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Nope!<br>
I have got a TAC open since the day before yesterday but i really
count on all of you guys!!<br>
:-)<br>
<br>
On 24/1/2012 9:29 ìì, Ruben Montes (Europe) wrote:
<blockquote
cite="mid:6988B59F48F66545A22C9871E09CE78B1843F297@eudefrasvex01.eu.didata.local"
type="cite">
<meta http-equiv="Content-Type" content="text/html;
charset=ISO-8859-7">
<style type="text/css" id="owaParaStyle"></style>
<div style="direction: ltr;font-family: Tahoma;color:
#000000;font-size: 10pt;">Is it working properly now?<br>
<div style="font-family: Times New Roman; color: #000000;
font-size: 16px">
<hr tabindex="-1">
<div id="divRpF665186" style="direction: ltr; "><font
color="#000000" face="Tahoma" size="2"><b>From:</b>
<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-abbreviated" href="mailto:cisco-voip-bounces@puck.nether.net">cisco-voip-bounces@puck.nether.net</a>] on behalf of Anthony
Kouloglou [<a class="moz-txt-link-abbreviated" href="mailto:akoul@dataways.gr">akoul@dataways.gr</a>]<br>
<b>Sent:</b> 24 January 2012 20:17<br>
<b>To:</b> Mike King<br>
<b>Cc:</b> <a class="moz-txt-link-abbreviated" href="mailto:cisco-voip@puck-nether.net">cisco-voip@puck-nether.net</a>; Mike<br>
<b>Subject:</b> Re: [cisco-voip] After upgrade to 8.6.2a
one way audio for some calls-No codec selected!<br>
</font><br>
</div>
<div><br>
<br>
Hi all,<br>
well, i have disabled any kind inspection on the ASA.Isn't
that enough?<br>
ASA does NOT NAT. Isn't that enough?<br>
However, i have to check some corporate linux based vpn
endpoints.<br>
<br>
Anthony<br>
<br>
On 24/1/2012 6:30 ìì, Mike King wrote:
<blockquote type="cite">Yes.
<div><br>
</div>
<div>But not just 8.6. </div>
<div><br>
</div>
<div><a moz-do-not-send="true"
href="https://supportforums.cisco.com/docs/DOC-8131"
target="_blank">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 moz-do-not-send="true"
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-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"
target="_blank">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 moz-do-not-send="true"
href="mailto:akoul@dataways.gr" target="_blank">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">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
moz-do-not-send="true"
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
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><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
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 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
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 message
is for the
sole use of
the intended
recipient(s)
and may
contain
confidential
and privileged
information.
Any
unauthorized
review, use,
disclosure or
distribution
is prohibited.
If you are not
the intended
recipient,
please contact
the sender by
reply email
and destroy
all copies of
the original
message.</span></p>
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<br>
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</blockquote>
<br>
<br>
<br>
<font color="white">itevomcid</font> </div>
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