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CCIE Voice Lab Workbook Volume II Version 3.0 Lab 3 Tasks

Copyright © 2009 Internetwork Expert www.INE.com - 1 -

Internetwork Expert Voice Workbook Volume II

Lab 3

Difficulty Rating (10 highest): 8

Lab Overview: This lab scenario is a mock lab exam designed to simulate the conditions of Cisco Systems’ CCIE Voice Lab exam. This lab should be completed within 7.5 hours. The only resource that should be used while configuring this lab is Cisco’s documentation set. This documentation is available in both DVD format, and online at http://www.cisco.com/go/support. Lab Instructions: Prior to starting, ensure that the initial configuration scripts for this lab have been applied. Troubleshooting scenarios have been introduced into this lab. Ensure all scenarios are working before completing the lab. Grading: This practice lab consists of various sections totaling 79 points. A score of 59 points is required to achieve a passing score. A section must work 100% with the requirements given in order to be awarded the points for that section. No partial credit is awarded. If a section has multiple possible solutions, choose the solution that best meets the requirements.




1. Infrastructure Configuration 1.1. VLAN Assignment • Configure the on-site switches to support Cisco IP phones • Provide support for future PCs to be connected to the Cisco IP Phones internal

switch ports

2 Points

1.2. DHCP • Configure the routers at all sites to allocate IP addresses via DHCP per the

following requirements:

o Create three DHCP pools for voice VLANs in routers R1, R2 and R3 o Configure the DHCP pools to allocate IP addresses from the respective

/24 subnet using the diagram for reference o The Publisher is the TFTP server for IP Phones at HQ and BR1 sites o R3 is the TFTP server for IP Phones at BR2 site

• Configure the remaining DHCP settings as you find appropriate for other tasks

2 Points

2. Station Devices

2.1. CCM IP Phones

• Using the following table as your reference, register the IP phones at the HQ and BR1 sites with the CallManager cluster and assign directory numbers:

Phone Extension DID Number HQ IP Ph1 1001 7752011001 HQ IP Ph2 1002 7752011002 HQ IP Ph3 1003 7752011003 BR1 IP Ph1 2001 3123012001 BR1 IP Ph2 2002 3123012002

• By the end of this task you should be able to place calls between all IP Phones

and distinguish the phones by calling numbers

3 Points




2.2. CME IP Phones

• You company is planning to use Cisco CallManager Express at the BR2 location

for local phone services • Using the following table as your reference, register the IP phones at the BR2

location with the CME system and assign the following directory numbers:

Phone Extension DID Number BR2 IP Ph1 3001 21313001 BR2 IP Ph2 3002 21313002

3 Points 3. Trunks & Gateways

3.1. MGCP Digital Gateway • Register the digital gateway at the HQ location as a MGCP endpoint with the

CallManager cluster • Busy out the unused channels in the CallManager Service Parameters • Use Top-Down approach for B-channel selection • Refer to the respective table for information on the digital gateways settings

3 Points

3.2. H.323 Gateway • Refer to the respective table for information on the digital gateway settings • Configure the BR1 on-site router (GW_BR1) as an H.323 gateway with the

CallManager cluster • Ensure the CallManager may optionally use multiple IP addresses to access the

gateway • After you have finished with the task, you should be able to place a call from the

PSTN test phone towards the respective on-site IP Phones

3 Points

3.3. IP-to-IP Gateway • Configure the CallManager cluster to use SIP as the signaling protocol with the

CallManager Express system at the BR2 Location • At the same time, configure the CallManager Express system to use H.323 as

the signaling protocol with the CallManager cluster • Use the HQ router (R1) as a middleware agent to convert the signaling protocols

and DTMF relaying conventions

4 Points




4. Call Routing

4.1. Route Plan Description

• Phones at all locations use code “9” to access the PSTN. Outside dial tone should be provided as soon as “9” has been dialed

• Users should be able to dial “911” and “9911” to reach emergency services where applicable

• Make sure “9” is stripped for calls going to the PSTN • Local area PSTN calls are placed to 7-digit numbers at the HQ and BR1

locations, and to 8-digit numbers at BR2 location • Long-distance (national) calls at the HQ and BR1 are placed to 10-digit numbers

using code “1” for toll-alert, and to 10 digit numbers at BR2 using code “0” for toll-alert

• International calls at the HQ and BR1 sites are signaled using access code “011” along with the variable length number

• International calls at BR2 are placed using toll-alert code “00” and variable length number

• Users may terminate their international number dialing using the “#” sign, in order to place the call immediately, without waiting for the inter-digit timeout

0 Points




4.2. Basic Call Routing

• Use the respective on-site gateway to place calls at each location to the patterns

listed in the following table:

Call Type HQ Pattern BR1 Pattern BR2 Pattern Emergency 911 911 999 Local [2-9]XXXXXX [2-9]XXXXXX [2-8]XXXXXXX National/LD 1[2-9]XX[2-9]XXXXXX 1[2-9]XX[2-9]XXXXXX 0[2-8]XXXXXXXXX Toll-Free 1800XXXXXXX International 011 + Variable length 011 + Variable length 00 + Variable length

• Implement classes of restriction according to the following table:

Class Pattern Types Phone Premium

Emergency Local Long Distance Toll-Free International

HQ IP Phone 3 BR1 IP Phone 2 BR2 IP Phone 2

Restricted Emergency Abbreviated

BR1 IP Phone 1

Default Emergency Local Long Distance Toll-Free

All Other Phones including the EM Profile

• Use IOS CoR lists functionality to implement restrictions with the CallManager

Express system

3 Points




4.3. Toll Bypass

• In order to cut down costs for international calls, implement the following

additional call routing rules:

o Calls from the HQ and BR1 phones to the BR2 country and area code, should be first attempted over the IPIP gateway

o Telephone calls from the BR2 location to the HQ and BR1 area codes should be first routed using the IPIP gateway

• If for any reason a call fails, apply configuration to fall back to the respective on-site PSTN gateway

3 Points

4.4. Abbreviated Dialing

o Configure abbreviated dialing to “1XXX”, “2XXX” and “3XXX” numbers per the following requirements:

o When a user at the HQ or BR1 locations dials “3XXX”, as a first routing option, attempt to place the call over the IPIP gateway

o If the IPIP gateway rejects the call, reroute it as a PSTN international call o When a user at the BR2 location dials “1XXX” or “2XXX” place the call

over the IPIP gateway first, and in the case it fails, re-route the call as a PSTN international call

• For the HQ and BR1 phones permit PSTN fallback to all Classes of Restriction

3 Points

4.5. Loopback Call

• When somebody at the HQ or BR1 site calls extension “3003” at the BR2 location over IP, the call should be redirected to the HQ IP Phone 2 over IP

• Do not use the allow-connection command under voice service voip configuration mode in R3 to accomplish this task

• Replace the Caller ID for redirected calls with the extension number “3000”

3 Points




5. CAC & Codecs

5.1. Codec Selection • Use the G.711 codec within the HQ and BR1 locations, and G.729 between the

HQ and BR1 endpoints, including the IP-to-IP gateway • Make sure the CallManager uses the G.711 codec to place VoIP calls to BR2

phones (via the IP-to-IP gateway) • Configure CallManager Express to use G.729 for calls to the HQ location placed

via the IP-to-IP gateway

3 Points

5.2. Application Codecs • The CUCCX application is installed with G.711 codec support only • Configure so that G.729 is used as the codec between the BR1 on-site devices,

and IPCC Express CTI Ports • Additionally, configure so that the CallManager uses the G.729 codec for voice

sessions with the Cisco Unity Express system

2 Points

6. Media Resources

6.1. Transcoding • Allocate DSP resources in the R3 router (BR2 location) to register a transcoder

with the CallManager cluster • Register an additional transcoder with the CallManager using the available DSP

resources at the BR1 site • Configure so that HQ and BR1 devices use the BR1 transcoder, and BR2

devices use the R3 DSP resources for transcoding

3 Points

6.2. Music on Hold • Configure the Subscriber CallManager to source Music on Hold audio streams • Music should be sent using the G.711 codec inside the HQ region, using the

G.729 codec when streaming to BR1 and G.711 when streaming to the IP-to-IP gateway

• When streaming to the HQ devices, limit the number of simultaneous audio streams to 4

3 Points




7. Applications & Services

7.1. Basic ACD • Integrate the B-ACD script with the CallManager Express system in R3 per the


o Users should be offered the default options to press “0” (“Operator”), “1” (“Sales”) or ‘2” (“Customer Support”)

o When a user presses “1”, BR2 IP Phone 1 should ring o BR2 IP Phone 2 should be called when a user presses “2”

• Associate the ACD application with PSTN DID number 2Y3Y3000

3 Points

7.2. Basic ACD (Continued) • When a user presses “3”, they should be allowed to dial the internal number

directly • As soon as “0” has been pressed, redirect the incoming call to the voice mailbox

of BR2 IP Phone 1 • After waiting for a maximum time (600 seconds) on queue, a call contact should

be redirected to the voice mailbox of HQ IP Phone 1 (Note that it’s the HQ phone)

3 Points

8. High Availability

8.1. AAR • Enable Automated Alternate Re-Routing with the CallManager cluster • Make sure a user at BR1 IP Phone 2 is allowed to reach the voicemail over the

PSTN even when CallManager CAC reports insufficient WAN bandwidth • You are not required to configure AAR settings for other devices in the system

3 Points




9. QoS 9.1. Traffic Shaping

• Use the following table to obtain information on Frame-Relay PVC CIRs and

physical port speed values:

Connection Interface Speed/CIR R1 Physical Serial 0/0/1:0 1536K R1 to R2 Serial 0/0/1:0.1 512K R1 to R3 Serial 0/0/1:0.2 384K R2 Physical Serial 0/0/1:0 1536K R2 to R1 Serial 0/0/1:0.1 512K R3 Physical Serial 1/1:0 2048K R3 to R1 Serial 1/1:0.1 384K

• Configure the routers connected to the Frame-Relay cloud to conform to their provisioned CIR values

• Make sure you provided minimum scheduling latency for shaped traffic • Use only the Modular QoS CLI commands to accomplish this task, and do not

use the map-class frame-relay command

3 Points

9.2. Traffic Classification

• BR2’s connection to the HQ office is secured by means of the IPsec tunnel used to protect the company’s sensitive information

• Provide a solution to allow classification of encrypted traffic at the physical interface level

• Apply configuration to classify CTIQBE packets from the CUE voice mail system. You are allowed to use an access list to accomplish this

• Classify voice bearer and signaling traffic, but do not rely on the DSCP marking for classification, and do not use any access-lists

3 Points




10. Voice Mail 10.1. Unity VM Integration

• You are planning to deploy a centralized Voice Mail system for your sites using

the Unity server installed at the HQ location • Provide a configuration to integrate your Unity installation with the CallManager

cluster per the following requirements:

o Use the voice-mail pilot number “1500” o The voice-mail ports should be in range “1501-1502” o Use numbers “2998” and “2999” for MWI on and off settings

3 Points

10.2. Unity Subscribers

• Create three voice mailboxes: for HQ IP Phone 2, HQ IP Phone 3 and BR1 IP Phone 2

• Use the password value “12345” for every user and any names/aliases of your choice

• Make sure you can send voice-mails between HQ and BR1 Phones, and get MWI lights lit correctly

• The unanswered phone call or calls to a busy phone line on the respective phones should be redirected to the Voice Mail system

3 Points

10.3. CUE Initialization

• Configure the CallManager Express system for integration with Cisco Unity Express voice-mail per the following requirements:

o Use the voice mail pilot number “3500” o Use MWI on and off numbers “3998....” and “3999....” o Create an administrator user named administrator with the

password value “cisco” o Use number “3555” for Administration via Telephone (AvT)

• Allow up to three concurrent VM sessions to the CUE system, for VoIP and

PSTN callers simultaneously • Run the Web-init setup wizard to make sure CUE users may access their

voicemail profiles via the web

3 Points




10.4. CUE Subscribers

• Create voice mailboxes for BR2 IP Phone 1 and IP Phone 2, and make sure

users can log in without listening to the CUE tutorial • Use the password value of “cisco” and a PIN value of “12345” for all users • Apply configuration to reach the VM pilot number from the PSTN • Test your configuration by sending messages between two users and verifying

that the MWI light goes on • Allow the PSTN phone to call the AvT number and leave a broadcast message.

Use number “999” to identify the PSTN phone user

2 Points 11. CRS

11.1. CUCCX Express Integration

• CUCCX system is deployed to handle customer calls at the BR1 location • Integrate your existing CUCCX installation with the CallManager cluster per the


o JTAPI user name prefix/password: “jtapi/cisco” o RMCM user name prefix/password: “rmcm/cisco” o CTI ports range: “1401-1402” o Default application script name: “icd.aef” o CTI RP DN: “1400”

• Configure both BR1 IP Phones for ICD agents, using agent names “agent1” and

“agent2” and the password of “cisco” • Ensure the integration works correctly and the agents can see Enterprise Data

on their LCD screens or Desktop interfaces

3 Points

11.2. Skill Based Routing • Create two skills in the CUCCX Resource Manager, named “sales” and “support” • Create a new CSQ in the CUCCX system, and assign both agents to this queue • Make sure the agent assigned to BR1 IP Phone 1 is preferred over the other

when routing calls • The previously defined “support” skill should be weighted more than sales for this

CSQ

2 Points




12. Presence 12.1. Presence Integration • Integrate the Presence Server (177.1.10.50) with the CUCM Server

o Ensure that HQ User 1 can control their phone with the CUPC client o Ensure the user can reach voicemail from the softphone

4 Points

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