CCIE Voice Volume II Mock Lab 5:: Tasks: Difficulty Rating (1:lowest, 10:highest) : 7 Lab Overview

INE CCIE Voice Volume II
Version 3.5
Mock Lab 5
CCIE Voice Volume II

Mock Lab 5 :: Tasks
Difficulty Rating (1:Lowest, 10:Highest): 7
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 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 the initial router/switch configs from INEs Members
site.
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 5 Initial Configs. This will only load
your router and switch configs (pstn, r1, r2, r3, sw1, sw2).
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.
Copyright 2011 Internetwork Expert
www.INE.com
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Version 3.5
Mock Lab 5
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
Messaging
Contact Center
TOTAL
Point Value
4
10
9
9
9
23
8
9
12
7
100
If you have any questions related to the scenario solutions, please post a comment on our
INE CCIE support forum.
GOOD LUCK!
www.INE.com
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Version 3.5
Mock Lab 5
Network Infrastructure
5.1
Configure the CorpHQ 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
2 pts
5.2
Provision all sites phones with the following infrastructure configuration

o Ensure the phone-connected ports using 802.1Q trunking to provide a
Computer Data VLAN of 12, and a Phone Voice VLAN of 11
o Do not use any sort of specific trunking configuration commands
o Ensure that phones are talking on the network as fast as possible
Provision DHCP for each sites respective IP phones as follows:
o Subnet for CorpHQ Phones is 177.1.11.0/24 and should come from UCM
o Subnet for Branch1 Phones is 177.2.11.0/24 and should come from UCM
o Subnet for Branch2 Phones is 177.3.11.0/24 and should come from R3
Ensure that all phones receive both CUCM servers as TFTP servers
2pts
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Version 3.5
Mock Lab 5
CUCM Server and Phone Basics

5.3
Register all IP phones (except the PSTN phone) to the CUCM server, ensuring that
the CUCM Publisher is the primary server registered to and that the Subscriber can
always take over as a CPE should failover occur, 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 (allow for
approximately 20 mins for all phones to complete their firmware upgrades)
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 All phones should be allowed to place all calls
o When any phones place calls within any given site ensure that they are able to
use the G.722 codec; when any phones place calls between any two sites
ensure that they should use the G.729 codec
3pts
5.4
For voice and video calls coming from or going to Branch1, use RSVP to configure the
following:
o Allow for maximum of 20 G.729 voice calls (this is also the maximum BW for
the interface for all CAC, including video)
o Within that maximum BW stated above, allow for maximum of 2 video calls at
128kbps each
o In your routers RSVP configuration, ensure that each voice call does not
exceed the G.729 codec and that each video call does not exceed 128kbps
4pts
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Version 3.5
Mock Lab 5
5.5
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 users to their corresponding IP Phones using the below information:
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)
Ensure all IP Phones' primary line displays their User's First and Last Name along
with their extension (DN) in the following format:
o FName LName xYYYY (where YYYY is the 4 digit extension)
o (e.g. Jack Shepherd x1001)
Ensure two IP phones setting up a call (one phone dialing and the other phone
ringing) both display the full name (only FName LName) of the person that is either
calling or being called respectively
Ensure all names and telephone numbers show up properly in the Corporate Directory
for all IP phones
3pts
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Version 3.5
Mock Lab 5
CUCM Media Resources

5.6
Ensure that the RTP media stream for all calls to or from the CorpHQ Phone1 (x1001)
go through an MTP on the CorpHQ R1 at all times
You may not use any DSP resources to accomplish this task
3pts
5.7
Setup MTP, transcoding and conferencing where applicable and as necessary

Do not ever allow the usage of CUCM Software MTP or Conference Bridge resources
3pts
5.8
Provision unicast MoH for the CorpHQ site phones and PSTN gateway using the
G.711 codec
o Use the Publisher as the primary MoH server, and ensure the Subscriber will
act as a backup if necessary
Provision multicast MoH for the Branch1 site phones PSTN gateway, both using the
G.729 codec
o Use the Subscriber as the primary MoH server, and ensure the Publisher will
Provision multicast MoH for the Branch2 site phones and PSTN gateway both using
the G.729 codec
o Use the Subscriber as the primary MoH server, and ensure the Publisher will
3pts
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Version 3.5
Mock Lab 5
CUCM Features
5.9
Enable Desmond Humes (x3001) and James Ford's (x3002) phones so that they are
able to distribute firmware images to each other via a peer-to-peer protocol to reduce
WAN bandwidth consumption and CUCM TFTP server strain
3pts
5.10
Enable Auto-Answer for Hugo Reyes phone (x1002) with speakerphone

Only allow it to auto-answer the call after 2 full rings
Do not allow auto-answer to function if there is a call already on hold
3pts
5.11
SIP phone users are reporting that their phones are randomly restarting, change
necessary settings so that their phones attempt re-registration in half the time as is the
default
3pts
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Version 3.5
Mock Lab 5
Gateways and Trunks

5.12
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::Q.SIG
L3::PRI::BChannels::3
L3::PRI::CNAM::Supported
L3::VoIP::MGCP sourced from Voice VLAN
3pts
5.13
Provision the Branch1 router (R2) as a gateway to the PSTN for CUCM using the
L1::T1::Linecoding::B8ZS
L1::T1::Framing::ESF
L1::T1::Time Source::PSTN
L2/3::ISDN::PRI Switch-Type::NI
L3::VoIP::MGCP sourced from Loopback0
3pts
5.14
Provision the Branch2 router (R3) as a gateway to the PSTN for CUCM using the
L1::E1::Linecoding::HDB3
L1::E1::Framing::CRC4
L1::E1::Time Source::PSTN
L2/3::ISDN::PRI Switch-Type::NET5
L3::VoIP::SIP sourced from Loopback1
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 2/10 of a
second of a CUCM Sub setup failure)
3pts
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Version 3.5
Mock Lab 5
Dial Plan
Route Plan Description and Overall Guidelines
Phones in US locations use a prefix code of 9 to access a PSTN trunk. Phones at

the NL location use a prefix code of 0 to access a PSTN trunk. Outside dial tone
should be provided as soon as the prefixed PSTN Trunk Code has been dialed for
each site respectively
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 which has a city code (area code) of 206
and Branch1 is in Austin, TX which has a city code (area code) of 512.
Branch2 site is located in the Amsterdam, The Netherlands (NL) and has the country
code of 31 and the city code of 20.
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 Trunk Code of 9 for US and "0" for NL are each stripped for all
calls before going to PSTN (except for the 9 in 911)
Local area (Subscriber) PSTN calls are able to be placed from CorpHQ & Branch1
locations by dialing 10-digit numbers, and from at BR2 by dialing both 7-digit and 10digit 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, followed by the country code and variable length
number for the site/number you are trying to reach (this is in addition to the PSTN
Trunk Code of 0 for Branch2)
You should allow for any phone users to be able to terminate interdigit timeout on
international number dialing using the # sign, in order to place the call immediately
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Version 3.5
Mock Lab 5
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:

5126022XXX
5126022XXX
5126022XXX
5126022XXX
*675126022XXX
Dialing into the Branch2 GW:

0207033XXX
0207033XXX
0207033XXX
0207033XXX
*670207033XXX
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Version 3.5
Mock Lab 5
Inbound Calling from the PSTN

5.15
Globalize all calls coming inbound from the PSTN to gateways at all sites using the
proper Full E.164 numbering format (including preceding +) for each site
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 hardware IP phone when
the user at any phone looks at the Call History
4pts
5.16
Localize all calls inbound from the PSTN as they arrive at IP phones but only as
directed below
Local (Subscriber) Calls:
o During inbound alerting, IP Phones in the US sites should display calling party
numbers that are local to each site as 7 digits
o During inbound alerting, IP Phones in the NL site should display calling party
numbers that are local to that site as 10 digits
Long Distance (National) Calls:
numbers as 10 digits if that call is from the same country, but a different
geographic/area code
numbers as 10 digits if that call is from the same country, but a different
geographic/area code. This means that the calling party number needs to have
the national access code of "0" added back to the front of the geographic code
International Calls:
numbers with all digits, including the country code and + if that call is from a
different country
numbers with all digits, including the country code and + if that call is from a
different country
4pts
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Version 3.5
Mock Lab 5
Outbound Calling to the PSTN

5.17
Calls from CorpHQ phones to the PSTN should be routed out in the following manner:
o Do not globalize any calls at this site before routing to the PSTN
o Emergency Services calls dial 911
o Local calls should be a total of 10 digits, and only be allowed to begin with
a [2-9]
o National calls should be a total of 11 digits, including the 1 PSTN national
prefix, followed by 10 digits that begin with a [2-9]
o International calls could be any length, but should always begin with a
011 PSTN international prefix, followed by the country code and remaining
digits
o All calls dialed from phones at this site must only go out their local gateway
o All Emergency calls should have 10 calling party digits and type Unknown
o All Local calls should have 10 calling party digits and type Subscriber
o All National calls should have 10 calling party digits and type National
o All International calls should have 11 calling party digits and type Unknown
4pts
5.18
Calls from Branch1 phones to the PSTN should be routed out in the following manner:
o Do not globalize any calls at this site before routing to the PSTN
a [2-9]
digits
o Emergency calls dialed from phones at this site must only go out their local
gateway; Local calls must go out their local gateway and fall back to the
CorpHQ site; while all remaining calls must go out the CorpHQ gateway and fall
back to their own local gateway
o All Emergency calls should have 10 calling party digits and type Unknown
o All International calls should have 11 calling party digits and type International
4pts
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Version 3.5
Mock Lab 5
5.19
Calls from Branch2 phones to the PSTN should be routed out in the following manner:
o Globalize all calls at this site to +E.164 format before routing to the PSTN
a [1-8]
digits
o All calls dialed from phones at this site must only go out their local gateway
o All Emergency calls should have 10 calling party digits and type Subscriber
o All International calls should have 12 calling party digits and type International
o All calls must be globalized immediately once they are dialed to a true +E.164
format
4pts
5.20
Ensure that any user at any IP Phone at the CorpHQ and Branch2 sites who view
their Missed Calls can simply press the "Dial" softkey to return the call
SIP phones must work properly
You do not have to maintain the proper Calling and/or Called Digits and/or Types
described by the previous tasks
3pts
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Version 3.5
Mock Lab 5
Mobility
5.21
Configure CUCM such that when Ben Linus Branch1 Phone 1 at DN 2001 is called,
that his mobile phone is rung after 3 full rings at his desk phone
From the information below, calculate the time it takes those mobile and home phones
to ring to VM, and prevent this mechanism from ringing into any of their respective
voicemail boxes
o Bens Mobile #: +1 512 602 6262
-Forwards to VM after 4 rings
This phone should not be rung if the caller is CorpHQ Phone2
o Note: For testing purposes, these calls will both ring to the PSTN Phone Line
5, however they have different CFwdNoAn timeouts and simulated VM on
INE/GradedLabs Voice Racks, and will show the differing CallerID based on
the above requirements being met for Home and Mobile phones respectively
Allow Ben to be able to transfer calls from his mobile phone back to his desk phone,
and also from his desk phone back to his mobile phone with a single button
Also allow Ben to be able to login to his CCMUser page and setup, change or even
add any Mobile Connect remote phones
4pts
5.22
Allow Ben the ability to call into the company from his Mobile Phone (PSTN Line 5)
and make International calls back out to through the PSTN.
o He must be able to call into the DID of 512 602 2988 to trigger MVA
4pts
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Version 3.5
Mock Lab 5
High Availability
5.23
Provision Branch2 R3 as a SCCP server for CME as SRST phone fallback with the
following stipulations:
o Provision the SCCP server IP address as the same as the Loopback1 interface
o Normal calls to the PSTN should function the same way
You need not worry about inbound/outbound globalization/localization
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)
o Do not provision any upgrades to firmware for the fallen back phones
o You may not use any srst auto-provision commands, but instead you must
prebuild all necessary ephones and ephone-dns
You must also use the prebuilt dial-peers 101 and 102 to route calls
back to the CUCM when the WAN is up that is to say that you may not
make more specific or longer destination patterns or dial-peers to
route calls to CUCM normally
You must also ensure that proper call routing to the CUCM works
properly when the phones are not in fallback mode
o Ensure that when the CUCM is accessible again, that the phone register again
with it within 15 seconds or less
4pts
5.24
Provision Branch1 R2 as a SCCP server for SRST phone fallback with the following
stipulations:
o Normal calls to the PSTN should function the same way
You need not worry about inbound/outbound globalization/localization
2pts
5.25
No matter what network situation may occur (e.g. WAN down or low CAC bandwidth),
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
3pts
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Version 3.5
Mock Lab 5
Messaging
5.26
Integrate CUCM with Unity Connection using SCCP

Import both CorpHQ users via AXL and create mailboxes for them
Make all passwords 55555
MWI must work properly
3pts
5.27
Setup PhoneView for CorpHQ phones

3pts
5.28
Provision the Unity Express module on Branch2 (R3) with the IP address of
177.1.250.254, however you may not change the IP address or Subnet Mask on
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
MWI must work properly at all times
3pts
5.29
Setup VoiceView Express for Branch2 phones

3pts
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Version 3.5
Mock Lab 5
Troubleshooting
5.30
Setup a SIP Trunk from your CUCM cluster to an ITSP located at IP address
177.1.254.254
You must hardcode the use of the G.729 Annex B protocol
Send all calls destined for the country code of 44 (UK) across this SIP Trunk to the
ITSP you must send them with full E164+ dialing in this format:
o +44 xx xxxx xxxx
Troubleshoot why the calls are not working
Document your work by copying any relevant trace/debug information to a notepad
file, and save it for proctor review
In one or two sentences, at the top of your notepad file (above your traces/debugs),
write a brief summary of why calls to the 44 country code have failed
4pts
5.31
Leave RTMT running on your XP-Util desktop with all of the performance monitor
statistics (in numerical, not graph format) that you used to prove all your media
resources were utilizing the proper resources
3pts
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