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<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
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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
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