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CCIE Voice Volume II Mock Lab 1:: Tasks: Difficulty Rating (1:lowest, 10:highest) : 8 Lab Overview

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0% found this document useful (0 votes)
100 views18 pages

CCIE Voice Volume II Mock Lab 1:: Tasks: Difficulty Rating (1:lowest, 10:highest) : 8 Lab Overview

CCIE,Voice,INE,Workbook

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MagicianRecon
Copyright
© © All Rights Reserved
We take content rights seriously. If you suspect this is your content, claim it here.
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INE CCIE Voice Volume II

Version 3.5

Mock Lab 1

CCIE Voice Volume II


Mock Lab 1 :: Tasks
Difficulty Rating (1:Lowest, 10:Highest): 8
Lab Overview:
The following scenario is a practice lab exam designed to test your skills at configuring Cisco
Unified Communications devices. Specifically, this scenario is designed to assist you in your
preparation Ciscos CCIE Voice Lab exam. However, remember that in addition to being
designed as a simulation of the actual CCIE lab exam, this practice exam should be used as
a learning tool. Instead of rushing through the lab, in order to complete all of the
configuration steps, benefit yourself by taking the time to research the networking technology
in question in order to gain a much deeper understanding of the principles behind its
operation. Also, it does you very little good to simply look at the answer video(s), as it doesnt
challenge and stretch your mind. Remember that your mind is a muscle, and the more you
stretch and challenge it, the more you will get out of it in return no different than lifting
weights will do you far more good than simply lifting a remote control.

Lab Instructions:
Prior to starting this lab, ensure that you have both read through the Voice Rack Rental
Guide before beginning this lab, so that you are fully aware of how all equipment works and
is accessed, and that you have loaded both the initial router/switch configs from INEs
members site, and that you have manually loaded the CUCM initial configs as described
below.

Note
- To load your router and switch configs, first log into your INE.com Members account, then
navigate to the "Rack Rental" tab, and click on "Control Panel" >> "Click here to choose a
configuration to be loaded on your Voice Rack", then choose the selection near the bottom
labeled CCIE Voice Workbook Volume II Lab 1 Initial Configs. This will only load
your router and switch configs (pstn, r1, r2, r3, sw1, sw2).
- To load your CUCM server initial configs, please log into the CUCM Admin WebUI, and use
the Bulk Administration tool to upload import the " IEVO_WB2_Lab1_CUCM_Startup.tar"
file that was included with your Volume II Lab 2 workbook downloads. Here are two brief
demonstration videos on how to successfully accomplish this upload and import into CUCM:
Importing or Exporting Configurations in CSV file from CUCM (15 mins).

Copyright 2011 Internetwork Expert

www.INE.com
-1-

INE CCIE Voice Volume II

Version 3.5

Mock Lab 1

Grading:
This practice lab consists of various sections and tasks totally 100 points. A score of 80
points is required to pass the exam. Any given task must work 100% with the requirements
given in order to be awarded points for that task. No partial credit is given within any task. If a
task has multiple possible solutions, attempt to choose the solution that best meets the
requirements.

Point Values:
The point values for each aggregate section are as follows:
Section
Network Infrastructure
CUCM Server and Phone Basics
CUCM Media Resources
CUCM Features
Gateways and Trunks
Dial Plan
Mobility
High Availability
Quality of Service
Messaging
Presence
Contact Center
TOTAL

Point Value
6
13
12
6
9
15
9
6
12
6
3
3
100

If you have any questions related to the scenario solutions, please post a comment on our
INE CCIE support forum.

GOOD LUCK!

Copyright 2011 Internetwork Expert

www.INE.com
-2-

INE CCIE Voice Volume II

Version 3.5

Mock Lab 1

Network Infrastructure
1.1

Configure the CorpHQ router to be the NTP master clock for the network all phones
and routers should ultimately be kept in sync with the CorpHQ router, and every router
should retain the proper time after being rebooted but before NTP fully syncs up
including the CorpHQ router
o All Devices at the CorpHQ site should use Pacific Time Zone (GMT -8) and
should follow Daylight Savings Time
o All Devices at the Branch1 site should use Central Time Zone (GMT -6) and
should follow Daylight Savings Time
o All Devices at the Branch2 site should use Central European Time Zone (GMT
+1) and should follow Daylight Savings Time
3pts

1.2

Provision the CorpHQ router to distribute IP addressing and options via DHCP for all
of your phones, and begin handing out the host IP of .100 for all subnets
o Subnet for CorpHQ Phones is 177.1.11.0/24
o Subnet for Branch1 Phones is 177.2.11.0/24
o Subnet for Branch2 Phones is 177.3.11.0/24
3pts

CUCM Server and Phone Basics


1.3

Create three Date/Time Groups:


o One group is to be used by all CorpHQ devices and should be in the Pacific
Standard Time zone, and allow for Daylight Savings Time
o One group is to be used by all Branch1 devices and should be in the Central
Standard Time zone, and allow for Daylight Savings Time
o One group is to be used by all Branch2 devices and should be in the Central
Europe Standard Time zone, and allow for Daylight Savings Time
3pts

Copyright 2011 Internetwork Expert

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-3-

INE CCIE Voice Volume II

Version 3.5

Mock Lab 1

1.4

Register all IP phones (except the PSTN phone) to the CUCM server, ensuring that
the CUCM Subscriber is the primary server registered to, and ensure the following
stipulations and configurations are met:
o NOTE: This lab may require you to upgrade a phone with new firmware,
depending on where the last candidate in the rack left it (leave plenty of time for
that phone to complete its firmware upgrade if you have to perform one)
o CorpHQ Phone1 should use DN 1001 and use SCCP firmware
o CorpHQ Phone2 should use DN 1002 and use SIP firmware
o Branch1 Phone1 should use DN 2001 and use SCCP firmware
o Branch2 Phone1 should use DN 3001 and use SCCP firmware
o Branch2 Phone2 should use DN 3002 and use SCCP firmware
o Use the G711 codec within, and the G729 codec between each site for normal
calls
3pts

1.5

Explore the default settings of the "Hub_None" Location, but do not modify anything.
Use this Location for the CorpHQ site
Create two additional Locations to be later assigned to all Branch1 and Branch2
devices respectively. Name them intuitively following the format of the "L_Branch1"
and "L_Branch2"
Allow for maximum total of 8 G.729 calls between the CorpHQ (Hub_None) and
Branch1 sites
Allow for maximum total of 14 G.729 calls between the CorpHQ (Hub_None) and
Branch2 sites
You are not permitted to put any BW values in any of these, or any other Locations in
the CUCM server
4pts

1.6

Ensure that all IP Phones show their PSTN DID number in the top right of their display
o CorpHQ Phones should show this number: 206501100X
o Branch1 Phones should show this number: 512602200X
o Branch2 Phones should show this number: 020703300X
Associate existing users as needed to their corresponding phones:
o CorpHQ Phone1 belongs to Jack Shepherd (userid: jshepherd)
o CorpHQ Phone2 belongs to Hugo Reyes (userid: hreyes)
o Branch1 Phone1 belongs to Benjamin Linus (userid: blinus)
o Branch2 Phone1 belongs to Desmond Hume (userid: dhume)
o Branch2 Phone2 belongs to James Ford (userid: jford)
3pts

Copyright 2011 Internetwork Expert

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-4-

INE CCIE Voice Volume II

Version 3.5

Mock Lab 1

CUCM Media Resources


1.7

If either Ben Linus (x2001) Jack Shepherd (x1001) calls the DN of 2555, they should
hear the Annunciator playing a message telling them something to the effect of the
fact that they are not authorized for their line
3pts

1.8

If Jack sets up an ad-hoc conference between himself, Hugo and Benjamin, and then
James calls either Hugo or Benjamin, the latter two should have the ability to add
James into the existing conference
Do not allow any software conference bridging resources to ever be able to be used
you are also not permitted to put the CUCM Pub or Sub Software Conference Bridge
in any MRGs for isolation
3pts

1.9

Provision multicast MoH for the CorpHQ site phones and PSTN gateway using the
G.711 codec
Provision multicast MoH for the Branch1 site phones and PSTN gateway using the
G.729 codec
Provision multicast MoH for the Branch2 site phones and PSTN gateway while
ensuring that no WAN bandwidth is utilized
3pts

1.10

Assign any media resources so that devices at each site have access in this order:
1) Any necessary hardware resources use own local first
2) Only necessary software-based resources Pub first
3) Only necessary software-based resources Sub next
3pts

Copyright 2011 Internetwork Expert

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-5-

INE CCIE Voice Volume II

Version 3.5

Mock Lab 1

CUCM Features
1.11

Setup a call pickup group for the CorpHQ site, and assign both phones' primary DN's
to the group
Ensure that if either one of the CorpHQ phones are ringing, that after 3 rings the other
phone in the group hears "Are You There?" and simultaneously sees both the Calling
and Called party along the bottom of their phone display (e.g. If Desmond at 3001
were calling Hugo at 1002, then Jack Shepherd at 1001 would see: "Desmond Hume > Hugo Reyes")
Allow any person picking up a call to only have to press a single key to immediately
answer the other person's ringing phone
Allow Jack to immediately see if Hugo's phone is ringing
3pts

1.12

Create a DN on a new line on both Desmond Hume and James Fords phones
o Assign the new Line the DN of 3005
Allow Desmond the ability to barge into any active call on that line, undetected by the
third party IP phones display, simply by Desmond pressing that line button
Allow James the ability to barge into any active call on that line simply by pressing that
line button, however if the third party on the existing call with Desmond is using an IP
phone, they should be able to tell that the call has been barged in on
Allow Desmond the ability to "lock" the line so that James cannot barge in on any inprogress, or on-hold existing call
James should not have the ability to "lock" the line for any calls
No CUCM Conference Bridge is be allowed to be used
3pts

Copyright 2011 Internetwork Expert

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-6-

INE CCIE Voice Volume II

Version 3.5

Mock Lab 1

Gateways and Trunks


1.13

Provision the CorpHQ router (R1) as a gateway to the PSTN for CUCM using the
following specifications:
L1::T1::Linecoding::B8ZS
L1::T1::Framing::ESF
L1::T1::Time Source::PSTN
L2/3::ISDN::PRI Switch-Type::5ESS
L3::PRI::BChannels::3
L3::PRI::CNAM::Supported
L3::VoIP::SIP sourced from Loop0
o Ensure that if for some reason the Sub server doesnt respond within 1 second
of a setup failure, that call setup is retried quickly to the Pub server
o Ensure that DTMF works properly over the link back to CUCM, but do not
configure any type that requires the use of a MTP
3pts

1.14

Provision the Branch1 router (R2) as a gateway to the PSTN for CUCM using the
following specifications:
L1::T1::Linecoding::B8ZS
L1::T1::Framing::ESF
L1::T1::Time Source::PSTN
L2/3::ISDN::PRI Switch-Type::NI2
L3::PRI::BChannels::3
L3::PRI::CNAM::Supported
L3::VoIP::H.323 sourced from Loop0
o Ensure that if a WAN failure were to occur, all active calls to the PSTN would
remain up with RTP media
o Ensure that if for some reason the CUCM Sub server isn't responding fast
enough, that call setup is retried quickly to the CUCM Pub (within 2 seconds of
CUCM Sub setup failure)
o Ensure that DTMF works properly over the link back to CUCM using a method
that carries not only the tone's frequency, but also the tone's duration
o Provision bearer channels so that they are chosen for outbound calls in a
Bottom-Up fashion, and so that if one channel happens to not be available from
the PSTN, that the router will allow them to be negotiated rather than chosen in
an absolute fashion
3pts

Copyright 2011 Internetwork Expert

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-7-

INE CCIE Voice Volume II

Version 3.5

Mock Lab 1

1.15

Provision the Branch2 router (R3) as a gateway to the PSTN for CUCM using the
following specifications:
L1::E1::Linecoding::HDB3
L1::E1::Framing::CRC4
L1::E1::Time Source::PSTN
L2/3::ISDN::PRI Switch-Type::EURO
L3::PRI::BChannels::3
L3::PRI::CNAM::Supported
L3::VoIP::MGCP sourced from Loop0
3pts

Copyright 2011 Internetwork Expert

www.INE.com
-8-

INE CCIE Voice Volume II

Version 3.5

Mock Lab 1

Dial Plan
Route Plan Description and Overall Guidelines

Phones in US locations use a PSTN access code of 9 to reach an outside line.


Phones at the NL location use a PSTN access code of 0 to reach an outside line.
Outside Dial Tone should be provided as soon as the prefixed PSTN access code
has been dialed for each site respectively. In all cases, users should be able to dial
this prefix before dialing their PSTN number.
Both the CorpHQ and Branch1 sites are located in the US and have the country code
of 1. CorpHQ is in the city of Seattle, WA and Branch1 is in Austin, TX.
Branch2 site is located in the Netherlands (NL) and has the country code of 31.
Branch2 is in the city of Amsterdam.
Users should be able to dial 911 from the US locations (CorpHQ and Branch1), and
112 from the Netherlands location (Branch2), to reach emergency services
Make sure the PSTN access code of 9 for US and "0" for NL are each stripped for all
calls before reaching the PSTN carrier (except for the 9 in 911)
Local area (Subscriber) PSTN calls are placed from CorpHQ, Branch1 & Branch2
locations all by dialing 10-digit numbers
Long-distance (National) PSTN calls placed from the CorpHQ and Branch1 locations
are done so by dialing 11-digit numbers with the first number being the PSTN LD
access code 1, and at the Branch2 location by dialing 10 digit numbers with the first
number being the PSTN National access code of 0 (this is in addition to the PSTN
Trunk Code of 0 for Branch2)
International calls from the CorpHQ and Branch1 site should be prefixed using the
PSTN International access code of 011 along with the country code and variable
length number for the site/number you are trying to reach
International calls from the Branch2 site should be prefixed using the PSTN
International access code of 00 along with the country code and variable length
number for the site/number you are trying to reach (this is in addition to the PSTN
access code of 0 for Branch2)
You should allow for any phone users to be able to terminate inter-digit timeout on
international number dialing using the # sign, in order to place the call immediately

Copyright 2011 Internetwork Expert

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-9-

INE CCIE Voice Volume II

Version 3.5

Mock Lab 1

How To Dial Into Each Site's GW From the PSTN Phone


Dialing into the CorpHQ GW:
For Calling Party Type to show as "Subscriber" >> Select PSTN Line 1, and dial
2065011XXX

For Calling Party Type to show as "National" >> Select PSTN Line 2, and dial
2065011XXX

For Calling Party Type to show as "International" >> Select PSTN Line 3, and dial
2065011XXX

For Calling Party Type to show as "Unknown" >> Select PSTN Line 6, and dial
2065011XXX

For Calling Party Type to show as "Private" >> Select Any PSTN Line, and dial
*672065011XXX

Dialing into the Branch1 GW:


For Calling Party Type to show as "Subscriber" >> Select PSTN Line 2, and dial
5126022XXX

For Calling Party Type to show as "National" >> Select PSTN Line 1, and dial
5126022XXX

For Calling Party Type to show as "International" >> Select PSTN Line 3, and dial
5126022XXX

For Calling Party Type to show as "Unknown" >> Select PSTN Line 6, and dial
5126022XXX

For Calling Party Type to show as "Private" >> Select Any PSTN Line, and dial
*675126022XXX

Dialing into the Branch2 GW:


For Calling Party Type to show as "Subscriber" >> Select PSTN Line 3, and dial
0207033XXX

For Calling Party Type to show as "National" >> Select PSTN Line 4, and dial
0207033XXX

For Calling Party Type to show as "International" >> Select PSTN Line 1, and dial
0207033XXX

For Calling Party Type to show as "Unknown" >> Select PSTN Line 6, and dial
0207033XXX

For Calling Party Type to show as "Private" >> Select Any PSTN Line, and dial
*670207033XXX

Copyright 2011 Internetwork Expert

www.INE.com
- 10 -

INE CCIE Voice Volume II

Version 3.5

Mock Lab 1

Inbound Calling from the PSTN


1.16

Globalize all calls coming inbound from the PSTN to gateways at all sites using the
proper Full E.164 numbering format (including +) for each site
You are responsible for determining how calls are ringing into each site
with regards to Numbering Type and Digit Length (note: not every call rings-in in the
same fashion, so place multiple calls into each site and make notes to determine this)
The preceding 0 coming into the Branch2 Amsterdam site from the PSTN should not
be included in the globalized format of the number - drop this "0" before doing
anything else to the number
The new Globalized Calling number should display at every phone when the user at
any phone looks at the Call History
3pts

1.17

Localize calls inbound from the PSTN as they arrive at the Branch1 phones
Local (Subscriber) Calls:
o During inbound alerting, phones should display calling party numbers that are
local to each site as 10 digits
Long Distance (National) Calls:
o During inbound alerting, phones should display calling party numbers as 11
digits if that call is from the same country, but a different geographic/area code
International Calls:
o During inbound alerting, phones should display calling party numbers with all
digits, including the country code and the preceding PSTN international access
code of "011", if that call is from a different country
3pts

Copyright 2011 Internetwork Expert

www.INE.com
- 11 -

INE CCIE Voice Volume II

Version 3.5

Mock Lab 1

Outbound Calling to the PSTN


1.18

Create only one set of PSTN Patterns for each country (US and NL) based on the
information in the two tables below
Ensure these PSTN Patterns globalize every Called Party Number to a proper Full
E.164 number format before the call ever reaches a Route List in CUCM
o NOTE: Emergency numbers need to be globalized to match the global route
patterns - but will not be in "proper" E.164 format - this is OK
Ensure that the Calling and Called Party number are sent to the PSTN with the proper
Types as listed in the tables below
Dial Peers 10-13 have already been configured for you on your CorpHQ and Branch1
routers that you must use for PSTN calls. You may add, delete and/or change almost
any parameter(s) on these dial-peers, but you may not change any of the
destination-pattern commands
Any user dialing a PSTN number from an IP phone should not see the PSTN access
code of either 9 or 0 once the call has properly routed out to the PSTN
3pts

Table 1: Dialing from within US sites


Call Type
Emergency
Local
National/LD
International

PSTN Pattern
911
[2-9]XX[2-9]XXXXXX
1[2-9]XX[2-9]XXXXXX
011 + Variable length

Calling # Format
10 ANI Digits
10 ANI Digits
10 ANI Digits
CC+ 10 ANI Digits

Calling Party Type


Unknown
Subscriber
National
International

Table 2: Dialing from within NL sites


Call Type
Emergency
Local
National/LD
International

PSTN Pattern
112
020XXXXXXX
0[16]XXXXXXXX
00 + Variable length

Calling # Format
10 ANI Digits
10 ANI Digits
10 ANI Digits
CC+ 9 ANI Digits

Copyright 2011 Internetwork Expert

Calling Party Type


Unknown
Subscriber
National
International

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- 12 -

INE CCIE Voice Volume II

Version 3.5

Mock Lab 1

1.19

Ensure that any user at any IP Phone who presses either "Missed Calls" or
"Received Calls", sees any PSTN number in Full E.164 + Globalized format, and can
simply press the "Dial" softkey without the need to press the EditDial softkey
You must maintain the proper Calling Type described by the previous task
3pts

1.20

Provision a support hunt group for both Branch2 phones


Allow they both to have the ability to log in or out of the hunt group
Both phones should take turns ringing back and forth from each other whenever
anyone dials the DN of 3500, and this number should also be reachable by dialing in
from the PSTN
3pts

Copyright 2011 Internetwork Expert

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- 13 -

INE CCIE Voice Volume II

Version 3.5

Mock Lab 1

Mobility
1.21

Configure CUCM such that when James "Sawyer" Fords Branch2 Phone 2 at DN
3002 is called, that his mobile is rung after 1 ring
From the information below, calculate the time it takes his mobile phone to ring to VM,
and prevent this mechanism from ringing into his mobile voicemail box
o Sawyers Mobile #: +31 6 7035 7575 -Forwards to VM after 4 rings
Allow Sawyer to be able to transfer calls from his mobile phones back to his desk
phone, and also from his desk phone back to his mobile phone with a single button
Also allow Sawyer to be able to login to his CCMUser page and setup or change any
of his Mobile Connect remote phones, however you may not allow him to add any
more remote destinations from his user page
3pts

1.22

Ensure that when Mobile Connect is invoked, that the call being placed out to the
PSTN tries to ring out the gateway that is local to the Sawyers remote PSTN phone
Ensure that when Sawyer calls in to CorpHQ site from his mobile, that his desk DN
number and name show up on either Jack or Hugos IP phone display
3pts

1.23

When calls show up at Sawyers mobile phone, the calling number should be
displayed as follows:
o Calls coming from other IP phones internally registered to the same CUCM
cluster should show up as their 4digit desk DN
o Calls coming from anywhere within the NL should display as 10 digits
beginning with a 0 for the geographic/area code (omit the country code of 31)
o Calls coming from anywhere outside the NL should show the full number
including country code and preceding +
3pts

Copyright 2011 Internetwork Expert

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- 14 -

INE CCIE Voice Volume II

Version 3.5

Mock Lab 1

High Availability
1.24

Provision Branch2 R3 as a SCCP server for CME as SRST phone fallback with the
following requirements:
o Provision the SCCP server IP address as the same as the Loopback0 interface
and ensure that this is the only IP address that phones can register with
o Allow negotiation of the CODECs G711alaw, G711ulaw & G729 with G711alaw
preferred
o Calls to the PSTN should function in exactly the same way (you do not have to
worry about inbound or outbound globalization and localization, however
Calling Party Types should use same format requested in the DialPlan section)
o Ensure that the phone looks and behaves just as it would when registered to
CUCM (some features may not work, configure all that you can, and ask
specific questions for clarification on things you suspect may not work when in
fallback mode)
4pts

1.25

No matter what network situation may occur, all calls from any phone to any other 4
digit extension must continue to work seamlessly
All IP phones at all sites should see 4 digit ANI when receiving a call from any other IP
phone at any site at any time
You are not permitted to change any phones top-right display from previous task
requirements
2pts

Copyright 2011 Internetwork Expert

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- 15 -

INE CCIE Voice Volume II

Version 3.5

Mock Lab 1

Quality of Service
1.26

Create three classes on both the CorpHQ and Branch1 routers:


o Classify, via trust, all traffic with a DSCP PHB of Expedited Forwarding
o Explicitly classify, via NBAR, all voice-based signaling traffic to include:
H.323 both Slow & Fast Start and RAS
SIP
MGCP
JTAPI
SCCP
All IP traffic marked with EF should go into a class labeled VOICE-RTP
All voice-based signaling traffic should go into a class labeled VOICE-SIG
o Mark this traffic with the DSCP PHB of Class Selector 3
3pts

1.27

Configure LLQ on both the CorpHQ and Branch1 routers Frame-Relay PVC to one
another
Create a strict priority queue for all traffic labeled with the DSCP PHB of EF
Create an assured bandwidth queue for all traffic labeled with the DSCP PHB of CS3
To each queue assign a bandwidth value for 8 G.729 calls placed over the link, for all
media and signaling streams respectively
o You must use the "percent" argument after your bandwidth command in your
configuration for each of these queues
3pts

1.28

Shape all traffic on both sides of this CorpHQ-Branch1 FR PVC WAN link
The carrier guaranteed CIR speed is 768kbps
Use a Hierarchical QoS Policy for this task while providing the best options for voice
traffic
3pts

1.29

Utilize an LFI mechanism best suited for "Frame-Relay to Frame-Relay" type WAN
links on the CorpHQ-Branch1 FR PVC
3pts

Copyright 2011 Internetwork Expert

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- 16 -

INE CCIE Voice Volume II

Version 3.5

Mock Lab 1

Messaging
1.30

Create mailboxes for all users with phones at both CorpHQ and Branch1 on the Unity
Connection server
You are not permitted to use the Unity Connection interface to create or import the
users needed for any mailboxes
Make all passwords 55555, but do not touch any user web page in UC to
accomplish this task
MWI must work properly for these phones when a message is left or listened to
3pts

1.31

Provision the Unity Express module on Branch2 (R3) with the IP address of
177.1.250.254
You may not change the IP address or Subnet Mask on interface Loopback1
Create mailboxes that work properly at all times for the users of phones at Branch2
You may only import any users needed to create mailboxes from the CUCM
Make all passwords 55555, but do not touch any user web page in CUE to
accomplish this task
MWI must work properly at all times for these phones when a message is left or
listened to
3pts

Copyright 2011 Internetwork Expert

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- 17 -

INE CCIE Voice Volume II

Version 3.5

Mock Lab 1

Presence
1.32

Configure Personal Communicator for Hugo Reyes at DN 1002 using the client on
your XP Utility machine
Assign Jack and Bens userids to Hugos CUPC client so that he sees them with their
proper name and any state changes that may occur when they get CUPC clients
Ensure that Hugos VMs show up on his CUPC client display
3pts

Contact Center
1.33

Create two CSQs called Oceanic and Others


CSQ Oceanic should be triggered any time someone calls any unused DID number
at either the CorpHQ or Branch2 sites (DID ranges of 2065011XXX and 0207033XXX)
CSQ Others should be triggered any time someone calls any unused DID number at
the Branch1 site (DID ranges of 5126022XXX)
You may not use any Translation Patterns to accomplish this task
Take steps to ensure that the following occur:
o For the Oceanic CSQ, only Jack and Hugo should be permitted to be call
taking agents, however Hugo should always be elected to receive calls first if
he is available
o For the Others CSQ, only Ben and Hugo should be permitted to be call taking
agents, however Ben should always be elected to receive calls first if he is
available
3pts

Copyright 2011 Internetwork Expert

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- 18 -

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