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 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:<u></u><u></u></span></p>
<div class="im"><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></span></p><p class="MsoNormal">Then, we moved it to another subnet.<br>
It got registered but not audio in one way!<u></u><u></u></p><p class="MsoNormal"><u></u> <u></u></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"><u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></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 href="mailto:cisco-voip-bounces@puck.nether.net" target="_blank">cisco-voip-bounces@puck.nether.net</a> [mailto:<a 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 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!<u></u><u></u></span></p>
</div></div><div><div class="h5"><p class="MsoNormal"><u></u> <u></u></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: <u></u><u></u></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><u></u><u></u></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"> </span><u></u><u></u></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 href="mailto:cisco-voip-bounces@puck.nether.net" target="_blank">cisco-voip-bounces@puck.nether.net</a> [<a 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 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><u></u><u></u></p>
</div></div><p class="MsoNormal"> <u></u><u></u></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<u></u><u></u></p><div><p class="MsoNormal"><span style="font-size:10.0pt;font-family:"Calibri","sans-serif""><br><br><br><br></span><u></u><u></u></p></div><div><p class="MsoNormal">
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