CCIE Voice Lab Workbook Volume II

Version 3.0

Lab 6

Internetwork Expert Voice Workbook Volume II Lab 6

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 8 hours. The only resource that should be used while configuring this lab is Ciscos 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. For a current copy of these scripts, see the Internetwork Expert members site at http://members.internetworkexpert.com. The initial configurations for all routers and switches include IP addressing and routing configuration. Refer to the lab diagram available under your members account for information on the topology and IP addressing. In addition to the lab diagram, please download and reference the Table Reference Guide while completing the lab. Both the diagram and reference guide are available under the Voice Volume II section of your members account. The lab could be completed using either soft-phones (e.g. Cisco IP Communicator or IP Blues VTGO) or by using Cisco 7960, 61, 62, and 65 IP phones. Refer to your rack rental users guide for detailed instructions on connecting the remote soft-phones to the rack.

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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.

Point Values:
The point values for each section are as follows: Section Infrastructure Configuration Station Devices Trunks & Gateways Call Routing CAC & Codecs Media Resources Applications & Services High Availability QoS Voice Mail CUCCX Total: Point Value 9 9 6 15 3 7 7 3 3 12 6 79

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1. Infrastructure Configuration
You company has recently acquired two other smaller companies and is in the process of merging their IT infrastructures with its own. The new acquisitions are located at BR1 and BR2 sites

1.1. VLAN Assignment

Configure the on-site switches to support Cisco IP phones Make sure you provided support for future PCs to be connected to the Cisco IP Phones internal switch port 3 Points

1.2. DHCP
Configure the routers at all three 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 rack topology diagram as reference o Set the TFTP servers IP to the Publisher CallManagers IP address for the HQ and BR1 devices o Set DHCP lease duration to 15 minutes for all sites Configure all other DHCP settings as you find appropriate Make sure you only allocate IP addresses with the last octet in range 50-100 3 Points

1.3. DNS Service

In order to provide added redundancy use a DNS name subsystem with your phones at HQ and BR1 locations Configure the respective on-site routers to act as DNS servers, and provide configuration to resolve the following hostnames to respective IP addresses of the CUCM Publisher, Subscriber, Unity, and Presence server: CUCM7-PUB, CUCM7-SUB, UNITY7, CUPS7 The name CUCM-CLUSTER should resolve to the IP address of the Subscriber and Publisher in the order mentioned 3 Points

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2. Station Devices
2.1. CCM IP Phones
Perform the following with the CallManager IP Phones o Register the IP phones at the HQ and BR1 sites with the CallManager cluster o Assign directory numbers to the IP Phones Use the following table as your reference to accomplish this task Phone HQ IP Ph1 HQ IP Ph2 HQ IP Ph3 BR1 IP Ph1 BR1 IP Ph2 Extension 2001 2002 2003 2001 2002 DID Number 7752011001 7752011002 7752011003 3123012001 3123012002

Configure the IP Phones so that they use consistent calling names across the sites, so that you can distinguish a calling phone by name. An example of an acceptable name is HQ IP Ph1 By the end of this task you should be able to place VoIP calls to/from all IP Phones and distinguish them by calling names/numbers 3 Points

2.2. CME IP Phones

Register the IP phones at the BR2 location with the CallManager Express system running in R3 Assign directory numbers to the IP Phones, create users and associate them with respective phones Use the following table as a reference for information needed to accomplish this task Phone BR2 IP Ph1 BR2 IP Ph2 Extension 3001 3011 3002 3012 User br2user1 br2user2 DID Number 2Y3Y3001 2Y3Y3011 2Y3Y3002 2Y3Y3012

Allow for call-waiting functionality with each IP Phones primary line only 3 Points

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2.3. CME IP Phones Layout

Configure the IP Phones so that they use meaningful calling names for their primary lines. Example of acceptable names are BR2 Ph1 for BR2 IP Phone1 Apply configuration to use line labels and other features in accordance with the display layout samples below: The display layout for BR2 IP Phone 1:
HH:MMp MM/DD/YY 21313001 Manager 3001 3011

CallManager Express at BR2 Redial Pickup CFwdAll

The display layout for BR2 IP Phone 2:

HH:MMp MM/DD/YY 21313002 3002 Assistant 3012

CallManager Express at BR2 Redial Pickup CFwdAll

3 Points

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3. Trunks & Gateways

3.1. MGCP Digital Gateways
Register HQ and BR1 digital gateways with the CallManager cluster: o Use the Loopback0 interface of R2 to source MGCP signaling and media packets o The CallManager should use the Top-Down approach, when looking for available ISDN bearer channels Refer to the following command output to configure your ISDN PRI settings:

PSTN#show isdn service ! ! BR1 connection ! PRI Channel Statistics: ISDN Se0/0/0:23, Channel [1-24] Configured Isdn Interface (dsl) 0 Channel State (0=Idle 1=Proposed 2=Busy 3=Reserved 4=Restart 5=Maint_Pend) Channel : 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 State : 0 0 3 3 3 3 3 3 3 3 3 3 3 3 3 3 3 3 3 3 3 3 3 3 Service State (0=Inservice 1=Maint 2=Outofservice 8=MaintPend 9=OOSPend) Channel : 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 State : 0 0 2 2 2 2 2 2 2 2 2 2 2 2 2 2 2 2 2 2 2 2 2 2 .... ! ! HQ connection ! ISDN Se0/3/0:23, Channel [1-24] Configured Isdn Interface (dsl) 2 Channel State (0=Idle 1=Proposed 2=Busy 3=Reserved 4=Restart 5=Maint_Pend) Channel : 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 State : 0 0 0 3 3 3 3 3 3 3 3 3 3 3 3 3 3 3 3 3 3 3 3 3 Service State (0=Inservice 1=Maint 2=Outofservice 8=MaintPend 9=OOSPend) Channel : 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 State : 0 0 0 2 2 2 2 2 2 2 2 2 2 2 2 2 2 2 2 2 2 2 2 2

Upon completion of the task, you should be able to place a call from the PSTN test phone to any on-site IP phone 3 Points

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3.2. SIP Gateway

Configure the router at the BR2 location (R3) as a SIP gateway and add this gateway to the CallManager cluster Configure SIP UA settings in accordance with the following command output
Rack01R1#show sip-ua status SIP User Agent Status SIP User Agent for UDP : DISABLED SIP User Agent for TCP : ENABLED SIP User Agent bind status(signaling): ENABLED 177.3.11.1 SIP User Agent bind status(media): ENABLED 177.3.11.1 SIP early-media for 180 responses with SDP: DISABLED <snip>

3 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 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 sites 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 are signaled using access code 011 along with the variable length number International calls at BR2 are placed using the toll-alert code 00 and variable length number Users may terminate their international numbers dialing using the # sign, in order to place the call immediately, without waiting for the inter-digit timeout 0 Points

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4.2. HQ Basic Call Routing

Use the following table for reference on dialing patterns for the HQ site Call Type Emergency Local National/LD International Toll-Free HQ Pattern 911 [2-9]XXXXXX 1[2-9]XX[2-9]XXXXXX 011 + Variable length 1800XXXXXXX

Configure call routing for the HQ Phones per the following requirements: o Route calls to the HQ local area code using HQ gateway as the primary gateway and BR1 gateway as the secondary gateway, in case the primary fails o Calls to toll-free numbers should have any type of caller ID information removed and should be placed using the HQ gateway only o Use the on-site gateway to place Emergency, Long Distance and International calls o Apply configuration to reach BR2 on-site phones via abbreviated number dialing by using VoIP calls only (no PSTN backup required) Only allow HQ IP Phone 3 to call the International destination 3 Points

4.3. BR1 Basic Call Routing

Configure call routing for BR1 phones per the following requirements: o The phones at the BR1 location are only allowed to use the on-site gateway as the PSTN gateway o The phones should be able to fall back to the HQ gateway as a backup PSTN access point for local calls only o Apply configuration to reach the BR2 on-site phones via abbreviated number dialing by using VoIP calls only (PSTN backup option is not required) Use the following table as a reference on dialing patterns for the BR1 site: Call Type Emergency Local National/LD International BR1 Pattern 911 [2-9]XXXXXX 1[2-9]XX[2-9]XXXXXX 011 + Variable length

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4.4. Toll Bypass

Provide a toll-bypass solution for calls originated from HQ and BR1 phones per the following requirements: o Calls from HQ phones to the BR1 area code should first be placed using the BR1 gateway and fall back to the HQ gateway in case of failure o Calls from HQ and BR1 phones to BR2 country code should be first placed using the BR2 gateway and use HQ as a backup gateway When a TEHO call is placed via the backup gateway, change the calling number to make the call look like it comes from the tail-end site 3 Points

4.5. BR2 Call Routing

Ensure no access restrictions apply to BR2 phones, every phone is allowed to call any PSTN number Configure call routing for BR2 phones per the following requirements: o The IP Phones should use R2 as the only PSTN gateway o Provide configuration for tail-end hop-off routing towards HQ and BR1 area codes, so that international calls from BR2 to the mentioned areas are first attempted via the respective HQ and BR1 on-site gateways o A failed hop-off call should be re-routed as a regular international call The IP Phones should present their calling names when calling any PSTN number, but apply configuration to hide the calling number 3 Points

4.6. Call Coverage

o Provide call coverage to BR2 Phone 1 per the following requirements: o Calls going to BR2 Phone 1s primary line should ring BR2 Phone 2 if the line is busy or user does not answer o If the primary line is busy, or user does not answer within 12 seconds, forward calls to BR2 Phone 2s secondary line o If BR2 Phone 2s secondary line is busy or does not answer, forward the call to the voice mailbox of BR2 Phone 1 Disable the call-forward-all function for BR2 IP Phone 2 3 Points Copyright 2009 Internetwork Expert -9www.INE.com

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5. CAC & Codecs

5.1. Codec Manipulation
Ensure you use the G.711 codec for voice calls within each site, and use G.729 between the HQ and BR1 sites All VoIP calls between the HQ and BR2 locations should use the G.711 codec Additionally, VoIP calls between the BR1 and BR2 locations should use the G.711 codec 3 Points

6. Media Resources
6.1. Hardware Resources
Configure hardware DSP resources per following requirements: o Register the HQ on-site DSP resource as a conference bridge with the CallManager o Register the BR1 on-site DSP resources as a transcoder with the CallManager cluster Ensure that HQ and BR1 devices only use the BR1 transcoding resources Only allow HQ users access to the conference bridge 3 Points

6.2. Music On Hold

Stream Music on Hold from the CallManager Publisher and Subscriber servers per the following requirements: o Use G.711 as the codec to stream MoH for the HQ and BR1 devices and stream music using multicast delivery Lastly, only configure the Subscriber MoH server for multicasting 2 Points

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6.3. Advanced Music on Hold

In order to conserve bandwidth, the Publisher and Subscriber CallManagers servers were configured to stream multicast Music on Hold to BR1 endpoints Ensure that multicast streams from the HQ location do not transverse the WAN cloud Configure the BR1 router to substitute the multicast stream with a local multicast feed out of the routers flash memory 2 Points

7. Applications & Services

7.1. Personal Directory
Configure the CallManager Express system so that users do not need to edit a stored number - e.g. prepend 9 - before dialing received or missed call stored in the phones directory This requirement applies only for calls coming from within the BR2 area and PSTN calls from the HQ and BR1 areas Test your configuration by verifying the phone number stored in the Personal Directory for calls coming from the above mentioned areas 4 Points

7.2. CME Overhead Paging

Add a new speed-dial button to BR2 IP Phone 2, and configure multicast paging as follows: o When a user at BR2 IP Phone 2 presses the new button, it should be immediately connected to a paging loudspeaker o Using multicast group 239.1.1.1 configure all phones in the CallManager Express system to be members of the paging group Make sure no other users are able to dial the paging DN directly from their phones Use CoR features to accomplish this task 3 Points

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8. High Availability
8.1. Basic SRST
Configure the BR1 router to provide SRST services for local phones. Users should be able to access the dial plan patterns they have access to in normal mode (including abbreviated dialing and access to overlapping numbers) Strip Called ID and Caller Name when calling 1-800 Toll Free numbers Configure the IP Phones to re-home back to the CallManager after 60 seconds of the connection being available again Phones status line should display SRST Services at BR1 SRST server should provide outside dial tone when a user dials 9 Do not use the dialplan-pattern command to provide DID services to on-site phones 3 Points

9. QoS

9.1. Frame-Relay Traffic Shaping

Refer to the Port Tables document for information on Frame-Relay PVC CIRs and physical port speed Configure the routers connected to the Frame-Relay cloud to conform to their provisioned CIR values Use only legacy FRTS commands to configure the traffic shaping 4 Points

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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 1998 and 1999 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

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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 3 Points

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11. CRS
11.1. IPCC Express Integration
Apply configuration to integrate your existing IPCCX installation at the HQ location with the CallManager cluster per the following requirements: o o o o o o JTAPI user name prefix/password: jtapi/cisco RMCM user name prefix/password: rmcm/cisco CTI port range: 2401-2402 Default application script name: icd.aef CTI RP DN: 2400 ICD Agent username/password: icdagent/cisco

Configure the HQ IP Phone 3 for ICD agent and assign it to the call service queue. Subscribe the phone to the IP Phone agent service 3 Points

11.2. IPCC Customization

Configure the IP Phone Agent Service to permit single-touch sign-on into the IPCC call service group When a call arrives to the IP Phone agent, call data should display Custom_ICD_Script among the others A contact placed on queue should listen to music on hold while waiting 3 Points

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