transfer is not
working corr
Hamu,
Are you saying then, that the issue is not DTMF? Does the system take your
input without error?
There's no magic bullet to fix transfers, you need to see what's happening,
and prescribe a fix. Can you share the SIP flow from the CUBE for the
entire call duration? Feel free to censor the s
Thank you very much for your help here. Is there anyway you could share the few
setting I could change on CUBE or CUCM if the issue is transffering?
thanks
From: Anthony Holloway
Sent: Friday, April 24, 2020 8:55 AM
To: Hamu Ebiso
Cc: cisco-voip@puck.nether.ne
Given the statement, “We ported our numbers to SIP yesterday. Now, their main
menu is not transferring numbers correctly”, I’m taking the change in ingress
signaling as the change agent and assuming nothing was changed in CUC/CUCM.
I’d suspect DTMF to be the cause in this case, as this can be a
The reason I ask is that the troubleshooting is a little different for each
issue.
*DTMF*
You would know if it's DTMF if for example, you push the button and the
voice recording just keeps on going. Most recordings are set such that if
you barge in on them, the recording ends abruptly to process
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,
I was thinking it might be Transfer issue. What makes you ask that question
Anthony?
thanks
Hamu
From: Anthony Holloway
Sent: Thursday, April 23, 2020 2:54 PM
To: Hamu Ebiso
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Ported Numbers to SIP call ha
14 matches
Mail list logo
