Your replication is OK. You do not have any server with status 3, you
have a server with ID = 2 and one with ID = 3, STATE is for both active.
The 3 you mean has nothing to do with a replication status. It is just
an ID.
Matthew Berry wrote:
Can someone explain this to me?
Below is my
In Brussels, yes.
Omotayo wrote:
In all the locations ( Brussels)?
On Fri, Nov 27, 2009 at 9:48 AM, Paul Kruger pauld.kru...@gmail.com
mailto:pauld.kru...@gmail.com wrote:
All the phones are 7965's. All of them are in front of you.
On Fri, Nov 27, 2009 at 3:16 AM, Chuck
The functionality is when you answer the call on the cell phone, the
line button LED on your desk phone should light red. If you press the
line button, you should get into remote in use-state. As far as i know
it is impossible to get into remote in use state with just answering
the call on the
Congratulations!
So it seems to be that IPX has the best study materials for the V3-Lab.
Since July there has been 12 Voice-CCIEs on IPX Success Stories. Half of
them passed in the last 3 weeks.
Checking other vendors, none of them has so many passing students for
the new lab.
Aamir
Congratulations!
___
For more information regarding industry leading CCIE Lab training, please visit
www.ipexpert.com
Congratulations!
October seems to be a good month for Voice CCIEs.
;-)
___
For more information regarding industry leading CCIE Lab training, please visit
www.ipexpert.com
,
if you analyse your attempt inside out and learn what your weak topics
are and what topics are OK. You must know everything what has not worked
in your failed attempt and and you must learn in your lab how it is
working. Don't make the same mistakes twice.
HTH,
Phil G
Phil G wrote:
Hi!
I
Hi!
I took my second V3-lab attempt on Thursday in Brussels (my third
attempt in sum) and finally i got my number! As you can imagine i am
very happy that i finally nailed it down. I want to thank everybody on
this list, which has been a very use- and helpful resource during
intensive
This is a kind of workaround for backup the speed dials of IPMA. I got
this from TAC in a case where the speed dials got lost. Maybe this works
for you:
1. Login to the assistant console application and configure all the
speed dials;
2. Once you have all the speeddials remove the network
Hi Mark!
#1
I have just checked the new config of the PSTN-Simulator. I think there
is still a error in the WB: External phone number mask for HQ phones
should be +44205943. Question 2.3 wants to have calls from
PSTN-London-UK should be displayed in call history as +4402059432785.
The
Mark Snow wrote:
I believe this did change in 7.0.2.
But either way, in SP you can configure inbound RD matching to be Full
Match (default) or Partial Match and if you do the latter, the next
field allows you to configure how many digits from the right to match
(default of 10). Unless
Hi!
Your prompt-location seems to be wrong:
application
service aa-drop flash:bacdprompts/app-b-acd-aa-2.1.2.2.tcl
paramspace english location flash:
It should be
application
service aa-drop flash:bacdprompts/app-b-acd-aa-2.1.2.2.tcl
paramspace
Can you post a sh flash:?
Omotayo wrote:
Hello,
i have same issue with the aa
REgards
On Wed, Oct 14, 2009 at 8:00 AM, Phil G pgciscov...@gmx.net
mailto:pgciscov...@gmx.net wrote:
Hi!
Your prompt-location seems to be wrong:
application
service aa-drop
Hello!
I have implemented CUPS/CUPC with CUCM and one user is configured with
IPPM. I can send IM between IPPM and CUPC and vice versa, but if i send
a message from CUPC to IPPM, there is no pop-up on the phone with the
message, instead i have to go into the IPPM-application and have to
The application user has all phones associated. I even do not get an
audible alert, even though the audilble-alert is enabled.
Steve Slater wrote:
Hi Phil,
Do you have the phone that is running IPPM associated to the IPPM App user?
This should allow you to enable audible alerts on the
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Phil G
Sent: Miércoles, 14 de Octubre de 2009 04:25 a.m.
To: Steve Slater
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] CUPS - IPPM
The application user has all phones associated. I even do not get an
audible alert
Calling Party manipulations performed on the RP or RL/RG details will
NOT override your Calling Party XForms. Transformations will override
any manipulations in RP/RL-Level.
In your case:
Your Calling Party XForm Pattern 212394500X will never be matched,
because your calling party is 5001.
There are 2 steps missing:
Under Application/Deskphone Control/User Assingments give the user the
permission to use Deskphone Control.
Under Application/CUPC/CTI Profile assing user to appropriate profile.
Nara Shikamaru wrote:
In working though this section and doublechecking my work in
Hi!
I have tried to configure my CME with a SCCP phone and a SIP phone. Both
phones are 7965 as in the real lab. I have SCCP load 8-3-3 and SIP load
8-3-3 downloaded and installed on flash in 2 different directories. Now
i wanted to configure the TFTP-Server statements, but there are some
My thoughts:
1. You are reffering to the wrong load:
voice register global
load 7960-7940 P003-8-12-00
P003-8-12-00 is the SCCP-image.
You need the SIP-image which should be: P0S3-8-12-00
2. You have configured authenticate register, but there is no
username/password config on your phone:
I agree. I did my very first attempt on 14th of January this year and
was very upset that i got no more date for the v2-lab. First i wanted to
quit studying for this year, but then i made a v3-attempt in August in
Brussels, with not much preparation due to vacation time and a lot of
work to
Ben Ng has confirmed some things on Networkers.
These confirmations are on the IPExpert Blog.
My hint: Do not believe everything.
Aamir Panjwani wrote:
I was watching gateway video in new voice VOD today, Mark Snow mentioned
that Ben Ng at networkers has publicly said that new voice lab will
I have the same issue every time i do the CUE-setup. At this stage i
have never made any QOS config.
Aamir Panjwani wrote:
It seems to happen when you have QOS setup on one of the PVC’s. remove
all the QOS config and try again
*From:* ccie_voice-boun...@onlinestudylist.com
I got my report on next business day at 6pm.
Kevin Damisch wrote:
When you take a voice lab, the proctor grades it either the same day or next
day at the same site using the seat you sat in and does test calls. On my
first voice attempt, I got my results the next day around 2:00pm. On my
Do you really see the 7 digit number or the 7digit number with the
leading 9? (I am asking because i am searching for an answer how i can
configure the system so that i only see the 7D without the leading nine)
Kumar, Narinder wrote:
Yes I do have separate TEHO translation pattern. But the
I think the INITIAL-configuration of the PSTN-router is wrong.
I have changed it in my home-lab:
voice translation-rule 1
rule 2 /^212\(3942123\)$/ /\1/ type any subscriber plan any isdn
With this config the debug isdn q931 looks like the output on page 63.
regards
Philipp
R Sam wrote:
Hello!
When i call the AA from the PSTN phone and dial 5002, the AA transfers
the call to HQ Phone 2, but on this phone the calling party number is
+1+12123942123, which is obviously wrong. If i call 5002 directly from
the PSTN phone, the calling party number is like it should be: 3942123
, Phil G pgciscov...@gmx.net
mailto:pgciscov...@gmx.net wrote:
Some minor changes in the PSTN-Router, there should be Vol2 INITIAL
configs available in your download section. Otherwise: contact support.
Nara Shikamaru wrote:
Has anyone worked through the 3 mock labs yet
Yeah, that would be cool!
Actually i print the workbook and then manualy xerox 4 pages to one (so
2 pages on each side).
Unfortunately i printed the LAB 5 WB as it had only section 5A and 5B,
and since they added section 5C i cannot print those pages because of
the printing restriction. Maybe
I took my first v3-lab attempt in Brussels two weeks ago, i was the only
voice-candidate and all tracks in sum we were 8 (with 2 women).
Brett Saling wrote:
v3 is not as popular as of yet. I was the only candidate a week ago in
SJ so I made sure the ringer volume was low for the RS folks :)
Some minor changes in the PSTN-Router, there should be Vol2 INITIAL
configs available in your download section. Otherwise: contact support.
Nara Shikamaru wrote:
Has anyone worked through the 3 mock labs yet? I'm preparing to focus
on them for the next few months and am preparing the home
Hello!
Are there any initial configs for the VOL 2 Labs released? I am working
on Vol2/Lab3 and have loaded the PSTN-RTR-config of Vol1 Lab13, but this
is obviously not the right one.
regards,
Philipp
___
For more information regarding industry
Week Lab experience, but
check with support.
-Original Message-
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Phil G
Sent: Thursday, August 27, 2009 9:40 AM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] Vol2
Hello!
When is a line Remote in use, so that this text it is shown on the
phone? I just thought it should appear on shared lines, when all lines
have max calls = 1 and a call is active, but thats not true.
Regards
Philipp
___
For more information
Phil G wrote:
Hi again!
I wanted to start with UCCX-lab. I did the server setup and now i want
to add a application, but i cannot see any scripts. When i click at
Applications/Script Managment i get the following message:
Script Management
Aamir Panjwani wrote:
Yes I had the exact same issue a while ago, just remove the application
sip.app command from under voice register pool and it should work fine.
Not sure what's the purpose of this command though
Hi!
Yeah, solved the problem.
Thank you very much.
Hello!
I have a problem with SIP SRST. I have configured SIP SRST like in the
PG, but it is not working. The SIP-phone is registering with SRST-Router
but as soon as i dial a number, after the first digit the call ends and
the router shows the following error message:
*Aug 11 09:02:20.634:
Hi again!
I wanted to start with UCCX-lab. I did the server setup and now i want
to add a application, but i cannot see any scripts. When i click at
Applications/Script Managment i get the following message:
Script Management
Kevin Damisch wrote:
As
an example, by going to:
http://www.cisco.com/cisco/web/psa/default.html?mode=prod
Then
choose:
Products
Voice
and Unified Communications
IP
Telephony
Call
Control
Cisco
Unified Communications Manager Express
You
are then at
Jonathan Charles wrote:
!
voice register dn 1
number 3005
call-forward b2bua busy 3100
call-forward b2bua mailbox 3005
call-forward b2bua noan 3100 timeout 12
allow watch
name Phone1
voice register pool 1
id mac 0016.9DEF.16E5
type 7961
number 1 dn 1
template 1
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