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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:
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<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?<o:p></o:p></span></p>
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<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 class="moz-txt-link-abbreviated" href="mailto:cisco-voip-bounces@puck.nether.net">cisco-voip-bounces@puck.nether.net</a>
[<a class="moz-txt-link-freetext" href="mailto:cisco-voip-bounces@puck.nether.net">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 class="moz-txt-link-abbreviated" href="mailto:cisco-voip@puck-nether.net">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!<o:p></o:p></span></p>
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<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<o:p></o:p></p>
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