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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">
<p class="MsoNormal"><span
style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"> </span></p>
<p class="MsoNormal">Then, we moved it to another
subnet.<br>
It got registered but not audio in one way!</p>
<p class="MsoNormal"> </p>
</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>
<p class="MsoNormal"><span
style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"> </span></p>
<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">
<p class="MsoNormal"> </p>
<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>
<p class="MsoNormal"><span
style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"> </span></p>
<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>
<p class="MsoNormal"> </p>
<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">
<span style="color:#666666"><br>
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