<html><head></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">With every major CUCM version there comes the possibility of SCCP version increments.  With new features comes new SCCP messages or even changes to existing messages.  Any change like that requires testing and upgrades by any device doing inspection on the protocol and those tend to lag the CUCM releases.<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>
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<br><div><div>On Jan 24, 2012, at 11:00 AM, Anthony Kouloglou wrote:</div><br class="Apple-interchange-newline">
  
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    Hi Mike,<br>
    i have completely disabled inspection on an ASA that i have that
    does only routing.<br>
    The question is: has something changed in SCCP negotiation in CUCM
    8.6?<br>
    The whole setup has been working for 3 years!!<br>
    <br>
    Anthony<br>
    <br>
    On 24-Jan-12 16:34, Mike King wrote:
    <blockquote cite="mid:CANtPpk5+YbJ6MuVbiWu+2eZ7F4nGceOS+rjgX3Z24VghTssu-Q@mail.gmail.com" type="cite">Having been bitten by this, Check for this.
      <div><br>
      </div>
      <div>Specifically, do you have ASA's doing site to site VPN's?  By
        default they do INSPECTION, which can drop SCCP packets they
        don't recoginize.</div>
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        <br>
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      <div>Mike<br>
        <br>
        <div class="gmail_quote">2012/1/23 Dennis Heim <span dir="ltr"><<a moz-do-not-send="true" href="mailto:Dennis.Heim@cdw.com">Dennis.Heim@cdw.com</a>></span><br>
          <blockquote class="gmail_quote" style="margin:0 0 0
            .8ex;border-left:1px #ccc solid;padding-left:1ex">
            <div style="WORD-WRAP:break-word">
              <div style="direction:ltr;font-size:10pt;font-family:Tahoma"><p>This may have already been mentioned but building on
                  what Ryan said... probably between 6.1(2) and 8.6.x
                  you had a firmware change, probably from around 8.4ish
                  to 9.x. The sccp version changes, and it sounds like
                  you might have some firewall/security device in the
                  way that is not opening the ports because it is used
                  to the older version of skinny.</p><div> <br class="webkit-block-placeholder"></div><p>-Dennis-</p>
                <div style="font-size:16px;font-family:Times New Roman">
                  <hr>
                  <div style="DIRECTION:ltr"><font color="#000000" face="Tahoma"><b>From:</b> <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">cisco-voip-bounces@puck.nether.net</a>]
                      on behalf of Ryan Ratliff [<a moz-do-not-send="true" href="mailto:rratliff@cisco.com" target="_blank">rratliff@cisco.com</a>]<br>
                      <b>Sent:</b> Monday, January 23, 2012 2:05 PM<br>
                      <b>To:</b> Anthony Kouloglou<br>
                      <b>Cc:</b> Mike; <a moz-do-not-send="true" href="mailto:cisco-voip@puck-nether.net" target="_blank">cisco-voip@puck-nether.net</a>
                      <div>
                        <div class="h5"><br>
                          <b>Subject:</b> Re: [cisco-voip] After upgrade
                          to 8.6.2a one way audio for some calls-No
                          codec selected!<br>
                        </div>
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                    <div class="h5">
                      <div>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 style="border-collapse:separate;text-indent:0px;letter-spacing:normal;text-transform:none;font:medium
Helvetica;white-space:normal;word-spacing:0px">
                              <div>-Ryan</div>
                            </span></div>
                          <br>
                          <div>
                            <div>On Jan 23, 2012, at 1:48 PM, Anthony
                              Kouloglou wrote:</div>
                            <br>
                            <div bgcolor="#FFFFFF">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 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" target="_blank">mikeeo@msn.com</a>></span>
                                  wrote:<br>
                                  <blockquote class="gmail_quote" style="PADDING-LEFT:1ex;MARGIN:0px
                                    0px 0px 0.8ex;BORDER-LEFT:#ccc 1px
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                                      <div><p class="MsoNormal"><span style="FONT-SIZE:11pt;COLOR:#1f497d;FONT-FAMILY:'Calibri','sans-serif'">Your
                                            key statement is this:</span></p>
                                        <div>
                                          <div><span style="FONT-SIZE:11pt;COLOR:#1f497d;FONT-FAMILY:'Calibri','sans-serif'"></span><br>
                                             </div><p class="MsoNormal">Then, we
                                            moved it to another subnet.<br>
                                            It got registered but not
                                            audio in one way!</p>
                                          <div><br>
                                             </div>
                                        </div><p class="MsoNormal">Check your
                                          routing path to the CM.<span style="FONT-SIZE:11pt;COLOR:#1f497d;FONT-FAMILY:'Calibri','sans-serif'"></span></p>
                                        <div><span style="FONT-SIZE:11pt;COLOR:#1f497d;FONT-FAMILY:'Calibri','sans-serif'"></span><br>
                                           </div>
                                        <div>
                                          <div style="BORDER-RIGHT:medium
                                            none;PADDING-RIGHT:0in;BORDER-TOP:#b5c4df
                                            1pt
                                            solid;PADDING-LEFT:0in;PADDING-BOTTOM:0in;BORDER-LEFT:medium
                                            none;PADDING-TOP:3pt;BORDER-BOTTOM:medium
                                            none"><p class="MsoNormal"><b><span style="FONT-SIZE:10pt;COLOR:windowtext;FONT-FAMILY:'Tahoma','sans-serif'">From:</span></b><span style="FONT-SIZE:10pt;COLOR:windowtext;FONT-FAMILY:'Tahoma','sans-serif'">
                                                <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>
                                            <div><br>
                                               </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:11pt;COLOR:#1f497d;FONT-FAMILY:'Calibri','sans-serif'">Are
                                                your phones running
                                                firmware you expect them
                                                to be?</span></p>
                                            <div><span style="FONT-SIZE:11pt;COLOR:#1f497d;FONT-FAMILY:'Calibri','sans-serif'"></span><br>
                                               </div>
                                            <div>
                                              <div style="BORDER-RIGHT:medium
                                                none;PADDING-RIGHT:0in;BORDER-TOP:#b5c4df
                                                1pt
                                                solid;PADDING-LEFT:0in;PADDING-BOTTOM:0in;BORDER-LEFT:medium
                                                none;PADDING-TOP:3pt;BORDER-BOTTOM:medium
                                                none"><p class="MsoNormal"><b><span style="FONT-SIZE:10pt;COLOR:windowtext;FONT-FAMILY:'Tahoma','sans-serif'">From:</span></b><span style="FONT-SIZE:10pt;COLOR:windowtext;FONT-FAMILY:'Tahoma','sans-serif'">
                                                    <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>
                                               </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:10pt;FONT-FAMILY:'Calibri','sans-serif'"><br>
                                                  <br>
                                                  <br>
                                                  <br>
                                                </span></p>
                                            </div>
                                            <div><p class="MsoNormal"><span style="COLOR:#666666"><br>
                                                  <br>
                                                  NOTICE: This email
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