<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=Content-Type content="text/html; charset=iso-8859-1">
<META content="MSHTML 5.50.4937.800" name=GENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=#ffffff>
<DIV><SPAN class=520323313-26072004><FONT face=Arial color=#0000ff
size=2>Lelio,</FONT></SPAN></DIV>
<DIV><SPAN class=520323313-26072004><FONT face=Arial color=#0000ff
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=520323313-26072004><FONT face=Arial color=#0000ff
size=2>UnknownCallerId - this is when a call arrives inbound to CM without any
calling party name or number (inbound through an MGCP FXO or through a t1-cas
link), what number should be displayed on the ipphone. the other Unknown*
params address this same situation.</FONT></SPAN></DIV>
<DIV><SPAN class=520323313-26072004><FONT face=Arial color=#0000ff
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=520323313-26072004><FONT face=Arial color=#0000ff
size=2>UserUserIEStatus - I have never seen callerID information passed in
user-user IE. This is typically for vendor extensions such as how CM instructs
h323 gateways to provide ringback tone during a transfer.</FONT></SPAN></DIV>
<DIV><SPAN class=520323313-26072004><FONT face=Arial color=#0000ff
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=520323313-26072004><FONT face=Arial color=#0000ff size=2>You
can use the "<FONT face="Times New Roman" color=#000000 size=3>Caller ID DN"
field on the gateway configuration page overwrite the calling pary number on
outbound calls. Using this parameter will cause the calling party name to
be blanked out. </FONT></FONT></SPAN></DIV>
<DIV><SPAN class=520323313-26072004><FONT face=Arial color=#0000ff
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=520323313-26072004><FONT face=Arial color=#0000ff size=2>When
the call is routed over the PSTN, the final PSTN switch before hopoff to
the destination CPE will perform a name lookup via the SS7 cloud on the
calling party number to determine what calling party name to send to the
destination CPE. If you choose your "main number" and register it with
your telco in the ss7 name database, then when you use the "caller id dn" field
on the gateway to overwrite calling party name with your "main number" all PSTN
destinations should receive the correct calling party name.</FONT></SPAN></DIV>
<DIV><SPAN class=520323313-26072004><FONT face=Arial color=#0000ff
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=520323313-26072004><FONT face=Arial color=#0000ff size=2>As far
as overwriting calling party name when we're just connected CCM---PBX, well, I
haven't found a way to do that yet.</FONT></SPAN></DIV>
<DIV><SPAN class=520323313-26072004><FONT face=Arial color=#0000ff
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=520323313-26072004><FONT face=Arial color=#0000ff
size=2>/Wes</FONT></SPAN></DIV>
<BLOCKQUOTE dir=ltr
style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px solid; MARGIN-RIGHT: 0px">
<DIV class=OutlookMessageHeader dir=ltr align=left><FONT face=Tahoma
size=2>-----Original Message-----<BR><B>From:</B>
cisco-voip-bounces@puck.nether.net
[mailto:cisco-voip-bounces@puck.nether.net]<B>On Behalf Of </B>Lelio
Fulgenzi<BR><B>Sent:</B> Thursday, July 22, 2004 6:52 PM<BR><B>To:</B>
cisco-voip@puck.nether.net<BR><B>Subject:</B> [cisco-voip] Unknown CallerID
System Parameters<BR><BR></FONT></DIV>
<DIV><FONT face=Arial size=2>Just wondering if anyone has successfully used
the following system parameters for
manipulating <STRONG>outbound</STRONG> callerID information. The
description is not all to clear to me, and I'm not sure if these
parameters modify the unknown callerID information that leaves the call
manager out to our PSTN trunks, or the other way around. Currently,
our PSTN service (Bell Canada Megalink PRI DMS-100) does not support
service level text entry, and opening a case with the TAC said the only
way we could get this working would be to enable the display of the individual
line's caller information field, but then were told this is not compatible
with DMS-100, either way, it's not what we want. We'd like to get all calls to
display one name to outbound calls, i.e. "University of Guelph". I came
across this in the forums and am hoping this will do what we
want.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2><STRONG>UnknownCallerId: </STRONG></FONT><FONT
face=Arial size=2>The directory number to be displayed. Valid value is any
numeric value representing a general number for your system (if you wish to
provide caller ID functionality to called parties). Valid value is any valid
telephone number.</FONT></DIV>
<DIV dir=ltr><FONT face=Arial size=2><STRONG></STRONG></FONT> </DIV>
<DIV dir=ltr><FONT face=Arial size=2><STRONG>UnknownCallerIdFlag:</STRONG>
This parameter is related to the Unknown CallerId field. We strongly recommend
using the default setting since this can now be configured using Cisco
CallManager Administration. <U><EM>Default: T</EM> <BR></U></FONT></DIV>
<DIV dir=ltr><FONT face=Arial
size=2><STRONG>UnknownCallerIdText:</STRONG> The text to be displayed to
called parties having caller ID capability. The first line is 20 characters
and the second line is 14 characters. Try to get a saying which looks OK in
the display when broken into two lines having the specified number of
characters per line. <EM><U>Default: Unknown</U></EM></FONT></DIV>
<DIV dir=ltr><FONT face=Arial
size=2><EM></EM><EM></EM><EM></EM><BR><STRONG>UserUserIEStatus:</STRONG>
If the user to user information element (UUIE) is passed in the system,
enabling UUIE status allows ISDN PRI messages to include them outbound PRI
calls. <EM><U>Default: F<BR></U></EM> <BR></DIV></FONT>
<DIV>-----
-----<BR>Lelio Fulgenzi,
B.A.
<A href="mailto:lelio@uoguelph.ca.eh">lelio@uoguelph.ca.eh</A><BR>Network
Analyst (CCS)<BR>University of
Guelph
FAX:(519) 767-1060 JNHN<BR>Guelph, Ontario N1G
2W1
TEL:(519) 824-4120
x56354<BR>^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^<BR>
remove the 1st letter of the canadian alphabet from my email, eh!</DIV>
<BLOCKQUOTE
style="PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
<DIV style="FONT: 10pt arial">----- Original Message ----- </DIV>
<DIV
style="BACKGROUND: #e4e4e4; FONT: 10pt arial; font-color: black"><B>From:</B>
<A title=wsisk@cisco.com href="mailto:wsisk@cisco.com">Wes Sisk</A> </DIV>
<DIV style="FONT: 10pt arial"><B>To:</B> <A
title=vandy.hamidi@markettools.com
href="mailto:vandy.hamidi@markettools.com">Vandy Hamidi</A> ; <A
title=cisco-voip@puck.nether.net
href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</A>
</DIV>
<DIV style="FONT: 10pt arial"><B>Sent:</B> Thursday, July 22, 2004 5:52
PM</DIV>
<DIV style="FONT: 10pt arial"><B>Subject:</B> RE: [cisco-voip] Weird
Conference Voice Quality</DIV>
<DIV><BR></DIV>Vandy,<BR><BR>This looks mostly correct - just a cpl
questions-<BR><BR>Can we see the rest of the config for the route-map on the
FastE<BR>"ip policy route-map SET-IP-QOS" This may be re-marking your
VOIP traffic<BR>so it does not match your outbound QOS policy, which
identifies VOIP bearer<BR>traffic based on:<BR><BR>ip access-list extended
VOIP-Classify-Voice<BR> permit ip any any precedence
critical<BR> permit ip any any dscp ef<BR><BR><BR>Also, your QOS-POLICY
appears to allocate 650kbps, this ususally should only<BR>be 75% of
available bw, so do your links offer 650/0.75= 866kbps at least?<BR><BR>This
is discussed in the QOS srnd at <A
href="http://www.cisco.com/go/srnd">http://www.cisco.com/go/srnd</A> if
you'd<BR>like to read more.<BR><BR>/Wes<BR><BR><BR><BR>> -----Original
Message-----<BR>> From: Vandy Hamidi
[mailto:vandy.hamidi@markettools.com]<BR>> Sent: Thursday, July 22, 2004
5:27 PM<BR>> To: Wes Sisk; <A
href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</A><BR>>
Subject: RE: [cisco-voip] Weird Conference Voice
Quality<BR>><BR>><BR>> Thanks Wes, that explains a lot and confirms
a hunch I had. It looks<BR>> like the 6608 waiting for a jittery
stream will cause choppiness for all<BR>> participants. That's a
real great design...<BR>><BR>> Because conference voice quality is
always GOOD when data isn't<BR>> competing, the QOS classifying and
policing must not be setup right. We<BR>> had outside consultants
install our base system and I've been using<BR>> their QOS to stamp out
other remote sites.<BR>> Something must be misconfigured.<BR>><BR>>
Can anyone just give it a once over. I cut and paste just the
pertinent<BR>> parts of my 2651XM config below.<BR>><BR>> If anyone
could email me their QOS setup, I would be extremely grateful.<BR>> This
problem is high priority and high visibility and I've be very<BR>>
thankful.<BR>> Thank you in advanced,<BR>><BR>>
-=Vandy-=<BR>><BR>> class-map match-all VOIP-VOICE_CONTROL<BR>>
match access-group name VOIP-Classify-Control<BR>> class-map
match-all VOIP-VOICE<BR>> match access-group name
VOIP-Classify-Voice<BR>> !<BR>> !<BR>> policy-map
QOS-POLICY<BR>> class VOIP-VOICE<BR>> priority
600<BR>> class VOIP-VOICE_CONTROL<BR>> bandwidth
50<BR>> class class-default<BR>>
fair-queue<BR>> !<BR>> interface Multilink1<BR>> description Market
Tools PL431562<BR>> ip address 172.16.16.70 255.255.255.252<BR>>
service-policy output QOS-POLICY<BR>> load-interval 30<BR>> no peer
neighbor-route<BR>> no cdp enable<BR>> ppp multilink<BR>> ppp
multilink fragment delay 20<BR>> ppp multilink interleave<BR>> ppp
multilink group 1<BR>> !<BR>> interface FastEthernet0/0<BR>>
description User/Server VLAN<BR>> encapsulation dot1Q 1 native<BR>> ip
address 10.21.0.1 255.255.255.0<BR>> ip policy route-map
SET-IP-QOS<BR>> !<BR>> interface FastEthernet0/1<BR>> description
VOIP Phone VLAN<BR>> encapsulation dot1Q 4<BR>> ip address 10.21.4.1
255.255.252.0<BR>> ip policy route-map SET-IP-QOS<BR>> !<BR>> ip
access-list extended VOIP-Classify-Control<BR>> permit ip any any
precedence flash<BR>> permit ip any any dscp af31<BR>> ip access-list
extended VOIP-Classify-Voice<BR>> permit ip any any precedence
critical<BR>> permit ip any any dscp ef<BR>> ip access-list extended
VOIP-Control<BR>> permit tcp any any range 2000 2002<BR>> permit tcp
any any eq 1720<BR>> permit tcp any any range 11000 11999<BR>> permit
udp any any eq 2427<BR>> ip access-list extended VOIP-Routine<BR>>
permit ip any any<BR>> ip access-list extended VOIP-Voice<BR>> permit
udp any any range 16384 32767<BR>><BR>><BR>> -----Original
Message-----<BR>> From: Wes Sisk [mailto:wsisk@cisco.com]<BR>> Sent:
Thursday, July 22, 2004 6:07 AM<BR>> To: Vandy Hamidi; <A
href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</A><BR>>
Subject: RE: [cisco-voip] Weird Conference Voice Quality<BR>><BR>>
Vandy,<BR>><BR>> If audio coming from the source device to the
conference device has<BR>> large,<BR>> irregular jitter, then everyone
listening to the conference will be<BR>> affected.<BR>><BR>> Simple
way to test this out:<BR>> setup conference with remote users and 1
central user.<BR>> have all remote users go on "mute" (no RTP stream
generated on mute)<BR>> have central user speak.<BR>> How is voice
quality?<BR>><BR>> Now, have everyone except 1 remote user go on
mute.<BR>> How is voice quality?<BR>><BR>> We need to make sure
those streams coming from the remote sites are<BR>> "good"<BR>> with
very low jitter as the 6608 uses a static dejitter buffer in<BR>>
contrast<BR>> with most ipphones and gateways which use dynamic jitter
buffers. This<BR>> makes fragmentation critical on the WAN
links.<BR>><BR>> CSCdx78486 6608 configured as CFB has a static jitter
buffer, needs<BR>> adaptive<BR>> Problem Description: Voice
quality suffers with choppiness when<BR>> conference<BR>> particants
are located across low speed links and the device providing<BR>>
for<BR>> conferencing is the 6608.<BR>><BR>> Workaround:
None<BR>><BR>> /Wes<BR>><BR>> > -----Original
Message-----<BR>> > From: <A
href="mailto:cisco-voip-bounces@puck.nether.net">cisco-voip-bounces@puck.nether.net</A><BR>>
> [mailto:cisco-voip-bounces@puck.nether.net]On Behalf Of Vandy
Hamidi<BR>> > Sent: Wednesday, July 21, 2004 11:30 PM<BR>> > To:
<A
href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</A><BR>>
> Subject: [cisco-voip] Weird Conference Voice Quality<BR>>
><BR>> ><BR>> > Hey guys,<BR>> > I'm having a weird
problem that isn't making sense<BR>> ><BR>> > I have a 6608-T1
blade in my main office performing conferencing for<BR>> > most of the
remote offices in the company. I know, not recommended,<BR>>
but<BR>> > read on.<BR>> ><BR>> > All the offices are
connected Hub/Spoke to the main office and I have<BR>> > QOS setup
(supposedly correctly) on my wan links and call quality is<BR>> >
great. I've tested the QOS by maxing out the line during voice
calls.<BR>> ><BR>> > We've been receiving user complaints of
voice quality problems.<BR>> Cutting<BR>> > out, delays, overall
chopping communication. All these complaints are<BR>> > when
they are on conference calls.<BR>> ><BR>> > We've performed
testing and during high wan usage the conference call<BR>> > quality
becomes horrible to/from the remote sites and sometimes for<BR>>
all<BR>> > users for example I'll have a hard time understanding a
conference<BR>> > attendee in my own office.<BR>> > We're able
to reproduce with windows copies during conference calls.<BR>>
><BR>> > 1) Why is the voice quality ok with phone to phone calls,
but become<BR>> > horrible during a conference call.<BR>> > 2)
Does the 6608-T1 blade change/alter the QOS/TOS bits on the<BR>>
packets?<BR>> > 3) Why would this issue affect local attendee
conference quality when<BR>> > everyone is connected to the same
switch where the 6608 blade is.<BR>> ><BR>> > TIA,<BR>>
><BR>> > -=Vandy=-<BR>> ><BR>> ><BR>> >
_______________________________________________<BR>> > cisco-voip
mailing list<BR>> > <A
href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</A><BR>>
> <A
href="https://puck.nether.net/mailman/listinfo/cisco-voip">https://puck.nether.net/mailman/listinfo/cisco-voip</A><BR>><BR>><BR><BR>_______________________________________________<BR>cisco-voip
mailing list<BR><A
href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</A><BR><A
href="https://puck.nether.net/mailman/listinfo/cisco-voip">https://puck.nether.net/mailman/listinfo/cisco-voip</A><BR></BLOCKQUOTE></BLOCKQUOTE></BODY></HTML>