Nate thanks for your quick response. This clears up the confusion its like routing voice mail calls from CUCM over the SIP trunk to user's voice mailbox. The only thing we have to keep in mind is like other SIP Call routing is max number of calls. No voice mail ports...<br>
<br>Thanks Again much appreciated. <br><br><div class="gmail_quote">On Sat, Feb 18, 2012 at 9:02 AM, Nate VanMaren <span dir="ltr"><<a href="mailto:VanMarenNP@ldschurch.org">VanMarenNP@ldschurch.org</a>></span> wrote:<br>
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<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Yea there isn’t really “ports” that you have to worry about on the SIP integrations, just max number of calls.<u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">You will still need a VM pilot and profile, and then a route pattern that points to the sip trunk that is pointed at exchange UM.<u></u><u></u></span></p>
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<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><a href="http://www.agileit.com/Blog/Lists/Posts/Post.aspx?ID=820" target="_blank">http://www.agileit.com/Blog/Lists/Posts/Post.aspx?ID=820</a><u></u><u></u></span></p>
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<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><a href="http://www.microsoft.com/download/en/details.aspx?id=13591" target="_blank">http://www.microsoft.com/download/en/details.aspx?id=13591</a><u></u><u></u></span></p>
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<p class="MsoNormal"><b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif"">From:</span></b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif""> Gr [mailto:<a href="mailto:grccie@gmail.com" target="_blank">grccie@gmail.com</a>]
<br>
<b>Sent:</b> Friday, February 17, 2012 3:00 PM<br>
<b>To:</b> Jason Aarons (AM); Nate VanMaren<br>
<b>Cc:</b> <a href="mailto:cisco-voip@puck.nether.net" target="_blank">cisco-voip@puck.nether.net</a></span></p><div><div></div><div class="h5"><br>
<b>Subject:</b> Re: [cisco-voip] CUCM 8.5 integration with Exchange 2010 for Voice mail<u></u><u></u></div></div><p></p>
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<p class="MsoNormal">Thanks Nate, Jason! Valuable information, I will keep this in mind.<u></u><u></u></p>
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<p class="MsoNormal"><u></u> <u></u></p>
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<p class="MsoNormal">Another question is we just create voice mail pilot in cucm and route it to sip trunk and then in exchange 2010 we create voice mail pilot and the actual voice mail ports?<u></u><u></u></p>
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<p class="MsoNormal"><u></u> <u></u></p>
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<p class="MsoNormal">Thanks,<u></u><u></u></p>
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<p class="MsoNormal">GR<u></u><u></u></p>
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Sent from my iPhone<u></u><u></u></p>
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On 18/02/2012, at 4:35 AM, "Jason Aarons (AM)" <<a href="mailto:jason.aarons@dimensiondata.com" target="_blank">jason.aarons@dimensiondata.com</a>> wrote:<u></u><u></u></p>
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<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">I think I understand that Exchange 2010 has a crappy sip stack. Good info. <lol></span><u></u><u></u></p>
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<p class="MsoNormal"><b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif"">From:</span></b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif"">
<a href="mailto:cisco-voip-bounces@puck.nether.net" target="_blank">cisco-voip-bounces@puck.nether.net</a>
<a href="mailto:[mailto:cisco-voip-bounces@puck.nether.net]" target="_blank">[mailto:cisco-voip-bounces@puck.nether.net]</a>
<b>On Behalf Of </b>Nate VanMaren<br>
<b>Sent:</b> Friday, February 17, 2012 11:03 AM<br>
<b>To:</b> gr11; <a href="mailto:cisco-voip@puck.nether.net" target="_blank">cisco-voip@puck.nether.net</a><br>
<b>Subject:</b> Re: [cisco-voip] CUCM 8.5 integration with Exchange 2010 for Voice mail</span><u></u><u></u></p>
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<p class="MsoNormal"> <u></u><u></u></p>
<p class="MsoNormal" style="margin-bottom:12.0pt"> <u></u><u></u></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Two things off the top of my head.</span><u></u><u></u></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"> </span><u></u><u></u></p>
<p><u></u><span>1.<span style="font:7.0pt "Times New Roman"">
</span></span><u></u><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"> Exchange has a crappy sip stack. So you have to use a MTP on the SIP trunk because it won’t deal with RTP source/destination changes in a session. Like
when someone does a supervised transfer to voicemail.</span><u></u><u></u></p>
<p><u></u><span>2.<span style="font:7.0pt "Times New Roman"">
</span></span><u></u><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Exchange has a crappy sip stack. So if you want correct caller name on the voicemail on call transferred to voicemail, you have to run the transfer through
an app that waits for the transferee to complete the transfer to send the call to exchange.</span><u></u><u></u></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"> </span><u></u><u></u></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Voicemail preview takes a lot of hardware. I think our boxes are quad core with 8/16gb of ram and 4-5 calls will max out the CPU.</span><u></u><u></u></p>
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<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">-Nate</span><u></u><u></u></p>
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<p class="MsoNormal"><b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif"">From:</span></b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif"">
<a href="mailto:cisco-voip-bounces@puck.nether.net" target="_blank">cisco-voip-bounces@puck.nether.net</a>
<a href="mailto:[mailto:cisco-voip-bounces@puck.nether.net]" target="_blank">[mailto:cisco-voip-bounces@puck.nether.net]</a>
<b>On Behalf Of </b>gr11<br>
<b>Sent:</b> Thursday, February 16, 2012 5:17 PM<br>
<b>To:</b> <a href="mailto:cisco-voip@puck.nether.net" target="_blank">cisco-voip@puck.nether.net</a><br>
<b>Subject:</b> [cisco-voip] CUCM 8.5 integration with Exchange 2010 for Voice mail</span><u></u><u></u></p>
<p class="MsoNormal"> <u></u><u></u></p>
<p class="MsoNormal">Hi List,<br>
<br>
I am providing the CUCM8.5 integration with exchange 2010 for a customer for their voice mail needs. The customer has an old unity server that will be decommissioned and voice mail functionality will be provided by exchange 2010 UM.<br>
<br>
Anyone who has done this before, any pitfalls or things to be aware of? We are going to use a third party gateway for SIP Trunk termination to/from CUCM and exchange<br>
<br>
Thanks,<br>
GR<u></u><u></u></p>
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<br>
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the sender by reply email and destroy all copies of the original message.</span><u></u><u></u></p>
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<br>
</span><span style="color:white">itevomcid</span><span style="color:#666666"> </span>
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