<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'>Thanks Brian.<br><br>I did engage the TAC and they were able to track it down to Nuance sending back an "inactive media" parameter setting. Nuance has confirmed that they've seen something like this before and are working at possibly recreating a patch they did for a previous version.<br><br>The TAC did mention something about the order preference of MR(G)L being done differently though. <br><br>Lelio<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>Sunday, April 27, 2014 8:12:09 PM<br><b>Subject: </b>Re: [cisco-voip] transfer to SIP endpoint (Nuance Speech Attendant) no longer working after upgrade from v7.1(5)SU3 to v9.1(2)SU1<br><br><div dir="ltr">Lelio,<div><br></div><div>That bug was fixed in 9.1.2 so you should have the fix as well. We'll probably need to take a look st the SIP messaging either from a packet capture or CallManager traces ideally to see what is going on with these calls.</div>
<div><br></div><div>Brian</div></div><div class="gmail_extra"><br><br><div class="gmail_quote">On Sun, Apr 27, 2014 at 12:01 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, transfers to Cisco SIP devices work fine.<br>
<br>found this bug after looking at list archives...<br><br><a href="https://tools.cisco.com/bugsearch/bug/CSCug50634" target="_blank">https://tools.cisco.com/bugsearch/bug/CSCug50634</a><br><br><div>one-way voice with diverted 3rd party phones, from sip via mgcp/qsig<br>
<br>not sure if it's related....<br><br><br></div><div class=""><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></div><br></div><hr><div style="font-size:12pt;font-style:normal;font-family:Helvetica,Arial,sans-serif;text-decoration:none;font-weight:normal">
<b>From: </b>"Lelio Fulgenzi" <<a href="mailto:lelio@uoguelph.ca" target="_blank">lelio@uoguelph.ca</a>><br><b>To: </b>"Cisco VOIP" <<a href="mailto:cisco-voip@puck.nether.net" target="_blank">cisco-voip@puck.nether.net</a>><br>
<b>Sent: </b>Sunday, April 27, 2014 11:40:23 AM<br><b>Subject: </b>transfer to SIP endpoint (Nuance Speech Attendant) no longer working after upgrade from v7.1(5)SU3 to v9.1(2)SU1<div><div class="h5"><br><br><div style="font-size:10pt;font-family:verdana,helvetica,sans-serif">
<br>OK, this is a weird one. It took a while to isolate, but it seems like calls that are transferred to SIP endpoints are not working after an upgrade from v7.1(5)SU3 to v9.1(2)SU1.<br><br>We don't have many SIP endpoints, in fact, we only have 3, but two of those are our Nuance Speech Attendants. The other is an ATA-187.<br>
<br>We're going to bring our spare ATA-187 up for additional testing to see if it's one way audio, or no audio, etc. But that may take a while since we're not too used to bringing these up quickly.<br><br>I'm also going to consider opening a TAC case, cause I have no idea where to look for something like this. Not being familiar with SIP endpoints too much. And never seeing something break after a transfer. :(<br>
<br>Any ideas anyone?<br><br>Lelio<br><br><br><br><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>
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