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

[Ryan Huff]

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 at 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 at hotmail.com<mailto:hebiso2010 at 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 at gmail.com<mailto:avholloway%2Bcisco-voip at gmail.com>>
Sent: Thursday, April 23, 2020 2:54 PM
To: Hamu Ebiso <hebiso2010 at hotmail.com<mailto:hebiso2010 at hotmail.com>>
Cc: cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net> <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.

Is this a DTMF issue, or a transfer issue?

On Thu, Apr 23, 2020 at 2:51 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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