<html><head><style type='text/css'>p { margin: 0; }</style></head><body><div style='font-family: verdana,helvetica,sans-serif; font-size: 10pt; color: #000000'>Quite honestly, I have no idea what they look like. :(<br><br>I haven't spent any time collecting logs or traces. I was hoping for a quick comment (such as yours) to point them in the right direction.<br><br>I honestly don't think it's going to be anything special that Cisco needs, just common information any point-to-point connection would need to get that information across.<br><br>More than likely, we're going to be out of luck. Which sucks.<br><br>Lelio<br><br><br><div><span name="x"></span>---<br>Lelio Fulgenzi, B.A.<br>Senior Analyst, Network Infrastructure<br>Computing and Communications Services (CCS)<br>University of Guelph<br><br>519‐824‐4120 Ext 56354<br>lelio@uoguelph.ca<br>www.uoguelph.ca/ccs<br>Room 037, Animal Science and Nutrition Building<br>Guelph, Ontario, N1G 2W1<span name="x"></span><br></div><br><hr id="zwchr"><div style="color:#000;font-weight:normal;font-style:normal;text-decoration:none;font-family:Helvetica,Arial,sans-serif;font-size:12pt;"><b>From: </b>"Brian Meade" <bmeade90@vt.edu><br><b>To: </b>"Lelio Fulgenzi" <lelio@uoguelph.ca><br><b>Cc: </b>"Cisco VOIP" <cisco-voip@puck.nether.net><br><b>Sent: </b>Monday, April 28, 2014 1:42:19 PM<br><b>Subject: </b>Re: [cisco-voip] how to populate a route pattern with a call display / connected name<br><br><div dir="ltr">Usually going to be the P-Asserted-Identity or the Remote-Party-ID headers that are used for this. They may have privacy enabled or something like that. What does the SIP messaging look like?<div><br></div>
<div>Brian</div></div><div class="gmail_extra"><br><br><div class="gmail_quote">On Mon, Apr 28, 2014 at 1:35 PM, Lelio Fulgenzi <span dir="ltr"><<a href="mailto:lelio@uoguelph.ca" target="_blank">lelio@uoguelph.ca</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="font-size:10pt;font-family:verdana,helvetica,sans-serif"><br>OK, so we've moved our Nuance Speech Attendant solution from a SIP endpoint to a SIP trunk solution. One of the differences we see is now, when people dial our speech attendant, they only see number information, and no name information. I understand that typically (always?) it's the far end that would supply that information back, but I'm wondering if there is a way to force CallManager to display something?<br>
<br>The Nuance tech I was working with is going to ask about that, but he has never seen any configuration option on their system to push that information back.<br><br>Is there a SIP parameter that I should be asking them to configure or make available? <br>
<br>Thoughts?<br><br><div><span></span>---<br>Lelio Fulgenzi, B.A.<br>Senior Analyst, Network Infrastructure<br>Computing and Communications Services (CCS)<br>University of Guelph<br><br><a href="tel:519%E2%80%90824%E2%80%904120%20Ext%2056354" target="_blank">519‐824‐4120 Ext 56354</a><br>
<a href="mailto:lelio@uoguelph.ca" target="_blank">lelio@uoguelph.ca</a><br><a href="http://www.uoguelph.ca/ccs" target="_blank">www.uoguelph.ca/ccs</a><br>Room 037, Animal Science and Nutrition Building<br>Guelph, Ontario, N1G 2W1<span></span><br>
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