<html><head></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">If the phone don't show a codec when the call is set up then this isn't a typical routing issue. The most obvious reason for the phone not sending audio is it isn't getting the skinny StartMediaTransmission message from CUCM. <div>Have you looked at ccm traces for one of these calls? When you do look at the messages going to and from the phones in the call. Compare/contrast what you see there to a working call and call out what's different.</div><div><br></div><div>You can get a packet capture at the phone as well to see what it is being told to send to from CUCM. I'd also double check there's nothing in the network doing sccp inspection. You can get a simultaneous packet capture at the phone and cucm to make sure every packet leaving the server gets to the phone (intact).</div><div><br><div>
<span class="Apple-style-span" style="border-collapse: separate; color: rgb(0, 0, 0); font-family: Helvetica; font-size: medium; font-style: normal; font-variant: normal; font-weight: normal; letter-spacing: normal; line-height: normal; orphans: 2; text-align: auto; text-indent: 0px; text-transform: none; white-space: normal; widows: 2; word-spacing: 0px; -webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: 0px; -webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px; "><div>-Ryan</div></span>
</div>
<br><div><div>On Jan 23, 2012, at 1:48 PM, Anthony Kouloglou wrote:</div><br class="Apple-interchange-newline">
<meta content="text/html; charset=ISO-8859-1" http-equiv="Content-Type">
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There is no way that this is the problem.<br>
In one remote site i had only one 7911 working fine with CUCM 6.1.2.<br>
After the upgrade to 8.6.2a, even this old phone is having the same
issue!<br>
I keep having on the phone status: failed to update itl .<br>
<br>
On 23/1/2012 8:09 μμ, Peter Slow wrote:
<blockquote cite="mid:CAMa5Jw5zwkwzshWMd-yGWHyfA4Dtyp0VUT4iP3YrGC=Bu4AAGA@mail.gmail.com" type="cite">I think what MIke meant was "Check the routing path
between the two phones."<br>
<br>
-Peter<br>
<br>
<br>
<div class="gmail_quote">On Mon, Jan 23, 2012 at 12:41 PM, Mike <span dir="ltr"><<a moz-do-not-send="true" href="mailto:mikeeo@msn.com">mikeeo@msn.com</a>></span>
wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0
.8ex;border-left:1px #ccc solid;padding-left:1ex">
<div bgcolor="white" link="blue" vlink="purple" lang="EN-US">
<div><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Your
key statement is this:</span></p>
<div class="im"><div><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"> </span><br class="webkit-block-placeholder"></div><p class="MsoNormal">Then, we moved it to another
subnet.<br>
It got registered but not audio in one way!</p><div> <br class="webkit-block-placeholder"></div>
</div><p class="MsoNormal">Check your routing path to the CM.<span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"></span></p><div><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"> </span><br class="webkit-block-placeholder"></div>
<div>
<div style="border:none;border-top:solid #b5c4df
1.0pt;padding:3.0pt 0in 0in 0in"><p class="MsoNormal"><b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif";color:windowtext">From:</span></b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif";color:windowtext">
<a moz-do-not-send="true" href="mailto:cisco-voip-bounces@puck.nether.net" target="_blank">cisco-voip-bounces@puck.nether.net</a>
[mailto:<a moz-do-not-send="true" href="mailto:cisco-voip-bounces@puck.nether.net" target="_blank">cisco-voip-bounces@puck.nether.net</a>]
<b>On Behalf Of </b>Anthony Kouloglou<br>
<b>Sent:</b> Monday, January 23, 2012 10:15 AM<br>
<b>To:</b> Nate VanMaren<br>
<b>Cc:</b> <a moz-do-not-send="true" href="mailto:cisco-voip@puck-nether.net" target="_blank">cisco-voip@puck-nether.net</a><br>
<b>Subject:</b> Re: [cisco-voip] After upgrade to
8.6.2a one way audio for some calls-No codec
selected!</span></p>
</div>
</div>
<div>
<div class="h5"><div> <br class="webkit-block-placeholder"></div><p class="MsoNormal">Yes!<br>
Everything seems to be as it supposed to be!<br>
One Phone got registered at the main site. Worked
fine.<br>
Then, we moved it to another subnet.<br>
It got registered but not audio in one way!<br>
<br>
Can't this ITL/CTL feature/bug be disabled?<br>
<br>
On 20-Jan-12 17:26, Nate VanMaren wrote: </p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Are
your phones running firmware you expect them to
be?</span></p><div><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"> </span><br class="webkit-block-placeholder"></div>
<div>
<div style="border:none;border-top:solid #b5c4df
1.0pt;padding:3.0pt 0in 0in 0in"><p class="MsoNormal"><b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif";color:windowtext">From:</span></b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif";color:windowtext">
<a moz-do-not-send="true" href="mailto:cisco-voip-bounces@puck.nether.net" target="_blank">cisco-voip-bounces@puck.nether.net</a>
[<a moz-do-not-send="true" href="mailto:cisco-voip-bounces@puck.nether.net" target="_blank">mailto:cisco-voip-bounces@puck.nether.net</a>]
<b>On Behalf Of </b>Anthony Kouloglou<br>
<b>Sent:</b> Friday, January 20, 2012 1:33 AM<br>
<b>To:</b> <a moz-do-not-send="true" href="mailto:cisco-voip@puck-nether.net" target="_blank">cisco-voip@puck-nether.net</a><br>
<b>Subject:</b> [cisco-voip] After upgrade to
8.6.2a one way audio for some calls-No codec
selected!</span></p>
</div>
</div><div> <br class="webkit-block-placeholder"></div><p class="MsoNormal">Hi all,<br>
here is a tough one! <br>
I recently upgraded my 6.1 cluster to 8.6.2a.<br>
Since my Hardware was 7825H3 typically it was not an
upgrade rather than a fresh install using a usb
drive (cisco has this procedure for these type of
servers)<br>
The upgrade was smooth for pub and one sub.<br>
All phones reregistered and upgraded.<br>
In the main site there are 20 devices (7975, 7961,
7911) and at 2 remote sites 2 devices (one at each
site).<br>
After the upgrade:<br>
all phones in the main site can talk to each other.<br>
The two remote phones can talk to each other.<br>
Each of the remote phones when talking to main site
have one way audio!<br>
The remote site does not hear the main site always.<br>
There is no firewall/NAT between the sites.<br>
I noticed that there is no codec selected for the
audio stream that has the problems and so no
transmit (or received packets for the other).<br>
And i explain: in an active call between the main
site and a remote i checked the send/received codecs
and statistics.<br>
the main site had g711 as received codec and of
course the received packets augmented<br>
but there was none as send codec and of course no
packets transmited.<br>
In the remote site the findings were inversed (no
receive codec and no receive packets<br>
<br>
lease advise<br>
<br>
BR<br>
Anthony</p>
<div><p class="MsoNormal"><span style="font-size:10.0pt;font-family:"Calibri","sans-serif""><br>
<br>
<br>
<br>
</span></p>
</div>
<div><p class="MsoNormal">
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