What specific debugs should be run ? <br><br>Sent from my HTC bolt of lightning...<br><br><div id="htc_header" style="">----- Reply message -----<br>From: "ccieid1ot" <ccieid1ot@gmail.com><br>To: "Jay Stants" <jaystants@rogers.com><br>Cc: "Peter Slow" <peter.slow@gmail.com>, "cisco-voip@puck.nether.net" <cisco-voip@puck.nether.net><br>Subject: [cisco-voip] [Bulk]redirect Voicemail to CUE from external SIP number<br>Date: Thu, Jul 28, 2011 10:53 am<br><br></div><br>Run some debugs to see what's the redirecting number when it's forwarding to voice mail. Might need a redirecting translation<br><br><div class="gmail_quote">On Tue, Jul 26, 2011 at 11:30 AM, Jay Stants <span dir="ltr"><<a href="mailto:jaystants@rogers.com">jaystants@rogers.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;"><div><div style="color:#000;background-color:#fff;font-family:times new roman, new york, times, serif;font-size:12pt">
<div><span>Peter,</span></div>
<div><span></span> </div>
<div><span>I have the following within my config</span></div>
<div><span></span> </div><span>
<div><span style="color:#1f497d">no voice register global</span></div>
<div><span style="color:#1f497d"> </span></div>
<div><span style="color:#1f497d">voice service voip</span></div>
<div><span style="color:#1f497d"> ip address trusted list</span></div>
<div><span style="color:#1f497d"> ipv4 0.0.0.0 0.0.0.0</span></div>
<div><span style="color:#1f497d"> allow-connections h323 to h323</span></div>
<div><span style="color:#1f497d"> allow-connections h323 to sip</span></div>
<div><span style="color:#1f497d"> allow-connections sip to h323</span></div>
<div><span style="color:#1f497d"> allow-connections sip to sip</span></div>
<div><span style="color:#1f497d"> supplementary-service h450.12</span></div>
<div><span style="color:#1f497d"> sip</span></div>
<div><span style="color:#1f497d"> registrar server expires max 600 min 60</span></div></span><div class="im">
<div></div>
<div> </div>
<div><span style="color:rgb(0,96,191)">Regards,</span><br style="color:rgb(0,96,191)"><span style="color:rgb(0,96,191)">Jay Stants</span><br><a style="color:rgb(0,64,127)" href="mailto:jaystants@rogers.com" rel="nofollow" target="_blank">jaystants@rogers.com</a><br>
<br></div>
</div><blockquote style="border-left:rgb(16,16,255) 2px solid;padding-left:5px;margin-left:5px">
<div style="font-family:times new roman, new york, times, serif;font-size:12pt">
<div style="font-family:times new roman, new york, times, serif;font-size:12pt"><font face="Arial" size="2">
<div style="border-bottom:#ccc 1px solid;border-left:#ccc 1px solid;padding-bottom:0px;line-height:0;margin:5px 0px;padding-left:0px;padding-right:0px;min-height:0px;font-size:0px;border-top:#ccc 1px solid;border-right:#ccc 1px solid;padding-top:0px" readonly>
</div><b><span style="font-weight:bold">From:</span></b> Peter Slow <<a href="mailto:peter.slow@gmail.com" target="_blank">peter.slow@gmail.com</a>><br><b><span style="font-weight:bold">To:</span></b> "<a href="mailto:jaystants@rogers.com" target="_blank">jaystants@rogers.com</a>" <<a href="mailto:jaystants@rogers.com" target="_blank">jaystants@rogers.com</a>><br>
<b><span style="font-weight:bold">Cc:</span></b> <a href="mailto:cisco-voip@puck.nether.net" target="_blank">cisco-voip@puck.nether.net</a><br><b><span style="font-weight:bold">Sent:</span></b> Monday, July 25, 2011 5:30:45 AM<br>
<b><span style="font-weight:bold">Subject:</span></b> Re: [cisco-voip] [Bulk]redirect Voicemail to CUE from external SIP number<br></font><div><div></div><div class="h5"><br>your sample config does not have<br><br>voice service voi<br>
allow s t s<br><br>under it. you need the sip to sip command to
allow it to function as a<br>CUBE in your case, since we're connecting two SIP dial-peers together.<var></var><br><br>Is that done?<br><br>-Peter<br><br>On Fri, Jul 22, 2011 at 4:30 PM, <a href="mailto:jaystants@rogers..com" target="_blank">jaystants@rogers.com</a><br>
<<a href="mailto:jaystants@rogers.com" target="_blank">jaystants@rogers.com</a>> wrote:<br>> Any one able to assist with this issue?<br>><br>> Regards,<br>><br>> Jay Stants<br>><br>> Sent from my Thunderbolt 4G LTE<br>
><br>><br>> ----- Reply message -----<br>> From: "Jay Stants" <<a href="mailto:jaystants@rogers.com" target="_blank">jaystants@rogers.com</a>><br>> To: <<a href="mailto:cisco-voip@puck.nether.net" target="_blank">cisco-voip@puck.nether.net</a>><br>
> Subject: [Bulk] [cisco-voip] redirect Voicemail to CUE from
external SIP<br>> number<br>> Date: Thu, Jul 21, 2011 8:41 am<br>><br>><br>> I've managed to get VM working internally on test phones but when i dial in<br>> from my DID, call gets redirected to a specific phone (which is the<br>
> behaviour i want) but after timeout duration call is not transfered to<br>> voicemail, instead just rings busy . Can someone point out anything i may be<br>> missing (translation or dial-peer or possibly something else)<br>
><br>> Help is much appreciated<br>><br>> Config Details<br>> ---------------<br>> snet-wan2#sh run<br>> Building configuration.....<br>> Current configuration : 6793 bytes<br>> !<br>> ! Last configuration change at 08:17:38 EST Thu Jul 21 2011 by netmgmt<br>
> ! NVRAM config last updated at 08:17:38 EST Thu Jul 21 2011 by netmgmt<br>> !<br>> version 15.1<br>> voice translation-rule 1<br>> rule 1 /^9/ //<br>>
!<br>> voice translation-rule 3<br>> rule 1 /4.../ /<a href="tel:5856786019" value="+15856786019" target="_blank">5856786019</a>/<br>> !<br>> !<br>> voice translation-profile <a href="http://voip.ms" target="_blank">voip.ms</a><br>
> translate calling 3<br>> translate called 1<br>> !<br>> !<br>> dial-peer voice 1 voip<br>> description **SIP Trunk to <a href="http://newyork.voip.ms" target="_blank">newyork.voip.ms</a>**<br>> translation-profile outgoing <a href="http://voip.ms" target="_blank">voip.ms</a><br>
> destination-pattern 9[2-9].[2-9].......<br>> session protocol sipv2<br>> session target dns:<a href="http://newyork.voip.ms" target="_blank">newyork.voip.ms</a><br>> dtmf-relay rtp-nte sip-notify<br>> codec g711ulaw<br>
> no vad<br>> !<br>> dial-peer voice 2 voip<br>> description **Incoming SIP Trunk - Voip.ms**<br>> translation-profile incoming <a href="http://voip.ms" target="_blank">voip.ms</a><br>> session protocol sipv2<br>
> session target ipv4:10.50.1.2<br>> incoming called-number <a href="tel:5856786019" value="+15856786019" target="_blank">5856786019</a><br>> dtmf-relay sip-notify<br>> codec g711ulaw<br>> no
vad<br>> !<br>> dial-peer voice 20 pots<br>> destination-pattern 5.T<br>> direct-inward-dial<br>> no sip-register<br>> !<br>> dial-peer voice 3 voip<br>> description ** Voicemail **<br>> destination-pattern 4000<br>
> session protocol sipv2<br>> session target ipv4:1.1.1.2<br>> dtmf-relay sip-notify<br>> codec g711ulaw<br>> no vad<br>> !<br>> !<br>> sip-ua<br>> credentials username {removed} password 7 {removed} realm <a href="http://newyork.voip.ms" target="_blank">newyork.voip.ms</a><br>
> authentication username {removed} password 7 {removed}<br>> no remote-party-id<br>> retry invite 2<br>> retry register 10<br>> timers connect 100<br>> mwi-server ipv4:1.1.1.2 expires 3600 port 5060 transport tcp unsolicited<br>
> registrar dns:<a href="http://newyork.voip.ms" target="_blank">newyork.voip.ms</a> expires 180<br>> sip-server dns:<a href="http://newyork.voip.ms" target="_blank">newyork.voip.ms</a><br>>
host-registrar<br>> !<br>> !<br>> !<br>> telephony-service<br>> authentication credential {username password}<br>> max-ephones 15<br>> max-dn 15<br>> ip source-address 10.50.1.2 port 2000<br>
> system message Cisco CME 8.1<br>> url services <a href="http://1.1.1.2/voiceview/common/login.do" target="_blank">http://1.1.1.2/voiceview/common/login.do</a><br>> url authentication <a href="http://1.1.1.1/CCMCIP/authenticate.asp" target="_blank">http://1.1.1.1/CCMCIP/authenticate.asp</a><br>
> cnf-file location flash:<br>> cnf-file perphone<br>> time-zone 12<br>> dialplan-pattern <a href="tel:1%205856786019" value="+15856786019" target="_blank">1 5856786019</a> extension-length 4<br>> voicemail 4000<br>
> max-conferences 8 gain -6<br>> dn-webedit<br>> time-webedit<br>> transfer-system full-consult<br>> secondary-dialtone 9<br>> create cnf-files
version-stamp Jan 01 2002 00:00:00<br>> !<br>> !<br>> ephone-dn-template 1<br>> call-forward busy 4000<br>> call-forward noan 4000 timeout 18<br>> !<br>> !<br>> ephone-dn 1 dual-line<br>> number 4005 secondary <a href="tel:5856786019" value="+15856786019" target="_blank">5856786019</a> no-reg<br>
> label 4005<br>> name Jay Stants<br>> ephone-dn-template 1<br>> !<br>> !<br>> ephone-dn 13<br>> !<br>> !<br>> ephone-dn 14<br>> number 8000...........<br>> mwi on<br>> !<br>> !<br>
> ephone-dn 15<br>> number 8001...........<br>> mwi off<br>> !<br>> !<br>> ephone 1<br>> device-security-mode none<br>> mac-address 001E.7AC5.896A<br>> username "user" password {removed}<br>
> type 7961GE<br>> button 1:1<br>><br>><br>> Regards,<br>> Jay Stants<br>> <a href="mailto:jaystants@rogers.com" target="_blank">jaystants@rogers.com</a><br>><br>> _______________________________________________<br>
> cisco-voip mailing list<br>> <a href="mailto:cisco-voip@puck.nether.net" target="_blank">cisco-voip@puck.nether.net</a><br>> <a href="https://puck.nether.net/mailman/listinfo/cisco-voip" target="_blank">https://puck.nether.net/mailman/listinfo/cisco-voip</a><br>
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<br></blockquote></div><br><br clear="all"><br>-- <br>duy<br>CCIE #27737 Voice<br><br>