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Hello Anthony and all,</div>
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I have found more information about this.</div>
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<span>3 call handlers are involved with this.<br>
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<div>Location 1 Main call handler<br>
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<div>Clasified department call handler<br>
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<div>Location Main call handler <br>
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<div>call to location 1 Main call handler >> option 3 to Classified advertising >> option 1 to going back to Nortel phone system and works fine.<br>
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<div>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.<br>
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<div>Location on SIP trunk and CUCM<br>
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<div>Location 1 is on PRI and Nortel phone system<br>
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<div>some how Location 1 phone system is connected to Unity connection.<br>
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<div>Dial-peer is configured with DTMF correctly<br>
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<span>Everything in unity is configured correctly but the transfer doesn't work.</span></div>
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<span>thanks</span></div>
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<span>Hamu</span><br>
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<div id="divRplyFwdMsg" dir="ltr"><font style="font-size:11pt" face="Calibri, sans-serif" color="#000000"><b>From:</b> Anthony Holloway <avholloway+cisco-voip@gmail.com><br>
<b>Sent:</b> Friday, April 24, 2020 9:32 AM<br>
<b>To:</b> Hamu Ebiso <hebiso2010@hotmail.com><br>
<b>Cc:</b> cisco-voip@puck.nether.net <cisco-voip@puck.nether.net><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="ltr">Hamu,
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<div>Are you saying then, that the issue is not DTMF? Does the system take your input without error?</div>
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<div>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.</div>
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<div dir="ltr" class="x_gmail_attr">On Fri, Apr 24, 2020 at 9:25 AM Hamu Ebiso <<a href="mailto:hebiso2010@hotmail.com">hebiso2010@hotmail.com</a>> wrote:<br>
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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?</div>
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thanks<br>
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<div id="x_gmail-m_1090757636184374103divRplyFwdMsg" dir="ltr"><font style="font-size:11pt" face="Calibri, sans-serif" color="#000000"><b>From:</b> Anthony Holloway <<a href="mailto:avholloway%2Bcisco-voip@gmail.com" target="_blank">avholloway+cisco-voip@gmail.com</a>><br>
<b>Sent:</b> Friday, April 24, 2020 8:55 AM<br>
<b>To:</b> Hamu Ebiso <<a href="mailto:hebiso2010@hotmail.com" target="_blank">hebiso2010@hotmail.com</a>><br>
<b>Cc:</b> <a href="mailto:cisco-voip@puck.nether.net" target="_blank">cisco-voip@puck.nether.net</a> <<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="ltr">The reason I ask is that the troubleshooting is a little different for each issue.
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<div><b><u>DTMF</u></b></div>
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<div>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.</div>
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<div>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.</div>
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<div><b><u>Transfer</u></b></div>
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<div>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.</div>
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<div>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.</div>
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<div dir="ltr">On Fri, Apr 24, 2020 at 7:26 AM Hamu Ebiso <<a href="mailto:hebiso2010@hotmail.com" target="_blank">hebiso2010@hotmail.com</a>> wrote:<br>
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I was thinking it might be Transfer issue. What makes you ask that question Anthony?</div>
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thanks</div>
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Hamu<br>
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<font style="font-size:11pt" face="Calibri, sans-serif" color="#000000"><b>From:</b> Anthony Holloway <<a href="mailto:avholloway%2Bcisco-voip@gmail.com" target="_blank">avholloway+cisco-voip@gmail.com</a>><br>
<b>Sent:</b> Thursday, April 23, 2020 2:54 PM<br>
<b>To:</b> Hamu Ebiso <<a href="mailto:hebiso2010@hotmail.com" target="_blank">hebiso2010@hotmail.com</a>><br>
<b>Cc:</b> <a href="mailto:cisco-voip@puck.nether.net" target="_blank">cisco-voip@puck.nether.net</a> <<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="ltr">Is this a DTMF issue, or a transfer issue?</div>
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<div dir="ltr">On Thu, Apr 23, 2020 at 2:51 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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