Yes.<div><br></div><div>But not just 8.6. </div><div><br></div><div><a href="https://supportforums.cisco.com/docs/DOC-8131">https://supportforums.cisco.com/docs/DOC-8131</a> </div><div><br></div><div>(Hey Wes, can you fix the link on that to remove the partner only link (
<span style="color:rgb(51,51,51);font-family:Arial,verdana,sans-serif;font-size:12px;text-align:left;background-color:rgb(255,255,255)"> </span><a class="jive-link-external-small" href="http://www.cisco.com/en/US/partner/docs/voice_ip_comm/cucm/rel_notes/7_0_1/cucm-rel_notes-701.html#wp584451" style="background-image:initial;background-color:rgb(255,255,255);border-top-width:0px;border-right-width:0px;border-bottom-width:0px;border-left-width:0px;border-style:initial;border-color:initial;border-collapse:collapse;font-size:12px;list-style-type:none;list-style-position:initial;margin-top:0px;margin-right:0px;margin-bottom:0px;margin-left:0px;outline-width:initial;outline-style:none;outline-color:initial;padding-top:0px;padding-right:0px;padding-bottom:0px;padding-left:0px;color:rgb(47,102,129);text-decoration:none;font-family:Arial,verdana,sans-serif;text-align:left">SCCPv17 significantly changes message formats from previous versions</a> )</div>
<div><br></div><div>It's when you upgraded the firmware on the Phones.</div><div><br></div><div>The SCCP protocol has version numbers. I'm finding references all the way up to SCCP version 20 (in 8.5.1).</div><div>
<br></div><div>Looks like ASA version 8.3 only supports up to version 19.</div><div><br></div><div>ASA version 8.4 supports SCCP v2.0 (Don't know what that means)</div><div><br></div><div>Mike</div><div><br><div class="gmail_quote">
2012/1/24 Anthony Kouloglou <span dir="ltr"><<a href="mailto:akoul@dataways.gr">akoul@dataways.gr</a>></span><br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div bgcolor="#FFFFFF" text="#000000">
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!!<span class="HOEnZb"><font color="#888888"><br>
<br>
Anthony</font></span><div><div class="h5"><br>
<br>
On 24-Jan-12 16:34, Mike King wrote:
<blockquote 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>
<div>
<br>
</div>
<div>Mike<br>
<br>
<div class="gmail_quote">2012/1/23 Dennis Heim <span dir="ltr"><<a href="mailto:Dennis.Heim@cdw.com" target="_blank">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>
<p> </p>
<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 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">cisco-voip-bounces@puck.nether.net</a>]
on behalf of Ryan Ratliff [<a 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 href="mailto:cisco-voip@puck-nether.net" target="_blank">cisco-voip@puck-nether.net</a>
<div>
<div><br>
<b>Subject:</b> Re: [cisco-voip] After upgrade
to 8.6.2a one way audio for some calls-No
codec selected!<br>
</div>
</div>
</font><br>
</div>
<div>
<div>
<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 ìì, 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 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 solid">
<div bgcolor="white" lang="EN-US">
<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 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!</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 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></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>
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