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INE CCIE Voice Volume II Version 3.5 Mock Lab 1 Copyright © 2011 Internetwork Expert www.INE.com - 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 Cisco’s 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 doesn’t 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 INE’s 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).

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INE CCIE Voice Volume II Version 3.5 Mock Lab 1

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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 Cisco’s 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 doesn’t 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 INE’s 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).

INE CCIE Voice Volume II Version 3.5 Mock Lab 1

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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 Point Value Network Infrastructure 6 CUCM Server and Phone Basics 13 CUCM Media Resources 12 CUCM Features 6 Gateways and Trunks 9 Dial Plan 15 Mobility 9 High Availability 6 Quality of Service 12 Messaging 6 Presence 3 Contact Center 3 TOTAL 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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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

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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 it’s 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

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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

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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 phone’s 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 in-progress, 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

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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 doesn’t 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

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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

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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

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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

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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

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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 PSTN Pattern Calling # Format Calling Party Type Emergency 911 10 ANI Digits Unknown Local [2-9]XX[2-9]XXXXXX 10 ANI Digits Subscriber National/LD 1[2-9]XX[2-9]XXXXXX 10 ANI Digits National International 011 + Variable length CC+ 10 ANI Digits International

Table 2: Dialing from within NL sites

Call Type PSTN Pattern Calling # Format Calling Party Type Emergency 112 10 ANI Digits Unknown Local 020XXXXXXX 10 ANI Digits Subscriber National/LD 0[16]XXXXXXXX 10 ANI Digits National International 00 + Variable length CC+ 9 ANI Digits International

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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

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Mobility

1.21 • Configure CUCM such that when James "Sawyer" Ford’s 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 Sawyer’s 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 Sawyer’s 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 Hugo’s IP phone display 3pts

1.23

• When calls show up at Sawyer’s 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

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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 phone’s top-right display from previous task

requirements 2pts

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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

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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

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Presence

1.32 • Configure Personal Communicator for Hugo Reyes’ at DN 1002 using the client on

your XP Utility machine • Assign Jack and Bens’ userid’s to Hugo’s CUPC client so that he see’s them with their

proper name and any state changes that may occur when they get CUPC clients • Ensure that Hugo’s 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