Cvoice v60 Lab Guide

CIPT1

Cisco Voice over IP Version 6.0

Lab Guide

Editorial, Production, and Web Services: 02.15.08

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Table of Contents Lab Guide 1

Overview 1 Outline 1

Lab Topology 2 Job Aids 5

Dial Plan 5 Dial Plan Conventions 5 PSTN Configuration 6

Lab 2-1: Configuring Analog Voice Interfaces 7 Activity Objective 7 Visual Objective 7 Required Resources 7 Command List 8 Task 1: Configure an Analog Voice Port 8 Task 2: Verify Analog Port Settings 10

Lab 2-2: Configuring POTS Dial Peers 12 Activity Objective 12 Visual Objective 12 Required Resources 12 Command List 12 Job Aids 13 Task 1: Configure POTS Dial Peers 13

Lab 2-3: Configuring VoIP Dial Peers 15 Activity Objective 15 Visual Objective 15 Required Resources 15 Command List 15 Job Aids 16 Task 1: Configuring VoIP Dial Peers 17 Task 2: Configure Codecs 18 Task 3: Calculate Bandwidth Requirements 19 Task 4: Configure Codec Negotiation 20

Lab 2-4: Configuring Digital Voice Interfaces 21 Activity Objective 21 Visual Objective 21 Required Resources 21 Command List 21 Job Aids 22 Task 1: Configure the BR T1 CAS Trunk to the PSTN 23 Task 2: Configure the HQ ISDN PRI Trunk to the PSTN 25

Lab 3-1: Implementing H.323 Gateways 27 Activity Objective 27 Visual Objective 27 Required Resources 27 Command List 27 Job Aids 28 Task 1: Configure an H.323 Gateway 29

Lab 3-2: Implementing SIP Gateways 30 Activity Objective 30 Visual Objective 30 Required Resources 30 Command List 30 Job Aids 31 Task 1: Configure a Gateway for SIP 31

Lab 4-1: Implementing Numbering Plans 33 Activity Objective 33 Visual Objective 33


ii Cisco Voice over IP (CVOICE) v6.0 © 2008 Cisco Systems, Inc.

Required Resources 33 Command List 33 Job Aids 35 Task 1: Configure a Numbering Plan for HQ and Branch Offices 36

Lab 4-2: Implementing PSTN Dial Plans 37 Activity Objective 37 Visual Objective 37 Required Resources 37 Command List 37 Job Aid 38 Task 1: Configure Dial Peers 39 Task 2: Configure Digit Manipulation 39

Lab 4-3: Configuring Path Selection 41 Activity Objective 41 Visual Objective 41 Required Resources 41 Command List 41 Job Aid 42 Task 1: Configure Hunt Groups 42 Task 2: Configure Redundant Calling Paths 43 Task 3: Configure TEHO 44

Lab 4-4: Implementing Calling Privileges 45 Activity Objective 45 Visual Objective 45 Required Resources 45 Command List 45 Job Aid 46 Task 1: Create COR Labels 46 Task 2: Configure COR Lists 47 Task 3: Configure Dial Peers to Use COR Lists 49

Lab 5-1: Configuring Basic Gatekeeper Functionality 50 Activity Objective 50 Visual Objective 50 Required Resources 50 Command List 51 Job Aid 52 Task 1: Configure Local Zones 52 Task 2: Configure Zone and Technology Prefixes 53 Task 3: Configure Gateways to Register with a Gatekeeper 53

Lab 5-2: Implementing Gatekeeper-Based CAC 56 Activity Objective 56 Visual Objective 56 Required Resources 56 Command List 56 Job Aids 57 Task 1: Calculate Bandwidth Requirements 58 Task 2: Configure Zone Bandwidth 58

Lab 6-1: Configuring a Cisco UBE 59 Activity Objective 59 Visual Objective 59 Required Resources 59 Command List 59 Job Aids 60 Task 1: Configure a Cisco UBE for Protocol Interworking and Address Hiding 60

Answer Key 61 Lab 2-1 Answer Key: Configuring Analog Voice Interfaces 61 Lab 2-2 Answer Key: Configuring POTS Dial Peers 62 Lab 2-3 Answer Key: Configuring VoIP Dial Peers 63 Lab 2-4 Answer Key: Configuring Digital Voice Interfaces 64 Lab 3-1 Answer Key: Implementing H.323 Gateways 67


© 2008 Cisco Systems, Inc. Cisco Voice over IP (CVOICE) v6.0 iii

Lab 3-2 Answer Key: Implementing SIP Gateways 68 Lab 4-1 Answer Key: Implementing Numbering Plans 69 Lab 4-2 Answer Key: Implementing PSTN Dial Plans on Cisco IOS Gateways 70 Lab 4-3 Answer Key: Configuring Path Selection 74 Lab 4-4 Answer Key: Implementing Calling Privileges on Cisco IOS Gateways 76 Lab 5-1 Answer Key: Configuring Basic Gatekeeper Functionality 79 Lab 5-2 Answer Key: Implementing Gatekeeper-Based CAC 81 Lab 6-1 Answer Key: Configuring a Cisco UBE 82


iv Cisco Voice over IP (CVOICE) v6.0 © 2008 Cisco Systems, Inc.


CVOICE

Lab Guide

Overview This guide presents the instructions and other information concerning the lab activities for this course. You can find the solutions in the lab activity Answer Key.

Outline This guide includes these activities:

Lab 2-1: Configuring Analog Voice Ports

Lab 2-2: Configuring POTS Dial Peers

Lab 2-3: Configuring VoIP Dial Peers

Lab 2-4: Configuring Digital Voice Ports

Lab 3-1: Implementing H.323 Gateways

Lab 3-2: Implementing SIP Gateways

Lab 4-1: Implementing Numbering Plans

Lab 4-2: Implementing PSTN Dial Plans on Cisco IOS Gateways

Lab 4-3: Configuring Path Selection

Lab 4-4: Implementing Calling Privileges on Cisco IOS Gateways

Lab 5-1: Configuring Basic Gatekeeper Functionality

Lab 5-2: Implementing Gatekeeper-Based CAC

Lab 6-1: Configuring a Cisco UBE to connect to an ITSP

Answer Key


2 Cisco Voice over IP (CVOICE) v6.0 © 2008 Cisco Systems, Inc.

Lab Topology This is a diagram of the lab topology used in the labs for Cisco Voice over IP (CVOICE) v6.0.

© 2008 Cisco Systems, Inc. All rights reserved. CVOICE v6.0—3

POD-1

Phone1-17941/61

1101

HQ-1

BR-1

PSTN Phone-1 PSTN Phone-2Phone5-27941/61

2201

Phone1-27941/61

2101

HQ-2

BR-2

POD-2

T1/E1

Phone5-17941/61

1201

(7941/61 or PC with NetMeeting)

PSTN

Serial

IP WAN

FXS FXS

Phone4-11104

Phone3-22103

Phone4-22104

Phone3-11103

Phone2-17941/61

1102

Phone2-27941/61

2102

Phone6-11202

FXS

Phone6-22202

FXS

GK1 GK2

Two-Pod Group

CVOICE 6.0 Lab Topology

The addressing of the classroom pods is described in the following tables.


© 2008 Cisco Systems, Inc. Lab Guide 3

Address Table for Odd-Numbered Classroom Pods

Device Name Device Name Abbreviation

Assigned Pod Interface

Network Address

Additional Information

Headquarters Pod X Gateway

HQX X Fa0/0.X02 10.X.2.101 HQX Voice VLAN

Fa0/0.X03 10.X.3.101 HQX Data VLAN

Fa0/0.198 3.X.0.1 Allows Telnet to switch from PodX

Loopback 0 10.X.250.101 Used as source interface

Serial 0/0/0 10.XY.XY.101 To PodY, HQY

Serial 0/0/1 10.X.4.101 To PodX, BRX

T1 01/0 T1 PRI trunk to PSTN:

■ PRI

■ Framing: ESF

■ Line code: B8ZS

Pod X Switch PodXSW X VLAN198 3.X.0.2 Management VLAN

VLANX02 HQX voice VLAN

VLANX03 HQX data VLAN

Branch Pod X Gateway

BRX X Fa0/0.X05 10.X.5.102 BRX voice VLAN

Fa0/0.X06 10.X.6.102 BRX data VLAN


Loopback 0 10. X.250.102 Used as source interface

Branch Pod X Switch

BRXSW X VLANX05 BRX voice VLAN

VLANX06 BRX data VLAN

Gatekeeper Pod X

GKX X Fa0/0. X03 10. X.3.102 HQX data VLAN




Note X = Pod number 1, 3, or 5. Y = Even number pod in pod pair.

Address Table for Even-Numbered Classroom Pods

Device Name Device Name Abbreviation

Assigned Pod Interface

Network Address

Additional Information

Headquarters Pod Y Gateway

HQY Y Fa0/0.Y02 10.Y.2.101 HQY Voice VLAN

Fa0/0.Y03 10.Y.3.101 HQY Data VLAN

Fa0/0.198 3.Y.0.1 Allows Telnet to switch from PodY

Loopback 0 10.Y.250.101 Used as source interface

Serial 0/0/0 10.XY.XY.101 To PodX, HQX

Serial 0/0/1 10.Y.4.101 To PodY, BRY

T1 01/0 T1 PRI trunk to PSTN: PRI

Framing: ESF

Line code: B8ZS

Pod Y Switch PodYSW Y VLAN198 3.Y.0.2 Management VLAN

VLANY02 HQY Voice VLAN

VLANY03 HQY Data VLAN

Branch Pod Y Gateway

BRY Y Fa0/0.Y05 10.Y.5.102 BRY Voice VLAN

Fa0/0.Y06 10.Y.6.102 BRY Data VLAN

Fa0/0.198 3.Y.0.3 Allows Telnet to switch from PodY

Loopback 0 10. Y.250.102 Used as source interface

Branch Pod Y Switch

BRYSW Y VLANY05 BRY Voice VLAN

VLANY06 BRY Data VLAN

Gatekeeper Pod Y

GKY Y Fa0/0. Y03 10. Y.3.102 HQY Data VLAN

Note Y = Pod number 2, 4, or 6. X = Odd number pod in pod pair.



Job Aids These job aids are available to help you complete lab activity.

Dial Plan The table represents the dial plan that will be used in the labs.

Classroom Dial Plan

Pod Number 1 2 X

HQX 1101(IP) 1102(IP) 1103(Analog)1104(Analog)

2101(IP) 2102(IP) 2103(Analog)2104(Analog)

X101(IP) X102(IP) X103(Analog) X104(Analog)

BRX 1201(IP) 1202(Analog)

2201(IP) 2202(Analog)

X201(IP) X202(Analog)

PSTN 331-555-1001 332-555-2001 33X-555-X001

Area code and exchange

HQ

BR

331-551

331-552

332-551

332-552

33X-551

33X-552

Dial Plan Conventions As with IP addresses, the dial plan convention should allow you to anticipate the number of any telephones in the classroom. The highlights of the classroom dial plan strategy include those listed here:

The classroom will use a 10-digit dial plan to call the PSTN.

— Calls to the pod PSTN phones should require a 9 and then the telephone number.

The classroom will use an 11-digit dial plan to call other pod phones through the PSTN.

— Calls to the other pod phones require a 9 + 1 and then the telephone number.

International calls should require a 011 international code.

The classroom uses four extension digits to place local calls.

— The first digit identifies the pod number (1 to 4).

— The second and third digits identify the device (PSTN = 00, HQ = 10, BR = 20).

— The fourth digit identifies the telephone.

Initially, a caller should be able to complete calls between same pod sites using just the extension number. Site codes will be assigned in the dial plan labs.



PSTN Configuration The HQ router will connect to the PSTN using an ISDN PRI connection while the BR router will connect to the PSTN using a T1 CAS.

HQ to PSTN The HQ PSTN parameters are as follows:

ISDN switch type is primary-ni

ISDN T1 connection

Timeslots 1 to 4

ESF framing

B8ZS line code

BR to PSTN The BR PSTN parameters are as follows:

CAS incoming T1 connection:

— Type E&M FGD

— Timeslots 1 to 2

— ESF framing B8ZS line code

— B8ZS line code

CAS outgoing T1 connection:

— Type FGD-EANA

— Timeslots 3 to 4

— ESF framing

— B8ZS line code

.



Lab 2-1: Configuring Analog Voice Interfaces Complete this lab activity to practice what you learned in the related module.

Activity Objective In this activity, you will become familiar with existing analog voice ports. You will learn how to customize your analog ports by configuring various port parameters. After completing this activity, you will be able to meet these objectives:

Identify default voice port settings

Customize and verify analog port operations

Visual Objective The figure illustrates what you will accomplish in this activity.


POD-1

Phone1-17941/61

1101

HQ-1

BR-1

PSTN Phone-1 PSTN Phone-2Phone5-27941/61

2201

Phone1-27941/61

2101

HQ-2

BR-2

POD-2

T1/E1

Phone5-17941/61

1201


PSTN

Serial

IP WAN

FXS FXS

Phone4-11104

Phone3-22103

Phone4-22104

Phone3-11103

Phone2-17941/61

1102

Phone2-27941/61

2102

Phone6-11202

FXS

Phone6-22202

FXS

GK1 GK2

Two-Pod Group

Lab 2-1: Configuring Analog Voice Interfaces

Required Resources These are the resources and equipment that are required to complete this activity:

Analog telephones

RJ-11 cables



Command List The table describes the commands that are used in this activity.

Voice Port Commands

Command Description

cptone country-code Sets the regional analog voice interface-related tone, ring, and cadence setting.

default parameter Resets the value of the parameter to its default value.

ring cadence {pattern-number | define pulse interval}

Specifies the ring cadence for an FXS voice port.

timeouts initial secs Sets the number of seconds that the system will wait for the caller to input the first digit.

voice-port port-number Enters voice-port configuration mode.

show voice port port-number (summary)

Views the voice port status and settings. Displays all default settings for each port. Specify a particular voice port to view only its settings. Use the summary option to view a summary table of the voice ports.

Task 1: Configure an Analog Voice Port In this task, you will examine analog voice ports and configure analog voice port parameters.

Activity Procedure Complete these steps:

Step 1 On both the HQ and BR routers, connect the analog phones to the FXS ports of the voice-enabled router using RJ-11 cables.

Step 2 Verify that the connections are correct by lifting the handset on both telephones and listening for the dial tone. If the dial tone is not present, troubleshoot the problem. Make sure that the router is powered on and that the cable is firmly seated. If the problem persists, ask your instructor for help.

Step 3 Using show commands, identify the available voice ports, their type, and their default settings.

Note Not all settings are applicable to all types of ports; for example, ring frequency and ring cadence apply only to FXS ports.



Step 4 Record the information here.

Voice Port 1 Voice Port 2 Voice Port 3 Voice Port 4

Port type __________ __________ __________ __________

Port number __________ __________ __________ __________

Phone number __________ __________ __________ __________

Operational state __________ __________ __________ __________

Echo cancellation __________ __________ __________ __________

Echo cancel coverage __________ __________ __________ __________

Initial timeout __________ __________ __________ __________

Region tone __________ __________ __________ __________

Signal type __________ __________ __________ __________

Ring frequency __________ __________ __________ __________

Ring cadence __________ __________ __________ __________

Step 5 Because many companies have international offices, it is important to know how to configure the voice port to match the standard signaling of a country. For this step, assume that you are configuring this router for Australia. On one FXS port that your telephone is connected to, configure the call progress tone setting for Australia using the cptone command. Notice that when you change the call progress tone setting, it automatically changes the ring cadence setting to match. Test the change by lifting the handset. You should hear a different dial tone.

Verify the changes with show commands and record the new settings here.

Region tone ________________________

Ring cadence _______________________

Once you have tested the tones for Australia, experiment with settings for other countries.

Step 6 What is the default initial timeout setting from Step 4? ___________ On at least one FXS port on both the HQ and BR routers, change the initial timeout value to 4 seconds. Lift the handset and listen for more than 4 seconds. Can you dial digits after the dial tone stops? Reset the initial timeout to the default.

Step 7 Because you will be working with two telephones at the HQ router all week, you may want one telephone to have a distinctive ring. Configure the ring cadence on one FXS port using the ring cadence pattern-X command. You will be able to test this ring cadence in the next lab. What ring cadence pattern did you specify? ____________



Activity Verification You have completed this task when you attain this result:

You have configured and verified analog voice port parameters.

Task 2: Verify Analog Port Settings In this task, you will verify your analog voice port operation and configuration.

Activity Procedure Complete this step:

Step 1 Use show voice port commands to verify that your analog voice ports are up and running.


Your output should resemble this.

HQ-1# show voice port summary

IN OUT

PORT CH SIG-TYPE ADMIN OPER STATUS STATUS EC

========= == ============ ===== ==== ======== ======== ==

0/0/0 -- fxs-ls up dorm on-hook idle y

0/0/1 -- fxs-ls up dorm on-hook idle y

HQ-1# show voice port 0/0/0

Foreign Exchange Station 0/0/0 Slot is 0, Sub-unit is 0, Port is 0

Type of VoicePort is FXS VIC2-2FXS

Operation State is DORMANT

Administrative State is UP

No Interface Down Failure

Description is not set

Noise Regeneration is enabled

Non Linear Processing is enabled

Non Linear Mute is disabled

Non Linear Threshold is -21 dB

Music On Hold Threshold is Set to -38 dBm

In Gain is Set to 0 dB

Out Attenuation is Set to 3 dB

Echo Cancellation is enabled

Echo Cancellation NLP mute is disabled

Echo Cancellation NLP threshold is -21 dB

Echo Cancel Coverage is set to 64 ms



Echo Cancel worst case ERL is set to 6 dB

Playout-delay Mode is set to adaptive

Playout-delay Nominal is set to 60 ms

Playout-delay Maximum is set to 250 ms

Playout-delay Minimum mode is set to default, value 40 ms

Playout-delay Fax is set to 300 ms

Connection Mode is normal

Connection Number is not set

Initial Time Out is set to 10 s

Interdigit Time Out is set to 10 s

Call Disconnect Time Out is set to 60 s

Supervisory Disconnect Time Out is set to 750 ms

Ringing Time Out is set to 180 s

Wait Release Time Out is set to 30 s

Companding Type is u-law

Region Tone is set for US

Analog Info Follows:

Currently processing none

Maintenance Mode Set to None (not in mtc mode)

Number of signaling protocol errors are 0

Impedance is set to 600r Ohm

Station name None, Station number None

Translation profile (Incoming):

Translation profile (Outgoing):

Voice card specific Info Follows:

Signal Type is loopStart

Ring Frequency is 25 Hz

Hook Status is On Hook

Ring Active Status is inactive

Ring Ground Status is inactive

Tip Ground Status is inactive

Digit Duration Timing is set to 100 ms

InterDigit Duration Timing is set to 100 ms

Hookflash-in Timing is set to max=1000 ms, min=150 ms

Hookflash-out Timing is set to 400 ms

No disconnect acknowledge

Ring Cadence is defined by CPTone Selection

Ring Cadence are [20 40] * 100 msec

Ringer Equivalence Number is set to 1



Lab 2-2: Configuring POTS Dial Peers Complete this lab activity to practice what you learned in the related module.

Activity Objective In this activity, you will configure POTS dial peers to establish locally terminated calls and calls to the PSTN. After completing this activity, you will be able to meet these objectives:

Configure dial peers for locally terminated calls

Determine the appropriate method of digit forwarding and manipulation

Verify basic call setup through debug commands

Use appropriate show and debug commands to monitor and troubleshoot the connections



POD-1

Phone1-17941/61

HQ-1

BR-1

PSTN Phone-1 PSTN Phone-xPhone5-x7941/61

Phone1-x7941/61

HQ-2

BR-2

POD-x

T1/E1

Phone5-17941/61


PSTN

Serial

IP WAN

FXS FXS

Phone4-11102

Phone3-xX101

Phone4-xX102

Phone3-11101

Phone2-17941/61

Phone2-x7941/61

Two-Pod Group

Lab 2-2: Configuring POTS Dial Peers


One PSTN device (instructor-controlled)

Local telephones




Dial Peer Commands

Command Description

debug vpm signal Displays real-time voice port module signaling; displays digits as they are received by the voice port

debug voip ccapi inout Displays real-time call control processing and call leg information

destination-pattern string Configures a telephone number for this dial peer

dial-peer voice tag pots Enters dial-peer configuration mode

forward-digits Specifies which digits to forward for voice calls

port port-number Configures the port for this dial peer

show call active voice Displays information on active calls

show dial-peer voice (tag)|(summary)

Displays dial-peer configuration information

show dialplan number number

Displays which dial peers are matched when a particular telephone number is dialed

show voice call summary Displays summary information on active calls

Job Aids There are no job aids for this lab activity.

Task 1: Configure POTS Dial Peers In this task, you will configure dial peers so that you can make calls between two telephones connected to your voice-enabled router.


Step 1 Using the “Classroom Dial Plan” table in the “Job Aids” topic, verify the telephone numbers that you will use for your local telephones. Note these numbers here.

HQ IP Phone 1 ____________________ HQ IP Phone 2 ____________________

HQ Analog Phone 1 ________________ HQ Analog Phone 2 __________________

BR IP Phone 1 ____________________ BR Analog Phone 1 __________________

Step 2 Configure POTS dial peers to enable calls between the locally terminated POTS telephones using their four-digit extension. Normally, you want the dial peer tag (name) to be as descriptive as possible (for example, “dial-peer 1201” for extension 1201, “dial-peer 911” for emergency services, and so on). Place calls in both directions to test the configuration.

Note Do not create dial peers 1 through 9, 97, 910, 9110, 9911, or 911 because these dial peers will be used in future labs.

Step 3 Use the appropriate show commands to view your newly configured dial peers.



Step 4 Place a call between local telephones and leave them both off hook. Use the appropriate show commands to view your active call. Find this information.

How many and what type of call legs were created? ____________

Which codec is being used for this call? ____________

What is the translated calling number? ___________ called number? ___________

Step 5 Use the debug command to see digits being collected by the voice port. Once debugging is turned on, place a call between telephones to view the digit collection.

Note Make sure terminal monitor is turned on to view the debug output.

Step 6 Verify and experiment with previously configured voice port parameters such as cptone and ring cadence. Listen to the distinctive rings you have configured. You may have to shut down and restart the voice port for the new ring cadence to take effect.


You have made calls between your locally terminated telephones in both directions.



Lab 2-3: Configuring VoIP Dial Peers Complete this lab activity to practice what you learned in the related module.

Activity Objective In this activity, you will establish basic VoIP connectivity between telephones in your pod and investigate the use of appropriate VoIP dial-peer parameters. After completing this activity, you will be able to meet these objectives:

Configure basic VoIP dial peers using the default parameters to process a call

Change the codec settings on VoIP dial peers and investigate how this change affects the ability to make calls

Determine the required bandwidth for a voice call using G.711 or G.729 codecs and test call quality

Configure codec negotiation



POD-1

Phone1-17941/61

HQ-1

BR-1


Phone1-x7941/61

HQ-2

BR-2

POD-x

T1/E1

Phone5-17941/61


PSTN

Serial

IP WAN

FXS FXS

Phone4-11102

Phone3-xX101

Phone4-xX102

Phone3-11101

Phone2-17941/61

Phone2-x7941/61

Two-Pod Group

Lab 2-3: Configuring VoIP Dial Peers


No new resources are required.




VoIP Commands

Command Description

codec codec-name Specifies which codec is to be used for calls matching this dial peer.

codec preference 1-14 codec-name

Configures one entry in the codec list under the voice class codec command. Repeat this command as many times as you need to specify codecs in this list.

debug voip ccapi inout Displays real-time call control processing and call leg information.

default parameter Sets the specified parameter back to its default setting. For example, default codec will set the dial peer to use the default codec for that device.

dial-peer voice tag voip Enters dial-peer configuration mode and specifies VoIP.

session target ipv4:x.x.x.x

Specifies the destination IP address for the gateway terminating a VoIP call.

show voice dsp Displays DSP usage.

show call active voice Displays information on active calls.


Displays dial-peer configuration information.


Displays which dial peers are matched when a particular telephone number is dialed.

voice class codec tag Enters voice class codec configuration mode.

voice-class codec tag Applies a predefined codec list to a dial peer. The tag must match the tag of the defined codec class.




Task 1: Configuring VoIP Dial Peers In this task, you will configure basic VoIP dial peers using the default parameters to process a call. You will verify configuration by placing calls across the IP network from your HQ to BR sites.


Step 1 Configure VoIP dial peers to reach the router-attached telephones connected to the equipment belonging to your pod BR and HQ routers using their extension numbers. In preparation, note the two telephone numbers for your HQ and BR here. Also note a valid IP address on the other router.

HQ telephone numbers ___________________ ______________________

HQ IP address __________________________

BR telephone numbers ___________________ ______________________

BR IP address __________________________

Step 2 Test your configuration by placing calls between the BR and HQ telephones.

Step 3 Configure VoIP dial peers to reach both of the router-attached telephones that are connected to the equipment belonging to your partner pod. In preparation, note the two telephone numbers for your partner here. Also note a valid IP address on the partner router.

Partner HQ telephone numbers___________________ ______________________

Partner HQ IP address __________________________

Partner BR telephone numbers___________________ ______________________

Partner BR IP address __________________________

Step 4 Test your configuration by placing calls to both of the partner telephones.

Step 5 Use show commands to verify the following:

Which dial peer will be matched when a specific number is dialed

Active call parameters

Which DSP resources are being used for the call

Step 6 Use debug commands to verify the following:

The calling number

The called number

Which dial peer was matched




You have placed calls to both of the partner telephones.

Task 2: Configure Codecs In this task, you will change the codec settings on VoIP dial peers and investigate how it affects the ability to make calls. When doing this lab, make sure that codecs are configured correctly on both sides of the connection before proceeding.


Step 1 Use show commands to verify the default codec setting and note it here.

Default codec___________________

Step 2 Change the codec on the HQ VoIP dial peer pointing to BR. Set the codec to g723r53.

Step 3 Place a call from one site to the other.

Step 4 Both HQ and BR should verify that the call was successful, and if so, which codec was used.

Was the call successful? ______________ Codec used _______________

Step 5 Change the codec on the BR VoIP dial peer pointing to HQ to match the HQ codec.

Step 6 Verify whether the codec was successful, and if so, which codec was used.

Was the call successful? ______________ Codec used _______________

Step 7 Change the codec setting back to default.


You have successfully configured the codec setting for VoIP calls.



Task 3: Calculate Bandwidth Requirements In this task, you will determine the required bandwidth for a voice call using G.711 or G.729 codecs, and test call quality.


Step 1 Using the bandwidth requirement concepts, determine the required bandwidth for G.711 and G.729 codecs using a 20-ms sample size, without header compression on these links.

G.711 PPP link ___________________________ G.711 Frame Relay link ____________________

G.729 PPP link ___________________________ G.729 Frame Relay link ____________________

Note Frame Relay and PPP have the same overhead.

Step 2 Change the codec on both HQ and BR to g711ulaw.

Step 3 Place a call from HQ to BR over the IP network.

Is the quality acceptable? If not, why not? _______________________________

Step 4 Leave the first call up and place a second call from HQ to BR over the IP network.

Is the quality acceptable? If not, why not? _______________________________

Step 5 Change the codec on both ends of the connection to g729r8.

Step 6 Repeat Steps 3 and 4 using the G.729 codec.

Has the quality improved? _______________

Why or why not? _______________________________________

Step 7 Change the codec on both HQ and BR back to the default setting.

What command did you use to reset to the default? __________________________


You can explain how and when to use RTP header compression.



Task 4: Configure Codec Negotiation As you saw in the previous labs, configuring a specific codec at the dial-peer level restricts that dial peer to responding with only a single codec choice during negotiation. At times, it is desirable to respond with a list of codecs to match the incoming call. For example, when a call is coming from the LAN segment, it may negotiate a G.711 codec for better voice quality because there is enough bandwidth to carry it. For a call coming into the router from a WAN segment, you may want to match a codec for compressed voice. For a single dial peer to match more than one codec, you must configure a list of codecs to negotiate. In this task, you will configure codec negotiation.


Step 1 Define a codec preference list. On HQ, select g711ulaw as the first choice and g729r8 as the second choice. On BR, select g729r8 as the first choice and g711ulaw as the second choice.

Step 2 Apply the codec list to the VoIP dial peer pointing to the IP address of your partner.

Step 3 Test calls in both directions. Use show commands to determine the order of preference for codec selection. Discuss the results with your partner.

Step 4 Remove the codec list and its application in the dial peer.

Step 5 Save your configuration.


You can explain how and when to configure codec negotiation.



Lab 2-4: Configuring Digital Voice Interfaces Complete this lab activity to practice what you learned in the related module.

Activity Objective In this activity, you will configure T1 CAS and PRI trunks to the PSTN on the HQ and BR gateway routers. After completing this activity, you will be able to meet these objectives:

Configure a T1 CAS interface for correct signaling, framing, and timeslot association

Configure a PRI interface for correct signaling, framing, and timeslot association

Configure the appropriate ports and dial peers to place calls to the PSTN



POD-1

Phone1-17941/61

HQ-1

BR-1


Phone1-x7941/61

HQ-2

BR-2

POD-x

Phone5-17941/61


PSTN

Serial

IP WAN

FXS FXS

Phone4-11102

Phone3-xX101

Phone4-xX102

Phone3-11101

Phone2-17941/61

Phone2-x7941/61

Two-Pod Group

T1/E1 T1/E1

CAS

ISDN

CAS

ISDN

Lab 2-4: Configuring Digital Voice Interfaces


A phone at the PSTN




T1 Interface Commands

Command Description

controller {t1 | e1} slot/subslot/port

Enters controller configuration mode

forward-digits {num-digit | all | extra}

Specifies which digits to forward for voice calls

framing {sf | esf} Selects the frame type for the E1 or T1 data line

linecode {ami | b8zs} Selects the line-code type for T1 or E1 lines

clear interface slot/port Resets the specified ISDN interface

clock source {[primary] line | internal | free-running}

Sets the clocking for individual T1 or E1 links

network-clock-participate {aim | slot | wic} slot-number

Allows the ports on a specified network module or VWIC to use the network clock for timing

ds0-group ds0-group-number timeslots <timeslot-list> type {e&m-delay-dial | e&m-fgd | e&m-immediate-start | e&m-wink-start | fxo-ground-start | fxo-loop-start | fxs-ground-start | fxs-loop-start} dtmf dnis

Specifies the DS0 time slots that make up a logical voice port on a T1 controller; also specifies the signaling type by which the router communicates and defines T1 channels for voice calls and the CAS method by which the router connects to the PBX or PSTN

isdn switch-type {country-specific-switch-type}

Defines the telephone company switch type

interface {bri | pri} interface-number

Enters interface configuration mode

isdn incoming-voice voice Routes all incoming voice calls to the modem and determines how they will be treated

isdn protocol-emulate {user | network}

Defines Layer 2 and Layer 3 user- or network-side emulation

prefix <string> Specifies the prefix of the dialed digits for a dial peer

pri-group timeslots timeslot-range [nfas_d {backup | none | primary {nfas_int number | nfas_group number | rlm-group number}} | service]

Specifies an ISDN PRI group on a channelized T1 or E1 controller.

network-clock-participate {aim | slot | wic} slot-number

Specifies which clock source to use for DSP clocking

show voice port [slot/port:ds0-group | summary]

Displays configuration information about a specific voice port or a summary of all voice ports

show controllers t1 number [bert] Displays information about the T1 links

Job Aids These job aids are available to help you complete the lab activity.

PSTN requirements:

— Framing = ESF

— Line coding = B8ZS



— Clock source = line

— Branch digital:

DS0 group 1 = timeslots 1 to 2, type E&M FGD

DS0 group 2 = timeslots 3 to 4, type FGD-EANA

— HQ digital = PRI timeslots 1 to 4, type National ISDN

Task 1: Configure the BR T1 CAS Trunk to the PSTN In this task, you will configure a T1 CAS trunk on the BR router to connect to the PSTN. In the process of configuring the T1 for voice calls, two logical voice ports will be created. You will be able to view these newly created digital voice ports with the same commands as for analog ports.


Step 1 Connect the PSTN device to the BR router T1 interface with a crossover T1 cable. Which port is your T1 cable plugged into? _________________

Step 2 Use the show controller T1 command to view the default settings for framing, line code, and clock source. Note these settings here.

Framing ______________________

Line code ______________________

Clock source ____________________

Step 3 Configure your T1 controller to complement the settings of the PSTN as shown in the Job Aids section. Use ESF framing and B8ZS line coding. Set the clock source to the PSTN. Verify that the settings match by checking that both controller LEDs are green. Use the show controller T1 command to view status and new settings.

Step 4 When the T1 is functional, create digital voice ports using the DS0-group command. Define CAS group 1 using channels 1 to 2 with E&M FGD signaling. Define CAS group 2 using channels 3 to 4 with FGD-EANA signaling. Use the show commands to verify the newly created digital voice ports. Fill in the information here.

How many voice ports were created? ____________

What are the voice port numbers? __________________

How many channels were created? _______________

Which command would you use to view the voice port and the channels? __________________________________________________________

What is the current status of these channels? ______________________

Step 5 Configure a POTS dial peer 1 with an incoming called number “.”, DID, and no port number.



Step 6 Configure a POTS dial peer 9 with an incoming called number “33X55YXY..” (where X = Pod # and Y = HQ or BR) and DID. Use the voice port previously defined by DS0 group 1. This dial peer will be used for incoming calls from the PSTN to the branch.

The following example is for the pod 1 BR router:

Router(config-dial-peer)# incoming called-number 33155212..

Step 7 Configure a POTS dial peer 97 using destination pattern “9[2-9]......”. Use the voice port previously defined by DS0 group 2. This dial peer will be used for seven-digit local dialing from the branch through the PSTN.

Step 8 Configure a POTS dial peer 910 using destination pattern “9[2-9]..[2-9]......”. Use the voice port previously defined by DS0 group 2. This dial peer will be used for 10-digit local dialing from the branch through the PSTN.

Step 9 Configure a POTS dial peer 9110 using destination pattern “91[2-9]..[2-9]......”. Use the voice port previously defined by DS0 group 2. This dial peer will be used for 11-digit long distance dialing from the branch to other pod phones through the PSTN.

Step 10 Test this call by dialing “9”. You should get a secondary dial tone from the PSTN. You can then dial your 3-, 7- or 10-digit number to complete the call. This is referred to as two-stage dialing.

Step 11 Change dial peers 97, 910, and 9110 so that users can dial “9” plus the 7, 10, or 11 digits without getting a secondary a dial tone. This is referred to as one-stage dialing.

What command did you use to enable one-stage dialing? ____________________

What other method could you have used? __________________________________

Step 12 Test this functionality by dialing “9” and the 10 or 11 digits and verify that you do not get a secondary dial tone.

Activity Verification You have completed this task when you attain these results:

You have verified the T1 CAS connection to the PSTN.

You have verified the existence of the newly created digital voice port or ports.

You have placed calls to the PSTN telephone over the CAS with both one-stage and two-stage dialing.



Task 2: Configure the HQ ISDN PRI Trunk to the PSTN In this task, you will configure an ISDN PRI trunk connection from HQ to the PSTN.


Step 1 Connect the PSTN device to the HQ router T1 interface with a crossover T1 cable. Which port is your T1 cable plugged into? _________________

Step 2 On the HQ router, ensure that the DSPs are correctly clocked.

Step 3 Specify the correct global ISDN switch type. Use primary-ni.

Step 4 Configure the controller of the T1 PRI PSTN trunk. Use time slots 1-4 and 24.

Step 5 Configure the ISDN signaling interface for inbound voice calls.

Step 6 Configure a POTS dial peer 1 with an incoming called number “.”, DID, and no port number.

Step 7 Configure a POTS dial peer 9 with an incoming called number “33X55YXY..” (where X = Pod # and Y = HQ or BR) and DID. Use the voice port previously created by the pri-group command during interface configuration mode. This dial peer will be used for incoming calls from the PSTN to the HQ.

The following example is for the pod 1 HQ router:

Router(config-dial-peer)# incoming called-number 33155111..

Step 8 Configure a POTS dial peer 97 using destination pattern “9[2-9]......”. Use the voice port previously created by the pri-group command during interface configuration mode. Make sure to strip the leading 9 before sending the call to the PSTN using the forward-digits 7 command. Also make sure that DID is enabled with the direct-inward-dial command. This dial peer will be used for seven-digit local dialing from the HQ through the PSTN.

Step 9 Configure a POTS dial peer 910 using destination pattern “9[2-9]..[2-9]……”. Use the voice port previously created by the pri-group command during interface configuration mode. Make sure to strip the leading 9 before sending the call to the PSTN using the forward-digits 11 command. Also make sure that DID is enabled with the direct-inward-dial command. This dial peer will be used for 10-digit local dialing from the HQ through the PSTN.

Step 10 Configure a POTS dial peer 9110 using destination pattern “91[2-9]..[2-9]……”. Use the voice port previously created by the pri-group command during interface configuration mode. Make sure to strip the leading 9 before sending the call to the PSTN using the forward-digits 11 command. Also make sure that DID is enabled with the direct-inward-dial command. This dial peer will be used for 11-digit long-distance dialing from the HQ through the PSTN.

Step 11 Test this functionality by dialing “9” and the 7- or 10-digit number between your HQ and BR or the 11-digit number to reach the phone of another pod.

Step 12 Use show commands to ensure a PRI channel has been seized.




You have verified the HQ T1 PRI connection to the PSTN.

You have verified the existence of the newly created digital voice ports.

You have placed calls to the PSTN telephone using the HQ ISDN PRI connection.



Lab 3-1: Implementing H.323 Gateways Complete this lab activity to practice what you learned in the related module.

Activity Objective In this activity, you will configure and verify the operation of H.323 on a gateway. After completing this activity, you will be able to meet these objectives:

Configure H.323 voice service parameters on the HQ and BR gateways

Use debug and show commands to monitor the status and progress of call setup procedures in an H.323 environment



POD-1

Phone1-17941/61

HQ-1

BR-1


Phone1-x7941/61

HQ-2

BR-2

POD-x

T1/E1

Phone5-17941/61


PSTN

Serial

IP WAN

FXS FXS

Phone4-11102

Phone3-xX101

Phone4-xX102

Phone3-11101

Phone2-17941/61

Phone2-x7941/61

Two-Pod Group

Lab 3-1: Implementing H.323 Gateways


Pod routers




H.323 Gateway Configuration Commands

Command Description

voice service voip Enters voice-service-VoIP configuration mode. This command is executed in global configuration mode.

gateway Enables gateway configuration. This command is executed in global configuration mode.

h323 Enables H.323 signaling for the VoIP service and enters H.323 configuration mode.

h225 timeout setup <secs> Configures the timeout value for the response of the outgoing SETUP message. This command is executed in global configuration mode.

show gateway Shows if the gateway is connected to the gatekeeper. This command is executed in user mode on the gateway.

show dial-peer voice tag Displays detailed information about the specified dial peer. This command is executed in user mode.

show dial-peer voice summary

Displays a summary of all active dial peers. This command is executed in user mode.

debug cch323 h225 Traces the state transition of the H.225 state machine based on the processed event. This command is executed in privileged mode.

debug voice dialpeer Displays default debug output for all active POTS dial peers. This command is executed in privileged mode.

debug voip dialpeer Displays default debug output for all active VoIP dial peers. This command is executed in privileged mode.

debug h225 q931 Displays H.225 Q931 debug information. This command is executed in privileged mode.

csim start called-number Initiates a test call using the gateway to verify call routing. This command is executed in privileged mode.




Task 1: Configure an H.323 Gateway In this task, you will configure voice service parameters on the HQ and BR gateways.


Step 1 Modify the default H.323 gateway operation of your HQ and BR routers as H.323 gateways.

Step 2 Change the H.225 setup timeout value to 3 seconds.

Step 3 Make sure you have dial peers that will route extension number calls correctly between the HQ and BR routers in your pod.

Step 4 Establish a voice call to a telephone in your pod. Use the show call active voice command to provide the information that follows:

How many and what type of call legs are established? _______________________

Calling number ________________________

Called number ________________________

Remote IP address _____________________

Remote UDP port ______________________

Step 5 Place another call to a telephone outside your own pod. Use the debug cch323 h225 command to provide the information that follows:

What is the source address of the call? ____________________________

What is the destination address of the call? ________________________


You have configured and verified that your gateway is configured for H.323.

The out from the show gateway command should resemble the following:

router# show gateway

H.323 ITU-T Version: 4.0 H323 Stack Version: 0.1

H.323 service is up

This gateway is not registered to any gatekeeper

Alias list (CLI configured)

E164-ID 1003

E164-ID 1001

E164-ID 1002

Alias list (last RCF) is empty

H323 resource thresholding is Disabled



Lab 3-2: Implementing SIP Gateways Complete this lab activity to practice what you learned in the related module.

Activity Objective In this activity, you will use SIP direct call procedures (UA to UA) to establish VoIP calls. After completing this activity, you will be able to meet this objective:

Configure dial peers to use SIP call control procedures to set up VoIP calls



POD-1

Phone1-17941/61

HQ-1

BR-1


Phone1-x7941/61

HQ-2

BR-2

POD-x

T1/E1

Phone5-17941/61


PSTN

Serial

IP WAN

FXS FXS

Phone4-11102

Phone3-xX101

Phone4-xX102

Phone3-11101

Phone2-17941/61

Phone2-x7941/61

Two-Pod Group

Lab 3-2: Implementing SIP Gateways


Pod routers

IP phones




SIP Gateway Configuration Commands

Command Description

voice service voip Enters VoIP configuration mode.

sip Enables SIP signaling for the VoIP service and enters SIP configuration mode.

session protocol sipv2 Specifies SIP to be the protocol used on the VoIP dial peer.

show sip-ua calls Displays active SIP calls.

debug ccsip options Displays various real-time SIP call information. This command is executed in privileged mode.


Task 1: Configure a Gateway for SIP In this task, you will configure your router to initiate calls with the router of your partner using SIP. For this activity, you will use SIP direct (UA to UA).


Step 1 On the HQ and BR routers, enable SIP services.

Step 2 Modify the existing VoIP dial peers that point to the telephones belonging to your partner to use direct call connect (UA to UA) SIP call control procedures when establishing voice calls. For direct calls, the IP address in the session target command will be a valid address of your HQ or BR router.

Step 3 Use the show call active voice command to verify that you now have SIP call legs when placing a call to a partner telephone.

Step 4 Enable SIP debugging and place a call between your telephone and a partner telephone. Observe the call setup, capabilities negotiation, and assignment of ports for the call.

Step 5 Investigate the status of SIP with variations of the show sip-ua command.


You have established voice calls between telephones connected to your routers by way of direct SIP call control procedures.

While a call from a local phone to a remote site is active, the show sip-ua calls command will display an active call. Your output should resemble the following:

HQ-1# show sip-ua calls

SIP UAC CALL INFO

Number of SIP User Agent Client(UAC) calls: 0



SIP UAS CALL INFO

Call 1

SIP Call ID : [email protected]

State of the call : STATE_ACTIVE (7)

Substate of the call : SUBSTATE_NONE (0)

Calling Number : 2818902001

Called Number : 1003

Bit Flags : 0x1212003A 0x100000 0x488

CC Call ID : 1

Source IP Address (Sig ): 10.10.10.1

Destn SIP Req Addr:Port : 10.10.10.2:5060

Destn SIP Resp Addr:Port: 10.10.10.2:56884

Destination Name : 10.10.10.2

Number of Media Streams : 1

Number of Active Streams: 1

RTP Fork Object : 0x0

Media Stream 1

State of the stream : STREAM_ACTIVE

Stream Call ID : 1

Stream Type : voice-only (0)

Negotiated Codec : g729r8 (20 bytes)

Codec Payload Type : 18

Negotiated Dtmf-relay : inband-voice

Dtmf-relay Payload Type : 0

Media Source IP Addr:Port: 10.10.10.1:18050

Media Dest IP Addr:Port : 10.10.10.2:16522

Orig Media Dest IP Addr:Port : 0.0.0.0:0

Number of SIP User Agent Server(UAS) calls: 1



Lab 4-1: Implementing Numbering Plans Complete this lab activity to practice what you learned in the related module.

Activity Objective In this activity, you will design a basic numbering plan composed of PSTN access codes, site codes, extensions, and dialing prefixes, and build out the numbering plan using dial peers. After completing this activity, you will be able to meet these objectives:

Design a private numbering plan

Use site codes to route calls between sites



POD-1

Phone1-17941/61

HQ-1

BR-1


Phone1-x7941/61

HQ-2

BR-2

POD-x

T1/E1

Phone5-17941/61


PSTN

Serial

IP WAN

FXS FXS

Phone4-11102

Phone3-xX101

Phone4-xX102

Phone3-11101

Phone2-17941/61

Phone2-x7941/61

Two-Pod Group

Lab 4-1: Implementing Numbering Plans


No new resources are required.




Numbering Plan Commands

Command Description

dial-peer voice tag voip Enters dial-peer configuration mode and specifies VoIP

session target ipv4:x.x.x.xSyntax Content Text style

Specifies the destination IP address for the gateway terminating a VoIP call


voice translation-rule rule-tag

Defines a voice translation rule for voice calls

rule precedence /match/ /replace/ [type {match-type replace-type} [plan {match-plan replace-plan}]]

Defines a rule within a voice translation rule

voice translation-profile profile-name

Specifies a translation profile for all incoming VoIP calls

translate {called | calling | redirect-called} translation-rule-number

Associates a translation rule with a voice translation profile

translation-profile {incoming | outgoing} name

Assigns a translation profile

test voice translation-rule number input-test-string [type match-type [plan match-type]

Tests the functionality of a translation rule



Job Aids These job aids are available to help you complete the lab activity.

Site Access Code and Site Numbers

Router Site Access Code Site Number

HQ1 8 11

BR1 8 12

HQ2 8 21

BR2 8 22

HQ3 8 31

BR3 8 32

HQ4 8 41

BR4 8 42

HQ5 8 51

BR5 8 52

HQ6 8 61

BR6 8 62

Telephone Numbers and Site Codes

Phone Telephone Number Site Code

HQ IP phone 1

HQ IP phone 2

HQ IP analog phone 1

HQ IP analog phone 2

BR IP phone

BR analog phone



Task 1: Configure a Numbering Plan for HQ and Branch Offices In this task, you will create a basic numbering plan for the HQ and branch offices. The HQ and branch sites should have different site codes so that users may dial either site using a site code and extension.


Step 1 Assign extensions to phones in your pod. Use the “Classroom Dial Plan” (at the beginning of this lab guide) for the extension numbers for your pod.

Step 2 Assign site codes to the HQ and branch sites. Use the “Site Access Code and Site Numbers” table (in the Job Aids section of this lab) for site access codes for your pod.

Step 3 Configure dial peers on the HQ and BR routers according to the numbering plan.

Note Use dial peers and voice translation patterns to simplify the implementation.


Users can call other users at the same site using the extension numbers and they can call remote site users using a combination of the site access code and the appropriate site number to create a unique site code for each site.



Lab 4-2: Implementing PSTN Dial Plans Complete this lab activity to practice what you learned in the related module.

Activity Objective In this activity, you will implement a proper PSTN dial plan to ensure correct DNIS, ANI, and TON type presentation for inbound and outbound PSTN calls on a gateway. After completing this activity, you will be able to meet these objectives:

Configure dial peers for inbound and outbound PSTN routing

Configure voice translation rules and profiles for correct ANI and DNI digit manipulation



POD-1

Phone1-17941/61

HQ-1

BR-1


Phone1-x7941/61

HQ-2

BR-2

POD-x

T1/E1

Phone5-17941/61


Serial

IP WAN

FXS FXS

Phone4-11102

Phone3-xX101

Phone4-xX102

Phone3-11101

Phone2-17941/61

Phone2-x7941/61

Two-Pod Group

PSTN

Lab 4-2: Implementing PSTN Dial Plans


IP phones

PSTN phones




Digit Manipulation Commands

Command Description

digit-strip Strips all the digits that explicitly match a POTS dial peer. Digit stripping is enabled by default on POTS dial peers.

prefix digits Specifies the prefix of the dialed digits for a dial peer.

forward-digits [0-32]|all|extra Specifies which digits to forward for voice calls.

num-exp dialed-digits substitution

Defines how to expand a telephone extension number into a particular destination pattern.

voice translation-rule rule tag Defines a voice translation rule for voice calls.

rule precedence /match/ /replace/ [type {match-type replace-type} [plan {match-plan replace-plan}]]

Defines a rule within a voice translation rule.

voice translation-profile profile-name

Specifies a translation profile for all incoming VoIP calls.

translate {called | calling | redirect-called} translation-rule-number

Associates a translation rule with a voice translation profile.

translation-profile {incoming | outgoing} name

Assigns a translation profile.

test voice translation-rule number input-test-string [type match-type [plan match-type]

Tests the functionality of a translation rule.

Call Routing and Path Selection Commands

Command Description

destination-pattern [+]string[T]

Specifies either the prefix or the full E.164 telephone number to be used for a dial peer.

incoming called-number [+]string[T]

Specifies a digit string that can be matched by an incoming call to associate the call with a dial peer.

answer-address [+]string[T] Specifies the full E.164 telephone number to be used to identify the dial peer of an incoming call.

direct-inward-dial Enables the DID call treatment for an incoming called number.

preference [0-9] Indicates the preferred order of a dial peer within a hunt group.

no dial-peer outbound status-check pots

Checks the status of outbound POTS dial peers during call setup and to disallow, for that call, any dial peers whose status is down.

Job Aid This job aid is available to help you complete the lab activity.

Voice configuration library at: http://www.cisco.com/en/US/products/ps6441/prod_configuration_guide09186a0080565f8a.html



Task 1: Configure Dial Peers In this task, you will configure dial peers for inbound and outbound PSTN routing.


Step 1 Verify that the controllers and voice ports are configured correctly.

Step 2 On the HQ and BR gateways, configure or verify the incoming PSTN dial peers have the incoming called number “33X55XXX..” format and DID. Use the port that connects to the PSTN for this dial peer.

Step 3 On the HQ and BR gateways, configure or verify these dial peers using the T1 PSTN trunk:

Local calls:

— Dial peer 97, destination pattern “9[2-9]......”

— Dial peer 910, destination pattern “9[2-9]..[2-9]......”

National calls: dial peer 9110, destination pattern “91[2-9]..[2-9]......”

International calls: dial peer 9011, destination pattern “9011T”

Services:

— Emergency services:

Dial peer 911, destination pattern “911”

Dial peer 9911, destination pattern “9911”

— Directory services: dial peer 411, destination pattern “9411”

— Repair services: dial peer 611, destination pattern “9611”


You can place and receive PSTN calls.

Callers can access emergency services by dialing “911” or “9911”.

Callers can access directory and repair services by dialing “411” and “611”.

Callers can place international calls by dialing “9011” plus any combination of digits.

Task 2: Configure Digit Manipulation In this task, you will configure voice translation rules and profiles for correct ANI and DNIS digit manipulation.


Step 1 Use the following configuration commands to remove the automatic number translation function that is used by Cisco Unified Communications Manager Express, where X is your pod number:



telephony-service

no dialplan-pattern 1 33X555X2.. extension-length 4

Step 2 On the HQ gateway, configure a voice translation profile pstn-in that performs these digit manipulations:

Any calling number starting with 33X should be prefixed with 91 so that a user may use the missing directory number to dial that number back.

The called DID numbers 33X55XX… should be cut down to the four-digit extension. For example, 3315511101 should be modified to 1101.

Step 3 Bind the voice translation profile pstn-in to the T1 PSTN trunk. The profile should be applied to incoming calls. Verify that inbound PSTN calls are working again.

Step 4 Configure a voice-translation profile pstn-out that performs this digit manipulation:

The calling number 1… should be prefixed with the gateway DID range 33X555. For example, if a call is placed from HQ phone 1-1, the calling number should be modified from 1101 to 3315551101.

Step 5 Bind the voice translation profile pstn-out to the T1 PSTN trunk. The ANI should now be correctly modified to include the DID range for outgoing calls.

Step 6 Complete the same exercise on the BR router by repeating Steps 1 through 5, and assign the pstn-in profile to voice port 0/0/0:1 and the pstn-out profile to voice port 0/0/0:2.


Inbound calls to HQ and BR should include 91 in the calling number. For example, a call from the PSTN 3345554101 to HQ-1 phone 1-1 should be displayed as 913345554101.

Outbound calls from HQ-1 should have the calling number 33155511XX.



Lab 4-3: Configuring Path Selection Complete this lab activity to practice what you learned in the related module.

Activity Objective In this activity, you will implement toll bypass and other path selection techniques. You will also experiment with two different configurations to control hunt capabilities. After completing this activity, you will be able to meet this objective:

Create a hunt group using the preference command and determine hunting behavior



POD-1

Phone1-17941/61

HQ-1

BR-1


Phone1-x7941/61

HQ-2

BR-2

POD-x

T1/E1

Phone5-17941/61


PSTN

Serial

IP WAN

FXS FXS

Phone4-11102

Phone3-xX101

Phone4-xX102

Phone3-11101

Phone2-17941/61

Phone2-x7941/61

Two-Pod Group

Primary PathBackup Path

Lab 4-3: Configuring Path Selection


One PSTN device per pod

One telephone per pod




Hunt Group Commands

Command Description

debug vpm signal Displays real-time voice port module signaling; displays digits as they are received by the voice port


dial-peer hunt 0-7 Specifies a hunt selection order for dial peers

dial-peer voice tag pots Enters dial-peer configuration mode

forward-digits Specifies which digits to forward for voice calls

port port-number Specifies the port for a dial peer

preference 0-9 Specifies the preferred order of a dial peer within a hunt group

huntstop Disables all dial-peer hunting if a call fails on the specified dial peer

show call active voice Displays information on active calls


Displays dial-peer configuration information.


Displays which dial peer is matched when a particular telephone number is dialed

show voice call summary Displays summary information on active calls


Voice configuration library at: http://www.cisco.com/en/US/products/ps6441/prod_configuration_guide09186a0080565f8a.html

Task 1: Configure Hunt Groups In this task, you will configure a hunt group to send calls to both of your locally terminated telephones using the same number.


Step 1 For the hunt group number, you will use the same first three digits as your telephones are now configured for. The fourth digit will be 9. For example, if your telephone numbers are 1103 and 1104, you will use 1109 as the hunt group number. Write your hunt group number here.

Hunt group number ______________________

Step 2 Ensure that both telephones are still connected to your router voice ports.

Step 3 Configure additional dial peers for your local FXS ports to reach both voice ports using the same new hunt group number for each dial peer. Do not edit existing dial peers for this activity because they will be needed for future labs. This means that



you will have two new dial peers, each for the same hunt group number (such as 1109), assigned to each of your FXS ports.

Step 4 To properly test the hunt behavior between your two phones, you will use the PSTN telephone. To reach the hunt group number from the PSTN phone, dial the seven-digit PSTN number of the hunt group number. Repeat this many times, taking note of any patterns as to how calls are allocated to each telephone. The default behavior at this point will be random choosing of a port. However, to demonstrate this randomness with only two telephones, you may have to make a number of calls.

Step 5 One method that is used to control the order of hunting is through the use of the preference command. What is the default setting for preference on a dial peer? _____________

Which command did you use to verify the setting? _______________________

Change the preference on one of your hunt dial peers to 1. Which setting is preferred, the 0 or 1 setting? ___________________

Test the hunt group again by calling several times, making note of which telephone rings first. Is this what you expected? Now take the preferred telephone off hook and dial the hunt group number again. Did the other telephone ring? ________________

On the dial peer with the preference set to 0, change the preference to 2. Which telephone should always ring first? Test the hunt group again. Is the outcome what you expected? _______________________________________________________

Step 6 You can configure hunt behavior for all dial peers globally with the dial-peer hunt command. Before changing this setting, find out what the default setting is and note it here.

Default dial-peer hunt setting _______________________

Which command did you use to view this? ___________________________

Configure the dial-peer hunt setting to 7. How do you expect your hunt group to choose which telephone to ring now? Test the hunt group again. Is the outcome what you expected? ________________________________________________________

Step 7 Delete the dial peers you created for the hunt groups.


You can control your hunt group behavior and explain how each command works.

Task 2: Configure Redundant Calling Paths In this task, you will configure redundant calling paths between your BR and HQ gateways.




Step 1 In case the WAN is down, you will need a dial peer that points directly to the PSTN. Create a dial peer that points to the PSTN but that will not be used unless the IP WAN is down or congested. This enables redundant calling paths between sites.

Step 2 Test that calls are going across the primary path (the WAN) between HQ and BR.

Step 3 Shut down the IP connection between the HQ and BR and test that calls are going across the PSTN.

Step 4 Bring the serial port back up.


The ISDN path is being used only when the WAN is down or congested.

Task 3: Configure TEHO In this task, you will configure TEHO to the PSTN phone of your partner.


Step 1 To configure TEHO, you will create a dial peer to route PSTN calls to your partner pod PSTN telephone. Write down the number of your partner pod PSTN phone.

What is your partner pod PSTN phone number? ________________________

Step 2 Create a dial peer that will route calls to your partner pod PSTN phone through the IP WAN instead of the PSTN.

Step 3 Test by placing a call to the PSTN phone of your partner.


The WAN path is being used when you call the PSTN phone of your partner.



Lab 4-4: Implementing Calling Privileges Complete this lab activity to practice what you learned in the related module.

Activity Objective In this activity, you will implement calling privileges on a gateway using COR. After completing this activity, you will be able to meet these objectives:

Prepare the dial peers for COR and create COR labels

Create COR lists and assign members

Assign COR lists to the appropriate dial peers



POD-1

Phone1-17941/61

HQ-1

BR-1


Phone1-x7941/61

HQ-2

BR-2

POD-x

T1/E1

Phone5-17941/61


PSTN

Serial

IP WAN

FXS FXS

Phone4-11102

Phone3-xX101

Phone4-xX102

Phone3-11101

Phone2-17941/61

Phone2-x7941/61

Two-Pod Group

Lab 4-4: Implementing Calling Privileges


IP phones

PSTN phone




COR Commands

Command Description

dial-peer cor custom Specifies that named class of restrictions (COR) apply to dial peers

name cor-name Creates a named COR

dial-peer cor list list-name

Defines a COR list name

corlist incoming cor-list-name

Specifies the COR list to be used when a specified dial peer acts as the incoming dial peer

corlist outgoing cor-list-name

Specifies the COR list to be used by outgoing dial peers


Phone number for international calls: 90113337775555

Task 1: Create COR Labels In this task, you will configure COR labels. Four different classes will be required: emergency, local, national, and international.


Step 1 On the HQ gateway, verify that you have a unique dial peer for local calls, national calls, international calls, and emergency calls.

Step 2 Create these COR labels:

local

ld

intl

911

int


Use the show dial-peer cor command to verify your configuration. Your output should resemble the following example:

router# show dial-peer cor

Class of Restriction

name: 911

name: local

name: ld



name: intl

name: int

Task 2: Configure COR Lists In this task, you will create COR lists and assign members.

Complete these steps:

Step 1 Configure the incoming COR lists and members:

key-int: int

key-local: int, local, 911

key-ld: int, local, 911, ld

key-intl: int, local, 911, ld, intl

Step 2 Configure the outgoing COR lists and members:

lock-911: 911

lock-int: int

lock-local: local

lock-ld: ld

lock-intl: intl


Use the show dial-peer cor command to verify your configuration. Your output should resemble this.

router# show dial-peer cor

Class of Restriction

name: 911

name: int

name: local

name: ld

name: intl

COR list key-int

member: int

COR list key-local

member: local

member: 911

member: int



COR list key-ld

member: local

member: 911

member: int

member: ld

COR list key-intl

member: local

member: 911

member: int

member: ld

member: intl

COR list lock-int

member: int

COR list lock-911

member: 911

COR list lock-local

member: local

COR list lock-ld

member: ld

COR list lock-intl

member: intl



Task 3: Configure Dial Peers to Use COR Lists In this task, you will assign COR lists to the appropriate dial peers. A user with the local class should not be able to place any national or international calls, a user with the national class should not be able to place any international calls, and a user with the international class should be able to place any call.

Complete these steps:

Step 1 Assign these outgoing COR lists to the appropriate PSTN dial peers:

lock-int

lock-local

lock-ld

lock-911

lock-intl

Step 2 Apply the COR list to the individual analog phones. Assign the incoming national COR list to the line of phone Phone3-1 and the incoming international COR list to the line of phone Phone4-1.

Step 3 (Optional) Apply the COR list to the individual ephone-dns. Assign the incoming national COR list to the line of phone Phone1-1 and the incoming international COR list to the line of phone Phone2-1.

Step 4 Create the COR members and COR lists on the BR router and apply the same method to the IP and analog phones.


You cannot place international calls using Phone1-1, Phone3-1, and Phone5-1.

You can call any PSTN destination using Phone2-1, Phone4-1, and Phone6-1.




Complete this lab activity to practice what you learned in the related module.

Activity Objective In this activity, you will configure the corporate gatekeeper Corp-GK as a gatekeeper controlling two zones: HQ and BR. Call routing between San Jose and Chicago will be done via the gatekeeper. After completing this activity, you will be able to meet these objectives:

Configure an H.323 gatekeeper to support multiple local zones

Register gateways at the gatekeeper

Test the gatekeeper configuration and gateway registration



(7941/61 or PC with NetMeeting)POD-1

Phone1-17941/61

HQ-1

BR-1


Phone1-x7941/61

HQ-2

BR-2

POD-x

T1/E1

Phone5-17941/61

PSTN

Serial

IP WAN

FXS FXS

Phone4-11102

Phone3-xX101

Phone4-xX102

Phone3-11101

Phone2-17941/61

Phone2-x7941/61

GK1

Two-Pod Group

GK2



Corporate gatekeeper Corp-GK

HQ phones

BR phones




Gatekeeper Configuration Commands

Command Description

gatekeeper Enters gatekeeper configuration mode

zone local zone-name domain-name [ras-IP-address]

Specifies a zone controlled by a gatekeeper

no shutdown Brings the gatekeeper online

zone prefix gatekeeper-name e164-prefix [blast | seq] [gw-priority priority gw-alias [gw-alias, ...]]

Adds a prefix to the gatekeeper zone list

gw-type-prefix type-prefix [hopoff gkid1] [hopoff gkid2] [hopoff gkidn] [seq | blast] [default-technology] [gw ipaddr ipaddr [port]]

Adds a technology prefix to the gatekeeper configuration list

show gatekeeper calls Displays the status of each ongoing call of which a gatekeeper is aware

show gatekeeper status Displays the overall gatekeeper status, including the authorization and authentication status and zone status

show gatekeeper endpoints Displays the status of all registered endpoints for a gatekeeper

show gatekeeper gw-type-prefix Displays the gateway technology prefix table

show gatekeeper zone prefix [all]

Displays the zone prefix table

show gatekeeper zone status Displays the status of zones related to a gatekeeper

Gateway Configuration Commands

Command Description

gateway Enters gateway configuration mode and enables the gateway to register with a gatekeeper

h323-gateway voip interface Identifies this as a VoIP gateway interface

h323-gateway voip id gatekeeper-id {ipaddr ip-address [port]| multicast} [priority priority]

Defines the name and location of the gatekeeper for this gateway



Command Description

h323-gateway voip h323-id interface-id

Defines the H.323 name of the gateway, identifying this gateway to its associated gatekeeper

h323-gateway voip tech-prefix prefix

Defines the numbers that are used as the technology prefix that the gateway registers with the gatekeeper

session target ras Enables RAS signaling; meaning that a gatekeeper is consulted to translate the E.164 address into an IP address


Gatekeeper address: Fa0/0.103-10.X.3.102

Task 1: Configure Local Zones In this task, you will configure an H.323 gatekeeper to support multiple local zones. You will also configure appropriate dial plan information for the gatekeeper to resolve calls between sites San Jose and Chicago.


Step 1 On the HQ gateway, configure a gatekeeper using these parameters:

Local zone HQ, domain cisco.com, IP address of loopback 0 interface

Local zone BR, domain cisco.com

Step 2 Enable the gatekeeper process.


The HQ-GK gatekeeper is up and running with two local zones: HQ and BR.

The output from a show gatekeeper status command should resemble the following:

HQ-1# show gatekeeper status

Gatekeeper State: UP

Load Balancing: DISABLED

Flow Control: DISABLED

Zone Name: HQ

Zone Name: BR

Accounting: DISABLED

Endpoint Throttling: DISABLED

Security: DISABLED

Maximum Remote Bandwidth: unlimited



Current Remote Bandwidth: 0 kbps

Current Remote Bandwidth (w/ Alt GKs): 0 kbps

Task 2: Configure Zone and Technology Prefixes In this task, you will configure technology prefixes and interzone routing on the San Jose gatekeeper. Call routing will use the site extension range. For example, any call starting with 1 should be routed to the HQ zone, and any call starting with 2 should be routed to the BR zone.


Step 1 On the HQ gatekeeper, configure the zone prefix for the HQ zone. Any call to a San Jose extension starting with 11 should be routed to this zone.

Step 2 On the HQ gatekeeper, configure the zone prefix for the BR zone. Any call to a Chicago extension starting with 12 should be routed to this zone.

Step 3 Configure a default gateway technology prefix of 1#.


The zone prefixes for the HQ and BR zones are configured correctly.

The default technology prefix is 1#.

The output from a show gatekeeper zone prefix command should resemble the following:

HQ-1# show gatekeeper zone prefix

ZONE PREFIX TABLE

=================

GK-NAME E164-PREFIX

------- -----------

HQ 11..

BR 12..

Task 3: Configure Gateways to Register with a Gatekeeper In this task, you will configure the gateway in Chicago to register with the gatekeeper in San Jose.


Step 1 Using the bulleted items here, configure your router as an H.323 gateway. Analyze the debug command output to observe the interactions between your router and the gatekeeper. These tasks are necessary to register with a gatekeeper:

Configure your router to be a gateway.



Specify which interface IP address will be used to register with the gatekeeper and also specify the identity of the gatekeeper. Use your Ethernet interface for gateway configuration.

Specify an H.323 ID for your gateway by combining “gw” with your pod and router number. For example, if you were pod 1 with the HQ router, your H.323 ID would be gwP1HQ.

Use the show gateway command on your router to verify that you have registered with the gatekeeper.

What is the gatekeeper name you have registered with? ______________________

Under the CLI alias list, which numbers register with the gatekeeper? ___________

Step 2 On the HQ router, configure the H.323 gateway process. The gateway should register with zone HQ and technology prefix 1#. Use the loopback interface as the H.323 source interface and ensure that you configure a proper H.323 gateway ID.

Step 3 Connect to the gatekeeper and use variations of the show gatekeeper command to verify that your router is registered and that your router has registered the destination patterns from your POTS dial peers.

Which IP address is registered to be used for calls to your device? ______________

Which port number is being used for call signaling? __________________________

What is the H.323 ID of your router? ________________________________

Which zone is your gateway part of? __________________________

Step 4 Reconfigure the existing VoIP dial peers that point to Chicago. Instead of routing directly to Chicago, the call should be routed to the corporate gatekeeper Corp-GK. Remove the redundant second VoIP dial peer.

Step 5 Create a dial peer to use RAS for all calls to destinations outside the zone.

Step 6 Ensure that Cisco Unified Communications Manager Express does not register any ephone-dn with the gatekeeper.

Step 7 On the BR router, configure the H.323 gateway process. The gateway should register with zone BR. Use the loopback interface as the H.323 source interface and ensure that you configure a proper H.323 gateway ID.

Step 8 Create a dial peer to use RAS for all calls to destinations outside the zone.


The HQ gateway is registered at the gatekeeper with a technology prefix of 1#.

Calls between the San Jose and Chicago sites are possible via the gatekeeper.

The output of a show gatekeeper endpoints command should resemble the following:

HQ-1# show gatekeeper endpoints

GATEKEEPER ENDPOINT REGISTRATION

================================



CallSignalAddr Port RASSignalAddr Port Zone Name Type Flags

--------------- ----- --------------- ----- --------- ---- -----

10.1.250.101 1720 10.1.250.101 58963 HQ H323-GW

H323-ID: GW-P1HQ

E164-ID: 1101

E164-ID: 1102

Voice Capacity Max.= Avail.= Current.= 0

10.1.250.102 1720 10.1.250.102 58306 BR VOIP-GW

H323-ID: GW-P1BR

E164-ID: 1201

E164-ID: 1202

Voice Capacity Max.= Avail.= Current.= 0

Total number of active registrations = 2



Lab 5-2: Implementing Gatekeeper-Based CAC Complete this lab activity to practice what you learned in the related module.

Activity Objective In this activity, you will configure the corporate gatekeeper to use CAC between SJC and CHI. After completing this activity, you will be able to meet these objectives:

Calculate the required zone bandwidth for a given scenario given call requirements

Configure zone bandwidth to enable proper CAC



POD-1

Phone1-17941/61

HQ-1

BR-1


Phone1-x7941/61

HQ-2

BR-2

POD-x

T1/E1

Phone5-17941/61


PSTN

Serial

IP WAN

FXS FXS

Phone4-11102

Phone3-xX101

Phone4-xX102

Phone3-11101

Phone2-17941/61

Phone2-x7941/61

Two-Pod Group

GK1 GK2

Lab 5-2: Implementing Gatekeeper-Based CAC


Corporate gatekeeper Corp-GK set up for call routing between San Jose and Chicago

San Jose IP phones

Chicago IP phones




Gatekeeper Configuration Commands

Command Description

bandwidth {interzone | total | session} {default | zone zone-name} bandwidth-size

Specifies the maximum aggregate bandwidth for H.323 traffic and verifies the available bandwidth of the destination gatekeeper.

For interzone and total bandwidth, the range is from 1 to 10,000,000. For session bandwidth, the range is from 1 to 5000.

show gatekeeper calls Displays the status of each ongoing call of which a gatekeeper is aware.

show gatekeeper status Displays the overall gatekeeper status, including the authorization and authentication status and zone status.

show gatekeeper endpoints Displays the status of all registered endpoints for a gatekeeper.

show gatekeeper gw-type-prefix

Displays the gateway technology prefix table.

show gatekeeper zone prefix [all]

Displays the zone prefix table.

show gatekeeper zone status

Displays the status of zones related to a gatekeeper.

Gateway Configuration Commands

Command Description

resource threshold [all] [high percentage-value] [low percentage-value]

Configures a gateway to report H.323 resource availability to its gatekeeper.

show call resource voice threshold

Displays the threshold configuration settings and status for an H.323 gateway.




Task 1: Calculate Bandwidth Requirements In this task, you will calculate the bandwidth requirements for one G.711 call from the San Jose site to the Chicago site.


Step 1 Calculate the bandwidth for the one G.711 call.


You have calculated the correct bandwidth for one G.711 call in a H.323 gatekeeper network.

Task 2: Configure Zone Bandwidth In this task, you will configure the correct zone bandwidth for calls from the San Jose zone to the Chicago zone.


Step 1 On the HQ gatekeeper, configure the interzone bandwidth for one G.711 call for the HQ zone and the BR zone.


The correct bandwidth is configured on the gatekeeper. Create a call from any phone in San Jose to a phone in Chicago. Use the debug h225 asn1 and the show gatekeeper status commands to verify correct bandwidth operation.

Place a second call between from Chicago to San Jose. The call should be rejected and routed over the PSTN.



Lab 6-1: Configuring a Cisco UBE Complete this lab activity to practice what you learned in the related module.

Activity Objective In this activity, you will configure a Cisco UBE to connect to an ISP. After completing this activity, you will be able to meet this objective:

Configure a Cisco UBE for protocol interworking and address hiding



POD-1

Phone1-17941/61

HQ-1

BR-1


Phone1-x7941/61

HQ-2

BR-2

POD-x

Phone5-17941/61


PSTN

IP WAN

FXS FXS

Phone4-11102

Phone3-xX101

Phone4-xX102

Phone3-11101

Phone2-17941/61

Phone2-x7941/61

Two-Pod Group

IPSIP

Lab 6-1: Configuring a Cisco UBE

IP


HQ-1 H.323 gateway router1

HQ-2 H.323 gateway router2




IP-to-IP Gateway Configuration Commands

Command Description

allow-connections from-type to to-type

Enables protocol interworking on an IP-to-IP gateway.

media [flow-around | flow-through]

Configures how media streams are handled on an IP-to-IP gateway in dial-peer, voice-class, or voice-service configuration mode.

codec transparent Enables codec capabilities to be passed transparently between endpoints on an IP-to-IP gateway. This can also be configured in a voice class.


Task 1: Configure a Cisco UBE for Protocol Interworking and Address Hiding

In this task, you will configure the San Jose gateway router to also act as a Cisco UBE. Calls to a SIP carrier will travel through the Cisco UBE, with H.323 being the internal call leg.


Step 1 On the San Jose gateway router, shut down the gatekeeper process.

Step 2 On the San Jose gateway router, enable interworking between H.323 and SIP.

Step 3 Ensure that the loopback 0 interface is used for H.323 signaling.

Step 4 Create a new outbound SIP VoIP dial peer that will route calls to the SIP carrier. The destination pattern should be a 9011T, G.711 mu-law codec.


You can place a call from San Jose to the international PSTN phone. Use the show call active voice brief command to verify H.323-to-SIP interworking.



Answer Key The correct answers and expected solutions for the activities that are described in this guide appear here.

Lab 2-1 Answer Key: Configuring Analog Voice Interfaces When you complete this activity, your configuration will be similar to the results here, with differences that are specific to your device or workgroup:

!

voice-port 0/2/1

cptone AU

ring cadence pattern AU

!



Lab 2-2 Answer Key: Configuring POTS Dial Peers When you complete this activity, your configuration will be similar to the results here, with differences that are specific to your device or workgroup:

HQ Router !

dial-peer voice 1103 pots

destination-pattern 1103

port 0/2/0

!



port 0/2/1

!

BR Router !



port 0/0/0

!



Lab 2-3 Answer Key: Configuring VoIP Dial Peers When you complete this activity, your configuration will be similar to the results here, with differences that are specific to your device or workgroup:

HQ !

voice class codec 1

codec preference 1 g711ulaw

codec preference 2 g729r8

!

dial-peer voice 1200 voip

destination-pattern 12..

voice-class codec 1

session target ipv4:10.1.4.102

!

BR !

voice class codec 1

codec preference 1 g711ulaw

codec preference 2 g729r8

!



voice-class codec 1


!



Lab 2-4 Answer Key: Configuring Digital Voice Interfaces Implementing CAS Trunks on BR

When you complete this task, your configuration will be similar to the results here, with differences that are specific to your device or workgroup:

Cisco IOS Configuration: T1 CAS network-clock-participate wic 0

network-clock-select 1 T1 0/0/0

!

controller T1 0/0/0

framing esf

linecode b8zs

clock source line

ds0-group 1 timeslots 1-2 type e&m-fgd

ds0-group 2 timeslots 3-4 type fgd-eana

!

Two-Stage Dialing dial-peer voice 3315511200 pots

incoming-called-number 33155112..

direct-inward-dial

port 0/0/0:1

!


destination-pattern 9[2-9]......

port 0/0/0:2

!


destination-pattern 9[2-9]..[2-9]......

port 0/0/0:2

!



port 0/0/0:2

!



One-Stage Dialing !



forward-digits 7

port 0/0/0:2

!



forward-digits 10

port 0/0/0:2

!



forward-digits 11

port 0/0/0:2

!

Implementing PRI Trunks on HQ When you complete this task, your configuration will be similar to the results here, with differences that are specific to your device or workgroup:

Cisco IOS Configuration: T1 PRI network-clock-participate wic 0

network-clock-select 1 T1 0/0/0

!

isdn switch-type primary-ni

!

controller T1 0/0/0

framing esf

linecode b8zs

pri-group timeslots 1-4

!

interface serial 0/0/0:23

isdn incoming-voice voice

!


incoming-called-number 33155111..

direct-inward-dial

port 0/0/0:23

!





forward-digits 7

port 0/0/0:23

!



direct-inward-dial

forward-digits 10

port 0/0/0:23

!



forward-digits 11

port 0/0/0:23

!


incoming-called-number .

direct-inward-dial

!



Lab 3-1 Answer Key: Implementing H.323 Gateways When you complete this activity, your configuration will be similar to the results here, with differences that are specific to your device or workgroup:

HQ voice service voip

h323

h225 timeout setup 3

!

gateway !

interface Loopback 0

ip address 10.1.250.101 255.255.255.0

h323-gateway voip interface

h323-gateway voip bind srcaddr 10.1.250.101

!

BR voice service voip

h323

h225 timeout setup 3

!

gateway !

interface Loopback 0

ip address 10.1.250.102 255.255.255.0





Lab 3-2 Answer Key: Implementing SIP Gateways When you complete this activity, your configuration will be similar to the results here, with differences that are specific to your device or workgroup:

HQ voice service voip

sip

!



session protocol sipv2


BR voice service voip

sip

!







Lab 4-1 Answer Key: Implementing Numbering Plans When you complete this activity, your configuration will be similar to the results here, with differences that are specific to your device or workgroup:

HQ voice translation-rule 1200 rule 1 /^812/ //

!

voice translation-profile strip-sites

translate called 1200

!


translation-profile outgoing strip-sites



!

BR voice translation-rule 1100 rule 1 /^811/ //

!



!







Lab 4-2 Answer Key: Implementing PSTN Dial Plans on Cisco IOS Gateways

When you complete this activity, your configuration will be similar to the results here, with differences that are specific to your device or workgroup:

HQ voice translation-rule 1000

rule 1 /^331551/ //

!

voice translation-rule 1001

rule 1 /.*/ /91\0/

!


rule 1 /11../ /331551\0/

!

voice translation-profile pstn-in

translate calling 1001


!

voice translation-profile pstn-out


!

voice-port 0/0/0:23

translation-profile incoming pstn-in

translation-profile outgoing pstn-out

!



incoming called-number 33155111..

direct-inward-dial

port 0/0/0:23

forward-digits 7

!




direct-inward-dial

port 0/0/0:23

forward-digits 10






direct-inward-dial

port 0/0/0:23

forward-digits 11

!


destination-pattern 9011T

port 0/0/0:23

prefix 011

!



no digit-strip

port 0/0/0:23

!



port 0/0/0:23

prefix 911

!



forward-digits 3

port 0/0/0:23

!



forward-digits 3

port 0/0/0:23

BR# voice translation-rule 1000

rule 1 /^331552/ //

!


rule 1 /.*/ /91\0/

!


rule 1 /12../ /331552\0/



!

voice translation-profile pstn-in



!

voice translation-profile pstn-out


!

voice-port 0/0/0:1

translation-profile incoming pstn-in

voice-port 0/0/0:2

translation-profile outgoing pstn-out

!



direct-inward-dial

port 0/0/0:1

!



port 0/0/0:2

forward-digits 7

!



port 0/0/0:2

forward-digits 10

!



port 0/0/0:2

forward-digits 11

!



port 0/0/0:2

prefix 011

!





no digit-strip

port 0/0/0:2

!



port 0/0/0:2

prefix 911

!



forward-digits 3

port 0/0/0:2

!



forward-digits 3

port 0/0/0:2



Lab 4-3 Answer Key: Configuring Path Selection When you complete this activity, your configuration will be similar to the results here, with differences that are specific to your device or workgroup:

Hunt Group Configuration !



port 0/2/0

!



preference 1

port 0/2/1

Redundant Path Configuration: HQ to BR !

voice translation-rule 1200 rule 1 /^812/ //

!



!

voice translation-rule 1300 rule 1 /^812/ /9133112/

!

voice translation-profile add-pstn


!





!


translation-profile outgoing add-pstn


preference 1

port 0/0/0:23



TEHO Configuration (Pod 1 to Pod 2) !




!



preference 1

forward-digits 11

port 0/0/0:23



Lab 4-4 Answer Key: Implementing Calling Privileges on Cisco IOS Gateways


COR Configuration dial-peer cor custom

name local

name intl

name ld

name 911

name int

!

dial-peer cor list lock-local

member local

!

dial-peer cor list lock-ld

member ld

!

dial-peer cor list lock-intl

member intl

!

dial-peer cor list lock-int

member int

!

dial-peer cor list lock-911

member 911

!

dial-peer cor list key-int

member int

!

dial-peer cor list key-local

member local

member 911

member int

!

dial-peer cor list key-ld

member local

member ld

member 911



member int

!

dial-peer cor list key-intl

member local

member intl

member ld

member 911

member int

!



corlist incoming key-ld

corlist outgoing lock-int

port 0/0/0

!



corlist incoming key-int

corlist outgoing lock-int

port 0/0/1

forward-digits 7

!


corlist outgoing lock-local


port 0/0/0:0

forward-digits 7

!


corlist outgoing lock-local


port 0/0/0:0

forward-digits 10

!


corlist outgoing lock-ld


port 0/0/0:0

forward-digits 11

!




corlist outgoing lock-intl


port 0/0/0:0

prefix 011

!


corlist outgoing lock-911


port 0/0/0:0

forward-digits 3

!


corlist outgoing lock-911


port 0/0/0:0

forward-digits 3

!

ephone-dn 1 dual-line

number 1001

description 13125551001

name HQ Phone1-1

corlist incoming ld

!


number 1002

description 13125551002

name HQ Phone2-1

corlist incoming intl

!



Lab 5-1 Answer Key: Configuring Basic Gatekeeper Functionality


Cisco IOS Configuration: Corp-GK (HQ-1) gatekeeper

zone local HQ cisco.com 10.1.250.101

zone local BR cisco.com enable-intrazone

zone prefix HQ 1...

zone prefix BR 2...

gw-type-prefix 1#* default-technology

bandwidth interzone default 384

no shutdown

Cisco IOS Gateway Configuration: HQ interface Loopback0

description Pod1 HQ loopback interface

ip address 10.1.250.101 255.255.255.255


h323-gateway voip h323-id GW-P1HQ

h323-gateway voip tech-prefix 1#


!

gateway

!



session target ras

!

!


number 1001 no-reg

!


number 1002 no-reg



Cisco IOS Gateway Configuration: BR interface Loopback0

description Pod1 BR loopback interface

ip address 10.1.250.102 255.255.255.255


h323-gateway voip id BR ipaddr 10.1.250.101

h323-gateway voip h323-id GW-P1BR

h323-gateway voip tech-prefix 1#


!



session target ras



Lab 5-2 Answer Key: Implementing Gatekeeper-Based CAC When you complete this activity, your configuration will be similar to the results here, with differences that are specific to your device or workgroup:

Cisco IOS Configuration: Gatekeeper gatekeeper

bandwidth interzone zone HQ 128

bandwidth interzone zone BR 128

no shutdown



Lab 6-1 Answer Key: Configuring a Cisco UBE When you complete this activity, your configuration will be similar to the results here, with differences that are specific to your device or workgroup:

Cisco UBE Configuration voice service voip

allow-connections h323 to sip

allow-connections sip to h323

!


rule 1 /^9011/ //

!

voice translation-profile sip-out


!

interface Loopback0

ip address 10.1.250.101 255.255.255.0


!


translation-profile outgoing sip-out



session target ipv4: 10.2.250.101

codec g711ulaw

!


destination-pattern .

incoming called-number 3315551...

direct-inward-dial

codec g711ulaw


Documents

Cvoice v60 Lab Guide