<html><head><style type='text/css'>p { margin: 0; }</style></head><body><div style='font-family: Verdana; font-size: 10pt; color: #000000'>Thanks for the update. <br><br>---<br>Lelio Fulgenzi, B.A.<br>Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1<br>(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)<br>^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^<br>"Bad grammar makes me [sic]" - Tshirt<br><br><br>----- Original Message -----<br>From: "Pete G. Ruttman" <PRuttman@foley.com><br>To: cisco-voip@puck.nether.net<br>Sent: Thursday, April 2, 2009 5:45:16 PM GMT -05:00 US/Canada Eastern<br>Subject: Re: [cisco-voip] SRST/Unity voicemail & vm-integration<br><br>
<title>SRST/Unity voicemail & vm-integration</title>
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<div dir="ltr" align="left"><span class="018473521-02042009"><font face="Arial"><font color="#0000ff"><font size="2">I am still working on <span class="330143621-02042009">the RDNIS method but I have solved the vm-integration
problem. For the scenario where you are calling off-premise into a phone
on SRST, you need to put the follow command into the
router:</span></font></font></font></span></div>
<div dir="ltr" align="left"><span class="018473521-02042009"><font face="Arial"><font color="#0000ff"><font size="2"><span class="330143621-02042009"></span></font></font></font></span> </div>
<div dir="ltr" align="left"><span class="018473521-02042009"><font face="Arial"><font color="#0000ff"><font size="2"><span class="330143621-02042009">voice-card
x</span></font></font></font></span></div>
<div dir="ltr" align="left"><span class="018473521-02042009"><font face="Arial"><font color="#0000ff"><font size="2"><span class="330143621-02042009">no
local-bypass</span></font></font></font></span></div>
<div dir="ltr" align="left"><span class="018473521-02042009"><font face="Arial"><font color="#0000ff"><font size="2"><span class="330143621-02042009"></span></font></font></font></span> </div>
<div><span class="330143621-02042009"></span><font face="Arial"><font color="#0000ff"><font size="2">Two POTS lines are used in this scenario. One for the inbound call and one for the outbound call to the datacenter with Unity.</font></font></font></div>
<div><span class="330143621-02042009"></span><font face="Arial"><font color="#0000ff"><font size="2">B<span class="330143621-02042009">y default, the router
will remove the DSPs from this scenario. vm-integration requires DSPs to
insert the DTMF tones and that is why it wasn't working. The command above
forces them back in play and vm-integration
works.</span></font></font></font></div>
<div><font face="Arial"><font color="#0000ff"><font size="2"><span class="330143621-02042009"></span></font></font></font> </div>
<div><font face="Arial"><font color="#0000ff"><font size="2"><span class="330143621-02042009">TAC says that it is still best to use RDNIS instead of
vm-integration, if possible, so I am still exploring that, but it is cool I have
a solution.</span></font></font></font></div>
<div><font face="Arial"><font color="#0000ff"><font size="2"><span class="330143621-02042009"><br>Pete</span></font></font></font></div>
<div><br></div>
<div class="OutlookMessageHeader" dir="ltr" align="left" lang="en-us">
<hr>
<font size="2" face="Tahoma"><b>From:</b> Lelio Fulgenzi [mailto:lelio@uoguelph.ca]
<br><b>Sent:</b> Friday, March 20, 2009 1:45 PM<br><b>To:</b> Ruttman, Pete
G.<br><b>Cc:</b> cisco-voip@puck.nether.net<br><b>Subject:</b> Re: [cisco-voip]
SRST/Unity voicemail & vm-integration<br></font><br></div>
<div></div>
<div style="font-size: 10pt; color: rgb(0, 0, 0); font-family: Verdana;">I've heard
that RDNIS is the hardest thing to ensure is working in the PSTN cloud. Your
service provider has to tell you it's turned on.<br><br>---<br>Lelio Fulgenzi,
B.A.<br>Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G
2W1<br>(519) 824-4120 x56354 (519) 767-1060 FAX
(JNHN)<br>^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^<br>"Bad
grammar makes me [sic]" - Tshirt<br><br><br>----- Original Message
-----<br>From: "Pete G. Ruttman" <PRuttman@foley.com><br>To:
cisco-voip@puck.nether.net<br>Sent: Friday, March 20, 2009 2:42:40 PM GMT -05:00
US/Canada Eastern<br>Subject: [cisco-voip] SRST/Unity voicemail &
vm-integration<br><br><!-- Converted from text/rtf format -->
<p><font size="2" face="Arial">I am trying to configure SRST and vm-integration to
dial our Unity in an offsite datacenter. We have RDNIS configured on both
PRIs in the picture here, but I do not see evidence that they are working so I
moved to trying the DTMF method.</font></p>
<p><font size="2" face="Arial">So while in SRST, I can hit the message button on the
phone and it will send DTMF tones to Unity and go all the way into
sign-in. I can also call this phone from another internal phone and if no
one answers it will redirect to Unity and play the personal greeting for that
phone. However, when I call from another office over the PSTN or
cellphone, the phone rings and sends the call to Unity, but I just hear the main
greeting instead of the phone's personal voicemail greeting. Does
vm-integration support offnet calls?</font></p>
<p><font size="2" face="Arial">Here's my configuration:</font> </p>
<p><font size="2" face="Courier New">voice translation-rule 2</font> <br><font size="2" face="Courier New"> rule 1 /\(^...........\)#\(.\)/ /\1,\2/</font>
<br><font size="2" face="Courier New">!</font> </p><br>
<p><font size="2" face="Courier New">dial-peer voice 100 pots</font> <br><font size="2" face="Courier New"> description FALLBACK LONG DISTANCE</font>
<br><font size="2" face="Courier New"> translation-profile outgoing
pause</font> <br><font size="2" face="Courier New"> destination-pattern
1XXX3274781</font> <br><font size="2" face="Courier New"> port 4/0:23</font>
<br><font size="2" face="Courier New"> forward-digits extra inband</font>
<br><font size="2" face="Courier New">!</font> <br><font size="2" face="Courier New">!</font> <br><font size="2" face="Courier New">!</font> <br><font size="2" face="Courier New">!</font> <br><font size="2" face="Courier New">call-manager-fallback</font> <br><font size="2" face="Courier New"> secondary-dialtone 9</font> <br><font size="2" face="Courier New"> max-conferences 8 gain -6</font> <br><font size="2" face="Courier New"> transfer-system full-consult</font> <br><font size="2" face="Courier New"> limit-dn 7935 1</font> <br><font size="2" face="Courier New"> limit-dn 7960 10</font> <br><font size="2" face="Courier New"> ip source-address 13.XXX.150.110 port 2000</font> <br><font size="2" face="Courier New"> max-ephones 250</font> <br><font size="2" face="Courier New"> max-dn 300 dual-line</font> <br><font size="2" face="Courier New"> default-destination 1034991</font> <br><font size="2" face="Courier New"> voicemail 1XXX3274781</font> <br><font size="2" face="Courier New"> alias 1 5035 to 1034991</font> <br><font size="2" face="Courier New"> call-forward busy 1XXX3274781</font> <br><font size="2" face="Courier New"> call-forward noan 1XXX3274781 timeout 3</font>
<br><font size="2" face="Courier New"> moh bignothing.ULAW.wav</font>
<br><font size="2" face="Courier New">!</font> <br><font size="2" face="Courier New">!</font> <br><font size="2" face="Courier New">vm-integration</font>
<br><font size="2" face="Courier New"> pattern direct * CGN
# </font><br><font size="2" face="Courier New"> pattern
ext-to-ext no-answer # FDN #2 </font><br><font size="2" face="Courier New"> pattern trunk-to-ext no-answer # FDN
#2 </font><br><font size="2" face="Courier New">!</font>
<br><font size="2" face="Courier New">!</font> </p>
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