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CCIE Voice Volume II Mock Lab 2:: Tasks: Difficulty Rating (1:lowest, 10:highest) : 9 Lab Overview
CCIE Voice Volume II Mock Lab 2:: Tasks: Difficulty Rating (1:lowest, 10:highest) : 9 Lab Overview
Version 3.5
Mock Lab 2
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 2 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_Lab2_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).
Grading:
Copyright 2011 Internetwork Expert
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Version 3.5
Mock Lab 2
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
TOTAL
Point Value
4
8
7
14
9
24
7
6
12
6
3
100
If you have any questions related to the scenario solutions, please post a comment on our
INE CCIE support forum.
GOOD LUCK!
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Version 3.5
Mock Lab 2
Network Infrastructure
2.1
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
2.2
Provision the CorpHQ and Branch1 site phones with the following infrastructure
configuration
o Voice VLAN for both sites = 11
o Data VLAN (to be pushed to phones PC Port) for both sites = 12
o Do not use any sort of trunking configuration for either sites switchports
o Ensure that phones are talking on the network as fast as possible
Provision each sites router to distribute IP addressing and options via DHCP for that
sites respective IP phones
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
o Your allowable hosts for each subnet are from .26 - .99 and from .110 - .254
Ensure that all phones receive the CUCM-Pub as their primary TFTP server and a
redundant CUCM-Sub as a secondary TFTP server
2pts
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Version 3.5
Mock Lab 2
2.4
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 When any phones at the CorpHQ and/or Branch1 site go to call each other
within a given site - they should use the G.711 CODEC; when they call each
other between sites, they should be using the G.729 CODEC
o Ensure that the G.722 voice codec is not able to be negotiated, however do not
disable it for any device
2pts
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Version 3.5
Mock Lab 2
2.5
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
2pts
2.6
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 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
2pts
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Version 3.5
Mock Lab 2
2.8
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 use any software conference bridging resources
Any IP Phones should be able to setup an ad-hoc conference, however if the person
who set's up the conference, leaves the call, then the conference bridge should
disintegrate for the remaining users
2pts
2.9
Use the Publisher as everyones primary MoH server, and the Subscriber as a backup
Provision multicast MoH for the CorpHQ site phones and PSTN gateway using the
G.711 codec
Provision unicast MoH for the Branch1 site phones and multicast MoH for the Branch1
PSTN gateway, both using the G.729 codec
o You must stream MMoH out to the Branch1 T1 PRI from the Loopback0
interface
Provision multicast MoH for the Branch2 site phones and PSTN gateway
o Stream all MoH traffic using the G.711 codec without any WAN bandwidth
o You must not stream MMoH out to the Branch2 E1 PRI from the Loopback0
interface
3pts
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Version 3.5
Mock Lab 2
CUCM Features
2.10
Provision Desmond Hume's phone for Line 1 (x3001) to go through to VM after 5 rings
or if he is on the phone, however he should still be able to conference in or transfer
calls and have up to 5 total active calls on his phone
o Allow Desmond to set his DND status, and make it such that if he is in DND
mode, that calls go through to VM without alerting him in any way
Provision two separate buttons on Jack Shepherds phone (x1001):
o Both buttons need to be useable when a call is ringing at his phone
o One should provide him the ability to press a single button to forward any
incoming call directly to his VM box
o One should provide him the ability to press a single button to forward any
incoming call directly to Hugo Reyes Line 1 (x1002)
Once the above button is pressed, then subsequent calls should also be
forwarded to Hugo (until Jack disables this button), however Jack wants
the ability to see that a call is alerting at his phone, but hear nothing
Hugo knows that calls will be forwarded to him, but is only comfortable
answering them if the calls come with the exact wording format of:
Forward <Calling Party> By Jack Shepherd
Also, if Hugo was too busy, or does not answer the call that was
forwarded by Jack in time, then this calling party should still reach Jacks
VM box, and not Hugos VM
4pts
2.11
Enable whisper intercom functionality between Jack Shepherd's (x1001) and Hugo
Reyes' phones (x1002) using the DNs of 1001 and 1002 respectively
Every other phone should be able to place a call to these phone's and trigger whisper
intercom
2pts
2.12
Ensure that any IP Phone user can park any call that is active on their phone
Ensure that phones anywhere are always able to park calls anywhere in the DN
range of 5500-5599, and they all must be in the <None> Partition
3pts
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Version 3.5
Mock Lab 2
2.13
Ensure that any IP Phone can choose to park a call by choosing the exact park DN,
rather than having one automatically assigned to them by CUCM
o They should be able to park the call anywhere in the range of 5600-5699
o Calls should be able to be retrieved by dialing *88<ParkDN>
2pts
2.14
Ensure that both SIP and SCCP phones can download and use custom ringtones and
phone desktop backgrounds at all times
Phone Desktop Backgrounds are not pre-loaded into the CUCM server, however the
images are provided for you in the proper size and format (for 7961 phones), however
the List.xml file is not - you need to create that on your own
o Load the "CiscoLogo" background on all SIP phones and load the "INELogo"
background on all SCCP Phones
3pts
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Version 3.5
Mock Lab 2
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
2.16
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
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Version 3.5
Mock Lab 2
2.17
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
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Version 3.5
Mock Lab 2
Dial Plan
Route Plan Description and Overall Guidelines
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Version 3.5
Mock Lab 2
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
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
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
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Version 3.5
Mock Lab 2
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
3pts
2.19
Localize all calls inbound from the PSTN as they arrive at every IP phone
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 10 digits
o During inbound alerting, IP Phones in the NL site should display calling party
numbers that are local to that site as 7 digits
Long Distance (National) Calls:
o During inbound alerting, IP Phones in the US sites should display calling party
numbers as 11 digits if that call is from the same country, but a different
geographic/area code
o During inbound alerting, IP Phones in the NL site should display calling party
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:
o During inbound alerting, IP Phones in the US sites 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
o During inbound alerting, IP Phones in the NL site should display calling party
numbers with all digits, including the country code and the preceding PSTN
international access code of "00", if that call is from a different country
3pts
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Version 3.5
Mock Lab 2
Create only one set of PSTN Patterns for each country (US and NL) based on the
information in the three tables below
Ensure these PSTN Patterns globalize every Called Party Number to a proper Full
E.164 (including preceding +) 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 (see next page)
Provision redundancy into the GW choice for PSTN calls in this manner:
o For calls made from one of the two US sites, all calls should use this order:
1) US Sites Local GW
2) Other US Sites GW
o For calls made from the NL site, all calls should use this order:
1) NL Local GW
2) CorpHQ Sites GW
o You are only permitted to create one Route List to satisfy the requirements of
these three tasks (T2.20, 2.21, & 2.22) Future tasks beyond T2.22 may
require more RLs, but this task should be fulfilled with only one RL
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 and/or change almost any
parameter(s) on these dial-peers, but you may not change any destination-patterns
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Version 3.5
Mock Lab 2
Table 1
Call Type
Emergency
Local
PSTN Pattern
911
[2-9]XX[2-9]XXXXXX
National/LD
International
Egress GW
(1) CorpHQ GW
(1) CorpHQ GW
(2) Branch1 GW
1[2-9]XX[2-9]XXXXXX (1) CorpHQ GW
(2) Branch1 GW
011 + Variable length (1) CorpHQ GW
(2) Branch1 GW
Calling # Format
10 ANI Digits
10 ANI Digits
11 ANI Digits
11 ANI Digits
11 ANI Digits
CC+ 10 ANI Digits
CC+ 10 ANI Digits
Table 2
Call Type
Emergency
Local
PSTN Pattern
911
[2-9]XX[2-9]XXXXXX
National/LD
International
Egress GW
(1) Branch1 GW
(1) Branch1 GW
(2) CorpHQ GW
1[2-9]XX[2-9]XXXXXX (1) Branch1 GW
(2) CorpHQ GW
011 + Variable length (1) Branch1 GW
(2) CorpHQ GW
Calling # Format
10 ANI Digits
10 ANI Digits
11 ANI Digits
11 ANI Digits
11 ANI Digits
CC+ 10 ANI Digits
CC+ 10 ANI Digits
Table 3
Call Type
Emergency
Local
PSTN Pattern
112
[1-8]XXXXXX
National/LD
0[16]XXXXXXXX
International
00 + Variable length
Egress GW
(1) Branch2 GW
(1) Branch2 GW
(2) CorpHQ GW
(1) Branch2 GW
(2) CorpHQ GW
(1) Branch2 GW
(2) CorpHQ GW
Calling # Format
10 ANI Digits
7 ANI Digits
CC+ 9 ANI Digits
10 ANI Digits
CC+ 9 ANI Digits
CC+ 9 ANI Digits
CC+ 9 ANI Digits
2.23
Ensure that any user at any hardware IP Phone who views their Missed Calls can
simply press the "Dial" softkey to return any call
Urgent Priority must be used in your patterns, and SIP phones must work properly
You must maintain the proper Calling Type described by the previous task
3pts
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Version 3.5
Mock Lab 2
Mobility
2.24
Configure CUCM such that when Hugo "Hurley" Reyes' CorpHQ Phone 2 at DN 1002
is called, that his home phone is rung immediately, and his mobile phone is rung after
1 full ring 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 Hurley's Home #: +1 206 501 5151 -Forwards to VM after 3 rings
This phone should only be rung Mon-Fri 07:00 08:00 local to his site
Mobile Connect calls displaying on this phone should show up in the
proper format so that he can simply press Dial from his home phone, if
the call came from inside or outside of the enterprise
o Hurley's Mobile #: +1 206 501 5555 -Forwards to VM after 4 rings
This phone should only be rung Mon-Fri 08:00 16:00 local to his site
Mobile Connect calls displaying on this phone should show up in the
globalized format if coming from outside the enterprise, and as the
standard 4 digit extension if coming from inside the company
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 Hurley 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 Hurley to be able to login to his CCMUser page and setup, change or even
add any Mobile Connect remote phones
5pts
2.25
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 Hurleys remote PSTN phone
o If Mobile Connect is ringing Hurley's mobile or home phone, the call should try
to ring out the CorpHQ GW primarily, if it is available
2pts
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Version 3.5
Mock Lab 2
High Availability
2.26
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 Loopback0 interface
o Normal calls to the PSTN should function the same way (you need not worry
about inbound or outbound globalization and 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)
3pts
2.27
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
3pts
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Version 3.5
Mock Lab 2
Quality of Service
2.28
Ensure that all routers sending any sort of signaling traffic to the CUCM or media
traffic to any IP phones directly, marks said traffic with CS3 and EF respectively
Trust only L2 CoS markings if sent from a Cisco IP phone at CorpHQ
3pts
2.29
On the CorpHQ switch (SW1), perform the following for the whole switch:
o Assign the following DSCP PHB's to the following egress queues and tail-drop
thresholds respectively:
DSCP CS3 >> Q2 T2
DSCP AF41 >> Q1 T2
DSCP EF >> Q1 T3
3pts
2.30
On the CorpHQ switch, assign the following configuration to only the uplink port going
to the CorpHQ router (R1):
o Of the available memory for all queues per interface, allocate:
55% to Q1
10% to Q2
30% to Q3
5% to Q4
o Of the available buffers for all WTD thresholds per queue, allocate:
20% to T1
80% to T2
100% as reserved for all thresholds
250% as the maximum that all thresholds can ever access
3pts
2.31
On the CorpHQ switch, enable expedited egress queuing for the port that connects to
your IP phones
3pts
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Version 3.5
Mock Lab 2
Messaging
2.32
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 page in UC to accomplish
this task
MWI must work properly for these phones when a message is left or listened to
3pts
2.33
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 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
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Version 3.5
Mock Lab 2
Presence
2.34
Configure Personal Communicator for Benjamin Linus at DN 2001 using the client on
your XP Utility machine
Assign all other IP phone users to Bens CUPC client so that he sees them with their
proper name and any state changes that occur with their primary extensions
Ensure that any VMs that Ben has left for him show up on his CUPC client display
3pts
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