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CIPT1

Cisco Voice over IP

Version 6.0

Lab Guide

Editorial, Production, and Web Services: 02.15.08

The PDF files and any printed representation for this material are the property of Cisco Systems, Inc.,for the sole use by Cisco employees for personal study. The files or printed representations may not beused in commercial training, and may not be distributed for purposes other than individual self-study.


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DISCLAIMER WARRANTY: THIS CONTENT IS BEING PROVIDED AS IS. CISCO MAKES AND YOU RECEIVE NO WARRANTIES IN

CONNECTION WITH THE CONTENT PROVIDED HEREUNDER, EXPRESS, IMPLIED, STATUTORY OR IN ANY OTHER PROVISION OF

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WARRANTIES, INCLUDING WARRANTIES OF MERCHANTABILITY, NON-INFRINGEMENT AND FITNESS FOR A PARTICULAR

PURPOSE, OR ARISING FROM A COURSE OF DEALING, USAGE OR TRADE PRACTICE. This learning product may contain early release

content, and while Cisco believes it to be accurate, it falls subject to the disclaimer above.



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Table of Contents

Lab Guide 1

Overview 1Outline 1

Lab Topology 2

Job Aids 5

Dial Plan 5

Dial Plan Conventions 5PSTN Configuration 6

Lab 2-1: Configuring Analog Voice Interfaces 7Activity Objective 7

Visual Objective 7

Required Resources 7

Command List 8Task 1: Configure an Analog Voice Port 8Task 2: Verify Analog Port Settings 10

Lab 2-2: Configuring POTS Dial Peers 12Activity Objective 12Visual Objective 12


Command List 12Job Aids 13Task 1: Configure POTS Dial Peers 13

Lab 2-3: Configuring VoIP Dial Peers 15

Activity Objective 15

Visual Objective 15Required Resources 15Command List 15

Job Aids 16Task 1: Configuring VoIP Dial Peers 17Task 2: Configure Codecs 18Task 3: Calculate Bandwidth Requirements 19

Task 4: Configure Codec Negotiation 20Lab 2-4: Configuring Digital Voice Interfaces 21


Visual Objective 21

Required Resources 21Command List 21Job Aids 22

Task 1: Configure the BR T1 CAS Trunk to the PSTN 23Task 2: Configure the HQ ISDN PRI Trunk to the PSTN 25

Lab 3-1: Implementing H.323 Gateways 27Activity Objective 27

Visual Objective 27Required Resources 27

Command List 27

Job Aids 28

Task 1: Configure an H.323 Gateway 29

Lab 3-2: Implementing SIP Gateways 30Activity Objective 30Visual Objective 30


Task 1: Configure a Gateway for SIP 31Lab 4-1: Implementing Numbering Plans 33


Visual Objective 33



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


Task 1: Configure a Numbering Plan for HQ and Branch Offices 36

Lab 4-2: Implementing PSTN Dial Plans 37

Activity Objective 37Visual Objective 37Required Resources 37

Command List 37Job Aid 38Task 1: Configure Dial Peers 39

Task 2: Configure Digit Manipulation 39

Lab 4-3: Configuring Path Selection 41Activity Objective 41Visual Objective 41


Command List 41Job Aid 42Task 1: Configure Hunt Groups 42Task 2: Configure Redundant Calling Paths 43Task 3: Configure TEHO 44

Lab 4-4: Implementing Calling Privileges 45Activity Objective 45

Visual Objective 45Required Resources 45

Command List 45

Job Aid 46

Task 1: Create COR Labels 46

Task 2: Configure COR Lists 47Task 3: Configure Dial Peers to Use COR Lists 49

Lab 5-1: Configuring Basic Gatekeeper Functionality 50

Activity Objective 50Visual Objective 50Required Resources 50

Command List 51

Job Aid 52

Task 1: Configure Local Zones 52Task 2: Configure Zone and Technology Prefixes 53

Task 3: Configure Gateways to Register with a Gatekeeper 53

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

Task 2: Configure Zone Bandwidth 58

Lab 6-1: Configuring a Cisco UBE 59Activity Objective 59

Visual Objective 59


Command List 59Job Aids 60Task 1: Configure a Cisco UBE for Protocol Interworking and Address Hiding 60

Answer Key 61Lab 2-1 Answer Key: Configuring Analog Voice Interfaces 61Lab 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 64Lab 3-1 Answer Key: Implementing H.323 Gateways 67



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

Lab 3-2 Answer Key: Implementing SIP Gateways 68Lab 4-1 Answer Key: Implementing Numbering Plans 69Lab 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 79Lab 5-2 Answer Key: Implementing Gatekeeper-Based CAC 81Lab 6-1 Answer Key: Configuring a Cisco UBE 82



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



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CVOICE

Lab Guide

Overview

This guide presents the instructions and other information concerning the lab activities for thiscourse. 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



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

Lab TopologyThis 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 v6.03

POD-1

Phone1-17941/611101

HQ-1

BR-1

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

2201

Phone1-2

7941/612101

HQ-2

BR-2

POD-2

T1/E1

Phone5-1

7941/611201

(7941/61 or PC withNetMeeting)

PSTN

Seria

l

IP WAN

FXSFXS

Phone4-11104

Phone3-22103

Phone4-22104

Phone3-1

1103

Phone2-1

7941/61

1102

Phone2-2

7941/61

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.



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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

HeadquartersPod XGateway

HQX X Fa0/0.X02 10.X.2.101 HQX VoiceVLAN

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

VLAN

Fa0/0.198 3.X.0.1 Allows Telnetto switch fromPodX

Loopback 0 10.X.250.101 Used assourceinterface

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 trunkto PSTN:

PRI

Framing:ESF

Line code:B8ZS

Pod X Switch PodXSW X VLAN198 3.X.0.2 ManagementVLAN

VLANX02 HQX voiceVLAN

VLANX03 HQX dataVLAN

Branch Pod XGateway

BRX X Fa0/0.X05 10.X.5.102 BRX voiceVLAN

Fa0/0.X06 10.X.6.102 BRX dataVLAN


Loopback 0 10. X.250.102 Used assourceinterface

Branch Pod X

Switch

BRXSW X VLANX05 BRX voice

VLAN

VLANX06 BRX dataVLAN

GatekeeperPod X

GKX X Fa0/0. X03 10. X.3.102 HQX dataVLAN




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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 YGateway

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

VLAN

Fa0/0.Y03 10.Y.3.101 HQY DataVLAN

Fa0/0.198 3.Y.0.1 Allows Telnetto switch fromPodY

Loopback 0 10.Y.250.101 Used assourceinterface

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 trunkto PSTN: PRI

Framing: ESF

Line code:B8ZS

Pod Y Switch PodYSW Y VLAN198 3.Y.0.2 ManagementVLAN

VLANY02 HQY VoiceVLAN

VLANY03 HQY DataVLAN

Branch Pod YGateway

BRY Y Fa0/0.Y05 10.Y.5.102 BRY VoiceVLAN

Fa0/0.Y06 10.Y.6.102 BRY DataVLAN

Fa0/0.198 3.Y.0.3 Allows Telnetto switch fromPodY

Loopback 0 10. Y.250.102 Used assourceinterface

Branch Pod YSwitch

BRYSW Y VLANY05 BRY VoiceVLAN

VLANY06 BRY DataVLAN

GatekeeperPod Y

GKY Y Fa0/0. Y03 10. Y.3.102 HQY DataVLAN

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



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Job AidsThese 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 theextension number. Site codes will be assigned in the dial plan labs.



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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

.



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Lab 2-1: Configuring Analog Voice InterfacesComplete 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 thisactivity, 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

Phone1-1

7941/61

1101

HQ-1

BR-1

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

2201

Phone1-2

7941/61

2101

HQ-2

BR-2

POD-2

T1/E1

Phone5-1

7941/61

1201

(7941/61 or PC with

NetMeeting)

PSTN

Seria

l

IP WAN

FXS FXS

Phone4-11104

Phone3-22103

Phone4-22104

Phone3-1

1103

Phone2-1

7941/61

1102

Phone2-2

7941/612102

Phone6-1

1202

FXS

Phone6-2

2202

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



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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, andcadence setting.

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

ring cadence {pattern-number| definepulseinterval}

Specifies the ring cadence for an FXS voice port.

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

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

show voice portport-number(summary)

Views the voice port status and settings. Displays all defaultsettings for each port. Specify a particular voice port to view

only its settings. Use the summary option to view a summarytable 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.



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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 dialdigits 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?

____________



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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 summaryIN 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



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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 USAnalog 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



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Lab 2-2: Configuring POTS Dial PeersComplete 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

Visual Objective



POD-1

Phone1-1

7941/61

HQ-1

BR-1

PSTNPhone-1 PSTN Phone-xPhone5-x

7941/61

Phone1-x

7941/61

HQ-2

BR-2

POD-x

T1/E1

Phone5-17941/61


PSTN

Seria

l

IP WAN

FXSFXS

Phone4-1

1102

Phone3-xX101

Phone4-xX102

Phone3-1

1101

Phone2-1

7941/61

Phone2-x

7941/61

Two-Pod Group

Lab 2-2: Configuring POTS Dial Peers

Required Resources


One PSTN device (instructor-controlled)

Local telephones

Command List




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Dial Peer Commands

Command Description

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

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

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

dial-peer voice tagpots Enters dial-peer configuration mode

forward-digits Specifies which digits to forward for voice calls

portport-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 numbernumber

Displays which dial peers are matched when a particulartelephone 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.

Activity Procedure


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.



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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. Oncedebugging 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.

Activity VerificationYou have completed this task when you attain this result:

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



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Lab 2-3: Configuring VoIP Dial PeersComplete 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, youwill 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

Visual Objective



POD-1

Phone1-1

7941/61

HQ-1

BR-1

PSTNPhone-1 PSTN Phone-xPhone5-x

7941/61

Phone1-x

7941/61

HQ-2

BR-2

POD-x

T1/E1

Phone5-1

7941/61

(7941/61 or PC with

NetMeeting)

PSTN

Serial

IP WAN

FXSFXS

Phone4-11102

Phone3-xX101

Phone4-xX102

Phone3-1

1101

Phone2-1

7941/61

Phone2-x

7941/61

Two-Pod Group

Lab 2-3: Configuring VoIP Dial Peers

Required Resources


No new resources are required.

Command List




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VoIP Commands

Command Description

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

codec preference 1-14codec-name

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

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

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

dial-peer voice tagvoip Enters dial-peer configuration mode and specifies VoIP.

session targetipv4:x.x.x.x

Specifies the destination IP address for the gatewayterminating a VoIP call.

show voice dsp Displays DSP usage.

show call active voice Displays information on active calls.

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

Displays dial-peer configuration information.

show dialplan numbernumber

Displays which dial peers are matched when a particulartelephone 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 mustmatch the tag of the defined codec class.

Job Aids




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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.

Activity Procedure


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



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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 correctlyon both sides of the connection before proceeding.

Activity Procedure


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.



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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.

Activity Procedure


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.



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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 willconfigure codec negotiation.

Activity Procedure


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 ofpreference 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.



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Lab 2-4: Configuring Digital Voice InterfacesComplete 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

Visual Objective



POD-1

Phone1-1

7941/61

HQ-1

BR-1

PSTNPhone-1 PSTN Phone-xPhone5-x7941/61

Phone1-x

7941/61

HQ-2

BR-2

POD-x

Phone5-17941/61


PSTN

Seria

l

IP WAN

FXSFXS

Phone4-11102

Phone3-xX101

Phone4-xX102

Phone3-1

1101

Phone2-1

7941/61

Phone2-x

7941/61

Two-Pod Group

T1/E1 T1/E1

CAS

ISDN

CAS

ISDN

Lab 2-4: Configuring Digital Voice Interfaces

Required Resources


A phone at the PSTN

Command List




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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 moduleor VWIC to use the network clock for timing

ds0-group ds0-group-numbertimeslots 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 alogical voice port on a T1 controller; alsospecifies the signaling type by which the routercommunicates and defines T1 channels for voicecalls 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 modemand determines how they will be treated

isdn protocol-emulate {user |network}

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

prefix Specifies the prefix of the dialed digits for a dialpeer

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

Specifies an ISDN PRI group on a channelizedT1 or E1 controller.

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

Specifies which clock source to use for DSPclocking

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

Displays configuration information about aspecific voice port or a summary of all voiceports

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



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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 PSTNIn 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.

Activity Procedure


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.



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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 isreferred 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

notget a secondary dial tone.


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.



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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.

Activity Procedure


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 fromthe HQ through the PSTN.

Step 9 Configure a POTS dial peer 910 using destination pattern 9[2-9]..[2-9].Usethe 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 thePSTN 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.



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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.



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Lab 3-1: Implementing H.323 GatewaysComplete 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

Visual Objective



POD-1

Phone1-1

7941/61

HQ-1

BR-1


Phone1-x

7941/61

HQ-2

BR-2

POD-x

T1/E1

Phone5-1

7941/61

(7941/61 or PC with

NetMeeting)

PSTN

Seria

l

IP WAN

FXS FXS

Phone4-11102

Phone3-xX101

Phone4-xX102

Phone3-1

1101

Phone2-1

7941/61

Phone2-x

7941/61

Two-Pod Group

Lab 3-1: Implementing H.323 Gateways

Required Resources


Pod routers

Command List




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H.323 Gateway Configuration Commands

Command Description

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

gateway Enables gateway configuration. This command is executedin global configuration mode.

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

h225 timeout setup Configures the timeout value for the response of theoutgoing SETUP message. This command is executed inglobal configuration mode.

show gateway Shows if the gateway is connected to the gatekeeper. Thiscommand 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 voicesummary

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

debug cch323 h225 Traces the state transition of the H.225 state machinebased on the processed event. This command is executedin privileged mode.

debug voice dialpeer Displays default debug output for all active POTS dialpeers. 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 isexecuted 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.

Job AidsThere are no job aids for this lab activity.



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Task 1: Configure an H.323 Gateway

In this task, you will configure voice service parameters on the HQ and BR gateways.

Activity Procedure


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



36/88


Lab 3-2: Implementing SIP GatewaysComplete 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

Visual Objective



POD-1

Phone1-1

7941/61

HQ-1

BR-1


Phone1-x

7941/61

HQ-2

BR-2

POD-x

T1/E1

Phone5-17941/61


PSTN

Seria

l

IP WAN

FXSFXS

Phone4-11102

Phone3-xX101

Phone4-xX102

Phone3-1

1101

Phone2-1

7941/61

Phone2-x

7941/61

Two-Pod Group

Lab 3-2: Implementing SIP Gateways

Required Resources


Pod routers

IP phones

Command List




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SIP Gateway Configuration Commands

Command Description

voice service voip Enters VoIP configuration mode.

sip Enables SIP signaling for the VoIP service and enters SIPconfiguration 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. Thiscommand is executed in privileged mode.

Job Aids


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).

Activity Procedure


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



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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



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Lab 4-1: Implementing Numbering PlansComplete 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. Aftercompleting this activity, you will be able to meet these objectives:

Design a private numbering plan

Use site codes to route calls between sites

Visual Objective



POD-1

Phone1-1

7941/61

HQ-1

BR-1


Phone1-x

7941/61

HQ-2

BR-2

POD-x

T1/E1

Phone5-1

7941/61

(7941/61 or PC with

NetMeeting)

PSTN

Seria

l

IP WAN

FXS FXS

Phone4-11102

Phone3-xX101

Phone4-xX102

Phone3-1

1101

Phone2-1

7941/61

Phone2-x

7941/61

Two-Pod Group

Lab 4-1: Implementing Numbering Plans

Required Resources


No new resources are required.

Command List




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Numbering Plan Commands

Command Description

dial-peer voice tagvoip Enters dial-peer configuration mode and specifies VoIP

session targetipv4:x.x.x.xSyntax ContentText style

Specifies the destination IP address for the gatewayterminating a VoIP call


voice translation-rulerule-tag

Defines a voice translation rule for voice calls

ruleprecedence/match//replace/ [type {match-type replace-type} [plan{match-plan replace-

plan}]]

Defines a rule within a voice translation rule

voice translation-profileprofile-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 numberinput-test-string[type match-type[plan match-type]

Tests the functionality of a translation rule



41/88


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



42/88


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.

Activity Procedure


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.



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Lab 4-2: Implementing PSTN Dial PlansComplete 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 completingthis 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

Visual Objective



POD-1

Phone1-1

7941/61

HQ-1

BR-1


Phone1-x

7941/61

HQ-2

BR-2

POD-x

T1/E1

Phone5-1

7941/61

(7941/61 or PC with

NetMeeting)

Seria

l

IP WAN

FXS FXS

Phone4-11102

Phone3-xX101

Phone4-xX102

Phone3-1

1101

Phone2-1

7941/61

Phone2-x

7941/61

Two-Pod Group

PSTN

Lab 4-2: Implementing PSTN Dial Plans

Required Resources


IP phones

PSTN phones

Command List




44/88


Digit Manipulation Commands

Command Description

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

prefix digits Specifies the prefix of the dialed digits for a dialpeer.

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

num-exp dialed-digitssubstitution

Defines how to expand a telephone extensionnumber into a particular destination pattern.

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

ruleprecedence/match//replace/ [type {match-typereplace-type} [plan {match-planreplace-plan}]]

Defines a rule within a voice translation rule.

voice translation-profileprofile-name

Specifies a translation profile for all incoming VoIPcalls.

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-rulenumberinput-test-string[typematch-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.164telephone number to be used for a dial peer.

incoming called-number[+]string[T]

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

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

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

preference [0-9] Indicates the preferred order of a dial peer within ahunt 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, anydial 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_guide09186a0080565f8

a.html



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Task 1: Configure Dial Peers

In this task, you will configure dial peers for inbound and outbound PSTN routing.

Activity Procedure


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

Activity VerificationYou have completed this task when you attain these results:

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.

Activity Procedure


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:



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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.



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Lab 4-3: Configuring Path SelectionComplete 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 completingthis activity, you will be able to meet this objective:

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

Visual Objective



POD-1

Phone1-1

7941/61

HQ-1

BR-1


Phone1-x

7941/61

HQ-2

BR-2

POD-x

T1/E1

Phone5-17941/61

(7941/61 or PC with

NetMeeting)

PSTN

Seria

l

IP WAN

FXSFXS

Phone4-11102

Phone3-xX101

Phone4-xX102

Phone3-1

1101

Phone2-1

7941/61

Phone2-x

7941/61

Two-Pod Group

Primary PathBackup Path

Lab 4-3: Configuring Path Selection

Required Resources


One PSTN device per pod

One telephone per pod

Command List




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Hunt Group Commands

Command Description

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


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

dial-peer voice tagpots Enters dial-peer configuration mode

forward-digits Specifies which digits to forward for voice calls

portport-number Specifies the port for a dial peer

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

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

cvoice60lg secured

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