IEVO_WB2_Lab2_Tasks.pdf

INE CCIE Voice Volume II Version 3.5 Mock Lab 2

Copyright © 2011 Internetwork Expert www.INE.com - 1 -

CCIE Voice Volume II

Mock Lab 2 :: Tasks Difficulty Rating (1:Lowest, 10:Highest): 9 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 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:



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 4 CUCM Server and Phone Basics 8 CUCM Media Resources 7 CUCM Features 14 Gateways and Trunks 9 Dial Plan 24 Mobility 7 High Availability 6 Quality of Service 12 Messaging 6 Presence 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!



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 site’s router to distribute IP addressing and options via DHCP for that site’s 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



CUCM Server and Phone Basics

2.3 • 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

Change this D/T group to allow for a format where the Day comes before the Month and a "dot" (.) is between each section of numbers

Do not allow "a" or "p" to display after the time on IP Phones that use this D/T group

2pts

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 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 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



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



CUCM Media Resources

2.7

• Setup a Meet-Me Conference at the DN of 4010 • You may only allow Jack or Desmond to setup this Meet-Me conference, however any

phone (up to a maximum of 8) should be able to dial into the conference • If any member of the conference puts the conference on hold, MoH should be allowed

to be played to the conference • Do not use any software conference bridging resources

2pts

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 everyone’s 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



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 Shepherd’s 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 Jack’s VM box, and not Hugo’s 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



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



Gateways and Trunks

2.15 • 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

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



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



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 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 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 placed from CorpHQ & Branch1 locations by dialing 10-digit numbers, and BR2 by dialing 7-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 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



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





*675126022XXX Dialing into the Branch2 GW: • For Calling Party Type to show as "Subscriber" >> Select PSTN Line 3, and dial





*670207033XXX



Inbound Calling from the PSTN

2.18 • 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



Outbound Calling to the PSTN

Tasks 2.20 – 2.22

NOTE: This begins the explanation for the next three tasks – one for each site. Each task has a point value independently from one another, however for each task, everything relevant to a given site (listed below in this explanation), must be configured properly to be awarded the points for that site.

• 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 Site’s Local GW 2) Other US Site’s GW

o For calls made from the NL site, all calls should use this order: 1) NL Local GW 2) CorpHQ Site’s 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 RL’s, 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”



2.20 Dialing From CorpHQ Phones out to the PSTN Table 1

Call Type PSTN Pattern Egress GW Calling # Format Calling Party Type Emergency 911 (1) CorpHQ GW 10 ANI Digits Unknown Local [2-9]XX[2-9]XXXXXX (1) CorpHQ GW 10 ANI Digits Subscriber

(2) Branch1 GW 11 ANI Digits National National/LD 1[2-9]XX[2-9]XXXXXX (1) CorpHQ GW 11 ANI Digits National

(2) Branch1 GW 11 ANI Digits National International 011 + Variable length (1) CorpHQ GW CC+ 10 ANI Digits International

(2) Branch1 GW CC+ 10 ANI Digits International 5pts

2.21 Dialing From Branch1 Phones out to the PSTN

Table 2 Call Type PSTN Pattern Egress GW Calling # Format Calling Party Type Emergency 911 (1) Branch1 GW 10 ANI Digits Unknown Local [2-9]XX[2-9]XXXXXX (1) Branch1 GW 10 ANI Digits Subscriber

(2) CorpHQ GW 11 ANI Digits National National/LD 1[2-9]XX[2-9]XXXXXX (1) Branch1 GW 11 ANI Digits National

(2) CorpHQ GW 11 ANI Digits National International 011 + Variable length (1) Branch1 GW CC+ 10 ANI Digits International

(2) CorpHQ GW CC+ 10 ANI Digits International 5pts

2.22 Dialing From Branch2 Phones out to the PSTN

Table 3 Call Type PSTN Pattern Egress GW Calling # Format Calling Party Type Emergency 112 (1) Branch2 GW 10 ANI Digits Unknown Local [1-8]XXXXXX (1) Branch2 GW 7 ANI Digits Subscriber

(2) CorpHQ GW CC+ 9 ANI Digits International National/LD 0[16]XXXXXXXX (1) Branch2 GW 10 ANI Digits National

(2) CorpHQ GW CC+ 9 ANI Digits International International 00 + Variable length (1) Branch2 GW CC+ 9 ANI Digits International

(2) CorpHQ GW CC+ 9 ANI Digits International 5pts

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



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 Hurley’s 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



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 phone’s top-right display from previous task

requirements 3pts



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



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



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 Ben’s CUPC client so that he see’s 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

Documents

IEVO_WB2_Lab2_Tasks.pdf