<div dir="ltr">Check your voicemail ports calling search space. Is the partition where the 4 IP phones reside listed there?</div><div class="gmail_extra"><br clear="all"><div><div dir="ltr"><font face="verdana, sans-serif"><b>Angel Roberto Castaneda</b></font></div>
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<br><br><div class="gmail_quote">On Fri, Mar 7, 2014 at 10:52 AM, Jeffrey Girard <span dir="ltr"><<a href="mailto:jeffrey.girard@girardinc.com" target="_blank">jeffrey.girard@girardinc.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div lang="EN-US" link="#0563C1" vlink="#954F72"><div><p class="MsoNormal"><span style="font-size:12.0pt;font-family:"Arial","sans-serif"">I have posted this same request for assistance in the Cisco IP Telephony Support Forum. So far, not hits<u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:12.0pt;font-family:"Arial","sans-serif""><u></u> <u></u></span></p><p class="MsoNormal" style="background:white"><span lang="EN" style="font-size:12.0pt;font-family:"Arial","sans-serif";color:#333333">CUCM 9 and CUC 9. 4 IP Phones registered and all have mailboxes. I have a System Call Handler configured to answer calls coming from a CTI Route Point.<u></u><u></u></span></p>
<p class="MsoNormal" style="background:white"><span lang="EN" style="font-size:12.0pt;font-family:"Arial","sans-serif";color:#333333"> <u></u><u></u></span></p><p class="MsoNormal" style="background:white">
<span lang="EN" style="font-size:12.0pt;font-family:"Arial","sans-serif";color:#333333">I have been playing with a scenario trying to learn all the capabilities of the various call handlers.<u></u><u></u></span></p>
<p class="MsoNormal" style="background:white"><span lang="EN" style="font-size:12.0pt;font-family:"Arial","sans-serif";color:#333333"> <u></u><u></u></span></p><p class="MsoNormal" style="background:white">
<span lang="EN" style="font-size:12.0pt;font-family:"Arial","sans-serif";color:#333333">Im currently trying to allow the caller to transfer to a phone number that is registered in CUCM - but does NOT have a mailbox.<u></u><u></u></span></p>
<p class="MsoNormal" style="background:white"><span lang="EN" style="font-size:12.0pt;font-family:"Arial","sans-serif";color:#333333"> <u></u><u></u></span></p><p class="MsoNormal" style="background:white">
<span lang="EN" style="font-size:12.0pt;font-family:"Arial","sans-serif";color:#333333">My CUC Configuration<u></u><u></u></span></p><p class="MsoNormal" style="background:white"><span lang="EN" style="font-size:12.0pt;font-family:"Arial","sans-serif";color:#333333"> Call Handler Basics<u></u><u></u></span></p>
<p class="MsoNormal" style="background:white"><span lang="EN" style="font-size:12.0pt;font-family:"Arial","sans-serif";color:#333333"> Callers Hear -> My Personal Recording<u></u><u></u></span></p>
<p class="MsoNormal" style="background:white"><span lang="EN" style="font-size:12.0pt;font-family:"Arial","sans-serif";color:#333333"> During Greeting -> Allow Transfers to Numbers Not Associate with Users or Call Handlers<u></u><u></u></span></p>
<p class="MsoNormal" style="background:white"><span lang="EN" style="font-size:12.0pt;font-family:"Arial","sans-serif";color:#333333"> After Greeting -> Conversation -> Caller System Transfer<u></u><u></u></span></p>
<p class="MsoNormal" style="background:white"><span lang="EN" style="font-size:12.0pt;font-family:"Arial","sans-serif";color:#333333"> <u></u><u></u></span></p><p class="MsoNormal" style="background:white">
<span lang="EN" style="font-size:12.0pt;font-family:"Arial","sans-serif";color:#333333"> System Settings -> Restriction Tables<u></u><u></u></span></p><p class="MsoNormal" style="background:white">
<span lang="EN" style="font-size:12.0pt;font-family:"Arial","sans-serif";color:#333333"> Default Transfer<u></u><u></u></span></p><p class="MsoNormal" style="background:white"><span lang="EN" style="font-size:12.0pt;font-family:"Arial","sans-serif";color:#333333"> I added a new entry and specifically entered the DN that I was trying to transfer to (123)<u></u><u></u></span></p>
<p class="MsoNormal" style="background:white"><span lang="EN" style="font-size:12.0pt;font-family:"Arial","sans-serif";color:#333333"> I unchecked the "blocked" for all entries<u></u><u></u></span></p>
<p class="MsoNormal" style="background:white"><span lang="EN" style="font-size:12.0pt;font-family:"Arial","sans-serif";color:#333333"> <u></u><u></u></span></p><p class="MsoNormal" style="background:white">
<span lang="EN" style="font-size:12.0pt;font-family:"Arial","sans-serif";color:#333333"> Call Management -> Call Routing -> Forwarded Routing Rules<u></u><u></u></span></p><p class="MsoNormal" style="background:white">
<span lang="EN" style="font-size:12.0pt;font-family:"Arial","sans-serif";color:#333333"> Attempt Forward Routing Rule -> Send Call to -> Conversation -> Attempt Forward<u></u><u></u></span></p>
<p class="MsoNormal" style="background:white"><span lang="EN" style="font-size:12.0pt;font-family:"Arial","sans-serif";color:#333333"> <u></u><u></u></span></p><p class="MsoNormal" style="background:white">
<span lang="EN" style="font-size:12.0pt;font-family:"Arial","sans-serif";color:#333333">From an IP Phone, I dial the CTI RP and get forwarded to the main / opening call handler. I traverse a couple of menu tree levels to get to my call handler where I am attempting to transfer back to CUCM. I hear my recorded personal greeting instructing me to enter my number. Whether I wait for the recording to end or whether I enter the digits during the greeting, the results are the same. The result is that nothing happens. No recorded message telling me that I cant be transferred to that number - nothing at all. The call just terminates.<u></u><u></u></span></p>
<p class="MsoNormal" style="background:white"><span lang="EN" style="font-size:12.0pt;font-family:"Arial","sans-serif";color:#333333"> <u></u><u></u></span></p><p class="MsoNormal" style="background:white">
<span lang="EN" style="font-size:12.0pt;font-family:"Arial","sans-serif";color:#333333">I started up Connection Remote Port Status monitor and I can watch the sequence of the call. When I enter the digits of the target phone (123), I see in the Remote Port Status Monitor that it sees and receives the DTMF digits, but then nothing happens. <u></u><u></u></span></p>
<p class="MsoNormal" style="background:white"><span lang="EN" style="font-size:12.0pt;font-family:"Arial","sans-serif";color:#333333"> <u></u><u></u></span></p><p class="MsoNormal" style="background:white">
<span lang="EN" style="font-size:12.0pt;font-family:"Arial","sans-serif";color:#333333">I next tested by adding a test user in CUC with the same DN of 123. As soon as I did that, I repeated the attempted call and when I dialed "123", I get the recorded message "Wait while I transfer that call" and then the call completes successfully.<u></u><u></u></span></p>
<p class="MsoNormal" style="background:white"><span lang="EN" style="font-size:12.0pt;font-family:"Arial","sans-serif";color:#333333"> <u></u><u></u></span></p><p class="MsoNormal" style="background:white">
<span lang="EN" style="font-size:12.0pt;font-family:"Arial","sans-serif";color:#333333">This is my home lab so I can make any changes that you suggest with no user implications.<u></u><u></u></span></p>
<p class="MsoNormal" style="background:white"><span lang="EN" style="font-size:12.0pt;font-family:"Arial","sans-serif";color:#333333"> <u></u><u></u></span></p><p class="MsoNormal" style="background:white">
<span lang="EN" style="font-size:12.0pt;font-family:"Arial","sans-serif";color:#333333">Jeff<u></u><u></u></span></p><p class="MsoNormal"><span style="font-size:12.0pt;font-family:"Arial","sans-serif""><u></u> <u></u></span></p>
<p class="MsoNormal"><span style="font-size:12.0pt;font-family:"Arial","sans-serif""><u></u> <u></u></span></p><p class="MsoNormal"><span style="font-size:12.0pt;font-family:"Arial","sans-serif";color:#1f497d">------------------------------------------------------------------------------------</span><span style="font-size:12.0pt;font-family:"Times New Roman","serif";color:#1f497d"><u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:12.0pt;font-family:"Arial","sans-serif";color:#1f497d">Dr. Jeffrey T. Girard (Jeff), PhD</span><span style="font-size:12.0pt;color:#1f497d"><u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:12.0pt;font-family:"Arial","sans-serif";color:#1f497d">Colonel, United States Army (Retired)<u></u><u></u></span></p><p class="MsoNormal"><span style="font-size:12.0pt;font-family:"Arial","sans-serif";color:#1f497d">Senior Network Engineer / VoIP Engineer - WireMeHappy.com</span><span style="font-size:12.0pt;color:#1f497d"><u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:12.0pt;font-family:"Arial","sans-serif";color:#1f497d">reply to: <a href="mailto:jeffrey.girard@wiremehappy.com" target="_blank"><span style="color:blue">jeffrey.girard@wiremehappy.com</span></a></span><span style="font-size:12.0pt;color:#1f497d"><u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:12.0pt;font-family:"Arial","sans-serif";color:#1f497d"><a href="tel:%28607%29835-0406" value="+16078350406" target="_blank">(607)835-0406</a> (home office)</span><span style="font-size:12.0pt;color:#1f497d"><u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:12.0pt;font-family:"Arial","sans-serif";color:#1f497d"><a href="tel:%28845%29764-1661" value="+18457641661" target="_blank">(845)764-1661</a> (mobile)<u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:12.0pt;font-family:"Arial","sans-serif";color:#1f497d"><a href="tel:%28607%29835-0458" value="+16078350458" target="_blank">(607)835-0458</a> (fax)</span><span style="font-size:12.0pt;color:#1f497d"><u></u><u></u></span></p>
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