Mailing List Archive

Is this a DTMF issue, or a transfer issue?

On Thu, Apr 23, 2020 at 2:51 PM Hamu Ebiso <hebiso2010@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
> _______________________________________________
> cisco-voip mailing list
> cisco-voip@puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

I was thinking it might be Transfer issue. What makes you ask that question Anthony?

thanks
Hamu

________________________________
From: Anthony Holloway <avholloway+cisco-voip@gmail.com>
Sent: Thursday, April 23, 2020 2:54 PM
To: Hamu Ebiso <hebiso2010@hotmail.com>
Cc: cisco-voip@puck.nether.net <cisco-voip@puck.nether.net>
Subject: Re: [cisco-voip] Ported Numbers to SIP call handler transfer is not working correctly.

Is this a DTMF issue, or a transfer issue?

On Thu, Apr 23, 2020 at 2:51 PM Hamu Ebiso <hebiso2010@hotmail.com<mailto:hebiso2010@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
_______________________________________________
cisco-voip mailing list
cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>
https://puck.nether.net/mailman/listinfo/cisco-voip

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@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
> _______________________________________________
> cisco-voip mailing list
> cisco-voip@puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

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 your input.

OR

You would know if it's a DTMF issue if for example, you press a button and
CUC processes it twice, as in double digits. This might be a little harder
to tell from UX, but it might be easier if you setup a test number to the
Opening Greeting and pressing * exists the app, versus taking you to Login.

*Transfer*

You would know if it's a Transfer issue, if it wasn't a DTMF issue. I.e.,
You press the button, the recording stops, or even says, "Wait while I
transfer your call" and then the failure happens.

Transfer failures could happen for a few different reasons, and there's a
few settings on CUBE and within CUCM which can affect how a transfer
functions, thus improving success with each knob turned.

On Fri, Apr 24, 2020 at 7:26 AM Hamu Ebiso <hebiso2010@hotmail.com> wrote:

> I was thinking it might be Transfer issue. What makes you ask that
> question Anthony?
>
> thanks
> Hamu
>
> ------------------------------
> *From:* Anthony Holloway <avholloway+cisco-voip@gmail.com>
> *Sent:* Thursday, April 23, 2020 2:54 PM
> *To:* Hamu Ebiso <hebiso2010@hotmail.com>
> *Cc:* cisco-voip@puck.nether.net <cisco-voip@puck.nether.net>
> *Subject:* Re: [cisco-voip] Ported Numbers to SIP call handler transfer
> is not working correctly.
>
> Is this a DTMF issue, or a transfer issue?
>
> On Thu, Apr 23, 2020 at 2:51 PM Hamu Ebiso <hebiso2010@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
> _______________________________________________
> cisco-voip mailing list
> cisco-voip@puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>

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 common symptom when switching to SIP from (I’m assuming this as well) TDM (PRI).

If DTMF were an issue, a likely and more immediate fix (though I’d only consider this temporary, I wouldn’t leave it this way) would be to check the “MTP Required” option on the ingress SIP trunk(s), then save/reset the SIP trunk(s) and test.

Back in the day, this was thought of as a solution, but it’s not, it’s just (if it works) masking the issue. It’s the difference between sweeping a dirty floor, or just laying new carpet on top of a dirty floor. Additional there are resource considerations within the CUCM cluster that you’d want to be concerned with because “MTP Required” in a scenario like this, would cause the media stream in every single call leg between the phone and (assuming CUBE) to terminate with CUCM.

If this were to work, then a DTMF mis-match would likely be the issue and that could a misconfiguration with EO, codecs... etc.

Thanks,

Ryan

On Apr 24, 2020, at 09:58, Anthony Holloway <avholloway+cisco-voip@gmail.com> wrote:

?
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 your input.

OR

You would know if it's a DTMF issue if for example, you press a button and CUC processes it twice, as in double digits. This might be a little harder to tell from UX, but it might be easier if you setup a test number to the Opening Greeting and pressing * exists the app, versus taking you to Login.

Transfer

You would know if it's a Transfer issue, if it wasn't a DTMF issue. I.e., You press the button, the recording stops, or even says, "Wait while I transfer your call" and then the failure happens.

Transfer failures could happen for a few different reasons, and there's a few settings on CUBE and within CUCM which can affect how a transfer functions, thus improving success with each knob turned.

On Fri, Apr 24, 2020 at 7:26 AM Hamu Ebiso <hebiso2010@hotmail.com<mailto:hebiso2010@hotmail.com>> wrote:
I was thinking it might be Transfer issue. What makes you ask that question Anthony?

thanks
Hamu

________________________________
From: Anthony Holloway <avholloway+cisco-voip@gmail.com<mailto:avholloway%2Bcisco-voip@gmail.com>>
Sent: Thursday, April 23, 2020 2:54 PM
To: Hamu Ebiso <hebiso2010@hotmail.com<mailto:hebiso2010@hotmail.com>>
Cc: cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net> <cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>>
Subject: Re: [cisco-voip] Ported Numbers to SIP call handler transfer is not working correctly.

Is this a DTMF issue, or a transfer issue?

On Thu, Apr 23, 2020 at 2:51 PM Hamu Ebiso <hebiso2010@hotmail.com<mailto:hebiso2010@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
_______________________________________________
cisco-voip mailing list
cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>
https://puck.nether.net/mailman/listinfo/cisco-voip<https://eur02.safelinks.protection.outlook.com/?url=https%3A%2F%2Fpuck.nether.net%2Fmailman%2Flistinfo%2Fcisco-voip&data=02%7C01%7C%7C5bf6ad819b3e4c4c966208d7e8578772%7C84df9e7fe9f640afb435aaaaaaaaaaaa%7C1%7C0%7C637233334936130722&sdata=t17te8vBHd1eEpHTgQMfQ9ra66Bf%2BCsD0gWEe77tcxw%3D&reserved=0>
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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 <avholloway+cisco-voip@gmail.com>
Sent: Friday, April 24, 2020 8:55 AM
To: Hamu Ebiso <hebiso2010@hotmail.com>
Cc: cisco-voip@puck.nether.net <cisco-voip@puck.nether.net>
Subject: Re: [cisco-voip] Ported Numbers to SIP call handler transfer is not working correctly.

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 your input.

OR

You would know if it's a DTMF issue if for example, you press a button and CUC processes it twice, as in double digits. This might be a little harder to tell from UX, but it might be easier if you setup a test number to the Opening Greeting and pressing * exists the app, versus taking you to Login.

Transfer

You would know if it's a Transfer issue, if it wasn't a DTMF issue. I.e., You press the button, the recording stops, or even says, "Wait while I transfer your call" and then the failure happens.

Transfer failures could happen for a few different reasons, and there's a few settings on CUBE and within CUCM which can affect how a transfer functions, thus improving success with each knob turned.

On Fri, Apr 24, 2020 at 7:26 AM Hamu Ebiso <hebiso2010@hotmail.com<mailto:hebiso2010@hotmail.com>> wrote:
I was thinking it might be Transfer issue. What makes you ask that question Anthony?

thanks
Hamu

________________________________
From: Anthony Holloway <avholloway+cisco-voip@gmail.com<mailto:avholloway%2Bcisco-voip@gmail.com>>
Sent: Thursday, April 23, 2020 2:54 PM
To: Hamu Ebiso <hebiso2010@hotmail.com<mailto:hebiso2010@hotmail.com>>
Cc: cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net> <cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>>
Subject: Re: [cisco-voip] Ported Numbers to SIP call handler transfer is not working correctly.

Is this a DTMF issue, or a transfer issue?

On Thu, Apr 23, 2020 at 2:51 PM Hamu Ebiso <hebiso2010@hotmail.com<mailto:hebiso2010@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
_______________________________________________
cisco-voip mailing list
cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>
https://puck.nether.net/mailman/listinfo/cisco-voip

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 sensitive bits.

On Fri, Apr 24, 2020 at 9:25 AM Hamu Ebiso <hebiso2010@hotmail.com> wrote:

> 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 <avholloway+cisco-voip@gmail.com>
> *Sent:* Friday, April 24, 2020 8:55 AM
> *To:* Hamu Ebiso <hebiso2010@hotmail.com>
> *Cc:* cisco-voip@puck.nether.net <cisco-voip@puck.nether.net>
> *Subject:* Re: [cisco-voip] Ported Numbers to SIP call handler transfer
> is not working correctly.
>
> 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 your input.
>
> OR
>
> You would know if it's a DTMF issue if for example, you press a button and
> CUC processes it twice, as in double digits. This might be a little harder
> to tell from UX, but it might be easier if you setup a test number to the
> Opening Greeting and pressing * exists the app, versus taking you to Login.
>
> *Transfer*
>
> You would know if it's a Transfer issue, if it wasn't a DTMF issue. I.e.,
> You press the button, the recording stops, or even says, "Wait while I
> transfer your call" and then the failure happens.
>
> Transfer failures could happen for a few different reasons, and there's a
> few settings on CUBE and within CUCM which can affect how a transfer
> functions, thus improving success with each knob turned.
>
> On Fri, Apr 24, 2020 at 7:26 AM Hamu Ebiso <hebiso2010@hotmail.com> wrote:
>
> I was thinking it might be Transfer issue. What makes you ask that
> question Anthony?
>
> thanks
> Hamu
>
> ------------------------------
> *From:* Anthony Holloway <avholloway+cisco-voip@gmail.com>
> *Sent:* Thursday, April 23, 2020 2:54 PM
> *To:* Hamu Ebiso <hebiso2010@hotmail.com>
> *Cc:* cisco-voip@puck.nether.net <cisco-voip@puck.nether.net>
> *Subject:* Re: [cisco-voip] Ported Numbers to SIP call handler transfer
> is not working correctly.
>
> Is this a DTMF issue, or a transfer issue?
>
> On Thu, Apr 23, 2020 at 2:51 PM Hamu Ebiso <hebiso2010@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
> _______________________________________________
> cisco-voip mailing list
> cisco-voip@puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>

Thank you Jason for your questions. how can you setup rfc2833 In CUCM trunks?

thanks

________________________________
From: Jason Aarons <scubajasona@gmail.com>
Sent: Friday, April 24, 2020 8:24 AM
To: Hamu Ebiso <hebiso2010@hotmail.com>
Cc: cisco-voip <cisco-voip@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@hotmail.com<mailto:hebiso2010@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
_______________________________________________
cisco-voip mailing list
cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>
https://puck.nether.net/mailman/listinfo/cisco-voip

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@hotmail.com> wrote:

> Thank you Jason for your questions. how can you setup rfc2833 In CUCM
> trunks?
>
> thanks
>
> ------------------------------
> *From:* Jason Aarons <scubajasona@gmail.com>
> *Sent:* Friday, April 24, 2020 8:24 AM
> *To:* Hamu Ebiso <hebiso2010@hotmail.com>
> *Cc:* cisco-voip <cisco-voip@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@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
> _______________________________________________
> cisco-voip mailing list
> cisco-voip@puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip@puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

RCF 4733, I believe.

Sent from my iPhone

On Apr 24, 2020, at 10:58, Anthony Holloway <avholloway+cisco-voip@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@hotmail.com<mailto:hebiso2010@hotmail.com>> wrote:
Thank you Jason for your questions. how can you setup rfc2833 In CUCM trunks?

thanks

________________________________
From: Jason Aarons <scubajasona@gmail.com<mailto:scubajasona@gmail.com>>
Sent: Friday, April 24, 2020 8:24 AM
To: Hamu Ebiso <hebiso2010@hotmail.com<mailto:hebiso2010@hotmail.com>>
Cc: cisco-voip <cisco-voip@puck.nether.net<mailto:cisco-voip@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@hotmail.com<mailto:hebiso2010@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
_______________________________________________
cisco-voip mailing list
cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>
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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 <ryanhuff@outlook.com> wrote:
>
> RCF 4733, I believe.
>
> Sent from my iPhone
>
>> On Apr 24, 2020, at 10:58, Anthony Holloway <avholloway+cisco-voip@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@hotmail.com <mailto:hebiso2010@hotmail.com>> wrote:
>> Thank you Jason for your questions. how can you setup rfc2833 In CUCM trunks?
>>
>> thanks
>>
>> From: Jason Aarons <scubajasona@gmail.com <mailto:scubajasona@gmail.com>>
>> Sent: Friday, April 24, 2020 8:24 AM
>> To: Hamu Ebiso <hebiso2010@hotmail.com <mailto:hebiso2010@hotmail.com>>
>> Cc: cisco-voip <cisco-voip@puck.nether.net <mailto:cisco-voip@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@hotmail.com <mailto:hebiso2010@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
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip@puck.nether.net <mailto:cisco-voip@puck.nether.net>
>> https://puck.nether.net/mailman/listinfo/cisco-voip <https://nam11.safelinks.protection.outlook.com/?url=https%3A%2F%2Fpuck.nether.net%2Fmailman%2Flistinfo%2Fcisco-voip&data=02%7C01%7C%7Ccdf3f9636d86481fa96008d7e85fef7a%7C84df9e7fe9f640afb435aaaaaaaaaaaa%7C1%7C0%7C637233371061347722&sdata=gQIs%2FfMD6wh1OiGlmZSljaGRaZwpL9ioN8Zltc%2BbBB0%3D&reserved=0>
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip@puck.nether.net <mailto:cisco-voip@puck.nether.net>
>> https://puck.nether.net/mailman/listinfo/cisco-voip <https://nam11.safelinks.protection.outlook.com/?url=https%3A%2F%2Fpuck.nether.net%2Fmailman%2Flistinfo%2Fcisco-voip&data=02%7C01%7C%7Ccdf3f9636d86481fa96008d7e85fef7a%7C84df9e7fe9f640afb435aaaaaaaaaaaa%7C1%7C0%7C637233371061347722&sdata=gQIs%2FfMD6wh1OiGlmZSljaGRaZwpL9ioN8Zltc%2BbBB0%3D&reserved=0>
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip@puck.nether.net
>> https://nam11.safelinks.protection.outlook.com/?url=https%3A%2F%2Fpuck.nether.net%2Fmailman%2Flistinfo%2Fcisco-voip&amp;data=02%7C01%7C%7Ccdf3f9636d86481fa96008d7e85fef7a%7C84df9e7fe9f640afb435aaaaaaaaaaaa%7C1%7C0%7C637233371061367711&amp;sdata=bYdGNM%2Fs795%2FjAJ%2FZFcXDZ41QJTGykNajY6IDAmSJeE%3D&amp;reserved=0 <https://nam11.safelinks.protection.outlook.com/?url=https%3A%2F%2Fpuck.nether.net%2Fmailman%2Flistinfo%2Fcisco-voip&amp;data=02%7C01%7C%7Ccdf3f9636d86481fa96008d7e85fef7a%7C84df9e7fe9f640afb435aaaaaaaaaaaa%7C1%7C0%7C637233371061367711&amp;sdata=bYdGNM%2Fs795%2FjAJ%2FZFcXDZ41QJTGykNajY6IDAmSJeE%3D&amp;reserved=0>
> _______________________________________________
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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@outlook.com> wrote:

> RCF 4733, I believe.
>
> Sent from my iPhone
>
> On Apr 24, 2020, at 10:58, Anthony Holloway <
> avholloway+cisco-voip@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@hotmail.com> wrote:
>
>> Thank you Jason for your questions. how can you setup rfc2833 In CUCM
>> trunks?
>>
>> thanks
>>
>> ------------------------------
>> *From:* Jason Aarons <scubajasona@gmail.com>
>> *Sent:* Friday, April 24, 2020 8:24 AM
>> *To:* Hamu Ebiso <hebiso2010@hotmail.com>
>> *Cc:* cisco-voip <cisco-voip@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@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
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip@puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>> <https://nam11.safelinks.protection.outlook.com/?url=https%3A%2F%2Fpuck.nether.net%2Fmailman%2Flistinfo%2Fcisco-voip&data=02%7C01%7C%7Ccdf3f9636d86481fa96008d7e85fef7a%7C84df9e7fe9f640afb435aaaaaaaaaaaa%7C1%7C0%7C637233371061347722&sdata=gQIs%2FfMD6wh1OiGlmZSljaGRaZwpL9ioN8Zltc%2BbBB0%3D&reserved=0>
>>
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip@puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>> <https://nam11.safelinks.protection.outlook.com/?url=https%3A%2F%2Fpuck.nether.net%2Fmailman%2Flistinfo%2Fcisco-voip&data=02%7C01%7C%7Ccdf3f9636d86481fa96008d7e85fef7a%7C84df9e7fe9f640afb435aaaaaaaaaaaa%7C1%7C0%7C637233371061347722&sdata=gQIs%2FfMD6wh1OiGlmZSljaGRaZwpL9ioN8Zltc%2BbBB0%3D&reserved=0>
>>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip@puck.nether.net
>
> https://nam11.safelinks.protection.outlook.com/?url=https%3A%2F%2Fpuck.nether.net%2Fmailman%2Flistinfo%2Fcisco-voip&amp;data=02%7C01%7C%7Ccdf3f9636d86481fa96008d7e85fef7a%7C84df9e7fe9f640afb435aaaaaaaaaaaa%7C1%7C0%7C637233371061367711&amp;sdata=bYdGNM%2Fs795%2FjAJ%2FZFcXDZ41QJTGykNajY6IDAmSJeE%3D&amp;reserved=0
>
>

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@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@outlook.com<mailto:ryanhuff@outlook.com>> wrote:
RCF 4733, I believe.

Sent from my iPhone

On Apr 24, 2020, at 10:58, Anthony Holloway <avholloway+cisco-voip@gmail.com<mailto:avholloway%2Bcisco-voip@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@hotmail.com<mailto:hebiso2010@hotmail.com>> wrote:
Thank you Jason for your questions. how can you setup rfc2833 In CUCM trunks?

thanks

________________________________
From: Jason Aarons <scubajasona@gmail.com<mailto:scubajasona@gmail.com>>
Sent: Friday, April 24, 2020 8:24 AM
To: Hamu Ebiso <hebiso2010@hotmail.com<mailto:hebiso2010@hotmail.com>>
Cc: cisco-voip <cisco-voip@puck.nether.net<mailto:cisco-voip@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@hotmail.com<mailto:hebiso2010@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
_______________________________________________
cisco-voip mailing list
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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 back to Nortel phone system and works fine.
Call to location 2 Main call handler >> option 2 to classified advertising >> option 1 go back to Nortel phone system. Plays wait while I transfer and then fail.

Location on SIP trunk and CUCM
Location 1 is on PRI and Nortel phone system

some how Location 1 phone system is connected to Unity connection.

Dial-peer is configured with DTMF correctly

Everything in unity is configured correctly but the transfer doesn't work.

thanks
Hamu

________________________________
From: Anthony Holloway <avholloway+cisco-voip@gmail.com>
Sent: Friday, April 24, 2020 9:32 AM
To: Hamu Ebiso <hebiso2010@hotmail.com>
Cc: cisco-voip@puck.nether.net <cisco-voip@puck.nether.net>
Subject: Re: [cisco-voip] Ported Numbers to SIP call handler transfer is not working correctly.

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 sensitive bits.

On Fri, Apr 24, 2020 at 9:25 AM Hamu Ebiso <hebiso2010@hotmail.com<mailto:hebiso2010@hotmail.com>> wrote:
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 <avholloway+cisco-voip@gmail.com<mailto:avholloway%2Bcisco-voip@gmail.com>>
Sent: Friday, April 24, 2020 8:55 AM
To: Hamu Ebiso <hebiso2010@hotmail.com<mailto:hebiso2010@hotmail.com>>
Cc: cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net> <cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>>
Subject: Re: [cisco-voip] Ported Numbers to SIP call handler transfer is not working correctly.

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 your input.

OR

You would know if it's a DTMF issue if for example, you press a button and CUC processes it twice, as in double digits. This might be a little harder to tell from UX, but it might be easier if you setup a test number to the Opening Greeting and pressing * exists the app, versus taking you to Login.

Transfer

You would know if it's a Transfer issue, if it wasn't a DTMF issue. I.e., You press the button, the recording stops, or even says, "Wait while I transfer your call" and then the failure happens.

Transfer failures could happen for a few different reasons, and there's a few settings on CUBE and within CUCM which can affect how a transfer functions, thus improving success with each knob turned.

On Fri, Apr 24, 2020 at 7:26 AM Hamu Ebiso <hebiso2010@hotmail.com<mailto:hebiso2010@hotmail.com>> wrote:
I was thinking it might be Transfer issue. What makes you ask that question Anthony?

thanks
Hamu

________________________________
From: Anthony Holloway <avholloway+cisco-voip@gmail.com<mailto:avholloway%2Bcisco-voip@gmail.com>>
Sent: Thursday, April 23, 2020 2:54 PM
To: Hamu Ebiso <hebiso2010@hotmail.com<mailto:hebiso2010@hotmail.com>>
Cc: cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net> <cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>>
Subject: Re: [cisco-voip] Ported Numbers to SIP call handler transfer is not working correctly.

Is this a DTMF issue, or a transfer issue?

On Thu, Apr 23, 2020 at 2:51 PM Hamu Ebiso <hebiso2010@hotmail.com<mailto:hebiso2010@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
_______________________________________________
cisco-voip mailing list
cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>
https://puck.nether.net/mailman/listinfo/cisco-voip

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 <hebiso2010@hotmail.com> wrote:

> 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 back to Nortel phone system and works fine.
> Call to location 2 Main call handler >> option 2 to classified advertising
> >> option 1 go back to Nortel phone system. Plays wait while I transfer and
> then fail.
>
> Location on SIP trunk and CUCM
> Location 1 is on PRI and Nortel phone system
>
> some how Location 1 phone system is connected to Unity connection.
>
> Dial-peer is configured with DTMF correctly
>
> Everything in unity is configured correctly but the transfer doesn't work.
>
> thanks
> Hamu
>
> ------------------------------
> *From:* Anthony Holloway <avholloway+cisco-voip@gmail.com>
> *Sent:* Friday, April 24, 2020 9:32 AM
> *To:* Hamu Ebiso <hebiso2010@hotmail.com>
> *Cc:* cisco-voip@puck.nether.net <cisco-voip@puck.nether.net>
> *Subject:* Re: [cisco-voip] Ported Numbers to SIP call handler transfer
> is not working correctly.
>
> 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 sensitive bits.
>
> On Fri, Apr 24, 2020 at 9:25 AM Hamu Ebiso <hebiso2010@hotmail.com> wrote:
>
> 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 <avholloway+cisco-voip@gmail.com>
> *Sent:* Friday, April 24, 2020 8:55 AM
> *To:* Hamu Ebiso <hebiso2010@hotmail.com>
> *Cc:* cisco-voip@puck.nether.net <cisco-voip@puck.nether.net>
> *Subject:* Re: [cisco-voip] Ported Numbers to SIP call handler transfer
> is not working correctly.
>
> 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 your input.
>
> OR
>
> You would know if it's a DTMF issue if for example, you press a button and
> CUC processes it twice, as in double digits. This might be a little harder
> to tell from UX, but it might be easier if you setup a test number to the
> Opening Greeting and pressing * exists the app, versus taking you to Login.
>
> *Transfer*
>
> You would know if it's a Transfer issue, if it wasn't a DTMF issue. I.e.,
> You press the button, the recording stops, or even says, "Wait while I
> transfer your call" and then the failure happens.
>
> Transfer failures could happen for a few different reasons, and there's a
> few settings on CUBE and within CUCM which can affect how a transfer
> functions, thus improving success with each knob turned.
>
> On Fri, Apr 24, 2020 at 7:26 AM Hamu Ebiso <hebiso2010@hotmail.com> wrote:
>
> I was thinking it might be Transfer issue. What makes you ask that
> question Anthony?
>
> thanks
> Hamu
>
> ------------------------------
> *From:* Anthony Holloway <avholloway+cisco-voip@gmail.com>
> *Sent:* Thursday, April 23, 2020 2:54 PM
> *To:* Hamu Ebiso <hebiso2010@hotmail.com>
> *Cc:* cisco-voip@puck.nether.net <cisco-voip@puck.nether.net>
> *Subject:* Re: [cisco-voip] Ported Numbers to SIP call handler transfer
> is not working correctly.
>
> Is this a DTMF issue, or a transfer issue?
>
> On Thu, Apr 23, 2020 at 2:51 PM Hamu Ebiso <hebiso2010@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
> _______________________________________________
> cisco-voip mailing list
> cisco-voip@puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>