CCNA Voice Study Guide (IIUC 640-460)

CCNA Voice
®
Study Guide
CCNA Voice
®
Study Guide
Andrew Froehlich
Acquisitions Editor: Jeff Kellum
Development Editor: Jim Compton
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ISBN: 978-0-470-52766-5
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Library of Congress Cataloging-in-Publication Data
Froehlich, Andrew, 1977CCNA voice study guide (640-460) / Andrew Froehlich. — 1st ed.
p. cm.
ISBN-13: 978-0-470-52766-5
ISBN-10: 0-470-52766-8
1. Internet telephony—Examinations—Study guides. I. Title.
TK5105.8865.F76 2010
004.69'5—dc22
2009047259
TRADEMARKS: Wiley, the Wiley logo, and the Sybex logo are trademarks or registered trademarks of John
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Dear Reader,
Thank you for choosing CCNA Voice Study Guide. This book is part of a
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Acknowledgments
I’d like to thank the entire team Sybex assembled for their hard work and dedication in
putting this book together. I wish to acknowledge Jeff Kellum, my acquisitions editor,
for giving me the opportunity to write my fi rst book for Sybex. A big thanks to my
development editor, Jim Compton. Jim’s tireless effort helped to shape the book into a
much more readable format. I’d also like to thank my technical editor, Scott Morris.
Having a multi-CCIE like Scott edit the book gave me a big reassurance that it was
accurately written. Also, thanks to Dassi Zeidel, my production editor, and copy editor
Linda Recktenwald. As is common with many books, the copy editor’s timeline is always
shrinking because of slowdowns in authoring and other edits. Dassi and Linda were able to
crank out the copy editing in record time so it could be placed into the readers’ hands
on schedule.
Finally, I’d like to thank my family and friends for all of their support and
encouragement. The writing and editing of this book over the past year for me took
place in multiple locations around the world including the United States, Colombia, and
Thailand. In each of these countries, I had support of family and/or friends to keep me
motivated and inspired. Starting with those in the United States, I’d specifically like to
thank my mother and father, Ron and Elaine Froehlich, as well my Chicago friends,
including Angie Barbini, Matt and Fabiana Liska, Kevin and Ruth Ann McQuire, and
Sean and Heather Uhles. Also in Chicago, my friends and co-workers at the University of
Chicago Medical Center. In Colombia, I want to thank my dear friend Adriana Castro.
Finally, in Thailand, I want to thank Manta Jambanja and the School of Information
Technology staff at Mae Fah Luang University.
About the Author
Andrew Froehlich, CCNA, CCDA, CCNA-Voice, CCNP, CCSP, CCDP, F5 systems
engineer, is the president of West Gate Networks, a network and IT consulting fi rm
based in Chicago. Andrew also holds the position of network architect at the University
of Chicago Medical Center. In the past, Andrew has performed network design and
support for large companies, including State Farm Insurance and United Airlines. In
addition to having more than 12 years of network experience, he holds a degree in
Management Information Systems from Northern Iowa University and a master of business
administration degree from Northern Illinois University. He is also a freelance writer for
IT publications, including Network World magazine. Andrew’s most recent work is as a
professor of Network Architecture at Mae Fah Luang University in Chiang Rai, Thailand.
Contents at a Glance
Introduction
xxiii
Assessment Test
xxx
Chapter 1
Cisco Unified Communication Solutions
Chapter 2
Traditional Telephony
35
Chapter 3
Voice over IP (VoIP)
75
Chapter 4
Configuring the Network Infrastructure for Voice
113
Chapter 5
CUCM Express Installation and Basic Configuration
173
Chapter 6
CUCM Express Advanced Configuration
237
Chapter 7
Configuring Voice Gateways for POTS and VoIP
299
Chapter 8
Unity Express Overview and Installation
361
Chapter 9
Unity Express Configuration
415
Chapter 10
Introducing the SBCS Platform and Cisco
Configuration Assistant
463
Configuring Telephony Functions Using the Cisco
Configuration Assistant
493
Design and Configuration Using the
CCA Telephony Setup Wizard
533
About the Companion CD
563
Chapter 11
Appendix A
Appendix B
1
Glossary
567
Index
585
Contents
Introduction
xxiii
xxx
Assessment Test
Chapter
1
Cisco Unified Communication Solutions
Why Should We Bother Integrating Voice and Data Services?
Communications Enhancements
Cost Savings
Introducing the Cisco Unified Communications Manager Lineup
Cisco Unified Communications Manager
Cisco Unified Communications Manager Business Edition
Cisco Unified Communications Manager Express
Comparing the Communications Manager Alternatives
Introducing the Cisco Unity Lineup
Cisco Unity
Cisco Unity Connection
Cisco Unity Express
Introducing Cisco IP Phones and User Applications
Cisco 7900 Series IP Phones
Cisco 7900 Expansion Modules
Cisco 6900 Series IP Phones
Cisco 3900 Series IP Phones
Cisco IP Communicator
Cisco 500 Series IP Phones
Cisco Analog Telephony Adapter
Cisco VG224 and VG248 Series Voice Gateway
Additional Unified Communications Applications
Using Voice Gateways
Introducing the Cisco Unified Communications 500 Series
Choosing an IP Telephony Deployment Option
Single Site with Centralized Call Processing
Multisite with Centralized Call Processing
Clustering over the Wide Area Network
Multisite with Distributed Call Processing
Summary
Exam Essentials
Written Lab 1.1
Review Questions
Answers to Review Questions
Answers to Written Lab 1.1
1
2
2
3
3
4
5
5
7
7
8
8
9
11
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13
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14
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17
19
20
22
22
23
24
25
26
26
27
28
32
33
xii
Chapter
Chapter
Contents
2
3
Traditional Telephony
35
Understanding Analog Network Signaling
Loop Start Signaling
Ground Start Signaling
Analog Network Event Signaling
Comparing Analog and Digital Circuits
The Analog Signal
Analog Voice Interfaces
The Analog-to-Digital Conversion Process
Digital Voice Interfaces
Multiplexing
Time-Division Multiplexing
Statistical Time-Division Multiplexing
Private Phone Switching
The Key System
Private Branch Exchange
PSTN Numbering Plans
The International Numbering Plan
The North American Numbering Plan
Combining the NANP with the International
Numbering Plan
Summary
Exam Essentials
Written Lab 2.1
Review Questions
Answers to Review Questions
Answers to Written Lab 2.1
36
36
38
38
41
42
43
47
52
58
59
60
60
60
61
62
62
63
65
65
66
67
68
72
73
Voice over IP (VoIP)
75
Understanding the Unified Communications Model
The Infrastructure Layer
The Call Control Layer
The Applications Layer
The Endpoints Layer
A Closer Look at Voice Gateways
Using DSP Resources on Voice Gateways to Connect
a CUCM to the PSTN
Using Voice Gateways to Connect a CUCM to a PBX
Voice Gateway Dial Peers
Dial Peers and Call Legs
Comparing Voice Gateway Communication Protocols
An Overview of Voice and Video Transport Protocols
The Real-Time Transport Protocol
76
77
78
78
79
79
79
82
83
84
85
88
88
Contents
Chapter
4
xiii
Compressed RTP
Real-Time Transport Control Protocol
Comparing VoIP Endpoint Signaling Protocols
SCCP
SIP
Voice Signaling Protocols in Review
Comparing the Common Voice Codecs
G.711
G.729
G.729a
iLBC
Which Codec Is Right for You?
Calculating IP Voice Packet Sizes
Voice Packet Payload
Layer 2 Header Information
Layer 3 Header Information
Special Case Packet Additions
Calculating Bytes per Second
Calculating Bits per Second
Size Calculation Examples
Reducing Voice Packet Sizes
Examples of When to Use Specific Codecs
Summary
Exam Essentials
Written Lab 3.1
Review Questions
Answers to Review Questions
Answers to Written Lab 3.1
90
91
92
92
93
94
95
95
95
96
96
97
98
98
99
99
99
100
100
101
103
104
104
105
106
107
111
112
Configuring the Network Infrastructure for Voice
113
Power Options for IP Phones
Power Brick
Powered Patch Panel/Power Injector
Power over Ethernet Switch
Understanding and Configuring VLANs and Voice VLANs
An Overview of VLANs
Configuring VLANs
Configuring VLAN Trunks
Implementing Inter-VLAN Routing
Using the VLAN Trunking Protocol
Configuring and Verifying Voice VLANs
Introduction to Quality of Service (QoS)
Traffic Classification
Traffic Marking
114
114
115
116
123
123
124
126
130
138
145
147
148
149
xiv
Contents
Traffic Queuing
Identifying QoS Trust Boundaries
Auto-QoS Implementation Options
Configuring Other Link Efficiency Techniques
Compression Techniques
Link Fragmentation Interleaving (LFI)
Network Infrastructure Services for VoIP support
Configuring DHCP for Voice Functionality
Monitoring and Troubleshooting the DHCP Service
Configuring the Network Time Protocol
Summary
Exam Essentials
Written Lab 4.1
Hands-on Labs
Hands-on Lab 4.1: Setting Power Options on
PoE Ethernet Interfaces
Hands-on Lab 4.2: Configuring Voice and Data
VLANs and Switchport Assignment
Hands-on Lab 4.3: Setting Up VTP
Hands-on Lab 4.4: Configuring Auto-QoS
Hands-on Lab 4.5: Setting Up a DHCP Server
Review Questions
Answers to Review Questions
Answers to Written Lab 4.1
Chapter
5
CUCM Express Installation and Basic
Configuration
Understanding CUCM Express Licensing
IOS Licenses for Voice
CUCM Express Feature Licenses
Cisco Phone User Licenses
Cisco CUCM Express License Bundles
Cisco Voice IOS and CUCM Express Software Installation
Initial CUCM Express Configuration
Configuring CUCM Express as a TFTP Server
Configuring the Mandatory CUCM Express
System Settings
Configuring Ephone and Ephone-DNs
Making Your First Call Powered by CUCM Express
Basic Configuration Using the Telephony Service Setup Script
Basic Configuration Using the GUI
Enabling the GUI Interface
CUCM Express Web GUI Basics
149
149
150
153
153
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155
155
157
157
158
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160
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165
166
170
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176
177
182
182
185
192
194
198
203
203
207
Contents
Using CUCM Express Verification and Troubleshooting
Commands
Troubleshooting Cisco Phone Registrations
Determining the State of an Ephone
Summary
Exam Essentials
Written Lab 5.1
Hands-on Labs
Hands-on Lab 5.1: Configuring the CUCM
Express as a TFTP Server
Hands-on Lab 5.2: Configuring the CUCM Express
for Basic Phone Operation
Hands-on Lab 5.3: Enabling HTTP/HTTPS GUI
Administration on the CUCM Express
Review Questions
Answers to Review Questions
Answers to Written Lab 5.1
Chapter
6
CUCM Express Advanced Configuration
Configuring Key System and PBX DNs and Ephones
Configuring Key Systems
Configuring PBX Systems
Configuring Ephone Button Options
Configuring Telephony Service Features
How to Configure User Locale and Network Locale
Configuring the Date and Time Format
Configuring the System Message
Configuring a Local Directory
Configuring Voice Productivity Features
Call Forwarding
Call Transfer
Call Pickup
Call Parking
Hunt Groups
Intercom
Paging
Configuring Voice Access and Accounting Features
on the CUCM Express
Call Blocking
Call Detail Records
Configuring Music on Hold (MoH)
Using the Multicast MoH Route Command
Disabling Multicast MoH on a Per-Ephone Basis
xv
211
211
218
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224
225
226
226
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268
268
270
271
273
275
275
277
281
282
283
xvi
Contents
Summary
Exam Essentials
Written Lab 6.1
Hands-on Labs
Hands-on Lab 6.1: Configuring a Hunt Group
Hands-on Lab 6.2: Configuring a Call Parking Slot
Hands-on Lab 6.3: Configuring Multicast Paging
Hands-on Lab 6.4: Configuring Multicast MoH
Review Questions
Answers to Review Questions
Answers to Written Lab 6.1
Chapter
7
Configuring Voice Gateways for POTS and VoIP
Configuring Analog FXS and FXO Ports with Basic
Dial Peers
Configuring FXS Ports
Reviewing FXS Port Configuration and Status
Configuring POTS Dial Peers for the FXS Ports
FXS PLAR Configuration
Configuring FXO Ports
Reviewing FXO Port Configuration and Status
Configuring POTS Dial Peers for the FXO Ports
FXO PLAR Configuration
FXO CAMA Configuration
Configuring Digital T1 Ports
Configuring T1 CAS Ports
Configuring POTS Dial Peers for T1 CAS Ports
Configuring T1 PRI Ports
Configuring POTS Dial Peers for T1 PRI Ports
Configuring VoIP Dial Peers over WAN Connections
Dial-Plan Strategy
Understanding the Dial-Peer Decision-Making Process
The Selection Process for Outbound Dial Peers
Selection Process for Inbound Dial Peers
When All Else Fails: Dial-Peer 0
Dial-Peer Digit Manipulation
POTS Digit Manipulation Using Stripped Digits
POTS Digit Manipulation Using Prefixes
POTS Digit Manipulation Using Forward-Digits
POTS and VoIP Digit Manipulation Using Number
Expansion
POTS and VoIP Digit Manipulation Using Translation
Profiles
Understanding the Digit-Manipulation Hierarchy
283
284
285
286
287
288
288
289
290
295
297
299
300
302
305
308
309
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329
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335
335
336
337
338
340
344
Contents
Configuring a Trunk between Voice Gateways using
H.323 and SIP Trunks
H.323 Trunking
SIP Trunking
Summary
Exam Essentials
Written Lab 7.1
Hands-on Labs
Hands-on Lab 7.1: Configuring FXS Interfaces for
Two Analog Phones
Hands-on Lab 7.2: Configuring a T1 PRI Interface
Hands-on Lab 7.3: Configuring an H.323 Trunk
Hands-on Lab 7.4: Configuring Translation Profiles
Review Questions
Answers to Review Questions
Answers to Written Lab 7.1
Chapter
8
xvii
345
345
346
347
348
349
349
350
351
352
353
354
358
360
Unity Express Overview and Installation
361
Understanding Unity Express Voice Mail Features
Users/Subscribers
Groups
Mailbox Owner Features
Understanding Distribution Lists
Mailbox Caller Features
Unity Express Advanced User Functionality
VoiceView Express
Integrated Messaging
Understanding Unity Express Auto Attendant Scripting
Methods
Preinstalled Scripts
Editor Express
Unity Express Editor Application
Understanding Unity Express Interactive Voice Response
Understanding CUCM Express Licensing
AIM-CUE
NM-CUE
NM-CUE-EC
NME-CUE
Installation and Initial Configuration of Unity Express
on CUCM Express Routers
Configuring IP Unnumbered to Use Existing IP
Network for Unity Express Connectivity
362
362
363
364
366
367
368
368
369
370
370
371
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373
374
375
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378
xviii
Contents
Configuring a Separate IP Network for Unity
Express Connectivity
Configuring Dial Peers for Unity Express Functions
Configuring MWI Ephone-DNs
Upgrading Unity Express Software
Unity Express Setup Using the Initialization Wizard
Restoring Unity Express to Factory Default Settings
Step 1: Suspend Unity Express Services
Step 2: Restore Factory Defaults on Unity Express
Summary
Exam Essentials
Written Lab 8.1
Hands-on Labs
Hands-on Lab 8.1: Configuring IP Network
Connectivity Using the ip unnumbered Command
Hands-on Lab 8.2: Configuring Unity Express Dial Peers
Hands-on Lab 8.3: Configuring MWI Ephone-DNs
Review Questions
Answers to Review Questions
Answers to Written Lab 8.1
Chapter
9
380
382
384
386
397
401
401
402
402
403
404
405
405
406
407
408
412
413
Unity Express Configuration
415
Configuring Unity Express System Settings and
Voice Mail Defaults
Configuring System Settings
Configuring Voice Mail Default Settings
Creating Users, Groups, and Mailboxes
User Creation with Mailbox
Group Creation with Mailbox
Group Creation for Administrative Roles
Configuring Auto Attendant
Administrating the Auto Attendant Application
Modifying the Business Hours Schedule
Configuring the Holiday Schedule
Creating Custom Prompts Using the AvT
Configuring Message Notification
Administrating and Troubleshooting Unity Express
Synchronizing Information
Backing Up and Restoring Configurations
Running a Unity Express Trace
Summary
Exam Essentials
Written Lab 9.1
416
416
422
426
426
432
436
439
440
441
442
442
443
448
448
449
450
452
452
453
Contents
Hands-on Labs
Hands-on Lab 9.1: Viewing Real-Time Trace Logs
Hands-on Lab 9.2: Saving and Retrieving Trace Log Files
Review Questions
Answers to Review Questions
Answers to Written Lab 9.1
Chapter
10
Introducing the SBCS Platform and Cisco
Configuration Assistant
The Smart Business Communications System
The SBCS Components
Using the UC500 Series Platform out of the Box
Introducing the Cisco Configuration Assistant
CCA Requirements
CCA Limitations Per Site
Setting Up CCA for Supporting the UC500 Series Platform
Installing the CCA Software
Navigating with the CCA User Interface
Adding a New CCA Site
Summary
Exam Essentials
Written Lab 10.1
Review Questions
Answers to Review Questions
Answers to Written Lab 10.1
Chapter
11
Configuring Telephony Functions Using
the Cisco Configuration Assistant
Telephony Initialization
Configuring the Telephony Region Using CCA
Configuring Telephony Voice Features Using CCA
Configuring Voice System Options
Configuring Voice Network Options
Configuring SIP Trunk Options
Configuring Voice Features Options
Configuring User Extensions Options
Configuring Telephony Voice Mail Features Using CCA
Voice Mail Setup Options
Voice Mail Mailbox Options
Configuring Telephony Phone Groups Features Using CCA
Hunt Groups
Paging Groups
Pickup Groups
xix
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463
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509
xx
Contents
Configuring Telephony Schedules Using CCA
Business Hours
Night Service
Holiday
Configuring Telephony Auto Attendant Features Using CCA
Auto Attendant
Prompt Management
Script Management
Configuring Telephony Dial Plans Using CCA
Creating an Incoming Dial Plan
Creating an Outgoing Dial Plan
Summary
Exam Essentials
Written Lab 11.1
Review Questions
Answers to Review Questions
Answers to Written Lab 11.1
Appendix
A
Design and Configuration Using the
CCA Telephony Setup Wizard
CCA Telephony Setup Wizard Overview and Requirements
The Information-Gathering Meeting for CC-NAV Inc.
Configuring Networking Parameters Using the TSW
Configuring System Access
Configuring the System Locale
Configuring WAN/LAN Settings
Configuring User and Extension Parameters Using the TSW
Configuring Internal Dialing
Configuring Analog Station (FXS) Ports
Configuring Phone Users and Extensions
Configuring Hunt Groups
Configuring Auto Attendant Parameters Using the TSW
Defining the AA and Setting Working Hours
Defining AA Prompts and Actions
Managing Auto Attendant Prompts
Configuring Analog PSTN Trunk Parameters Using the TSW
Configuring Call Routing Parameters Using the TSW
Final Review and Applying the Configuration Using the TSW
Summary
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546
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556
558
560
561
Contents
Appendix
Glossary
Index
B
xxi
About the Companion CD
563
What You’ll Find on the CD
Sybex Test Engine
PDF of the Book
Adobe Reader
Electronic Flashcards
System Requirements
Using the CD
Troubleshooting
Customer Care
564
564
564
565
565
565
565
566
566
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585
Introduction
Welcome to CCNA Voice Study Guide, a comprehensive guide that covers everything you
need for Cisco’s new exam 640 - 460. For readers who are new to Cisco certifications, there
is a well-defi ned structure to the different levels that network administrators can achieve.
Cisco’s current certification structure has the following five levels of certification:
■
Entry level
■
Associate
■
Professional
■
Expert
■
Architect
This book is written for the associate level of certification. Cisco considers this level to
be the “apprentice or foundation level” for network administrators.
Cisco has recently broadened its associate-level certifications to include not only a
certification for routing and switching (CCNA) and design (CCDA) but also more targeted
associate-level certifications for security (CCNA Security), wireless (CCNA Wireless), and
voice (CCNA Voice). These new certifications target specific areas of Cisco technology
and are to be used as stepping-stones for the professional and expert levels of certification
that Cisco offers.
Cisco’s Voice Certifications
Cisco offers three distinct levels of voice certifications. The following diagram shows that
the CCNA Voice certification is a building block to the professional- and expert-level voice
certifications:
CCIE
Voice
CCVP
CCNA Voice
This book covers the CCNA Voice certification exam 640 - 460. As of the writing of this
book, the exam costs $250 USD. The exam tests your knowledge a great deal in areas both
theoretical and technically specific to Cisco hardware and software.
Once you achieve your CCNA Voice certification, you can choose to continue on the
voice path and achieve higher certifications, such as the CCNP Voice or the ultimate CCIE
Voice Expert. But even if you stop after achieving your CCNA Voice certification, you
will have demonstrated to your current or prospective employers that you have a sound
knowledge of the interoperations of voice and Cisco voice technologies. This assurance to
employers will make it easier for you to land that dream job you’ve always wanted!
xxiv
Introduction
What Skills Do You Need to Become CCNA Voice Certified?
To meet the CCNA Voice certification skill level, you must possess the following skills:
■
■
A thorough knowledge of analog and voice technologies, including but not limited to
FXS, FXO, T1/E1, voice trunks, voice packetization, codecs, transcoding, PBX, key
systems, and multiplexing
The ability to install, configure, and operate Cisco Unified Communications Manager
Express hardware and software. In addition, you must be able to install, configure,
and manage Unity Express hardware and software to work in coordination with the
CUCM Express.
How Do You Become CCNA Voice Certified?
There are two ways to become CCNA Voice certified. This book provides one method,
which is to pass the 640 - 460 exam. This is considered the CCNA Voice Commercial
track, which covers the CUCM Express hardware and software that are commonly found
in small and medium-size organizations.
The other way to obtain a CCNA Voice Certification is to study for and
pass the 642- 436 exam. This exam is known as the CCNA Voice Enterprise
track, which covers the CUCM hardware and software used in large
organizations. This exam is also a requirement for those pursuing their
CCVP certification.
It is critical that you get some hands- on experience with a router installed with the
CUCM Express software. It would be even more valuable to acquire an SBCS UC500
device that contains both CUCM Express functionality and Unity Express. In addition, you
can practice working with the Cisco Configuration Assistant software to set up the UC500,
which is a critical skill to have before attempting to pass the exam.
Finding UC500 hardware at a low cost can be very difficult. If you cannot afford to
purchase a system, you’ll be happy to know that I’ve worked hard to provide configuration
examples and screenshots throughout this book to help test takers learn what they need to
pass the 640 - 460 exam.
What Does This Book Cover?
This book covers everything you need to know in order to pass the CCNA 640 - 460 exam.
In addition to this book, having the ability to study and practice with CUCM Express/
Unity Express hardware and software will provide you the confidence to complete the
simulation questions found in the exam.
Introduction
xxv
You will learn the following information in this book:
■
■
■
■
■
■
■
■
Chapter 1 introduces you to the Cisco hardware and software lineup for small to
medium-size businesses as well as large enterprise organizations. In addition, you will
learn about the different deployment options you can use to design your voice network.
Chapter 2 provides you with the background covering traditional telephony. Topics
such as analog network signaling, analog interface types, the analog-to -digital
conversion process, multiplexing, and numbering plans are detailed to give you a firm
foundation in traditional voice terminology and processes.
Chapter 3 introduces you to Voice over Internet Protocol in a Cisco network
environment. This chapter covers topics such as the Cisco Unified Communications
Model, voice gateway purpose and components, dial peers and call legs, and voice
gateway and endpoint communication protocols. You will also read about the protocol
that is responsible for transporting voice over an IP network— RTP. Finally, you will be
introduced to some of the more popular voice codecs used with Cisco voice equipment
and shown how to calculate voice packet sizes.
Chapter 4 provides you with the core networking skills required to design, configure,
and operate IPT equipment on a Cisco IP network. Cisco endpoint power options
are covered so you can appropriately plan for powering your IP phones in office
deployments. The chapter then goes on to discuss the importance of segmenting voice
traffic from data using voice VLANs and trunks. The chapter then describes some QoS
techniques that can be implemented on a Cisco IP network for more reliable delivery of
time-sensitive traffic such as voice.
Chapter 5 exposes readers to CUCM Express licensing options required to operate a
Cisco voice system. The chapter then moves on to describe how to install and set up
CUCM Express software on compatible Cisco hardware. At the end of this chapter,
readers will know how to configure ephones and ephone-DNs to the point where a
phone call can be successfully made from one Cisco IP phone to another.
Chapter 6 dives into more complex CUCM Express techniques that show readers how
to set up voice features such as ephone button options, user and network locales, and
user directories. This chapter also covers configuration of voice-productivity features
including call forwarding, call parking, hunt groups, and paging. Then you will learn
about voice accessibility and accounting settings such as call blocking and call detail
records. Lastly, you will learn how to configure Music on Hold settings for both
unicast and multicast MoH.
Chapter 7 covers the design and configuration of voice gateways, including how
to configure analog and digital interfaces as well as POTs and VoIP dial peers. In
addition, the chapter covers how to develop a dial-plan strategy to provide a simple
and expandable dial plan for current and future growth. Finally, the chapter covers
the dial-peer decision-making process of a voice gateway and how to manipulate dial
strings for proper call routing.
Chapter 8 introduces you to the various features found in the Unity Express voice
mail system. Those features include users/groups, message waiting indicators, message
xxvi
Introduction
notification, Auto Attendant (AA), and Interactive Voice Response (IVR). You will
then learn how to install and configure the Unity Express software to work with the
CUCM Express. Once Unity Express can interoperate with the CUCM Express, you
will learn how to set up Unity Express using the Unity Express Initialization Wizard.
■
■
■
■
Chapter 9 covers how to configure Unity Express using the web GUI. Specifically, you
will learn how to configure system settings such as NTP, time zone, and DNS. You will
also learn how to create and modify user and group mailbox settings. You will also
learn how to set up and modify the Auto Attendant feature as well as learn different
tools used to create AA scripts. You will see how to configure message notification
configuration to allow users to be remotely notified and listen to voice messages when
the user is away from their desk. Finally, you will learn techniques an administrator
can use to maintain and troubleshoot Unity Express.
Chapter 10 is an introduction to the Cisco Smart Business Communications System
(SBCS) lineup, including the UC500 Series hardware, which is an ideal voice/data
platform for small to medium-size businesses. This chapter also introduces the Cisco
Configuration Assistant (CCA), which is an innovative GUI tool used to configure and
maintain SBCS devices.
Chapter 11 goes into how to configure a SBCS UC500 device using the CCA.
Specifically, the chapter shows readers how to configure telephony regions, network
options, SIP trunking, voice features, voice mail features, Auto Attendant options, and
dial plans.
Appendix A provides a real-world scenario designing and implementing a small office
with a Cisco SBCS UC500 using the Cisco Configuration Assistant Telephony Setup
Wizard.
How to Use This Book
The CCNA Voice Study Guide is designed to prepare a reader to pass the 640 - 460 exam to
achieve the associate-level certification in Cisco voice technologies. To get the most out of
this book, I recommend you use the following study method:
1.
Take the assessment test provided to you prior to Chapter 1 of this book. Try to
answer each question without looking at the answers and explanations found in the
back of the book. This should give you an indication of your skill level prior to reading
the book. Once you have completed the assessment test and graded yourself, take time
to carefully read over the explanations for any question you get wrong and note the
chapters in which the material is covered. This information should help you identify
sections of the book that you need to spend additional time on. Keep in mind, however,
that the book was designed for you to read each chapter in order. Much of the material
found in the chapters builds on knowledge learned from previous chapters.
2.
Prior to reading each chapter, make sure to review the test objectives listed at the
beginning. These objectives are what the exam taker must ultimately know in order to
pass the CCNA Voice 640 - 460 exam.
Introduction
xxvii
3.
Complete each written lab at the end of each chapter. These labs are created to make
sure the reader fully understands key topics that are contained within that chapter.
Using a written format instead of multiple- choice format forces the reader to know the
answers off the top of their head instead of just eliminating options, as we often do
with multiple- choice questions.
4.
Work through and fully understand the commands found in the hands- on labs in the
chapter. Not all chapters have hands- on labs, but the book focuses on the important
tasks necessary for aspiring CCNA Voice – certified network engineers. See the accompanying sidebar for a recommended lab setup.
5.
Answer all of the review questions related to each chapter. Once you have finished
answering the questions, review the answers and explanations to not only understand
the correct answers but also understand why the incorrect answers are actually incorrect! Keep in mind that these review questions will not be the exact questions you will
find on the exam, but they will help you to understand the material that Cisco creates
the actual exam questions from.
6.
Take time to review the bonus exams that are included on the companion CD.
Questions in these exams appear only on the CD.
7.
Test yourself using all the flashcards on the included CD. These flashcards can be
viewed on both PCs and mobile devices, so now you can take your study material with
you wherever you go!
Recommended Home Lab Setup
As stated earlier, it is critical to get some hands-on experience with both CUCM
Express and Unity Express hardware and software. Following is a list of equipment
I recommend you try to acquire for your home lab studies. If you are concerned about
the high cost of purchasing the equipment, keep in mind that Cisco hardware can be
easily resold on used markets such as Craigslist or eBay. Combine that fact with
adding an extremely hot certification to your resume, and it’s an investment well
worth the initial cost.
Qty
Item
1
Cisco SBCS UC520
1
Cisco 7940 IP phone
1
Cisco 7965 IP phone
1
Windows PC loaded with the Cisco IP Communicator and Cisco
Configuration Assistant software
2
Analog telephones
xxviii
Introduction
Recommended Home Lab Setup (continued)
This equipment should give you the ability to practice configuring CUCM Express and
Unity Express using the command line, web GUI, and CCA methods detailed in this
book. The two different IP phones I recommend allow you to understand the
differences between two- and six-line phones as well as the fact that different Cisco
IP phones require different firmware files. A Windows PC will be needed to install both
the Cisco IP Communicator softphone and the Cisco Configuration Assistant software
used to configure SBCS devices such as the UC520. Finally, the analog phones in your
lab are useful for testing FXS configurations.
What ’s on the CD?
The CD included with this book includes many supplemental tools that you can use
to further your studies and achieve your goal of becoming a CCNA Voice – certified
administrator. The following content is provided for you to use to further your study.
The Sybex Test Engine
The Sybex test engine software lets readers practice all of the review and assessment
questions found in the book as well as two additional bonus exams that are found
only on the CD. The exams let potential test takers practice in an electronic test-taking
environment that is similar to the actual Cisco exam.
Electronic Flashcards for PCs and Handheld Devices
In addition to the Sybex test engine software, Sybex has included over 200 electronic
flashcards for you to test yourself with on PCs and compatible handheld devices.
These flashcards are designed to get the reader to quickly recognize and recall important
CCNA Voice information that will be useful for them when taking the 640 - 460 exam.
CCNA Voice: Cisco Certified Network Associate Voice Study
Guide in PDF
Finally, this book contains the entire CCNA Voice Study Guide in PDF format on the
included CD so you can read the book on your PC or laptop or any handheld devices that
reads PDF fi les such as a Blackberry or iPhone.
Tips for Taking Your CCNA Voice Exam
According to Cisco’s website at https://learningnetwork.cisco.com/community/
certifications/voice_ccna/iiuc?view=overview, the CCNA Voice exam contains
anywhere from 60 to 70 questions and must be completed in 90 minutes or less. The
languages this exam is offered in include English, Japanese, Chinese, Russian, Portuguese,
Introduction
xxix
Korean, French, and Spanish. This information can change per exam. A passing score
varies according to the types of questions found in the exam, but it is probably best to
assume you need to get approximately 85 percent of the questions correct to pass the exam.
When taking the exam, thoroughly read each question to make sure you know what
answer it is looking for. Cisco exam questions tend to have answers that look identical.
You will fi nd, however, that there are small differences in the answer that can determine a
correct or incorrect answer.
Also, keep in mind that you should choose the answer that Cisco believes is the correct
as opposed to what you or other vendors believe. This is a Cisco exam, after all, so the
right answer is the one that Cisco recommends!
The format of the 640 - 460 exam questions might include any of the following:
■
■
Multiple- choice single-answer
Multiple- choice multiple-answer — Cisco will always tell you to choose two or three,
depending on the proper number of multiple correct responses.
■
Drag-and-drop
■
Fill-in-the-blank
■
CUCM Express and Unity Express simulations
Test-Day Tips for Certification Success
■
■
■
■
Arrive at least 30 minutes early to the exam center. That way you can check in and
mentally prepare for the exam without having to rush.
Take the Cisco exam tutorial. This tutorial is offered prior to the official start of each
exam before the test timer starts. In this tutorial you will be given an interactive lesson
as to the format of the exam and how to navigate through the different question types,
including multiple- choice, drag-and-drop, fill-in-the-blank, and simulation questions.
Even if you have taken many Cisco exams, I highly recommend going through the
tutorial in case there is something new to the exam format since the last time you took
an exam.
Read both the questions and answers very carefully. Cisco often will intentionally lead
the hasty test taker, who simply glosses over a question, to quickly choose the incorrect
answer. Patience and careful thinking pay off greatly when taking Cisco exams!
Be aware that you cannot go back to change an answer once you have moved on to the
next question. Make sure that the answer you choose is the one you want to stick with,
because there is no way to change it later on.
Assessment Test
xxx
Assessment Test
1.
What two configuration steps are required for proper communication between the CUCM
Express and Unity Express?
A. A loopback interface needs to be configured.
2.
B.
A default gateway needs to be configured on the service module pointing to the IP of
the service engine (or integrated service engine).
C.
The service engine (or integrated service engine) must be on the same IP subnet as the
service module.
D.
A default gateway needs to be configured on the service engine (or integrated service
engine) pointing to the IP of the service module.
Which IP Telephony deployment model places independent call-processing and voice mail
devices at each remote site?
A. Multisite with distributed call processing
3.
B.
Single site
C.
Multisite with centralized call processing
D.
Single site with SRST
E.
Clustering over the WAN
What type of license might you need if you want to add a new Cisco IP phone to your existing CUCM Express system? Choose all that apply.
A. Cisco softphone license
4.
B.
CCME Express feature license
C.
Cisco IOS license for voice capabilities
D.
Individual user license
When you order a T1 circuit from a PSTN, you request that only 12 channels be available.
What term is used for this scenario?
A. Timeslots
5.
B.
Fractional
C.
LoopStart
D.
Ds0 -group
What is the 3.5 mm port used for on the UC500 system?
A. Wireless expandability
B.
ESW 500 series uplink
C.
Fax/modem connectivity
D.
External music source for MoH
Assessment Test
6.
xxxi
What Windows application can be used to configure a UC500 system?
A. CUCM Express GUI
7.
B.
CCA
C.
SIP
D.
SRST
Which two are peer-to -peer signaling protocols?
A. SCCP and H.323
8.
B.
SIP and MGCP
C.
SIP and H.323
D.
SCCP and MGCP
E.
H.323 and MGCP
What queuing technique is considered the best option for voice traffic?
A. FIFO
9.
B.
LIFO
C.
LLQ
D.
PQ
E.
CQ
What CUCM Express config-telephony command modifies tone and cadence differences
between geographic regions?
A. user-locale
B.
network-locale
C.
language-locale
D.
telephony-service-locale
10. What type of hunt group algorithm rings the hunt group members in the order in which
they were entered into the CUCM, always starting from the first number?
A. Longest idle
B.
Peer
C.
Round robin
D.
Sequential
E.
FIFO
Assessment Test
xxxii
11. Using the Unity Express CLI, what show command lets you view the maximum number of
configurable personal and GDM mailboxes?
A. router#show license
B.
router(config)#show license
C.
router#show software license
D.
router#show software mailbox
12. When configuring voice features by using CCA, what tab would you use to exclude an IP
address from the DHCP scope on a UC500?
A. System
B.
SIP Trunk
C.
Voice Features
D.
Network
E.
User Extensions
13. When SCCP is used between two endpoints that call each other, how is RTP data
transported when a connection is established?
A. It must be proxied throughout the CUCM.
B.
It must be sent through a voice gateway.
C.
It is sent directly from one endpoint to another.
D.
Each RTP stream terminates at the CUCM.
14. When a telephone handset is in its cradle, what state is it in?
A. Multiplex
B.
Dual-line
C.
Single-line
D.
Off-hook
E.
On-hook
15. What are the three different inter-VLAN routing methods that you can configure on your
network?
A. Layer 3 switching
B.
Layer 2 switching
C.
Individual router links
D.
Trunked router link
E.
Individual trunk link for each VLAN
Assessment Test
xxxiii
16. By default, what type of signaling are FSX ports configured for?
A. SIP
B.
GroundStart
C.
LoopStart
D.
SCCP
17. How are Unity Express backups run using the web GUI?
A. Backups can be set to run automatically each day.
B.
Backups can be run only during evening hours or holidays.
C.
Backups can be run using the TUI interface as long as the user is an AvT administrator.
D.
Backups are run by navigating to Administration Backup/Restore Start Backup.
18. Which of the following is not a Trunk Priority dial-plan rule for outgoing calls?
A. PSTN Only
B.
SIP Only
C.
FXO Only
D.
None
19. What does 2 represent in the command button 1:2?
A. Extension #2 on the phone
B.
A dual-line ephone
C.
Ephone-DN #2
D.
Ephone #2
20. Which voice signaling protocol is proprietary?
A. H.323
B.
SIP
C.
SCCP
D.
MGCP
E.
G.711
F.
G.729
21. What PBX service redirects a call to a different extension?
A. Extension dialing
B.
Call forwarding
C.
Hunt group
D.
Paging group
E.
Call park
xxxiv
Assessment Test
22. What step in the analog-to - digital conversion converts the data into binary?
A. Encode
B.
Quantize
C.
Compress
D.
Sample
23. What percentage of dropped packets can be allowed on a network and still have quality
voice calls according to Cisco?
A. Less than 1 percent
B.
Less than 4 percent
C.
Less than 8 percent
D.
Less than 2 percent
24. Given the following dir flash: output, what command would you use to view Cisco IP
phone firmware files?
Router#dir flash:
Directory of flash:/
1 drw0
13 -rw22224
readme-v.2.0.txt
14 drw0
27 drw0
45 -rw496521
hold.au
46 drw0
127 drw0
161 -rw47576204
Apr 7 2009 18:17:56 +00:00
Apr 7 2009 18:25:56 +00:00
bacdprompts
CME43-full-
Apr 7 2009 18:18:06 +00:00
Apr 7 2009 18:18:14 +00:00
Apr 7 2009 18:26:22 +00:00
Desktops
gui
music-on-
Apr 7 2009 18:18:28 +00:00
Apr 7 2009 18:31:02 +00:00
Apr 7 2009 18:37:22 +00:00
phone
ringtones
c3825-
ipvoicek9-mz.124-15.XZ2.bin
A. dir flash:/Desktops
B.
dir tftp:/Desktops
C.
dir tftp:/phone
D.
dir flash:/phone
Assessment Test
xxxv
25. What are three situations where a voice gateway is required?
A. Connecting a CUCM Express to the local IP network
B.
Connecting a CUCM Express to the PSTN
C.
Connecting a CUCM Express to an SIP phone
D.
Connecting a CUCM Express to a legacy PBX
E.
Connecting a CUCM Express to a second CUCM Express over an IP WAN
26. What is the absolute maximum number of devices supported by CCA within a single site?
A. 20
B.
15
C.
25
D.
50
27. When running through the Unity Express Initialization Wizard, when are changes actually
made to the configuration?
A. After pressing the Next button to move on to the next configuration screen.
B.
Changes are saved every 60 seconds.
C.
When the administrator selects the Save To Startup Configuration check box and
clicks the Finish button.
D.
When the administrator presses the Finish button at the Initialization Wizard Commit
page.
28. What are the two voice mail configuration tabs within the CCA?
A. Network
B.
Pilot
C.
Mailboxes
D.
Setup
E.
Users
29. Out of the box, if a Cisco IP phone is plugged into an Ethernet interface on a UC500 system, what extension is automatically assigned to it?
A. 2001
B.
3001
C.
201
D.
301
E.
101
xxxvi
Assessment Test
30. What is the default call-transfer method on the CUCM Express for dual-line DN phones?
A. local-consult
B.
full-consult
C.
blind
D.
full-blind
31. Which protocol is used to synchronize clocks on all voice and data network equipment?
A. VTP
B.
NTP
C.
CDP
D.
Timezone
E.
DST
32. What portion of the E.164 International code is actually assigned by the ITU board?
A. Country code
B.
National destination code
C.
Area code
D.
Station code
E.
Office code
33. Which Cisco Communications Manager runs on a router platform?
A. CUCM
B.
CUCMBE
C.
CUCM Express
D.
All of the above
34. At what layer of the UC model is the Cisco Unity voice mail solution found?
A. Infrastructure layer
B.
Data Link layer
C.
Call Control layer
D.
Applications layer
E.
Session layer
Assessment Test
xxxvii
35. Which protocol is used to monitor and provide detailed information about the quality of an
RTP stream?
A. cRTP
B.
RTCP
C.
UDP
D.
TCP
E.
H.323
36. When issuing a show ephone command on a CUCM Express system, you see that one of
your ephones is in a DECEASED state. What does this mean?
A. The phone unregistered abnormally because of a keepalive timeout.
B.
A hardware malfunction occurred at the phone endpoint.
C.
The phone unregistered normally and is not currently active.
D.
The phone unregistered abnormally because of a reverse proxy lookup.
E.
The phone unregistered normally because of a keepalive timeout.
37. Which of the following is not a required configuration setting when setting up a T1 CAS?
A. Pri-group options
B.
Framing type
C.
Clock source
D.
Ds0 -group options
E.
Linecode type
38. What methods are available to administrators when they want to run a trace on Unity
Express?
A. Using the Unity Express command line
B.
Using the Unity Express web GUI
C.
Using the CUCM Express command line
D.
Using the CUCM Express web GUI
39. What type of analog interface connects to standard analog telephones?
A. CO
B.
DID
C.
FXS
D.
FXO
xxxviii
Assessment Test
40. Given the following configuration output, what type of phone system does this represent?
Router(config)#ephone-dn 1
Router(config-dn)#number 5555558888
Router(config-dn)#ephone-dn 2
Router(config-dn)#number 5555559999
Router(config-dn)#exit
Router(config)#ephone 1
Router(config-ephone)#button 1:1 2:2
Router(config)#ephone 2
Router(config-ephone)#button
1:1 2:2
A. PBX system
B.
Key system
C.
Hybrid system
D.
Publisher system
41. If you try to use the web GUI for Unity Express Configuration but you are unable to
connect to the Unity Express web service, what must you do?
A. You must enable HTTP server on the CUCM Express.
B.
You must enable SSH on the CUCM Express.
C.
You must enable HTTP server on Unity Express.
D.
You must enable SSH on Unity Express.
E.
You must configure a username/password on Unity Express.
42. What is a function of a voice gateway?
A. Provides user authentication
B.
Provides call processing
C.
Translates between analog-to - digital and digital-to -analog connections
D.
Connects IP networks over a WAN
E.
Stores voice mail information
43. What limits the types of numbers that can be used by the message-notification feature?
A. Extension blocking
B.
Message expiry
C.
Number profile
D.
Restriction tables
Answers to Assessment Test
xxxix
Answers to Assessment Test
1.
B, C. The two interfaces must be located on the same IP subnet either by creating a brandnew subnet or by using a preexisting subnet and ip unnumbered. Also, the service module
on the CUCM Express side of the network must point to the IP address of Unity Express.
See Chapter 8 for more information.
2.
A. The multisite with distributed call processing model places all voice functions out to the
remote site edge. See Chapter 1 for more information.
3.
B, D. You will need individual user licenses for each phone, and you might need to purchase
additional feature licenses depending on how many feature licenses you have available. See
Chapter 5 for more information.
4.
B. When you order any capacity of lines on a T1 or E1 circuit that is less than the maximum,
the term used to describe this circuit is “fractional.” See Chapter 7 for more information.
5.
D. The 3.5 mm jack is used to connect an external audio source for MoH. See Chapter 10
for more information.
6.
B. The Cisco Configuration Assistant is a Windows application used for configuration of
small-business hardware such as the UC500. See Chapter 1 for more information.
7.
C. Both SIP and H.323 are considered peer-to -peer protocols. See Chapter 3 for more
information.
8.
C. Low Latency Queuing creates a priority queue for voice and sets aside a guaranteed rate
for this traffic. See Chapter 4 for more information.
9.
B. The network-locale command changes tone and tone cadences to match what users at
a given geographic location are accustomed to. See Chapter 6 for more information.
10. D. The sequential algorithm always rings the fi rst configured member fi rst, then the second
entered number, and so on. See Chapter 6 for more information.
11. C. The show software license command lets you view the maximum number of
configurable personal and GDM mailboxes the license allows. See Chapter 8 for more
information.
12. D. The Network tab includes DHCP configuration options. See Chapter 11 for more
information.
13. C. SCCP information is passed from the endpoint to the CUCM, and RTP data is sent
from one endpoint to another. See Chapter 3 for more information.
14. E. A phone is considered on-hook when the phone handset is in its cradle. See Chapter 2
for more information.
xl
Answers to Assessment Test
15. A, C, D. You can configure inter-VLAN routing using Layer 3 switches, individual
links per VLAN, or a trunked link in router- on-a-stick mode. See Chapter 4 for more
information.
16. C. The most common signaling for FXS ports is LoopStart, which is enabled by default.
See Chapter 7 for more information.
17. D. Backing up Unity Express using the web GUI requires that the administrator navigate to
Administration Backup/Restore Start Backup. See Chapter 9 for more information.
18. C . The Trunk Priority dial-plan rules can be set for PSTN Only, SIP Only, PSTN Then SIP,
SIP Then PSTN, or None. See Chapter 11 for more information.
19. C. This command tells the CUCM Express system that button 1 is to be set with
Ephone-DN 2. See Chapter 5 for more information.
20. C . The Skinny Call Control Protocol is proprietary to Cisco equipment. See Chapter 3 for
more information.
21. B. Call forwarding allows a user to redirect calls from one extension to another
automatically. See Chapter 2 for more information.
22. A. Encoding takes the quantized sample and converts it into binary code of 1s and 0s. See
Chapter 2 for more information.
23. A. You can have approximately 1 percent or less of dropped IP packets on a network and
still maintain a quality voice network. See Chapter 4 for more information.
24. D. This CUCME system used the archive tar /xtract command to uncompress fi les and
place them into a directory structure. All of the phone fi rmware fi les will be located in the
flash:/phone directory. See Chapter 5 for more information.
25. B, D, E . Voice gateways are required to connect to PSTNs, legacy PBX systems, or other
voice gateways/call managers across an IP network. See Chapter 7 for more information.
26. C . CCA supports a maximum of 25 devices in a single site. See Chapter 10 for more
information.
27. C . The Save To Startup Configuration check box must be checked on the Commit page for
the changes to be saved on Unity Express. See Chapter 8 for more information.
28. C, D. The two CCA voice mail configuration tabs are Mailboxes and Setup. See Chapter 11
for more information.
29. C . Auto -registration of Cisco IP phones is used to set up the phone, and it is assigned a
single extension beginning with 201. See Chapter 10 for more information.
30. B . Full-consult allows you to speak to the transfer number party prior to transferring
the call. See Chapter 6 for more information.
Answers to Assessment Test
xli
31. B . The Network Time Protocol is used to synchronize network equipment to extremely
accurate time sources on a network. See Chapter 4 for more information.
32. A . The ITU distributes the country code each country uses. See Chapter 2 for more
information.
33. C . CUCM Express runs on Cisco router hardware. See Chapter 1 for more information.
34. D. CUCM Express runs on Cisco router hardware. See Chapter 3 for more information.
35. B . RTCP provides information regarding the quality of the RTP stream it is responsible for.
See Chapter 3 for more information.
36. A . When a phone is in a DECEASED state, this means that the phone was on the network
but failed to return keepalives. This is common when a phone loses power. See Chapter 5
for more information.
37. A . The pri-group command is used on T1 PRI interfaces and not on T1 CAS interfaces.
See Chapter 7 for more information.
38. A . The only way to run a trace is to use the Unity Express command-line interface. See
Chapter 9 for more information.
39. C . FXS interfaces connect analog devices to the PSTN. See Chapter 2 for more information.
40. B. Key systems typically share lines, and each ephone is identically configured. See Chapter 6
for more information.
41. A . Your CUCM Express must have HTTP server enabled by issuing the ip http server
global configuration command. See Chapter 8 for more information.
42. C . A primary function of a voice gateway is to translate voice streams between analog and
digital. See Chapter 3 for more information.
43. D. Restriction tables protect the organization from users employing message notification to
forward calls to long- distance numbers. See Chapter 9 for more information.
Chapter
1
Cisco Unified
Communication
Solutions
THE FOLLOWING CCNA VOICE EXAM
OBJECTIVES ARE COVERED IN THIS
CHAPTER:
Describe the components of the Cisco Unified
Communications Architecture.
Describe the function of the infrastructure in a UC
environment.
Describe the function of endpoints in a UC environment.
Describe the function of the call processing agent in a UC
environment.
Describe the function of messaging in a UC environment.
Describe the applications available in the UC environment,
including Mobility, Presence, and TelePresence.
Describe and configure gateways, voice ports, and
dial peers to connect to the PSTN and service provider
networks.
Describe the differences between PSTN and Internet
Telephony Service Provider circuits.
Cisco Systems seems to have the market cornered when it
comes to product placement of telephones in television shows
and movies. If you look closely at shows such as The Office
and 24, Cisco is cleverly placing their phones in the camera shots. While that placement
is a good way to show off the sleekness of the phones, network engineers want to know
what’s powering the phones behind the scenes. This fi rst chapter begins CCNA: Voice
Study Guide with an overview of the Cisco Unified Communications (UC) hardware and
software currently available. Knowing all the equipment that is in Cisco’s IP telephony
(IPT) arsenal will enable you not only to prepare for the CCNA Voice exam but ultimately
to make intelligent engineering decisions for your company and clients.
Once the key hardware and software solutions are detailed, the discussion will
turn toward the various best-practice design and deployment strategies for the Unified
Communications system. This will help to clarify choices to be made regarding centralized
versus distributed call-processing designs.
Why Should We Bother Integrating
Voice and Data Services?
So, why are we here? What’s the point of ripping out our old phone handsets and PBX
hardware to replace everything with Cisco equipment that runs on our data network?
Fortunately, there are many advantages that provide cost savings as well as increased
capabilities that ultimately will change the way users communicate with each other. It’s no
secret that many businesses have already made the switch or are at least considering the
increased benefits of IP telephony. An IPT system provides many business drivers. Let’s
break down the communications enhancements and monetary reasons to switch to an IP
telephony–based solution such as one of the Cisco Unified Communications systems.
Communications Enhancements
IP telephony provides the following enhancements to communication:
Integration of voice and data networks Combining communications methods such as
voice, video, and data becomes much more feasible if all of them speak the same language.
Applications can now seamlessly integrate features such as email and instant messaging
into your voice functions to provide added functionality to users.
Introducing the Cisco Unified Communications Manager Lineup
3
Unified messaging of voice, email, and fax messages All of these once-separate communication methods can be combined into a single central repository. This allows users to have
a single location where they store and retrieve messages and greatly reduces the need for
communications in the workplace.
The ability to communicate while out of the office If your voice system runs over IP, you
can harness the power of the Internet to make remote connections back to your voice system while you are on the road. Cisco refers to this as mobility. The ability to function from
a remote location as if you were sitting in your office allows for a great amount of workforce flexibility, which can increase overall productivity.
Cost Savings
IP telephony can save money in the following ways:
Reduced cabling costs By integrating voice and data, businesses now maintain a single
cabling structure. Previously, there was separate physical cabling for voice systems and data
systems. Combining the two separate networks into a single integrated network can potentially cut cabling costs in half!
Reduction in telephone company charges If you have remote site or branch offices that
use telephones to communicate with one another, you can significantly reduce telephone
charges. Instead of transporting your voice calls over public telephone lines that incur high
monthly rates and long-distance charges, you can send them across your WAN links that
are currently transporting only data services. The elimination of public telephone lines
between branch sites can result in significant savings.
Preservation of investment in analog technology Many businesses have a significant
investment already in analog phones and other legacy phone technology. Cisco provides
several methods where you can continue to use legacy hardware over an IP telephony
network.
Now that you’re thoroughly convinced of the reasons to jump on board with an
integrated voice and data network, let’s look at the Cisco equipment offerings that meet
virtually any business need.
Introducing the Cisco Unified
Communications Manager Lineup
In Cisco’s Unified Communications architecture, Unified Communications Managers are
what makes IP telephony possible. These hardware/software devices are the brains that
handle IP call processing. The call-processing portion of a Unified Communications system
handles the sequence of operations from the time a user picks up a phone to make a call
to the time the user ends the call by hanging up. All of the signaling, dial interpretation,
ringing, and call connecting is performed by the call processor. From a phone user’s
4
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Cisco Unified Communication Solutions
standpoint, the call processor acts like a legacy-based analog or digital phone. All of the
basic phone functions such as dialing, ring signals, and interactions are the same as they ’ve
always been. This is obviously by design; because users are so familiar with using phones, it
would be very difficult to modify user behavior.
From an administrative standpoint, the call processor is where you configure dialing rules
for end users. Things like how to reach an outside line, internal extension dialing, and other
rules are configured and maintained in the call processor database. You can also administrate
the individual phones from the call-processing unit. Additions, changes, and deletions
of phone extensions, voice mail access, intercom, and other voice features are controlled
centrally at the call-processing unit. The configurations are then pushed out to the individual
phones on the network. All of these processes are transparent to the end users and require no
manual interaction from them.
There are three distinct Unified Communications Manager systems:
Cisco Unified Communications Manager (CUCM)
Cisco Unified Communications Manager Business Edition (CUCMBE)
Cisco Unified Communications Manager Express (CUCM Express)
Each of these voice solutions is feature rich and highly flexible. The major differences
from an end user’s point of view are the number of users that each solution can handle and
which solutions provide high availability (HA) and redundancy. When you take a closer
look at the hardware/software architecture, you will fi nd that the CUCM and CUCMBE
run on server-based hardware and a hardened Linux OS, while the CUCM Express runs on
Cisco routers and utilizes the Cisco IOS to run on. Let’s dive into the specifications of each
of these IP telephony call-processing solutions.
Cisco Unified Communications Manager
The Unified Communications Manager is Cisco’s IP telephony flagship system. Beginning
with software version 5.0, it runs on a Linux-based operating system. The current CUCM
version is 7.1. While Cisco supports a few select third-party hardware vendors to run the
Unified Communications Manager, typical enterprise- class implementations are appliancebased, running on the Cisco 7800 Series Media Convergence Servers (MCS). Older versions
of the CUCM ran on Windows 2000 Server operating systems. Cisco has moved away from
the Windows-based systems and now provides only a version that runs on Linux.
Cisco packs in virtually every possible voice and video feature capability you can think
of in the CUCM system. Each server appliance is capable of handling up to 7,500 endpoints
and can be clustered to support up to 30,000 endpoints. Scalability is the name of the
game here. If you have a large company or plan to grow quickly, the CUCM can grow right
along with you. When clustering multiple CUCM servers together, one CUCM Publisher
controls the read/write functions of the database. All other servers are called subscriber
servers. Subscriber servers handle additional call processing or sit idle as standby servers in
case an active subscriber were to fail. Subscriber servers are key components if your voice
environment requires high availability in the event of a hardware or software failure. By
Introducing the Cisco Unified Communications Manager Lineup
5
providing a clustered call-processing environment, you can have a call processor go offline,
whether because of failure or maintenance, and continue to process calls with the other
subscriber servers that are still operational on the network.
It is important to keep in mind that the CUCM appliance offers only call-processing
features. All voice mail functionality must be handled by a separate hardware/software
solution. The voice mail could be a Cisco Unity or Unity Express system or another
third-party voice mail solution. There is no integrated call-processing and voice mail
solution because the CUCM is targeted toward very large enterprise- class environments
with thousands of users and phones. In these environments, call-processing tasks require
dedicated hardware to be able to handle the large call volumes that CUCM was intended
for. Likewise, a large voice mail solution should also have its own dedicated hardware to
handle the high number of users and the degree of functionality that is required in such
large organizations.
When choosing a Cisco Unified Communications system for a particular
environment, it is important to consider company growth in the equation.
A communications system that meets the company ’s needs today may
not meet future needs. Estimate the percentage of growth over a five - to
seven-year period.
Cisco Unified Communications Manager
Business Edition
The business edition of the full-blown CUCM is geared toward medium-size companies
that require up to 500 endpoints. The CUCMBE is either a software- or appliance-based
solution, much like its big brother. The call-processing features and functionality are
identical to the CUCM. The only downside is that it does not provide redundancy and
cannot be clustered with other CUCM or CUCMBE systems to add additional users. This
lack of redundancy may or may not be an issue for your implementation, but it is important
to keep it in mind. If redundancy is not an issue, CUCMBE might be a great solution for
your medium-size environment because it offers many of the features of the larger CUCM
system at a vastly reduced cost.
One major benefit of the CUCMBE is that it offers an integrated Unity voice mail system
that runs on the same hardware as the call-processing system. This helps lower customer
costs by allowing them to use only one piece of hardware for both purposes. The integrated
Unity Connection is detailed later in this chapter.
Cisco Unified Communications Manager Express
The Unified Communications Manager Express solution is the call-processing system the
CCNA Voice exam is mainly focused on. This software runs on Cisco routers such as
the Integrated Service Router (ISR) line. Integrated Service Router is a term Cisco uses
6
Chapter 1
Cisco Unified Communication Solutions
for routers that integrate multiple services into a single chassis. For example, an ISR can
integrate full routing, switching, wireless, fi rewalling, and voice capabilities on a single
unit. You can mix and match the services that you need because the add- on capabilities are
hardware modules that are inserted into the router unit. This provides businesses with a
very flexible platform that will scale well for many years to come.
A special version of Internetwork Operating System (IOS) software must be licensed
for the router to run CUCM Express. The IOS is the software used by Cisco routers and
switches that performs routing, switching and telecommunications functionalities. The
Communications Manager Express software must also be installed in the router’s flash
memory. This software works alongside the router IOS and provides administrators
with a single configuration fi le to help simplify and consolidate changes. Chapter 5
details instructions on how to install the CUCM Express software on compatible Cisco
router hardware.
The ability to integrate voice, data, and security on a single Cisco platform appeals to
many small-business owners. No separate servers or hardware are needed to handle the
communications of a business up to 250 users. Here is a list of older IOS routers as well as
the newer Cisco ISR router lineup that fully support CUCM Express:
Non- ISR routers supporting CUCM Express 7.1
Unified Communications 500 Series (SBCS)
Cisco IAD 2430
Cisco 1751-V
Cisco 1760 Series
Cisco 2600XM Series
Cisco 2691
Cisco 3700 Series
ISR routers supporting CUCM Express 7.1 and above
Cisco 1800 ISR Series
Cisco 2800 ISR Series
Cisco 3200 ISR Series
Cisco 3800 ISR Series
Each router supports a different number of IP phones because of differences in hardware
specifications. Following are the maximum specifications of CUCM Express:
Integrated data, voice, and security on a single platform
Up to 250 users on a 3845 ISR router running CUCM Express 7.1
On-board Unity voice mail by installing either the NM- CUE module or AIM- CUE
card in a compatible router
Introducing the Cisco Unity Lineup
7
Comparing the Communications Manager Alternatives
As a summary, Table 1.1 compares the capabilities of the three CUCM systems.
TA B L E 1 .1
CUCM comparison
System
Platform
Max Endpoints
High
Availability
Unity Options
CUCM
Server/Linux
7,500–30,000
Yes
Unity, Unity Express
CUCMBE
Server/Linux
500
No
Unity Connection
CUCM Express
Router/IOS
250
No
Unity, Unity Express
These three call-processing platforms, CUCM, CUCMBE, and CUCM Express, give
businesses both small and large the opportunity to integrate voice with data and take
advantage of Unified Communications features. Now that you have an understanding of
the call-processing functionality that Cisco offers, we’ll take a look at the Unity voice mail
solutions that almost always accompany the CUCM.
Introducing the Cisco Unity Lineup
When a user cannot answer the phone because they are away from their desk or on another
call, the calling party is directed to leave a voice message. Typically, every phone system
offers its own voice mail box where users can log in and check personal messages. In the
Cisco world, the voice mail product is called Unity. Much like Cisco’s call-processing lineup,
the Unity voice mail solutions can be broken down into three main categories:
Cisco Unity
Cisco Unity Connection
Cisco Unity Express
The main differences are in the number of users supported and the platform the Unity
software resides on. Let’s take a closer look at each of these products. All of these products
offer standard voice mail. What sets Unity systems apart from legacy voice mail systems is
the fact that Unity can integrate with other communications mediums such as email, fax,
and instant messaging to give users increased flexibility in the ways they conduct business
on a daily basis.
8
Chapter 1
Cisco Unified Communication Solutions
Cisco Unity
The Unity product is Cisco’s largest and most robust voice mail and unified messaging
solution. The server-based appliance runs on the Microsoft Windows 2000 or 2003
operating system. A single server can support up to 15,000 mailboxes, and multiple
servers can be clustered to provide additional mailboxes. The amount of voice mail storage
depends on the size of the hard drives available on the server. The Unity product is the only
solution that offers all the Unified Messaging (UM) functionality because it fully integrates
with Microsoft Exchange. Unified Messaging includes a great number of value-added
features for voice mail users. Essentially, it creates a single message-storage database for
voice mail, fax, and email. All voice mail, email, and fax transmissions end up being stored
on the Microsoft Exchange email server. This migrating of messages allows you to mesh
voice with email to create some very helpful services. Here are some of the most popular
Unified Messaging features available with the product:
A single directory database for voice and email.
Listening to and deleting voice mail on Microsoft Exchange –powered email. Deleting
the voice mail on email also deletes the message from your message waiting box on the
phone and turns off the message waiting indicator (MWI) lamp on the phone.
Forwarding voice mail messages as email attachments.
Email messages being read to users over the phone using Cisco’s text to speech (TTS)
conversion services.
Reception of fax messages sent to the Unity system and the ability to view them as
image files.
Broadcasting of a voice mail message to multiple voice mail boxes.
The product can also be configured to be fully redundant. Unity servers are set up in
a primary and standby configuration, where the primary server handles all messaging
processes. In the event of a primary Unity server failure, the standby server automatically
takes over the primary server duties. The primary server stays inactive until an
administrator automatically brings it back online as the primary message-processing device.
Cisco Unity Connection
Unity Connection software runs on a Linux-based server platform. It often accompanies
CUCMBE implementations and can even reside on the same hardware. It can be installed
on separate hardware if desired. A Unity Connection server with full integrated messaging
can handle up to 7,500 mailboxes. Much like the Unity system, the amount of voice mail
storage available depends on the server the software resides on. Also similar to the Unity
system, Unity Connection is designed to be fully redundant. The main difference between
Introducing the Cisco Unity Lineup
9
Unity and Unity Connection from a functionality standpoint is that Unity Connection does
not integrate as seamlessly with Microsoft Exchange. Instead of having a single message
store between all voice mail, fax, and email messages, the Unity Connection server is
responsible for handling voice mail and fax services, and the Microsoft Exchange server
deals with email messages. The Unity Connection server does have the ability to send
voice mail messages as attachments, so users can receive voice mails through their email.
This process essentially makes a copy of the voice mail. Unfortunately, now you have
two copies of the message. One message is in your email inbox, and the other is on Unity
Connection. The deletion of one message does not delete all messages. These will have to
be individually deleted by the user.
Cisco Unity Express
As with all Express products in Cisco’s lineup, Unity Express runs on the same routers
that CUCM Express supports. The Unity Express system can be thought of as a “boltedon” solution. By this I mean the voice mail system resides on a piece of hardware that
must be installed into a compatible Cisco router in order to function. It integrates with the
router IOS to provide added services. This is where the Cisco marketing term Integrated
Services Router, or ISR, came from. All of these added services, such as Unity Express
voice mail, must integrate with the router hardware and software to function. Without this
integration, they cannot function. Technically, however, the Unity Express system
operates as a separate “server” within the router. It actually runs a hardened Linux OS for
voice mail processing. You’ll learn how to install and configure Unity Express properly in
Chapters 8 and 9.
Unity Express offers the fewest messaging-integration features, but for such a small
system, it packs quite a punch! Here are some of the voice messaging solutions available
with Unity Express:
Basic automated attendant (AA) functionality, in which the Unity Express system
automatically answers calls using a computerized voice. The user is then presented
with audible options from which to choose in order to direct their call to the proper
recipient. It essentially eliminates the need to have a human receptionist transfer calls.
Voice mail access through email. Unity Express can create a copy of the voice mail
message and forward it to an email address. This is similar to the Unity Connection
messaging process.
Voice mail access by dialing into the system from an external phone. This allows users
to receive messages when they cannot physically be at their desk.
Voice mail access over a web -based interface. Users can download and listen to their
voice mails using a web browser such as Internet Explorer.
Chapter 1
10
Cisco Unified Communication Solutions
Two hardware options are available for Unity Express:
NM- CUE Unity Express NM- CUE is a network module card that slides into an open bay
of the router. It contains a hard drive that stores the Unity Express software and provides
additional storage for up to 250 mailboxes. Of the two Unity Express options currently
available, this model offers the most storage for messages. The NM- CUE does come at a
cost, however, because you are required to use up a network module slot on your router,
which you may need for other services.
AIM- CUE Unity Express AIM- CUE is an advanced integration module. It is a card that
must be installed on the router motherboard. The module looks very similar to a RAM
chip that you install in a PC or server. The benefit of this module over the NM- CUE is
that you do not have to use up a module slot on your router. If you need to install other
Integrated Service modules on your router, the AIM- CUE might be a good option to save
precious NM slot space. The Unity Express software and mailbox storage space are housed
on a compact flash (CF) card on the AIM card. The card comes with either 512 MB or 1
GB of storage. This is enough storage to handle up to 50 mailboxes. So here again we see a
tradeoff between the AIM- CUE and the NM- CUE. While the AIM- CUE does not use up a
network module slot, the amount of storage available for voice mail is very limited.
It is easy to remember which Unity product offers full Unified Messaging
with Microsoft Exchange. Because Unity runs on the Microsoft Windows
Server platform, it can integrate more completely than the other two Unity
products, which run on Linux and IOS operating systems.
Table 1.2 compares the capabilities of the Unity systems.
TA B L E 1 . 2
Unity comparison
System
Platform
Max Mailboxes
Unity
Windows Server 2000/2003
15,000+
Unity Connection
Integrated CUCMBE server/Linux
7,500
Unity Express NM- CUE
Router NM slot/IOS
250
Unity Express AIM
Router AIM slot/IOS
50
This should give you a good idea of the Unity voice mail systems that Cisco offers today.
Later in the book you will learn more about the features offered with Unity Express and
how to configure it to work with CUCM Express. Right now, we’ll discuss what Cisco
refers to as VoIP endpoints. These are IP phones and other voice/video solutions.
Introducing Cisco IP Phones and User Applications
11
Introducing Cisco IP Phones
and User Applications
IP phones include hardware- and software-based voice solutions. These IP phones allow
the user to make phone calls on- and off-network just like any traditional telephone unit.
Having an end-to - end Cisco voice solution provides many benefits in regard to increased
functionality and ease of administration. Following are some examples of value -added
functionality that Cisco IP phones possess and that third-party phone vendors may or may
not provide:
The ability to power the phones using Power over Ethernet (PoE) Using PoE - capable
switches or powered patch panels, this allows the phones to be installed without the need
to plug the handset directly into a power source.
A built-in data PC Ethernet port Many phones allow you to use a single Ethernet connection to plug your phone into the access layer switch. The phone then has a second port that
acts as a mini switch. You can plug your desktop PC into this port. This allows you to add
phones to your network without the need to double your switch-port capacity.
Intelligent voice segmentation Being able to differentiate between voice and data allows
administrators to provide a higher quality of service (QoS). Being able to provide voice traffic with enough network bandwidth and throughput is essential for providing an acceptable
voice stream end to end. The Cisco phone talks to the access switch using a proprietary
communications protocol called Cisco Discovery Protocol (CDP) to tag traffic. You’ll learn
how this works in Chapter 4.
Fixed and programmable voice feature buttons These buttons allow for value-added services to tailor fit a particular voice environment.
Software-based phones These provide voice functionality using a standard PC with a
microphone and sound card, giving users the ability to have a “virtual” phone anywhere
they choose.
Integration with video services This can be in the form of either an add- on webcam and
software or a fully integrated video system housed within the phone unit itself.
Cisco has four main lines of IP hardware-based phones. The 7900 Series, 6900 Series,
3900 Series, and 500 Series phones offer different features and are designed for various
IP telephony deployment scenarios. Cisco also has its version of a softphone, called the IP
Communicator. This Microsoft Windows application emulates a full-featured 7900 Series
phone without the need for a physical unit. Let’s examine the capabilities of the various
Cisco phone lineups.
12
Chapter 1
Cisco Unified Communication Solutions
Cisco 7900 Series IP Phones
The 7900 Series phones are Cisco’s most popular line on the market today. Multiple
models are available, from entry-level phones to high- end IP telephony solutions with
built-in videoconferencing. There are also phones that are specifically designed to be used
completely hands free in conference rooms and boardrooms. The 7900 Series also offers
Wi-Fi – connected phones that operate using 802.11a and 802.11b/g radios. All 7900
Series phones support both the Session Initiation Protocol (SIP) and the Cisco proprietary
Skinny Call Control Protocol (SCCP) for call signaling in an IP voice deployment.
The other major distinction of the 7900 Series phones is its full support of advanced
Extensible Markup Language (XML) functionality. XML is a programming language
that can provide the phones with additional feature -rich services such as employee
directories, company information, and other web content that can be made instantly
visible to any phone in a CUCM cluster. Chapter 3 provides an in- depth discussion on
how these protocols work. Table 1.3 lists some of the Cisco 7900 Series phones and their
unique features. The list begins with the most basic phones and ends with the models
with the most advanced features:
TA B L E 1 . 3
7900 Series phone features
Display Type
Integrated
PC Port
Number
of Lines
Available
7906G
192 × 64 mono
None
1
Softkeys and basic XML
support
7911G
192 × 64 mono
10/100
1
Softkeys and basic XML
support
7931G
192 × 64 mono
10/100
24
Softkeys and basic XML
support
7941GE
320 × 222 mono
10/100/1000
2
Full-duplex speakerphone and
XML support
7945G
320 × 240 color
10/100/1000
2
Same features as 7941GE with
16-bit color display
7961GE
320 × 222 mono
10/100/1000
6
Full-duplex speakerphone and
XML support
7945G
320 × 240 color
10/100/1000
6
Same features as 7961GE
with 16-bit color display
Phone
Model
Other Features
Introducing Cisco IP Phones and User Applications
13
Display Type
Integrated
PC Port
Number
of Lines
Available
7971G
320 × 240 color
10/100/1000
8
Large color touch screen
display
7921G
176 × 220 color
None
1
802.11a/b/g wireless
7925G
176 × 220 color
None
1
802.11a/b/g wireless; rugged
shell and Bluetooth support
7937G
Conference
Station
192 × 64 mono
None
1
Advanced speakerphone
with 360-degree microphone
coverage
7985G
Large
multiresolution
color
None
8
Built-in video camera for integrated videoconferencing
Phone
Model
Other Features
Cisco 7900 Expansion Modules
If you need additional phone line extension buttons in situations where a receptionist
answers and transfers calls all day, you can use the 7900 Series expansion modules,
which can plug into 796X and 797X Series phones. Table 1.4 lists the current Cisco 7900
expansion module devices and how many additional lines they support.
TA B L E 1 . 4
7900 Series expansion modules
Model
Display Type
Number of Lines
7914
Mono LCD
14
7915
Mono LCD
24
7916
Color LCD
24
You can also connect as many as two of these expansion modules to a compatible Cisco
7900 Series phone to give you up to 48 additional lines.
Cisco 6900 Series IP Phones
Cisco’s 6900 Series phones are midrange business phones for users who don’t require all
the advanced features found in the 7900 Series phones. The 6900 Series phones feature a
straightforward design, which makes them very easy to use even for nontechnical users.
14
Chapter 1
Cisco Unified Communication Solutions
The phones feature fi xed keys as opposed to softkeys used on the 7900 Series phones for
features such as directory, call transfer, conference, hold, and voice mail. These fi xed keys
provide a clutter-free environment that is streamlined and easy to use. The 6900 Series
phones boast the following features:
Two, four, or twelve lines
Full-duplex speakerphone
396 × 81 or 396 × 162 backlit, monochrome display
Multilanguage support
Integrated 10/100Mbps switch
Power over Ethernet support
Cisco 3900 Series IP Phones
The 3900 Series phones are entry-level models that are targeted to areas where multiuse
phones are needed, such as lobbies, hallways, and manufacturing floors. The phones could
also be used as everyday cubicle phones, but some features are lacking, which may require
an upgrade to the 6900 or 7900 Series IP phones. The major difference between the 3900
Series phones and the 7900 Series phones is that the 3900 Series phones communicate using
the Session Initiation Protocol only and do not have the ability to run Cisco’s proprietary
SCCP signaling protocol. The 3900 Series phones also are not capable of running XML
services and utilize fi xed-feature buttons as opposed to the softkeys found on the
higher- end 7900 Series phones. At the time of this writing, the only 3900 Series model
available is the 3911. This phone has the following features:
Single-line phone
Half-duplex speakerphone
144 × 32 monochrome display
Power over Ethernet support
Cisco IP Communicator
The Cisco IP Communicator is a software-based phone that delivers the capabilities of
the 7900 Series phones through a PC running Microsoft Windows XP and Vista. This
solution is perfect for travelers who require advanced telecommuting features. Using the IP
Communicator gives users all the features of a 7970 IP hardware phone through a PC. In
fact, the interface looks exactly like a 7970 hardware phone, so end users will immediately
be familiar with how it works. All the features are available, including up to eight separate
lines, direct access to voice mail, and XML services. Most third-party microphones and
headsets are fully compatible with the IP Communicator. Figure 1.1 shows an image of the
Cisco IP Communicator running on Windows XP.
Introducing Cisco IP Phones and User Applications
F I G U R E 1 .1
The IP Communicator running on Windows XP
CISCO IP PHONE
Copyright 2002–2008,
Cisco Systems, Inc.
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The IP Communicator, a Telecommuter’s Best Friend
Jennifer is an IT hardware reseller at a midsize company. A typical day for Jennifer would
start off with a one-hour commute from her home to work. Many of Jennifer’s customers
contact her directly over the phone to place their orders.
This was no ordinary day, however. The temperatures outside were well below freezing,
and her car wouldn’t start. Fortunately for Jennifer, the company recently upgraded
its phone system to a Cisco Unified Communications Manager. In addition, all of the
salespeople were given IP Communicator software to use on the road and when getting
into the office is impossible. So on this particular morning, instead of attempting to get
her car started, Jennifer went back into her home and logged into the company network
through her home Internet connection. She loaded up her Cisco IP Communicator from
her laptop, and she was off and running.
All her calls to her office phone were forwarded to the softphone, and what started out to
be a disastrous day turned into just another day at the (virtual) office.
15
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Cisco 500 Series IP Phones
Cisco’s 500 Series IP phones are cost- effective voice hardware devices that are specifically
designed for the SBCS Unified Communication 500 Series hardware. In fact, the 500
Series phones will not function on any other Cisco IP telephony system. Unlike the 3900
Series phones, which use the SIP communications protocol, the 500 Series phones
run Cisco’s SCCP.
These are very low- cost phones geared toward small businesses. Just because they
are low in cost doesn’t mean they don’t have any bells and whistles, however. The 500
Series phones can handle up to five lines per phone and have support for features such as
conference calling, call parking, paging, and intercom. Table 1.5 lists the Cisco 500 Series
phones and their unique features.
TA B L E 1 . 5
500 Series phone features
Number
of Lines
Available
Other Features
Phone Model
Display Type
Integrated
PC Port
521G
128 × 64 mono
None
1
Softkeys, backlit LCD
521SG
128 × 64 mono
10/100
1
Softkeys, backlit LCD
524G
128 × 64 mono
10/100
4
Softkeys, backlit LCD
524SG
128 × 64 mono
10/100/1000
4
Softkeys, backlit LCD
SPA525G
320 × 240 color
10/100/1000
5
Operates in wired or
802.11b/g wireless mode
Cisco Analog Telephony Adapter
The Cisco Analog Telephony Adapter (ATA) is a device that allows standard analog
telephones and fax machines to operate on an Ethernet LAN. The ATA has one Ethernet
port for network connectivity and two RJ-11 analog ports to connect up to two analog
devices. This solution is good if the organization has analog telephony devices it wants
to leverage throughout an IP telephony migration. Figure 1.2 shows how an ATA device
connects an analog phone to the IP network.
Introducing Cisco IP Phones and User Applications
17
Cisco VG224 and VG248 Series Voice Gateway
A second Cisco analog-to -IP conversion technology is the VG224. This device actually runs
the Cisco IOS operating system that runs on a Cisco ISR router platform. Its sole purpose
is the conversion of multiple analog lines into IP packets for transport over a data network.
F I G U R E 1. 2
An ATA device connecting an analog phone to an IP network
CUCM Express
V
Analog
Phone
IP Network
Cisco
ATA
The VG224 can terminate up to 24 analog voice endpoints, while the VG248 can
terminate up to 48 devices. They provide a cost- effective way to migrate your current
analog voice system to IP. Following are some reasons to use the VG200 Series as a starting
point into an IP telephony solution:
Investment protection If you’ve invested a large amount of money into analog handsets
and fax machines, this is a way to prolong their useful life while moving toward an integrated voice and data solution.
High voice quality
You experience the same quality calls as with a pure IP-based handset.
High availability The VG224 and VG248 have the ability to detect WAN failures and
route calls over backup PSTN lines.
Reduced cost of entry to IP telephony You can lower the initial costs of an IP telephony
implementation by using VG200 Series devices to provide a gateway for your existing analog phones. Eliminating the need to purchase new handsets can greatly reduce costs.
Additional Unified Communications Applications
In addition to the hard phones and softphones that Cisco offers, several user applications
enhance the Unified Communications experience. In this section, I’m going to describe
several of the Cisco applications available to you.
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Cisco Video Advantage
The Cisco Video Advantage solution adds videoconferencing to your voice calls, a
capability that seamlessly integrates with your Cisco Unified Communications system.
Video Advantage comes with a USB web camera, called the Cisco VT Camera II. You
can connect this camera to your desktop PC. You can then install the Video Advantage
software onto your PC; this software integrates with your Cisco IP desk phone or IP
Communicator. You can then make and receive not only standard audio voice calls but also
videoconferencing calls with your Cisco phone. Video Advantage is an extremely scalable
and cost- effective videoconferencing solution.
Cisco Unified Personal Communicator
You can think of the Unified Personal Communicator as a one-stop shop for all of your
communication needs. Similar to the IP Communicator mentioned earlier, the Unified
Personal Communicator is a piece of communications software that is installed on a
computer and has the ability to make and receive standard phone calls. The Unified
Personal Communicator takes things a step or two further, however, because the software
is multiplatform and can be installed not only on Windows PCs but also on Apple
computers running OS X. This product is marketed to larger companies with many remote
employees who need to stay in constant contact with one another. Here are some of the
integrated features that the Cisco Unified Personal Communicator offers:
Softphone capabilities for standard phone calls and voice mail using any of the Cisco
Unity products
Instant messaging tool for real-time text-based communication
Presence integration to notify other users of your whereabouts and to keep track of the
availability of other Unified Personal Communicator users
Integrated directory and contact services
Videoconferencing using Cisco Video Advantage
Web conferencing using Cisco Unified MeetingPlace software
Cisco Unified CallConnector
The Unified CallConnector is a Windows-based software application that installs on user
PCs. This software is geared toward small and medium businesses. It fully integrates
with the Cisco Unified Communications Express solution. Following are just some of the
features available for use with the CallConnector applications:
Integrates with Microsoft Outlook and Internet Explorer to provide a seamless
integration of web, email, and voice functions
Tracks user presence information to notify you when a user is available in real time
Simple-to -use contact list that provides a single repository for your phone, email, and
instant message contacts
Lets users make and receive calls with the click of a mouse
Allows users to be completely mobile
Using Voice Gateways
19
This should give you a good overview of the Cisco endpoints and endpoint applications
that a Unifi ed Messaging system can handle. In the next section, we’ll discuss the
hardware that allows our voice calls to be routed out to the public phone network.
Cisco TelePresence
TelePresence is an enterprise-class hardware and software video conferencing solution that
uses large screen high-defi nition monitors to display remote video to simulate a virtual
meeting room. The tool is designed to simulate face-to-face meetings over an IP network
where members can be located all around the world. TelePresence hardware and software
fully integrates into a CUCM system that is responsible for video conference call processing
such as setups and tear-downs as well as scheduling.
Using Voice Gateways
The Cisco Unified Communications call-processing devices in the CUCM suite allow
engineers to configure and deploy a voice system for on-network calling. On-network
dialing means that the calls are contained within a private network fully controlled
by a private organization. But what if you need to make a call to someone outside the
local organization? This is where voice gateways come into play. Voice gateways are
responsible for the setup of off-network calling. A voice gateway is simply an IOS router
that is capable of translating legacy voice communication into IP packets for transport
over data networks. It sits right on the border between your IP network and standard
telephone networks. If you are familiar with configuring routers for Ethernet LANs and
WANs, then you are already familiar with most of the configuration processes. What
may be new to you are the voice interfaces that are needed to connect your IP network
to the public telephone network. You’ll learn more about these interfaces as well as other
hardware that voice gateways utilize in a VoIP environment. For now, it is important to
understand that voice gateways can connect your Cisco Unified Communications solution
to the public switched telephone network (PSTN ) either physically or virtually using a
variety of methods. Let ’s briefly discuss the physical and virtual methods that can be
used to connect to the PSTN.
Voice analog and digital interfaces are special PSTN connections configured on
voice gateway routers to bridge the VoIP and PSTN networks so off-network calls can
successfully be completed. Figure 1.3 shows how a voice gateway can be configured to
connect directly to the PSTN.
F I G U R E 1. 3
A voice gateway connecting directly to the PSTN
V
Customer
Site
PSTN
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A virtual method of connecting your voice gateway to the PSTN is to set up a VoIP
trunk to an Internet Telephony Service Provider (ITSP). This VoIP trunk essentially creates
a virtual tunnel from your local voice gateway to a service provider using your Internet
connection. The calls are then switched out onto the PSTN at the ITSP site. The concept
of using ITSPs for off-network calling is fairly new to the telephony world. This is because
you need your voice system to be able to transport calls over IP using the Internet. But
because you can leverage your existing Internet connection to provide transport for your
off-network calls, many businesses fi nd that they can eliminate expensive PSTN lines,
which can cut operating costs significantly. Figure 1.4 shows how a voice gateway can be
configured to connect virtually to the PSTN by tunneling voice traffic to an ITSP.
F I G U R E 1. 4
A voice gateway configured through an ITSP
is
ice ed
Vo nel
n
tu
Internet
PSTN
V
V
Customer
Site
ITSP
Provider
Voice gateways perform a great number of VoIP and PSTN services that will be covered
in greater detail in subsequent chapters of this book. This section is provided simply to
give you a 30,000 -foot view of the way a voice gateway is responsible for the translation of
voice between the IP network and legacy voice systems such as the PSTN.
Introducing the Cisco Unified
Communications 500 Series
Cisco Unified Communications requires its own section because of the unique functionality
that it brings to the table. The UC500 is part of Cisco’s Smart Business Communication
System (SBCS). Cisco has a unique suite of hardware that is geared toward small- to
medium-size businesses. Table 1.6 lists the SBCS suite of hardware and shows what services
each device performs.
Introducing the Cisco Unified Communications 500 Series
TA B L E 1 . 6
21
Cisco SBCS hardware platforms
Model
Functions
Unified Communications 500
Voice, data, security, and wireless
Catalyst Express 500
Layer 2 switching for desktops, servers, and IP
phones
Wireless Express 500
Flexible controller-based 802.11b/g wireless solution
Unified 500 Series IP phones
Hardware-based IP phones
Cisco 500 Series secure router
Advanced firewalling, wireless, and dynamic routing
protocol support for LANs and WANs
The CCNA Voice Exam focuses solely on the configuration of the UC520 hardware,
which offers voice capabilities to small businesses. It is important to know, however,
that other SBCS components are available that complement the UC520 in a network
environment. The UC500 is an all-in-one solution that combines call processing, voice mail,
voice gateway, standard IP data, optional Wi-Fi, and security into a single box. From a voice
standpoint, the system runs the CUCM Express and Unity Express software with support
for up to 48 IP phones. Because this is an all-in-one system, the UC500 has several voice
gateway methods available for connecting your CUCM Express system for off-network
calling using either physical or virtual methods. The SBCS Series is clearly targeted directly
toward small businesses of 50 or fewer users who want a single, easy-to-manage system for
all of their networking needs.
The SBCS UC500 can utilize any Cisco 7900 and 3900 Series IP phones. In addition,
the SBCS system is also the only solution that is compatible with the Cisco 500 Series
IP phones. These entry-level phones are ideal for small- office environments that are just
making the transition from a legacy key system to an integrated voice/data system over IP.
Here is a list of features that the UC500 small business solution can provide:
Support for 8 to 48 IP phones
Integrated Unity Express voice mail
Built-in eight-port Power over Ethernet (PoE) switch
Integrated Auto Attendant (AA)
Internal or external Music on Hold (MoH) sources
Integrated analog voice ports to connect to the PSTN and to utilize analog phones and
fax machines
Static routing and Network Address Translation (NAT) support
Virtual private network (VPN) support for up to 10 simultaneous remote access users
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Firewall configuration support using IOS firewall configuration commands
Optional wireless 802.11a/b/g integration
One of the key features that the UC500 Series offers is a simple-to -use graphical
configuration tool. The UC500 series can easily be managed using the Cisco Configuration
Assistant (CCA). The CCA is a Windows-based application that allows administrators to
connect to and configure the entire UC500 system using a graphical user interface (GUI).
This tool greatly improves the ease of management and support for small businesses. In
fact, the CCA can not only help you configure the UC500 device; it can also assist in the
configuration of any of the SBCS 500 Series devices listed in Table 1.6. The ability to
utilize the CCA for configuration and basic support sets the SBCS apart from Cisco’s other
offerings. Because everything can be run from a simple, graphics-driven interface on a PC,
it appeals greatly to small businesses that may not have a full-time Cisco engineer on staff.
Chapter 10 of this book is dedicated to the SBCS UC500 Series and its configuration using
the CCA 2.0 software. In addition, Appendix A presents a case study of a detailed design
and implementation of a UC500 system in a small- office environment.
Choosing an IP Telephony
Deployment Option
Now that you understand the basic components of the Unified Communications system,
let’s talk about the four Cisco-supported deployment options that we can use to design an
IP telephony solution. Each option has advantages and disadvantages that we will cover.
Most deployments will fall nicely within one of the options depending on the end-user
requirements. The four supported deployment options are as follows:
Single site with centralized call processing
Multisite with centralized call processing
Clustering over the wide area network
Multisite with distributed call processing
Single Site with Centralized Call Processing
The single-site deployment is a bit of a no-brainer. If all of your users are in the same
geographic region and are interconnected with a high-speed network, then a centralizedcall processing solution is the way to go. High availability can be achieved by clustering
multiple Communications Managers on the same local network. Voice gateway redundancy
can also be achieved by adding two or more voice gateway routers. The key factor to keep
in mind is that even though the system needs multiple CUCM systems for redundancy,
all users ultimately connect to a single system for their unified communication needs.
Figure 1.5 shows a single-site design.
Choosing an IP Telephony Deployment Option
F I G U R E 1. 5
23
A single site with CUCM
PSTN
V
Primary Site
Most of your small-office environments will utilize the single-site deployment, and the CCNA
Voice Exam does not go too far beyond this setup in the configuration knowledge it requires.
Multisite with Centralized Call Processing
The design options become more interesting when you need to provide unified
communications to multiple sites that are interconnected across a wide area network (WAN).
One option is to offer VoIP services to remote sites that communicate across the WAN to a
CUCM system located at a central site. Figure 1.6 depicts a typical multisite design.
F I G U R E 1. 6
A typical multisite WAN with a central CUCM for VoIP
Remote
Site A
Remote
Site B
IP WAN
V
PSTN
Primary Site
Phones located at Remote Site A and Remote Site B will communicate back to a single
call-processing cluster located at the primary site. This solution is best when the vast
majority of your users are located at a single site and you have a few remote sites. A reliable
and high-speed WAN connection is obviously beneficial because all voice traffic will need
24
Chapter 1
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to traverse the WAN to reach the primary site, where the call-processing, voice mail, and
other Unified Communications applications are located.
If your remote sites have a single WAN connection, this creates a single point of failure.
This may or may not be a concern to you. If it is critical that the remote sites maintain
the ability to have at least basic PSTN calling capabilities during a WAN outage, you can
implement the survivable remote site telephony service (SRST) to a voice-capable router at
each remote site. SRST is a cost-effective method to provide high-availability voice features
to remote sites that have a centralized call-processing design. Along with installing and
configuring SRST on the router, you will need to purchase one or more local PSTN services
to be used as a backup. When a WAN outage occurs, the remote site phones can no longer
communicate with the call processor. An SRST-configured router will detect the WAN failure
and begin routing calls out of the backup PSTN connections that are locally available. If you
get the opportunity to configure a router for SRST redundancy, you’ll be happy to find out that
the configuration is very similar to a standard voice router with the CUCM Express software.
Clustering over the Wide Area Network
If your organization has two to six large and geographically dispersed locations that are
interconnected by a high-speed and reliable WAN link, the clustering option may be the right
choice for you. Basically, you provision a CUCM publisher server at one location and up to five
subscriber servers at the remote site locations. The publisher and subscriber servers talk to one
another and work as a single unit. This enables administrators to maintain a single system, but
you must take care to ensure that the WAN is very stable and free of latency. On-network calls
are made across the WAN, while off-network calls are made over the local PSTN connections. If
the WAN were to fail, users would continue to connect to their local CUCM server, but all calls
placed would go over the PSTN. Figure 1.7 shows a typical WAN clustering deployment model.
F I G U R E 1.7
WAN clustering
Remote
Site A
Remote
Site B
Subscriber
Subscriber
V
V
IP WAN
PSTN
V
Publisher
Primary Site
Inter-cluster Communication
Choosing an IP Telephony Deployment Option
25
Because there is a limit to the number of active subscriber servers in a single CUCM
deployment, this solution does not scale beyond six total sites. Because the clustered
CUCM servers need to stay constantly updated and synchronized, the WAN links must
provide quick response times for updates and keepalives.
Multisite with Distributed Call Processing
The distributed call-processing deployment is the best choice when you have many users
who are geographically dispersed in multiple locations. The remote sites may also have
low-speed WAN connections that are not highly reliable. If WAN conditions are not ideal
for voice, then it might be best to distribute the call-processing and voice mail functionality
out to the remote site itself. Figure 1.8 shows a typical multisite distributed call-processing
deployment.
F I G U R E 1. 8
Distributed call processing
Remote
Site A
Remote
Site B
V
V
IP WAN
PSTN
V
Primary Site
As you can see in the figure, each remote site has a Communications Manager and voice
gateway to locally serve the remote-site phones, and each is completely independent of
every other. The design can be configured for VoIP traffic to traverse WAN links to reach
other internal destinations if possible. If the WAN is too congested, internal calls will be
made over the PSTN.
Clearly this deployment offers the utmost in reliability but it comes at a high cost. Not
only will the additional purchase of call-processing and voice mail services at each site be
required, but you also need to keep in mind that support costs go up because you have to
support multiple Unified Communications systems as opposed to one centralized system.
26
Chapter 1
Cisco Unified Communication Solutions
Summary
In this chapter, you learned the major components of a Cisco Unified Communications
system, namely, the Unified Communications Manager, Unity voice mail, and the voice
gateway. We discussed the features, architecture, and scaling capabilities of the three-tiered
Communications Manager and Unity systems. You learned the difference between on- and
off-network calling and therefore the need for a voice gateway to connect to the PSTN for
external calls. In addition, you learned about the Cisco UC500 Series SBCS, which rolls up all
three voice services along with security, wireless, and basic routing features into one little unit.
We also covered hardware- and software-based phones to give you an understanding of
the various types of voice/video capabilities a Unified Communications system provides.
Finally, this chapter discussed the three recommended deployment methods of the Cisco
Unified Communications system. You learned the pros and cons of each solution and when
it would be best to implement a particular method. You also learned about how SRST can
be used as a low- cost method to provide highly available voice service to remote sites that
are configured in a centralized call-processing environment.
Exam Essentials
Understand the differences between the main components of a Unified Communications
system. The Communications Manager handles all call-processing and setup functions.
The Unity system is responsible for the voice mail and unified or integrated messaging. The
voice gateway handles off-network calls on the PSTN. Finally, the various hard and softphones are end units that users interact with.
Know the number of end devices that each Cisco Unified Communications Manager can
handle. When choosing a system, it is vital that you understand how it will scale. The
CUCM supports a cluster environment of up to 30,000 end devices. The CUCMBE supports up to 500 end devices. Finally, the CUCM Express supports up to 250 end devices.
Understand the redundancy capabilities of each Communications Manager. Full redundancy may or may not be an important issue for a particular implementation. Typically,
users assume that when they pick up a phone, it will work. All three systems provide
some form of redundant system to keep calls moving in the event of a failure. The CUCM
uses clustered servers to provide call-processing redundancy. The CUCMBE and CUCM
Express solutions provide PSTN redundancy by implementing SRST at the voice gateway.
Know the two Unity Express hardware options. The NM- CUE takes up a module slot,
and the AIM- CUE is a card inserted on the router system board.
Know the difference between the 7900, 6900, 3900, and 500 Series IP phones. The 7900
supports both SCCP and SIP, the 6900 Series drops some of the advanced features of the
7900 but provides more features than the 3900 Series, the 3900 Series supports only SIP,
and the 500 Series supports only SCCP and can be used only with the UC500 SBCS Series.
Written Lab 1.1
27
Understand what an IP Communicator is used for. The IP Communicator is Cisco’s version of a softphone that is based on the 7970 hardware phone. It is excellent for people
who are traveling or spend much of their time away from the office.
Understand the function of the ATA hardware endpoint.
log phones and fax machines to an IP network.
The ATA helps to connect ana-
Know when to design a single -site IP telephony deployment. Single-site deployments offer
centralized calling to a group of users who are on the same high-speed network.
Know when to design a multisite centralized IP telephony deployment. When the majority of users are in a single location and you have a handful of remote sites, this design is
ideal. It also requires less capital and has lower support costs because of the centralized
nature of the voice equipment. WAN connections need to be fast and fairly reliable.
Know when to design an IP telephony cluster across an IP WAN. When you have two to
six large remote sites that are interconnected by a high-speed IP WAN, you can cluster six
CUCM servers together. This allows you to be able to administrate the cluster as one large
system that is geographically dispersed. In the event of a WAN failure, remote users continue to use the local CUCM and send calls over the PSTN.
Understand when to use SRST. In multisite centralized deployments, SRST can be used at
remote sites to give some voice functionality during a WAN outage.
Know when to design a multisite distributed IP telephony deployment. When users are
distributed at multiple sites, you should take a distributed approach. Each site has local call
processing, voice mail, and voice gateways. The downside is the increased cost and support
of multiple systems.
Written Lab 1.1
Write the answers to the following questions:
1.
What is the maximum number of endpoints a CUCMBE can handle?
2.
Which Cisco Unified Communications Manager runs on an IOS router?
3.
Which two Communications Managers and Unity systems can function within a
single unit?
4.
What is the name of Cisco’s software-based phone?
5.
What does ATA stand for?
6.
What types and numbers of ports does an ATA 180 Series have?
7.
What are the four IP telephony deployment types?
8.
What does ITSP stand for?
9.
Which Unity Express device uses up a router slot?
10. What is the maximum number of mailboxes Unity Express can support?
(The answers to Written Lab 1.1 can be found following the answers to the review
questions for this chapter.)
Chapter 1
28
Cisco Unified Communication Solutions
Review Questions
1.
Which Cisco IP phone series is not compatible with the CUCM Express system installed on
ISR routers?
A. 7900 Series IP phones
2.
B.
3900 Series IP phones
C.
500 Series IP phones
D.
Cisco IP Communicator
E.
Cisco ATA
Which Cisco voice mail system offers Unified Messaging?
A. Cisco Unity Express
3.
B.
Cisco Unity
C.
Cisco Unity Connection
D.
Cisco Unity and Cisco Unity Connection
Which of the following non-ISR routers is not compatible with CUCM Express?
A. Cisco 3725
4.
B.
Cisco 1761
C.
Cisco UC520
D.
Cisco 2621
What firmware is the Cisco IP Communicator based on?
A. Cisco 7965
5.
B.
Cisco 7970
C.
Cisco 7940
D.
Cisco 3925
E.
Cisco 524
Which Unified Communications Manager runs on a Linux operating system? Select all that
apply.
A. CUCM
6.
B.
CUCM Express
C.
CUCMBE
D.
None of the above
Which Unity solution uses compact flash (CF) for mailbox storage?
A. Unity Express NM- CUE
B.
Unity Connection
C.
Unity Express AIM- CUE
D.
Unity
Review Questions
7.
Which Communications Manager is best suited for large businesses that require high availability and uptime for thousands of phones?
A. CUCMBE
8.
B.
CUCM
C.
CUCM Express
D.
Unity
Which Cisco phone series can utilize SIP for voice signaling? Select all that apply.
A. 7900 Series
9.
29
B.
3900 Series
C.
500 Series
D.
2800 ISR
What device allows you to connect analog devices to an IP network?
A. IP Communicator
B.
SIP
C.
ATA
D.
SCCP
E.
UC500
10. What is responsible for handling off-network calls?
A. ATA
B.
SCCP
C.
CUCM Publisher
D.
Voice gateway
11. What is the name of the call-processing device that handles the database read/write functions of the CUCM?
A. Subscriber
B.
Publisher
C.
Unity Connection
D.
Unity
12. What is the maximum number of analog devices can you connect to a Cisco ATA?
A. One
B.
Two
C.
Four
D.
Eight
Chapter 1
30
Cisco Unified Communication Solutions
13. What method can be used to connect a voice gateway to the PSTN for off-network calling?
Select all that apply.
A. Physical connection to the PSTN
B.
VoIP trunk to an ITSP
C.
VoIP trunk to a PSTN
D.
Virtual connection to the ITSP
14. Which phone cannot run SCCP?
A. 7985
B.
7921 wireless phone
C.
524
D.
3951
E.
IP Communicator
15. What GUI tool can be used to configure the UC500?
A. Unity Express
B.
IP Communicator
C.
CCA
D.
SCCP
16. What platform does the CCA run on?
A. Internet Explorer 6.0 and up
B.
MS Windows
C.
Mac OS X
D.
Terminal Emulator
17. What IP telephony deployment model would you use if you have 10 remote sites that
require full availability of voice functions and applications in the event of a WAN outage?
Select all that apply.
A. Multisite with centralized call processing
B.
Single site
C.
Single site with distributed call processing
D.
Multisite with distributed call processing
18. What Cisco solution can be used at remote sites in a multisite with centralized callprocessing design to allow basic voice functionality to remote-site users in the event of a
WAN outage?
A. SCCP
B.
SRST
C.
SIP
D.
CUCM Express
Review Questions
19. Which of the following is not an IP telephony deployment model Cisco recommends?
A. Single site
B.
Multisite with centralized call processing
C.
Single site with distributed call processing
D.
Clustering over the WAN
E.
Multisite with distributed call processing
20. Which IP telephony deployment model is likely to cost the most?
A. Single site
B.
Multisite with centralized call processing
C.
Multisite with distributed call processing
D.
Multisite with centralized call processing and SRST
31
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Answers to Review Questions
1.
C . The 500 Series IP phones are compatible only with the UC500 SBCS solution.
2.
B . Only Unity offers Unified Messaging, which means it fully integrates with MS
Exchange. The other two Unity systems offer Integrated Messaging with the use of IMAP.
3.
D. The 2600 Series is not supported but the 2600XM Series is.
4.
B. The IP Communicator is a softphone that is based on the 7970 phone fi rmware.
5.
A, C. Both CUCM and CUCMBE run on a Linux OS. The CUCM Express runs on IOS.
6.
C. The Unity Express AIM-CUE has a 512MB or 1GB CF for mailbox storage.
7.
B. The CUCM allows for full clustering that offers complete HA for thousands of
endpoints.
8.
A, B. Both the 7900 and 3900 Series can run SIP for voice signaling.
9.
C. The Analog Telephony Adapter (ATA) connects standard analog devices to an
IP network.
10. D. The voice gateway handles off-network calling to the PSTN.
11. B. The publisher handles the read/write functions of the CUCM database.
12. B. The ATA 180 series has two RJ-11 interfaces to connect up to two analog phones.
13. A, D. You can connect to the PSTN by using a physical PSTN connection or by sending
traffic across an IP network to an ITSP.
14. D. The 3900 Series phones run only SIP.
15. C. The Cisco Configuration Assistant (CCA) is a Windows application that uses
a user-friendly GUI to help configure and support a number of Cisco products including
the UC500.
16. B. The CCA is a Windows application.
17. D. The multisite with distributed call-processing model pushes the call-processing and
other unified communications functions out to the remote site. In the event of a WAN
outage, no voice services are affected.
18. B. Survivable remote site telephony (SRST) can be implemented on a voice-capable remote
site router to set up basic voice calling using the PSTN in the event of a WAN outage.
19. C. It does not make sense to distribute call-processing functionality in a single site.
20. C. The distributed call-processing method forces you to purchase multiple CUCMs that are
deployed throughout your environment. This is typically the most expensive deployment
model.
Answers to Written Lab 1.1
Answers to Written Lab 1.1
1.
500 endpoints
2.
CUCM Express
3.
CUCMBE/Unity Connection and CUCM Express/Unity Express
4.
IP Communicator
5.
Analog telephony adapter
6.
One Ethernet and two analog
7.
Single site, multisite with centralized call processing, clustering over the WAN, and
multisite with distributed call processing
8.
Internet Telephony Service Provider
9.
NM- CUE
10. 250
33
C hapter
2
Traditional Telephony
THE FOLLOWING CCNA VOICE
EXAM OBJECTIVES ARE COVERED
IN THIS CHAPTER:
Describe PSTN components and technologies.
Describe the services provided by the PSTN.
Describe time division and statistical multiplexing.
Describe supervisory, informational, and address signaling.
Describe numbering plans.
Describe analog circuits.
Describe digital voice circuits.
Describe PBX, trunk lines, key- systems, and tie lines.
Describe VoIP components and technologies.
Describe the process of voice packetization.
Describe the components of the Cisco Unified
Communications Architecture.
Describe the function of auto attendants and IVRs in a
UC environment.
The 1980s were an interesting time to grow up, in regard to
technology. The personal computer was just being introduced,
and a public Internet was many years away. Nearly all distance
communication was done over the telephone. I was always fascinated with telephones and
how they worked. To me, the “phone guy” was my version of the American cowboy—
always out on the road and in the sun climbing phone poles to set up or restore service. The
boom trucks and butt sets were the telco version of horses and six-shooters. Now that voice
and data are converging, many aspects of the traditional telecommunications realm are
changing while many others remain the same. This chapter will discuss traditional public
switched telephone network (PSTN) components that are still being used today. This will
give you the background you need to understand how they are integrated in IP telephony
(IPT) networks.
We’ll fi rst discuss analog signaling and the different signaling functions each type
performs. Then I’ll describe analog and digital circuits and explain the differences between
the two. I’ll also describe multiplexing, to detail how multiple voice circuits can be
transported over a single cable. Next we’ll move on to PBX and key systems and how they
connect to the PSTN. Finally, we’ll end with a discussion of PSTN numbering plans and
why they are so important from a planning and design point of view.
Understanding Analog
Network Signaling
A standard analog telephone has such a simple interface that almost anyone can use it. All
you have to do is pick up the handset, punch in the number of the person you are trying
to reach, and all the wizards behind the curtain do the rest to connect your handset to
another person, who may be located down the street or on the other side of the planet!
We’re going to pull back the curtain to see exactly what it takes for the magic of telephones
to work. The most obvious place to start is with voice signaling.
Loop Start Signaling
When a telephone handset is sitting in the phone cradle, the telecommunications term for
this state is on - hook. If someone wishes to make a phone call, the fi rst thing they do is pick
up the handset. This action sets off a series of signaling processes that notifies the phone
Understanding Analog Network Signaling
37
switch that someone wishes to use the phone and places it in an off-hook state. In the
analog world, this is called a loop start because it opens a circuit loop back to the PSTN.
When the phone is on-hook, the loop is open. When you take the phone off-hook, it closes
the loop, sending the voltage back to the CO so they can detect that you went off-hook.
You’ll also receive audible feedback in the form of a dial tone when picking up a phone
handset. This is to signal that the phone system is ready for you to begin dialing a number.
You see, an analog line has two wires that plug into the back of your phone from the
wall jack. One wire is called the tip or ground wire; it is your link back to the telephone
company switch equipment. The second wire is called either the ring or the battery wire.
This wire provides a constant flow of low-voltage power (– 48 volts) to the phone. When
the phone is on-hook, the connection between the ring and tip wires is severed. As soon
as you pick up the handset and go off-hook, the connection between ring and tip on your
phone connects power from the ring wire, and that power then flows over the tip wire to
the phone company. Now that you have a fully powered loop, you can begin sending and
receiving signaling information to and from the phone company equipment in order to
make calls. Figure 2.1 displays the phone in an on-hook state where the loop is severed and
also in an off-hook state where the circuit connects and the loop start occurs.
F I G U R E 2 .1
Loop start signaling
On-Hook
Tip
Analog Phone
Ring
⫺48volt
Phone Company
Equipment
Off-Hook
Tip
Loop
Start
Analog Phone
On-Hook
Ring
⫺48volt
Phone Company
Equipment
Glare is an interesting and somewhat common anomaly that occurs on analog lines that
use loop start signaling. It occurs when both the local user and the telephone company
attempt to access or seize the analog circuit simultaneously, that is, when someone is
38
Chapter 2
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attempting to call your phone at the exact same time you want to use the line. When
the local user picks up the phone, they are not given any warning (the phone didn’t get the
chance to ring) that someone is on the other end. This can lead to confusion because
the local phone user is surprised to fi nd a call is already in progress. Glare occurs in loop
start signaling scenarios because the PSTN equipment does not check for a current on the
wire prior to seizing the line. Glare typically isn’t a problem in residential installations
because call volume is typically low. In business settings, however, glare can become a real
nuisance. To eliminate the glare problem, ground start signaling was developed.
Ground Start Signaling
A different approach to loop start in the analog world is ground start. From a pure
signaling standpoint, ground start momentarily grounds one side of the circuit. This
grounding is an indication to the PSTN equipment that the circuit is ready for use. This
added signaling process avoids the glaring situation found in loop start signaling. A good
example of ground start signaling is an analog pay phone. Pay phones require that the
calling party fi rst insert change prior to making a call; the physical insertion of coins into
the phone activates ground start signaling. When a coin is inserted into a pay phone, it
causes a lever within the phone to tip. This lever action is what grounds the circuit. Once
the signal is grounded, the PSTN knows that money has been inserted, and a call
can then be placed.
Analog Network Event Signaling
All analog phone calls need a way to signal events on the phone network in order to
establish communication between end devices. There are three distinct types of network
signaling in a voice network:
Address signaling
Informational signaling
Supervisory signaling
Each type provides vital functions for call setup and end-user feedback to inform us
that either a call is being processed properly or a problem has occurred. Let’s look at each
signaling type.
Address Signaling
Address signaling represents the transmission of digits to the remote party that the calling
party wishes to dial. Two types of address signaling are in use. One method is pulse
dialing, also called rotary dialing because the numbers are arranged on a round disk. The
dial uses a mechanical method of quickly going on-hook and off-hook within a certain
timeframe. Each digit 0 to 9 is represented by one of these on/off-hook transitions as it
spins around. A different way to look at this is to think about what on-hook and off-hook
sequences do. Basically we’re momentarily turning on and off the power current to the
Understanding Analog Network Signaling
39
phone switch. The phone switch then counts the number of starts and stops in the power
flow and determines the intended digit to dial based on this calculation. It’s very much like
simple Morse code. For example, a user dials 5 by inserting their fi nger into the properly
labeled slot and turning the dial clockwise until it stops. When the fi nger is removed, the
dial rotates back to its original position. During this return motion, a series of five
on/off-hook rapid transitions occurs. The phone recognizes these and collects the digits
that the user wishes to dial.
The second and far more popular type of address signaling is dual-tone multi-frequency
(DTMF ). It is often called touch-tone dialing. This form of address signaling uses very
specific audible tones that the phone network equipment recognizes. It combines two voiceband frequencies to represent twelve different numbers and symbols. The switch recognizes
the tones and properly interprets the intended destination phone number. Figure 2.2 shows
how the frequency combinations are used to represent the digits on a phone.
FIGURE 2.2
DTMF frequency creation
DTMF Frequencies
1209 Hz
697 Hz
770 Hz
852 Hz
941 Hz
1
1336 Hz
1663 Hz
2
3
ABC
DEF
4
5
6
GHI
JKL
MNO
7
8
9
PQRS
TUV
WXYZ
0
#
A Cisco voice gateway can be configured to recognize either DTMF or
pulse address signaling. The default signaling is DTMF.
40
Chapter 2
Traditional Telephony
Timeout values are associated with the maximum and minimum speeds a user can dial
numbers using DTMF. The quickest duration between dialed digits is 45 milliseconds,
whereas the longest time allowed between dialed digits is 3 seconds.
Informational Signaling
Informational signaling is all about letting the calling party know what is going on with
the phone system and the attempted call. As soon as you pick up the receiver of a phone,
you hear a dial tone. This tone informs you that the phone is operational and talking to the
phone switch. People commonly listen to make sure they hear the dial tone before dialing
a number. Informational feedback is generated from the phone switch to the user in the
form of audible tones and/or voice messages. Table 2.1 lists some of the most common
informational signals and their meanings.
TA B L E 2 .1
Informational signals
Informational Signal Type
Signal Meaning
Dial tone
Phone is in an off-hook state and ready to accept user input
with the keypad.
Busy
Called number phone is currently in use.
Number not in service
Called number is not available on the phone network.
Call waiting
An incoming call is being made to line 2 on the phone; line 1
is in use.
Ring-back
The phone company is attempting to establish the connection
to the called party.
Reorder
All local circuits are busy; thus the call cannot be completed.
This is also known as a “fast busy” signal.
Congestion
The long-distance company is unable to complete the call.
Handset off-hook
Someone has picked up the handset of a phone from
the cradle.
Informational signaling may be different depending on what region you
are in. A dial tone in the United States is different from a dial tone in
Mexico, for instance.
Comparing Analog and Digital Circuits
41
Supervisory Signaling
Supervisory signaling deals with the behind-the-scenes part of call setup and teardown.
There are many different types of supervisory signaling, depending on the types of circuits
being used and the type of phone equipment making the signals. This signaling is done
to ensure that the phone system properly interprets user input and that that user input is
properly handled. For example, a phone seizure signal is a very common supervisory signal.
When you pick up your phone, a seizure supervisory signal is fi rst sent to the telephone
switch to ensure that you have control over the analog circuit. As soon as the line is seized,
you receive an informational signal in the form of a dial tone.
Supervisory signaling can occur in or out of band depending on the type of circuit being
used. In-band signaling means that the signals are transported on the same wire as the
voice traffic. Out- of-band signaling refers to signaling that utilizes a separate transport
medium such as a separate pair of wires. Table 2.2 lists many common types of supervisory
signaling.
TA B L E 2 . 2
Supervisory signals
Supervisory Signal Type
Signal Meaning
Seizure
Signals the phone system to change the line/trunk state from
idle to active.
Wink/hook flash
Indicates that the phone system is ready to receive address
information in the form of DTMF or pulse digits.
Answer
Indicates when the remote-side phone is answered and
two-way communication is established.
Disconnect
Indicates that either phone in the two-way communication goes
on-hook. The call is torn down and the circuit returned to an
idle state.
Robbed-bit
In-band bits are used to signal the start and end of address
information.
Comparing Analog and Digital Circuits
The public switched telephone network established the framework for long-distance
communication long before the Internet. The PSTN is a global network of telephony
equipment that once was purely an analog system. Today the PSTN consists of mostly
digital circuit-switched phone systems that are interconnected. It is important to
42
Chapter 2
Traditional Telephony
understand both analog and digital circuits and why we’re moving to a system that will
eventually be 100 percent digital. Let’s fi rst look at analog circuits and how they transmit
voice from one phone to another. Then we’ll move on to digital circuits and how they
convert analog into digital.
The Analog Signal
The goal of any voice circuit is to transmit sounds (typically the human voice) from
one point to another. Using analog technology, the human voice is picked up from the
transmitter portion of a telephone and is translated into an electrical signal that varies
continuously with changes in the sound. Sounds such as the human voice are in analog
form to begin with. The changes in pitch and tone as we pronounce various words create
variations in the sound wave. A microphone and analog circuit are used to capture the
analog sound waves and transmit them in an electrical form over copper wiring. Once
the analog sound waves are in an electrical form, they can be transmitted across the PSTN
to the other end of the phone connection. When the electrical signal reaches the intended
destination, it is converted back into analog sound waves and sent through the receiver
speaker. Figure 2.3 depicts a typical analog sound wave.
FIGURE 2.3
Analog sound wave
Wavelength
Peak Amplitude
Peak-to-Peak Amplitude
Frequency
⫹1
0
Time
⫺1
Comparing Analog and Digital Circuits
43
You’ll notice that the figure shows the frequency of a sound over a period of time. A
wavelength of a sound wave is the distance between each wave collected. The frequency of
a wavelength is measured in cycles per second, or Hertz (Hz). You’ll also note two different
amplitudes listed. The peak-to - peak amplitude is the variation in the frequency over a onewavelength period. The peak amplitude is the same frequency variation, but it is measured
from the mean, which is 0 on the graph.
Analog circuits carry voice signals in a very pure form. A standard plain old telephone
service (POTS) line from the phone company is typically delivered to a home or business
on two wires. These two wires provide full-duplex voice conversation. The last hop on the
PSTN before it goes to the customer premises is called the central office (CO). The central
office bundles your circuit with other customers’ circuits and switches them as needed
to other COs on the PSTN. Your specific circuit coming into the central office is called a
local loop. On the customer side of the analog circuit, Cisco has several types of analog
interfaces to terminate various PSTN and analog endpoints on the voice gateway. Let’s
discuss each type.
Analog Voice Interfaces
Literally dozens of different analog interfaces can be used on Cisco hardware. The CCNA
Voice exam focuses on the most popular types in use today. This next section covers FXS,
FXO, and CAMA analog interfaces. Exam takers need to understand the situations where
each interface type is used.
Foreign Exchange Station Interface
The Foreign Exchange Station (FXS) is an interface that connects directly to an analog
endpoint such as an analog phone or fax machine. The connection handoff is a standard
RJ-11 port. Figure 2.4 shows how you connect an analog line to an FXS port that connects
to the PSTN.
FIGURE 2.4
An FXS interface
PSTN
2-wire POTS Line
FXS Port
Analog
Phone
FXS ports are commonly found in residential homes that require very few analog lines.
The interface provides voltage and signaling to analog devices. So in a residential home
situation, we assume that all of the voltage and signaling will be provided to us from the
PSTN switch equipment.
44
Chapter 2
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Analog devices don’t really care what signaling you use, so FXS interfaces can use either
loop start or ground start signaling from the source. Typically you will configure loop start
signaling when connecting to analog devices like phones or fax machines. FXS ports can
also be configured to use ground start. This is more common when you use an FXS port to
connect to a legacy PBX.
Foreign Exchange Office Interface
Instead of plugging directly into an analog phone like the FXS port does, a Foreign
Exchange Office (FXO) port connects to a PBX. The FXO interface assumes that all
dial tones, ring indicators, and other call-progress signaling are provided locally by
the equipment attached to it such as a key system or PBX. Contrast that with an FXS
connection where the device that plugs into the port does not provide any form of
signaling. Instead, it relies on the backend equipment to provide it. Typically, a business
will provision a handful of FXO ports from the PSTN and then connect the phones to
the PBX. If an analog phone or fax machine needs to make an off-network call, the PBX
switches the connection to one of the free analog lines. This is beneficial to businesses
because not every phone requires an off-network analog line at any given time. If you plan
accordingly, you can get away with paying for a fraction of the number of analog lines than
you have phones on your local network. Figure 2.5 shows a single analog line terminating
at a PBX with an FXO interface.
FIGURE 2.5
An FXO interface
PBX
PSTN
2-wire POTS Line
FXO Port
FXS Ports
Analog
Phones
With this type of setup, there will be only one PSTN number per line. As with the
FXS port, the connection handoff is RJ-11. The diagram also points out a simple way to
remember the difference between FXS and FXO ports. FXS interfaces connect analog
telephones, while FXO interfaces are used to connect to the PSTN. The fi rst hop out to the
PSTN cloud is called the central office. (A quick mnemonic: The central “o”ffice connects
to your FX“O” interface.) In this scenario, either the single number would ring all phones
on the PBX when dialed, or it would ring a single phone handled by a live operator, or the
PBX would utilize an auto attendant (AA). An auto attendant is much like a live operator,
but instead of a human, it is an automated system with voice prompts to assist the caller to
be routed to the proper contact using internal extensions.
Comparing Analog and Digital Circuits
45
Analog Direct Inward Dial Service Interface
The analog direct inward dial (DID) service is very similar in functionality to the FXO
circuit. The PSTN connection plugs into the PBX. The main difference between a
standard FXO line and an analog DID is how the phone company handles the sending
of digits. With an FXO connection, each physical analog line terminated at the PBX
corresponds to a specific phone number. With an analog DID, the phone company can
offer a service that bundles any number of analog lines and trunks them. Trunking means
that a call can be sent to a specific number across any of the analog lines instead of
specifically assigning them as in FXO connections. The phone company will then strip off
all but the extension digits at the local PSTN central office. So in North America, when an
off-network person calls your company’s number, say 555-123 -2221, the 555-123 digits
are removed at the last hop on the PSTN and only the 2221 is sent to your PBX. Your PBX
then takes that extension and switches it to the appropriate phone extension. Figure 2.6
shows this scenario.
FIGURE 2.6
Direct inward dial service interface
2
Local telco strips off all but last four digits and
sends to company PBX on single analog line.
22
21
PBX
3
PBX routes
to Ext 2221
Ext 2221
2-wire POTS Line
PSTN
Ext 2221
DID port
Ext 2221
1 User dials: 555-123-2221
Centralized Automatic Message Accounting Service Interface
All of the previous analog interfaces that we discussed connected to the PSTN. An analog
Centralized Automatic Messaging Accounting (CAMA) circuit is used exclusively for
Emergency 911 (E911) service in North America. Some states require businesses over a
certain size to connect directly to the E911 service. A CAMA link is one way to comply
with this law.
The E911 service is largely built outside the PSTN, and calls are routed differently
within the network. Typical PSTN phone calls are routed based on destination phone
number. With the E911 service, phone calls are routed based on source phone number. The
source number is used to route calls to an E911 because it can pinpoint the caller’s location.
When the location of the number has been determined, the call is then switched to the
proper public service answering point (PSAP), where the 911 operator can assist. Figure 2.7
depicts what happens when a user dials 911.
46
Chapter 2
F I GU R E 2 .7
Traditional Telephony
CAMA service interface
PBX
Normal offnetwork calls
PSTN
2
E911
3
E911 routes call to PSAP
based on source number
All 911 calls out CAMA
CAMA trunk
PSAP
1
User
dials 911
911 Operator
It is easy to understand why E911 calls rely so heavily on the source phone number. In
an emergency, it is far more important for emergency services to be able to fi nd you than
for you to know where they are located!
E911 to the Next Level
Voice engineers have just completed a major VoIP implementation at a major university
campus. One of the tasks on the test plan was to test the 911 service to ensure that
emergency services can correctly identify where users are calling from in any of the
multiple buildings that cover several city blocks. When performing various testing with
emergency services, they determined that no matter where users called from, it looked
like the calls were being made from a single building. This was a big problem because
some buildings were up to several miles away. Emergency services would not be able to
properly identify the location of users within a reasonable distance.
The E911 service relies heavily on using the source number as the location of the person
who requires emergency assistance. A major problem exists when you begin to build
out an enterprise Unified Communication system that spans multiple buildings over a
large geographical region. Depending on your design, you may have all of your PSTN
lines terminated into a single location, like our university campus situation. If a phone
user were to dial emergency services, that user could be several miles away from where
the PSTN lines are physically located. So how can we better pinpoint where a person
is on our network? Cisco to the rescue! Using what’s known as the Cisco Emergency
Responder, the server can dynamically track and update the location of Cisco IPT phones
and place it into a database. This information is used to ensure that the emergency call
is routed to the proper PSAP. It also better directs emergency personnel to the actual
location of the caller instead of the location of the terminated phone line.
Comparing Analog and Digital Circuits
47
The Analog-to-Digital Conversion Process
The analog circuits just described seem to work well, so why do we need digital circuits?
There are two main reasons. One deals with inefficiencies of analog, and the other deals
with its distance limitations. In regards to efficiency, analog simply does not scale well.
Based on what you’ve learned about analog signals, you know that for each voice call
made with analog, we need two wires: one for the ring and the other for the tip. With
digital signals, we have the capability to sample analog voice frequencies, turn the result
into binary, compress it, and send it across an IP network using less bandwidth than just
sending the analog waveform over the wire. Because of the smaller size, we can use other
techniques to send multiple voice streams over fewer pairs of wires.
With analog signals there is also a distance limitation to contend with. Because analog
signals are purely electrical on the wire, over longer distances these electrical signals become
degraded. To address the analog degradation problem, electrical repeaters can be used to help
extend the distance of analog. These repeaters sit on the wire at certain distance points. The
repeater’s job is to listen to the electrical signals coming in one end and reproduce the signals
out the other. While this may work to extend analog distances a bit farther, they stop becoming
productive at a certain point. This is because repeaters can interpret electrical pulses called
noise on the wire and falsely assume they are part of the signal to be repeated. This noise gets
retransmitted by the repeater. After the signal is repeated several times, a considerable amount
of electrical noise is now accompanying our legitimate analog voice signal. When it finally
reaches the other side, the electrical noise comes out as audible static on the receiving phone
handset. You’ve probably played the “grapevine” game as a child, where one person whispers
a message to the next, and then they whisper what they interpreted to the next child. By the
time the message is repeated to the last kid on the grapevine, the message is almost always
wrong! This game is very similar in concept to what happens with analog repeaters over time.
So now that you know we need digital circuits to overcome inefficiencies and distance
limitations of analog circuits, let’s look at how we can transform analog waves into a digital
format. Digitizing voice solves our distance problem because instead of transporting electrical
signals, we only have to worry about transporting numbers, as you will soon see. Then,
after you’ve seen how to digitize analog waves, I’ll explain how we can efficiently transport
multiple voice streams over the same pair of wires using a technique called multiplexing.
Four steps are necessary to transform an analog signal into a compressed digital signal.
The steps always occur in the following order:
1.
Sample the analog voice signal.
2.
Quantize the sample.
3.
Encode the digital sample.
4.
Compress the encoded sample.
Let’s take a closer look at each of these steps so we can fully understand the digitizing process:
Step 1: Sample the Analog Voice Signal
A standard analog telephone can pick up sound waves from 0 to 4000 Hertz. Using this
frequency range, the human voice is sampled 8,000 times per second. How did they come
48
Chapter 2
Traditional Telephony
up with this sample rate? In 1924, a Bell Labs engineer by the name of Dr. Harry Nyquist
found that by using a mathematical formula, he could fi nd the optimal relationship
between audio quality and acceptable bandwidth sample rates. Nyquist was doing
theoretical research in the field of improving transmission speeds of data using analog
lines. His research provided the base for digital transmissions that are currently being
used. While performing this research, Nyquist discovered the bandwidth-saving benefits of
continuously sampling analog signals and converting them to digital form. This theory is
now referred to as the Nyquist formula:
max data rate(bits/sec) = 2 × B × log2 V
B = bandwidth and V = number of voltage levels
So what does all of this mean? Nyquist found that if you sample a sound wave at two
times the highest frequency perceived, you can accurately reconstruct the signal digitally.
Since 4000 Hz is approximately the highest frequency a human voice can achieve, sounds
are sampled 8,000 times per second.
Even though the frequency range we use with the Nyquist formula is between 0 and
4000 Hz, the average human voice falls within the range of 200 to 2800 Hz. Filters are set
up on the phone to collect any sound that falls within the range of 300 to 3300 Hz. Sound
waves in the ranges 0 –299 and 3301– 4000 Hz are used for out- of- band signaling. Voice
traffic is considered to be in-band.
Once the analog sounds are fi ltered, a technique called Pulse Amplitude Modulation
(PAM) is performed on the waveform. PAM takes a slice of the wavelength at a constant
number of 8,000 intervals per second. Using these samples, it is possible to reconstruct
the entire wave on the other side of the connection without having to actually send the
complete wave. Figure 2.8 shows a waveform sample being taken.
FIGURE 2.8
A digital sample
Comparing Analog and Digital Circuits
49
Step 2: Quantize the Sample
The process of digitizing voice is called pulse code modulation (PCM). PCM uses a method
called quantization to encode the analog waveform into digital data for transport and
then to decode the data to turn it back into analog form (the DC voltages that drive phone
speakers). Quantization is the language used in this encoding process. Each analog
sample is given a quantized number code that gets as close as possible to the amplitude
of the signal. In the next step, these numbers will be used to encode the waveform for
transport. Figure 2.9 shows the analog sample being quantized.
FIGURE 2.9
Quantization
Quantize
Coding
4
4
0
3
9
3
6 2
3
3 3 2 8
3 2 2
2 4 5 7
1
Step 3: Encode the Digital Sample
After quantization has been completed, step 3 of the PCM process is to put the data in
a format that can be easily sent across the wire. We use the binary system to make this
happen. Binary is the numbering system used in digital electronic systems. It consists of a
series of 1s and 0s called bits to represent any numeric value.
No matter which PCM technique you use, the encoder uses the quantized numbers
that represent analog waveforms and converts these numbers into binary. The 8,000
sample rate is converted into an 8 -bit binary number. Therefore we need 64 Kbps of
bandwidth to transport a single digital call. We arrive at the 64 Kbps (or 64,000 bits)
using the following math:
8,000 samples × 8 bits per sample = 64,000 bits
50
Chapter 2
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There are multiple methods to encode the quantized signals. The intelligence or
algorithm behind encoding and decoding is called a codec (short for compressor/
decompressor). Depending on which codec is being used, the quality of the encoded
waveforms as well as the size of the encoded data stream can differ. Either of two
common types of PCM binary conversion techniques is used in most voice systems. The
fi rst PCM type is called u-law and is most commonly used in the United States, Canada,
and Japan. The second PCM binary conversion type is a-law. It is used just about
everywhere else on the planet. It is important to note that the two PCM techniques are not
compatible with each other and need to be transcoded for interoperability. Transcoding is
the process of translating one codec into another.
Now the signal is ready either to be sent across the wire immediately or to be optionally
compressed prior to transport. Figure 2.10 shows the binary encoding process.
F I G U R E 2 .1 0
Binary encoding
Quantize
Coding
3
9
3
6 2
3
3 3 8
3 2 2 2
2 4 5 7
1
4
4
0
Binary Encoding
101000101
101000100
101000001
Step 4: Compress the Encoded Sample (Optional)
Compression is all about getting the biggest bang for the buck. As mentioned earlier,
codecs are used for encoding and decoding digital voice data. Remember that even though
we’ve been discussing traditional telephony circuits up to this point, compression is
used only with newer voice technologies, which we’ll discuss in the next chapter. These
specifications also contain logic for compressing and decompressing this data so it can
be more efficiently sent across the wire. When fewer bits are used per voice conversation,
then more conversations can simultaneously exist on a fi nite amount of bandwidth.
Compression attempts to eliminate redundancy in the data that is sent. It attempts to match
your original encoded sample with something very similar to a known sample. It then uses
Comparing Analog and Digital Circuits
51
this known sample, which can be identified with a much smaller binary stream, to send
across the wire. The smaller the stream that is sent across the wire, the more individual
streams can be sent across the same wire at the same time!
There is a tradeoff to compression, however. Because the actual encoded sample is not
used, when the digital sample is decoded and turned back into analog, it is not an exact
reproduction of the original sampled source. Typically what people notice is that the audio
turns the human voice into a more robotic sound. And the more the sample is compressed,
the more it loses any kind of uniqueness on the other end.
Analog vs. Digital: You Can See the Difference!
Sometimes it’s difficult to understand the value behind the movement toward
abandoning analog for digital voice circuits. To better clarify why the change is good,
let’s look at a slightly different medium that might be easier to understand because
we can literally see the difference in full high definition! Of course, I’m referring to the
movement from analog to digital television transmission.
In the United States, a government regulation mandates that all over-the-air broadcasters
must broadcast solely digital television transmissions. This began on June 12, 2009, and
eliminated all analog forms of broadcasting.
It is interesting to compare the reception of digital and analog television transmissions
over the air. If you’ve ever used an antenna to receive analog signals, if you didn’t have
optimal reception, you’d literally see the distortion in the form of snow. Even though
there was distortion, however, you could still see the picture even though the clarity was
degraded. This is the same distortion that you can encounter using an analog voice line.
The signal might get through, but it may be full of distortion.
In contrast, when you pull in digital television signals over the air, it’s all or nothing.
Either you get a high-quality signal that is far superior to analog or you get a blank
screen. There is no snowy, degraded signal as you might find with analog. Moving to
digital is a bit of a tradeoff between quality and ability to receive the signal. If you have a
weak digital signal, you won’t see a picture, but if you have a strong signal, the picture is
unrivaled compared to analog.
So why would the U.S. government enforce a law that forced television broadcasters
to move to an all-digital transmission format? Just as we learned with digital voice, it is
possible to digitize and compress video transmissions to use less radio frequency space
when compared to analog television transmissions. Moving to all-digital transmissions
freed up UHF frequencies that were at one time consumed by larger analog transmissions.
These newly unused frequencies can then be redistributed for different purposes.
52
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Digital Voice Interfaces
Analog circuits are fi ne if you require only a few PSTN lines into your business. If you
need approximately 10 or more external lines, it is typically more cost effective to look
into a digital trunk circuit such as a T1 or E1. From a physical point of view, T1 and E1
circuits are typically terminated at the customer site in the form of copper wiring, usually
Category 5 cabling. This same cabling is used for Ethernet LAN connections. The circuits
are terminated using a standard RJ- 45 connector. Looking at the eight pinouts on the
RJ- 45, you can see that a T1 uses pins 1 and 2 for transmit and 4 and 5 for receive.
Figure 2.11 shows T1/E1 pinouts to give you a better understanding of how the
wiring is used.
F I G U R E 2 .11
T1 and E1 RJ-45 pinouts
RX
RX
TX
TX
RJ-45
Connector
Pin Number: 1 2 3 4 5 6 7 8
The digital circuit handed off to the customer is again called the local loop. Finally, most
digital circuits bundle multiple voice lines on a single trunk line that is handed off to the
customer. Let’s look at some of the more popular digital circuits that PSTNs offer.
ISDN Basic Rate Interface
The ISDN Basic Rate Interface (BRI) circuit offers the ability to make two simultaneous
calls on 64Kbps channels, called bearer channels, or B channels. The voice communication
itself uses the entire amount of the 64Kbps channel. All call signaling is performed outside
the voice channel. As you have already learned, this type of signaling is known as out- ofband signaling. On the ISDN BRI, signaling takes place on a third channel that has 16
Kbps of bandwidth. This signaling channel is referred to as the data, or D, channel. Thus, a
single ISDN BRI circuit offers two B channels plus one D channel for signaling both bearer
channels. The main type of BRI signaling used on the D channel is Q.931. This is the most
popular signaling format used by PSTNs around the world.
It is important to note that there is a difference between the ISDN bit rate and the
available bandwidth for making calls. The complete bit rate of an ISDN BRI circuit is 192
Kbps. This includes the 2 × 64Kbps B channels and 1 × 16Kbps D channel. The other
Comparing Analog and Digital Circuits
53
48 Kbps is used for framing and synchronization. So while the bit rate may be 192 Kbps for
an ISDN BRI, the bandwidth is 144 Kbps.
T1 Channel Associated Signaling
The T1 channel associated signaling (CAS) has 24 channels associated with it. Each one
of these channels can transport voice traffic. This means that 24 simultaneous voice calls
can occur at the same time. Signaling for the traffic occurs in-band, meaning that bits that
are typically used for voice are taken and reused to help with control and signaling of the
circuit. This is often referred to robbed - bit signaling (RBS). 8 Kbps of each 64Kbps channel
is used for signaling instead of utilizing an entire channel for shared out- of-band signaling.
Let’s break this down a bit further for your understanding.
Each T1 CAS has 24 channels that can transmit 8 bits per channel each. This gives us
a total of 192 bits. The T1 has one additional bit for framing, bringing the total to 193
bits. Two types of line coding can be used on a T1 CAS. The fi rst type of line coding is
called Super Frame (SF). This is an older and less- efficient type of framing. Super Frame
bundles 12 of these 193 -bit frames together for transport. It then uses the even-numbered
frames as signaling bits. The T1 CAS signaling then looks at every sixth frame for signaling
information. This comes out to be 2 bits that are referred to as the A and B bits, which
reside in frames 6 and 12. Figure 2.12 shows the robbed-bit framing process.
F I G U R E 2 .1 2
SF robbed-bit framing
Frame 12
Frame 11
Frame 10
Frame 9
Frame 8
Frame 7
Frame 6
Frame 5
Frame 4
Frame 3
Frame 2
Frame 1
Super
Frame
1 bit robbed
from 8 bits of
frames 6 and 12
A newer CAS framing method is called Extended Super Frame (ESF). This method
bundles 24 of the 193 -bit frames together. Because ESF bundles larger groups of frames,
this frees up additional bits. So now, with ESF we have 4 bits for signaling instead of the
2 that SF offered. The 4 bits are referred to as bits A, B, C, and D. They reside in frames
6, 12, 18, and 24. These extra framing bits allow for more intelligence and the ability to
process error checking using the cyclical redundancy check (CRC) method. The better
54
Chapter 2
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efficiency and error handling make ESF framing the far more optimal choice. Almost every
modern telephone provider now uses ESF for their T1 circuits. Figure 2.13 shows the bits
being robbed from the 4 frames using ESF.
F I G U R E 2 .1 3
ESF robbed-bit framing
Frame 24
Frame 23
Frame 22
Frame 21
Frame 20
Frame 19
Frame 18
Frame 17
Frame 16
Frame 15
Frame 14
Frame 13
Frame 12
Frame 11
Frame 10
Frame 9
Frame 8
Frame 7
Frame 6
Frame 5
Frame 4
Frame 3
Frame 2
Frame 1
Extended
Super
Frame
1 bit robbed
from 8 bits of
frames 6, 12, 18,
and 24
Three types of signaling methods are used on CAS circuits. These signaling methods use
the four A, B, C, and D ESF framing bits for synchronization, control, and error handling
of the circuit. When you provision a T1 CAS from the phone company, they need to tell
you what type of signaling they will be using. You will have to configure the correct type
of signaling on your T1 voice gateway interface. The three types of signaling methods are
Loop start
Ground start
E&M (A supervisory signaling mode uses DC signals called the E and M leads. They
were mostly found within the PSTN between phone switches. The technology is
becoming obsolete in favor of PRI circuits.)
Did you ever wonder why a T1 is said to be 1.544 Mbps? Now that you
understand framing, it is a bit easier to comprehend. We know that a T1
CAS has 24 channels at 8 bits per channel plus one framing bit. Because
we sample voice 8,000 times every second using the Nyquist formula, we
need to send 8,000 of these 193 -bit frames across a T1 every second. 8,000
× 193 is 1,544,000 bps or 1.544 Mbps. Of course, this is the complete bit
rate. To calculate the bandwidth rate, we must subtract the one bit that is
used for framing and synchronization. So the true bandwidth for a T1 is
8,000 × 192, which is 1,536,000 bps or 1.536 Mbps.
Comparing Analog and Digital Circuits
55
For a T1 CAS this means that all 24 channels can be utilized. A downside to the CAS is
that since signaling information is in-band, only 56 Kbps is available for voice calls. A more
CPU -intensive encoding technique will be required to compress the voice from the standard
64 Kbps to 56 Kbps. Table 2.3 lists the components of the T1 CAS.
TA B L E 2 . 3
T1 CAS components
Component
T1 CAS
Location used
North America
Total bit rate
1.544 Mbps
Total bandwidth
1.536 Mbps
Total number of channels
24
Number of usable voice channels
24
Voice bandwidth per channel
56 Kbps
Framing technique
SF or ESF
Signaling methods
Loop start, ground start, and E&M
E1 Channel Associated Signaling
The E1 CAS is a bit of an oddball. E1 circuits have a total of 32 channels, compared to
24 channels with a T1. Unlike the T1 CAS, which uses robbed-bit signaling for control and
signaling of the circuit, the E1 uses out- of-band signaling on channels 1 and 17. Figure 2.14
breaks down the channel responsibilities of an E1 CAS.
F I G U R E 2 .1 4
E1 CAS channels
Channel 1:
Framing
Channel 1
Channel 17:
Out-of-Band
Signaling
Channel 32
56
Chapter 2
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T1 and E1 Primary Rate Interface
The major difference between a T1 PRI and the previously mentioned CAS T1 is
the way signaling is handled. Unlike the T1 CAS circuits that use in-band signaling, the
T1 PRI circuits utilize out- of-band signaling. You might be surprised to learn that
two very popular framing methods for the T1 PRI are Super Frame (SF) and Extended
Super Frame (ESF). Although these are the way the circuit frames our voice for
transport, the process stops there, because it doesn’t have to bother with stealing a bit
from every sixth frame for signaling. Instead, all signaling occurs on the separate outof-band channel.
The T1 and E1 Primary Rate Interface (PRI) is the PSTN big brother of the ISDN
BRI circuit. T1 circuits are most often found in North America and parts of Asia,
while Europe and much of the rest of the world use the E1 PRI. This out- of-band
signaling is called common channel signaling (CCS). Each individual voice connection
has its own separate 64Kbps channel, and the signaling channel is also 64 Kbps. We’ll
fi rst look at how the T1 circuit works and then highlight the differences between it
and the E1 circuit.
T1 PRI Twenty-four logically unique 64Kbps circuits make up a T1 PRI. Channel 24
is designated as the signaling channel for the circuit. The PRI is said to be a 23B + 1D,
which means that there are 24 bearer channels for voice and 1 data channel for signaling.
T1 circuits are most commonly offered by public telephone companies that operate in the
United States, Canada, Japan, and South Korea. There is no real reason for this except that
phone companies that operate within a country usually standardize on the type of PRI
that is offered.
The T1 PRI uses Q.931 for signaling. As you can see, this is the same type of signaling
used in ISDN BRI connections. Because we have a full 64Kbps DS0 channel for signaling,
however, Q.931 can transmit all of the signaling information required for each of the
23 voice channels. In fact, enough bandwidth is available on the data channel that other
signaling functions can be sent across the D channel. Many telecommunications vendors
send proprietary signaling information across the D channel to add additional control
and services.
E1 PRI The E1 PRI bundles 32 logically unique 64Kbps channels. If we label
the channels 1– 32, channel 1 is responsible for framing and channel 17 is used
as out- of-band signaling. This is exactly the same method as the E1 CAS described
above. The difference is that the E1 PRI uses Q.931 signaling whereas the E1 CAS uses
one of three other signaling formats. So if we subtract 2 channels from our E1, this
leaves 30 channels with which to send voice traffi c. Another way to put it is that the
E1 PRI is a 30B + 2D circuit.
Table 2.4 lists the primary differences between a T1 and an E1 PRI circuit.
Comparing Analog and Digital Circuits
TA B L E 2 . 4
T1/E1 PRI components
Component
T1 PRI
E1 PRI
Location used
North America
Europe
Total bit rate
1.544 Mbps
2.048 Mbps
Total bandwidth
1.536 Mbps
1.984 Mbps
Total number of channels
24
32
Number of usable voice channels
23
30
Voice bandwidth per channel
64 Kbps
64 Kbps
Channel used for out-of-band signaling
24
17
Common framing signaling
Q.931
Q.931
Comparing CAS and CCS Circuits
So, if given the opportunity, which T1 should you order, the T1 PRI that uses CCS or the
T1 CAS that uses in-band RBS? Let’s break down the positives and negatives of both:
T1 PRI Pros
Full 64 Kbps for voice signals.
Uses Q.931 signaling protocol, which is universally used around the world.
Additional signaling bandwidth means more flexibility for vendors to communicate
proprietary signaling information.
Higher security because signaling is out- of-band.
T1 PRI Cons
Only 23 usable signals to transport voice
T1 CAS Pros
Can use all 24 channels for voice calls
More efficient signaling mechanism
Offers three different signaling methods
57
58
Chapter 2
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T1 CAS Cons
Slightly degraded call quality
Slightly increased router CPU utilization due to compression
Signaling not as widely used
If I were given the choice, I would choose to use the T1 PRI circuit for my connection
to the PSTN. Even though I do give up one full channel for signaling, it quite simply is the
preferred standard and makes it much easier to understand how signaling works!
Multiplexing
In the previous section we discussed how digital circuits such as ISDN BRI/PRI and CAS
T1s have multiple channels that divide the bandwidth into separate voice segments. For
example, you can think of a T1 PRI as having 23 physical D channels for voice traffic and
1 D channel for signaling information. Figure 2.15 shows the logical representation of
a T1 PRI.
F I G U R E 2 .1 5
PRI T1 circuit
1 D Channel
Logical T1 Circuit
23 D
Channels
While it is useful to think of these channels as physically separate wires, that is not
actually the case. Instead, all of the circuits are transmitted over the same pairs of copper
or fiber- optic connections. In reality, the telecommunications equipment uses what is
known as multiplexing to logically segment a single connection into multiple connections.
Multiplexing
59
Multiplexing is the digital circuit’s answer to the analog efficiency problem. With analog,
every phone call requires a pair of wires to transmit the signal. If we digitize our voice
calls, we can reduce the bandwidth requirements needed to transport the calls. And we can
fi nally use multiplexing to transport multiple calls over the same pair of wires.
While there are many different types of multiplexing, the two main types you should
become familiar with are Time-Division Multiplexing (TDM) and Statistical Time-Division
Multiplexing (STDM). Both types handle multiplexing slightly differently and ultimately
handle the circuit bandwidth in different ways. Now we’ll take a closer look at these
two methods.
Time-Division Multiplexing
Time - Division Multiplexing is often referred to as circuit mode multiplexing because of
the fi xed nature of the timeslots. Each timeslot reoccurs in a specific order. This means that
a limited number of circuits can be transmitted on a single connection. This is the type of
multiplexing typically found in current PSTN networks such as ISDN PRI circuits, where a
fi xed number of circuits or channels transmit voice in 64Kbps streams.
Let’s look at an example of TDM. Figure 2.16 shows the timeslots of a T1 PRI circuit.
F I G U R E 2 .1 6
Time-Division Multiplexing
Physical T1 Circuit with TDM
TimesIot 24
TimesIot 23
TimesIot 22
TimesIot 9
TimesIot 8
TimesIot 7
TimesIot 6
TimesIot 5
TimesIot 4
TimesIot 3
TimesIot 2
TimesIot 1
1 D Channel
23 D
Channels
Each of the 24 T1 channels is actually a multiplexed timeslot on the same set of four
wires. Each logical channel receives a timeslot at a specific interval. Once the voice
segments come out the other side of the circuit, they are reassembled and put onto a single
64Kbps circuit that reaches the phone on the other end.
60
Chapter 2
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Statistical Time-Division Multiplexing
Statistical Time - Division Multiplexing is sometimes called packet-mode multiplexing.
It is considered to be more advanced than standard TDM. While TDM reserves a timeslot
for a channel regardless of any data requiring it, STDM reserves a timeslot on the wire
only when the slot is required for sending or receiving data. Because of the bandwidth
savings, it is actually possible to oversubscribe the circuit to connect more end devices than
there is actual physical bandwidth for. The reasoning behind this is that it’s unlikely that
every phone would be in use at the exact same time, and therefore the circuit can be
better utilized.
Private Phone Switching
Now that you are familiar with some of the most popular types of digital and analog
circuits available from the PSTN, I’m going to move the focus back to the private portion
of a phone network. Private switching allows a business to lower costs by eliminating a
one-to - one ratio between PSTN extensions and telephone handsets. Using an intermediary
switch that is privately managed, you can configure a many-to -few ratio scenario. A private
phone switch also allows the administrator to configure advanced on-network functionality
that enhances the communication experience. In a sense, you’re becoming a mini-PSTN
because you can now offer services such as dual lines, voice mail, intercom, and so on. All
of these services are local to the key system or PBX. Let’s look at the two types of private
switching technologies found in most businesses: the key system and the private branch
exchange (PBX).
The Key System
Small businesses typically deploy this type of internal phone-switching system. A typical
key system has just a few analog or digital PSTN lines that are colocated in a single control
unit. Most commonly, these are simple analog lines with FXO connections into the key
system switch. Phones are then attached to the key system switch, which is also commonly
referred to as the control unit. Each phone is set up identically and has every PSTN line
available for use. This ensures that anyone in the office can answer an incoming call to
any line. It also means that no single person has a unique phone number to call their own.
Cisco calls this a shared-line scenario. Each PSTN extension is shared communally. When
a user wants to make a call, they manually choose one of the unused extensions from which
to place the call.
For users of key systems the vast majority of phone usage is for off-network calls. In
small-business environments, you usually don’t call from extension to extension using
on-network dialing. Instead, you simply walk over to the person you want to talk to. An
alternative method that is popular with key systems is the intercom feature. This feature is
Private Phone Switching
61
far more likely to be used than extension-to - extension dialing. The bottom line is that key
systems are shared-line phone systems where the phones are identically set up and provide a
small number of enhancement features.
Private Branch Exchange
Unlike a key system, where the end user manually selects an extension to use in order to
make a phone call, the user on a private branch exchange (PBX) has a specific extension
(or extensions) assigned to their phone. However, PBX resembles the key system in that all
PSTN lines are colocated to a control switch, and all internal phones communicate with it
to make on- and off-network calls. But unlike the key system, which most commonly uses
analog PSTN circuits, the PBX typically works with digital circuits and interfaces such as
the T1/E1 PRI. This is because PBX systems are usually in larger environments where more
than a handful of outside PSTN lines are needed.
While key system end phones are typically identical in setup and functionality, phones
connected to a PBX are often configured individually, depending on the voice functions
required. An extension configured on the phone may be a DID number accessible directly
from the PSTN, or it may be configured as internal only. In this type of setup, outside users
call a main DID number and are either manually transferred to an internal number by an
operator or transferred through the use of an auto attendant (AA).
PBX systems can also offer advanced services to their internal users. Table 2.5 briefly
describes some of the most popular PBX services in use today.
TA B L E 2 . 5
Common PBX services
Service Name
Function
Extension dialing
Truncated (typically 4–5 digits), used for on-network dialing
Call forwarding
Provides call redirection to a different extension that is either
on or off network
Hunt groups
Provides one extension to a group of phones that ring on a
rotating schedule
Conference bridge
Allows multiple extensions to participate in a single call
Call parking
Places a call on hold and resumes the call from a different phone
Paging
Provides notification of users/groups using phones
After hours support
Allows different dial rules depending on time of day/week
62
Chapter 2
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PSTN Numbering Plans
Public numbering plans are global and/or regional standard numbering formats created
so that long-distance calls can be properly routed throughout the public network. Similar
to networking, where public IP address spaces must not overlap, PSTN numbering plans
help to ensure that each region has its own identification system so all calls can properly
be routed. These numbering plans must be carefully maintained to insure that there is
no overlap. Every phone number on the planet must be unique. Numbering plans help to
group geographic blocks of users together to help assist with optimizing call routing on the
PSTN. For the purpose of the CCNA Voice certification, you should be familiar with both
the International Numbering Plan (ITU E.164) and the North American Numbering
Plan (NANP).
The International Numbering Plan
The International Numbering Plan is commonly known as the International
Telecommunications Union (ITU) E.164 standard. A globally recognized organization, the
ITU is responsible for creating inter-border communications standards. The E.164 standard
defi nes the format of PSTN numbers on a global scale. Table 2.6 details the ITU E.164
numbering system. As you can see, the structure consists of three distinct categories.
TA B L E 2 . 6
ITU E.164 structure
Structure
Format
Description
Country code (CC)
1–3 digits
Defines the country of origin
National destination code (NDC)
0–15 digits
Optional country/region-specific code
Subscriber code (SC)
1–15 digits
Central office significant code
Within a given country code, the national destination and subscriber codes
are primarily governed by the local country or region and can be in any
format. The only caveat is that the ITU E.164 numbering plan stipulates that
the maximum number of digits for an international call must be less than
or equal to 15 and must use the assigned country code at the beginning of
the dial string.
PSTN Numbering Plans
Table 2.7 lists a handful of country codes that the ITU has provisioned.
TA B L E 2 . 7
ITU country code sampling
Country or Region
E.164 Country Code
North America
1
Mexico
52
United Kingdom
44
France
33
Germany
49
India
91
Hong Kong
852
Spain
34
This is just a sample list of the country codes available. You can get the
most recent ITU -T E.164 country code assignments at this URL:
http://www.itu.int/publ/T-SP-E.164D-2009/
The North American Numbering Plan
The North American Numbering Plan (NANP) consists of a standard calling format for
24 countries and territories including the United States, Canada, and the Caribbean. The
numbering structure consists of the three segments described in Table 2.8.
TA B L E 2 . 8
NANP structure
Segment
Number Format
Description
Three-digit area code
[2-9][0-8][0-9]
Code dictated by geographic location
Three-digit office code
[2-9][0-9][0-9]
Code where circuit is terminated at the
central office
Four-digit station code
[0-9][0-9][0-9][0-9]
Locally unique code at the central office
63
64
Chapter 2
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As you can see, neither the area code nor the office code can begin with a 1. Also, the
area code can never have a 9 as the second digit. It is also important to keep in mind that
the central office code’s second and third digits cannot both be 1. This is because the X11
numbers are used for special purposes such as emergency services (911). Table 2.9 lists
several X11 numbers along with other NANP numbers that are reserved for special use.
TA B L E 2 . 9
NANP special numbers
Special Use Number
Description
0
Local operator
00
Long-distance operator
011
International access code
211
Community government information
311
City government information
411
Local/national directory assistance
511
Traffic and road conditions
611
Telephone repair service information
711
Hearing-disabled relay service
811
Underground pipe safety service
911
Emergency services
The rapid growth of additional PSTN numbers in the 1990s was due mainly to the
deregulation of local phone services and the introduction of cellular phones. Because of
this deregulation, multiple carriers received a portion of an area code’s numbers. These
numbers came in blocks of 10,000. Many cities quickly ran out of available numbers
and required additional area codes to provide coverage for the same geographical region.
These additional area codes are called overlay numbers. While overlay area codes fi xed
one problem, they introduced another. In order to call from one area code to another in
an overlay region, you would have to make sure to dial the three-digit area code if the area
Summary
65
codes of the called and calling party were not the same — even if the person you are trying
to call is right next door! Ten- or 11-digit dialing may not seem like a big deal today, but it
caused many problems when area code overlays were fi rst introduced.
Combining the NANP with the International
Numbering Plan
We know that there is an International Numbering Plan that all countries must abide
by. We have also looked at the North American Numbering Plan to see how its components
fit together. As an example, assume that we’re in Spain and need to make an international
phone call to a person In Chicago, Illinois. We have the NANP number of the person we
wish to call in Chicago. It is 312-555-1234. If we simply dial this number when we are in
Spain, we will not reach our intended destination. The most general piece of information
we have is the 312 area code, which tells us the geographic location of our intended party
while we are within the NANP calling area. But because our source phone is in Spain, we
need to tell the Spanish PSTN that we need to make a call outside of their nation. If we
don’t notify the Spanish PSTN of this, it assumes that we want to make a call within their
national boundaries. Since we are attempting to call a number in the United States, we can
use the assigned country code that the ITU designated for the NANP, the number 1. So
now we have all the pieces we need. Because we’re dialing internationally, we need to dial
the following number: 1-312-555-1234. Figure 2.17 breaks down the components of our
combined International and NANP structure.
F I G U R E 2 .17
International and NANP example
ITU
International
Numbering
1
Country
Code
North America Numbering Plan
–
312 – 555 – 1234
Area
Code
CO
Code
Station
Code
Summary
In this chapter you learned about traditional IPT technology. The chapter covered the
three types of PSTN network signaling and when they’re used. In addition I explained the
most common PSTN analog and digital circuits and how multiplexing is used to transport
multiple channels over the same wire.
Next, you learned about the two types of private switching and when you’ll typically
see a key system setup versus a PBX. Finally, I covered both the ITU E.164 and NANP dial
66
Chapter 2
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plans and explained why they are so important in order to create a uniform and scalable
global PSTN system.
Hopefully I ’ve uncovered much of the magic that the PSTN offers. While phone
systems are moving from a circuit-based system to a network-friendly packet system,
it is clear that the current PSTN system that fascinated me as a child is still alive and
functioning today.
Exam Essentials
Understand the three types of phone network signaling. Network signaling is broken
into three distinct categories of address, supervisory, and informational. Each category is
responsible for a part of each phone call made.
Know how to identify parts of an analog sound wave. The key to understanding how
voice travels across phone lines lies in understanding what analog sound waves look like
and how they can be turned into an electrical form for transport.
Know the four most common types of analog voice interfaces. Analog interfaces
come in many types, but there are four interfaces to be familiar with. Understand the
differences between FXS, FXO, DID, and CAMA interfaces and when each one is likely
to be used.
Understand the proper steps and purpose for converting analog signals into a digital
format. Know why we would want to convert analog signals into digital and know the
order of the four steps required to complete the process.
Know the four most common types of digital voice interfaces. Make sure you know each
digital interface type (ISDN BRI, T1 CAS, E1 CAS, and T1/E1 PRI), how they handle
signaling, and how many voice calls can be made at one time. Also be sure to understand
when you would utilize one digital circuit type over another.
Understand the purpose and types of multiplexing. Multiplexing is a method to send
multiple calls across the wire simultaneously. Understand that with TDM, a timeslot is
reserved for each call regardless of any voice data being sent. STDM, on the other hand,
requests a timeslot only if one is needed.
Know the difference between a PBX and a key system. For the most part, there are two
traditional private switches in use today. Understand the differences between a PBX and
a key system, and know what types of office environments would be best suited for one
system over the other.
Understand the International and NANP PSTN dial plans. Know the purpose for having
standards-based dial plans for PSTN networks, and understand the parts that make up the
numbering plan. Also be aware of any requirements that each plan has designated.
Written Lab 2.1
67
Written Lab 2.1
Write the answers to the following questions:
1.
What are the three types of voice network signaling?
2.
What type of analog interface is typically used to connect to E911 services?
3.
Name the type of PRI circuit typically used in Europe.
4.
What type of multiplexing is considered to be circuit mode?
5.
In what type of private phone system would the majority of calls made be
off-network calls?
6.
In what PBX service scenario will phone calls to a single extension rotate from phone
to phone?
7.
List the three categories of the ITU E.164 International Dial Plan.
8.
What are the three NANP Dial Plan categories?
9.
What term is used to describe the NANP designation when more than one area code is
required for a single geographical region?
10. What NANP special code will connect a caller to local/national directory services?
(The answers to Written Lab 2.1 can be found following the answers to the review
questions for this chapter.)
Chapter 2
68
Traditional Telephony
Review Questions
1.
A dial tone is considered to be what type of network signaling?
A. Address signaling
2.
B.
Informational signaling
C.
Notification signaling
D.
Supervisory signaling
Which address signaling methods can be used to interpret phone numbers to place a call?
Choose all that apply.
A. DTMF
3.
B.
Informational signaling
C.
Pulse dialing
D.
Supervisory signaling
Which multiplexing type is also referred to as packet-mode multiplexing?
A. FIFO
4.
B.
TDM
C.
STDM
D.
FHSS
E.
PTDM
What is the distance measured between each sound wave called?
A. Analog signal
5.
B.
Wavelength
C.
Peak-to -peak amplitude
D.
Peak amplitude
What is the variation in a sound wave amplitude over a one-wavelength period called?
A. Peak-to -peak amplitude
6.
B.
Peak amplitude
C.
Feedback
D.
POTS
E.
Hook flash
What is the name for an automated voice system that assists callers to their desired
extension destination?
A. DID
B.
FXO
C.
AA
D.
CO
E.
FXS
Review Questions
7.
69
Which of the following is an analog interface that is typically found in small businesses
with a few PSTN lines going into a PBX?
A. E &M
8.
B.
T1 PRI
C.
FXO
D.
FXS
E.
AA
Which of the following is an analog connection in which digits are stripped off at the PSTN
switch prior to being sent to a private PBX?
A. CAMA
9.
B.
DID
C.
Amplitude
D.
FXS
Which analog interface routes calls based on the calling number?
A. FXS
B.
FXO
C.
DID
D.
CAMA
E.
Trunk
10. What is the first step in the analog-to - digital conversion?
A. Encode
B.
Quantize
C.
Compress
D.
Sample
11. Name the technique used to gather samples of an analog sound wave.
A. POTS
B.
PAM
C.
Quantize
D.
Encode
E.
Compress
12. Which digital circuit consists of two B channels and one D channel?
A. T1 PRI
B.
E1 PRI
C.
E&M
D.
FXO
E.
ISDN BRI
Chapter 2
70
Traditional Telephony
13. Which step in the analog-to - digital conversion process is optional?
A. Compress
B.
Encode
C.
Quantize
D.
Codec
E.
Sample
14. How many voice calls can an E1 PRI handle at one time?
A. 23
B.
24
C.
2
D.
30
E.
32
15. Which signaling method is used for T1 PRI circuits?
A. Q.931
B.
HDB3
C.
ESF
D.
SF
E.
BRI
16. What type of T1 framing method is considered to be an older method?
A. SF
B.
ESF
C.
Q.931
D.
HDB3
17. Which of the following digital circuits uses robbed-bit signaling?
A. T1 PRI
B.
ISDN BRI
C.
T1 CAS
D.
E1 PRI
E.
ISDN CAS
18. Which type of multiplexing is considered to be more efficient?
A. TDM
B.
DID
C.
T1 CAS
D.
STDM
E.
CAMA
Review Questions
71
19. What type of private switch typically has unique extension numbers configured on
each phone?
A. PBX
B.
CO
C.
Key system
D.
T1 CAS
20. What is the maximum number of digits that an international number can have and abide by
the ITU E.164 numbering plan?
A. 3
B.
10
C.
11
D.
15
E.
18
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Traditional Telephony
Answers to Review Questions
1.
B . Informational signaling provides audible feedback to the called and/or calling party.
2.
A, C. Address signaling can use either DTMF or pulse dialing to transmit phone numbers
to the telecommunications switch.
3.
C . Statistical Time-Division Multiplexing is also referred to as packet mode.
4.
B . A wavelength of a sound wave is the distance between the two crests of each wave.
5.
A . The peak-to -peak amplitude measures the entire variation of amplitude (highest and
lowest points) over a one-wavelength period.
6.
C. An auto attendant is similar to an operator but there is no human interaction.
7.
C . FXO interfaces are individual lines that connect to a PBX.
8.
B . With DID connections, the phone company strips off all address information digits
except for the extension, which then are passed on to the private PBX.
9.
D. CAMA interfaces are typically used for E911. These calls are routed to the PSAP based
on the calling party’s phone number.
10. D. The fi rst step is to sample the analog signal.
11. B. Pulse Amplitude Modulation takes a slice of the analog sound wave at a constant
interval over a period of time.
12. E . ISDN BRI circuits are composed of two bearer channels and one data channel.
13. A. Depending on the codec being used, compression may or may not occur.
14. D. The E1 PRI has 32 total channels; 30 are dedicated to voice calls while the other 2 are
for framing and signaling.
15. A. Q.931 is used on both PRI and BRI digital circuits.
16. A . Super Frame is the older version for T1 CAS framing. Most T1 CAS circuits now use
Extended Super Frame.
17. C . The T1 CAS uses RBS for in-band signaling so it can utilize all 24 channels for voice
transport.
18. D. Statistical Time-Division Multiplexing is more efficient because it reserves a timeslot
only when it is required.
19. A. A PBX is typically configured so that each phone has at least one unique extension
assigned.
20. D. E.164 states that no phone number may exceed 15 digits including the country code.
Answers to Written Lab 2.1
Answers to Written Lab 2.1
1.
Address signaling, informational signaling, supervisory signaling
2.
CAMA
3.
E1
4.
Time-Division Multiplexing
5.
Key system
6.
Hunt group
7.
Country code, national destination code, subscriber code
8.
Area code, office code, station code
9.
Overlay
10. 411
73
Chapter
3
Voice over IP (VoIP)
THE FOLLOWING CCNA VOICE
EXAM OBJECTIVES ARE COVERED
IN THIS CHAPTER:
Describe the components of the Cisco Unified
Communications Architecture.
Describe how the Unified Communications components
work together to create the Cisco Unified Communications
Architecture.
Describe the function of Contact Center in a UC environment.
Describe VoIP components and technologies.
Describe RTP and RTCP.
Describe the function of and differences between codecs.
Describe H.323, MGCP, SIP, and SCCP signaling protocols.
Describe gateways, and dial peers to connect to the PSTN
and service provider networks.
Describe the function and application of voice gateways.
Describe the function and operation of call legs.
Describe voice dial peers.
Legacy voice and data networks consist of separate phone and
data systems that occupy completely independent physical
cabling and hardware and often have separate support staff.
As data networks matured throughout the ‘90s and became more stable and efficient,
it didn’t take a genius to figure out that one could run both voice and data on the same
infrastructure.
This chapter will focus on voice over IP (VoIP) technologies that allow phone calls to
be reliably made over a packet-switched network. We’re fi rst going to discuss the four
layers in the Unified Communications VoIP model. Next, we’ll reexamine voice gateways
in more detail to see exactly what services these devices can provide. We’ll then look at
the underlying voice transport protocols that provide a method for moving calls from
one phone to another on the IP network. Following that, we’ll look at different types of
signaling protocols that assist with the setup and teardown of phone calls. Finally, the
chapter will close with a look at various voice codecs, discussing what they do and when
they should be used.
Understanding the Unified
Communications Model
Cisco always has a knack for breaking up complex networking structures into simple,
easy-to -understand hierarchical models. The Cisco Unified Communications Model is
no exception. This model consists of four layers and their core components that build upon
each other to provide a complete VoIP solution. Figure 3.1 displays the four layers and the
core components within each layer.
Let’s examine each of the Unified Communications Model layers to see how they build
upon one another.
Understanding the Unified Communications Model
F I G U R E 3 .1
77
The Unified Communications Model
Endpoints
• IP Phones
• IP Communicator
Applications
• Unity Messaging
• Emergency Responder
• Unified Customer Contact
Solution
Call Control
• Unified
Communications
Manager
Infrastructure
•
•
•
•
•
Routing
Switching
QoS
Management
Security
The Infrastructure Layer
The Infrastructure layer of the Unified Communications Model is where you will fi nd your
routers, switches, and voice gateways. The Infrastructure layer is responsible for moving IP
packets from the source to the destination.
Because the Unified Communications Model is a converged network, multiple types of
traffic are sent over the same infrastructure. Data, voice, and video traffic are all running
over the same cabling and hardware in this layer. Because of the mixed traffic, there must
be a way to distinguish voice and video from data and prioritize the types. Voice and
video traffic are much more sensitive to delay on the network and ultimately must take
priority over data. The Infrastructure layer typically incorporates quality of service (QoS)
mechanisms to intelligently identify time-sensitive traffic and give it priority when there is
congestion on the network.
Security also plays a key role in the Infrastructure layer. All access- control and IP
restrictions are implemented at this layer. Typically, access control lists (ACLs) are created
and implemented to limit the type of TCP/UDP traffic and source IP addresses that can
access the Call Control layer, which is where the CUCM resides.
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Voice over IP (VoIP)
The Call Control Layer
Sitting on top of the Infrastructure layer is the Call Control layer. This is the heart of the
Unified Communications system and is where the Call Manager is found. The CUCM is
responsible for the following functions:
Call processing
Call signaling
Endpoint control
Dial plan control
Media resource management
User management
As you can see, the upper two layers of the Unified Communications Model fully rely on
the Call Control layer to provide the underlying foundation from which the Application and
Endpoint layers build. The majority of a UC phone system’s intelligence is configured and
maintained at this layer of the model.
The Applications Layer
The Applications layer builds on the Call Control layer to provide value-added functionality
that makes a Unified Communications system feature rich. The Cisco Unity voice mail
application is part of this layer. Not only does it provide full voice mail capabilities, but it can
also integrate with email systems to provide integrated or unified messaging features.
The Cisco Emergency Responder is another application independent from the CUCM.
This application is responsible for providing accurate information to emergency services in
the event that an end user dials 911. While the Emergency Responder relies heavily on the
Call Control layer, technically it is an independent application and sits one layer above
the Call Control layer in the Unified Communications Model.
A third popular Unified Communications application found in large call centers is the
Unified Customer Contact Center. Again, this separate application works in conjunction with
the CUCM, found in the Call Control layer. It adds additional functionality to the call center
in the form of collaboration and customer resource management tools to provide a more
personal service. Specifically, the Contact Center application performs the following functions:
Separates and delivers different customers to the proper call-handling representative
based on intelligent routing decisions
Monitors customer representative resource availability and idle times for proper
staffing needs
Utilizes detailed customer profiles using both dynamically and manually added
customer information
Integrates with Cisco Presence applications such as voice, video, email, instant
messaging, and web collaboration
A Closer Look at Voice Gateways
79
The Endpoints Layer
The Endpoints layer of the Unified Communications Model is probably the easiest to
understand. This is where all voice/video communications begin and end for the user. In
this layer you will fi nd devices such as Cisco IP phones, IP Communicators, soft phones,
Cisco Video Advantage devices, and Cisco ATA termination points. Essentially, any user
device that utilizes the following endpoint signaling protocols is considered to be an
endpoint in the Cisco Unified Communications Model:
Skinny Call Control Protocol (SCCP)
Session Initiation Protocol (SIP)
The users interact with these endpoints, and they require the support of the three
layers below the Endpoints layer. Without the Applications layer, there would be no
voice mail functions. Without the Call Control layer, there would be no intelligence
to place a phone call. And without the Infrastructure layer, there would be no way to
transport the traffic!
A Closer Look at Voice Gateways
Voice gateways are a vital part of VoIP communications and are often the most
misunderstood. The purpose of voice gateways is to interconnect a VoIP packet-based
network with a legacy phone network. A conversion process must take place for
the two different systems to communicate with each other properly. Voice gateways
can serve two primary functions on your network. They can be used to connect a
CUCM to the PSTN or to connect a CUCM with a legacy PBX. Both of these
services require a hardware component on the voice gateway, called a digital signal
processor (DSP). Let ’s fi rst look at how DSP resources can be used to provide
connectivity to the PSTN and legacy gateways. We’ll look at both setups to see
exactly how these two situations are used and the technology behind voice gateways.
Then we’ll move on to discuss voice gateway dial peers and call legs. Last, we’ll
cover the signaling protocols that facilitate the connection between the CUCM and
the voice gateway.
Using DSP Resources on Voice Gateways
to Connect a CUCM to the PSTN
The most common use for a voice gateway is to connect a CUCM to the PSTN. The
voice gateway handles termination of any analog or digital trunks that you are leasing
from the phone company. The magic of the voice gateway is how voice signals based on
one voice technology are converted to a different technology. The voice gateway uses
special hardware chips called digital signal processors (DSPs) to accomplish this goal.
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Chapter 3
Voice over IP (VoIP)
DSPs perform several functions. Basically, they ’re specialized processor hardware that
offloads voice processing services from the main router processor. After all, a router’s
main job is to route IP packets. We’re asking the router, as a voice gateway, to
provide additional services that go above and beyond its original intention. DSPs
are installed to help support the additional services. Let ’s look at the two primary
voice gateway functions: analog-to - digital conversion and digital transcoding.
Then we’ll look at echo cancellation and DTMF -relay, which are secondary voice
gateway services.
Analog-to-Digital Conversion
Because DSPs are responsible for analog-to -digital conversion, they are required on voice
gateways. Depending on the Cisco hardware used for a voice gateway, DSP modules can
be found either attached to a network module (NM), WIC, or VWIC card or plugged
directly into the motherboard of the router. It is important to note that a DSP resource
is required for every legacy PSTN- circuit-to -IP-packet conversion that takes place on the
voice gateway.
Your voice gateway router is also likely to include one or many different types of digital
or analog voice interfaces, such as FXO, ISDN BRI, or T1 PRI circuit. The voice gateway
sits between the legacy PSTN network and the VoIP network. The voice signals between
the two networks do not speak the same language in terms of signaling, coding, and
control. DSP resources are used to translate this information from one format to the other
and thereby bridge the two networks.
Digital Transcoding
Even when two devices speak natively on an IP network, they still may need DSP resources
to communicate properly. When two VoIP devices wish to talk to one another over the
IP network, both need to be able to understand the codec that is being used. If one VoIP
end unit uses a codec that the other end unit does not understand, a DSP can be used to
transcode the stream into a codec that is supported. If you require hardware transcoding in
your voice network, you can connect your CUCM to a DSP farm, which is normally found
on a voice gateway. Essentially, you configure SCCP signaling between the CUCM and the
voice gateway, which contains DSP resources. When the CUCM receives a voice stream
that requires transcoding, the stream is directed to the voice gateway using SCCP. While
it is important to understand the purpose of DSP farms, the actual configuration of a DSP
farm is outside the scope of this book.
The types of codecs you need to transcode are a factor in how many DSP resources
you need on your voice gateway. Usually, the more compressed the audio signal is,
the more DSP resources are required for transcoding into another codec. Voice
codecs are classifi ed by complexity; they are considered to be either medium or
high complexity.
A Closer Look at Voice Gateways
81
Calculating DSP Requirements Online
Simon is a network consultant working on his first voice implementation for a client
wanting to replace their current legacy PBX with a Cisco VoIP solution. During the
course of the initial conversation meant to flesh out system requirements, Simon has
determined that the site is best suited to utilize a 2800 series router with a T1 module, a
four-port FXO card, and an eight-port FXS card. The four-port FXO card will be used for
fax machine pass-through. Simon ended the meeting and compiled a bill of materials
for a senior consultant to review. Upon looking at the information gathered, Max (the
senior consultant) asked Simon if he knew how many DSP chips were required for this
project. Simon had no idea what the senior consultant was talking about. Noticing the
confused look on Simon’s face, Max opened a web browser and navigated to
http://www.cisco.com/cgi-bin/Support/DSP/dsp-calc.pl
Max informed Simon that Cisco provides a DSP calculator to engineers to help them
gauge which DSP chip will need to be ordered and in what quantity. The calculator asks
for variables including router module, IOS version, and installed voice components.
Simon entered all the information he gathered from the consulting session. The
application then calculated the approximate number of DSP resources required and the
DSP part numbers for easy ordering. Now that Simon has learned this little trick, he’ll
make sure to include this information in the bill of materials for future customers.
Let’s use a couple of examples to make sure you understand when DSP resources may
or may not be needed on your network. I’ll discuss specific codec types later in the chapter,
but to understand when they are required, it’s sufficient to know that they are inoperable
unless transcoding is performed. It is also important to note that all Cisco phones in use
today support the G.729 and G.711 codecs.
Example 1: Cisco Phones Running G.711 and G.729
Suppose we have two Cisco phones, both 7960G desk phones. They are configured and
running on a CUCM Express system. By default, the codec used for all VoIP on the CUCM
Express is G.711. Let’s say that we change the codec of one of the phones to G.729. Now
we have a situation where one phone is using the G.711 codec and the other is using
G.729. If one phone calls the other, we will need DSP resources to transcode one codec
into another, right? Well, not necessarily! Because the Cisco IP phones understand both
codecs, when the call setup occurs, the phones will actually negotiate which codec is used.
If they can both talk natively using one codec, that’s what they’ll do. If they do not have
a common codec, then DSP resources will be needed. In our example, the phones will
82
Chapter 3
Voice over IP (VoIP)
negotiate and end up using the G.729 codec. Why did they choose G.729? If there is a
choice between two or more codecs that both endpoints natively speak, they will choose
the one that offers the most compression and therefore uses the least bandwidth.
Example 2: One Cisco Phone Running G.729 and a
Third-Party Phone Running G.726
In this situation we have a Cisco phone that can understand G.729 and G.711. The thirdparty phone can use only the G.726 codec. In this situation, we’ll need to tap into the DSP
resources to perform transcoding between the two endpoints.
Echo Cancellation
Echo is the reflection of sound that arrives to the listener a period of time after the direct
sound is heard. A certain amount of echo is experienced on most voice calls and up to a
certain point is tolerated. When analog signals are converted to digital signals and then
compressed using codecs, echo is often amplified to the point where it severely degrades
the quality of the call. DSP resources are used to assist in the elimination of echo when
converting from one voice signal into another. Echo cancellation is performed by default.
DTMF-Relay Services
VoIP devices do not support traditional DTMF digits by default. It may be necessary to
allow your IP endpoints to use DTMF to communicate with non-VoIP-based services.
DTMF -relay can be used to facilitate this conversion. There are several methods for
configuring DTMF -relay. All of them require the use of DSPs to properly transport
the DTMF tone uncompressed over an IP network. Just like DSP farms, the
configuration of DTMF -relay is outside the scope of this book, but it is important
to know the service exists.
Media Termination Points
When using H.323 or SIP endpoints or gateways, you can use DSP resources to assist
with the process of functions such as call holds, parks, transfers, and conferences. These
voice services are all very commonly used in most CUCM implementations. These
supplementary services are referred to as media termination points. The DSPs are used
to either help “park” calls while on hold or parking or provide audio multiplexing into a
single audible stream for conference calls.
Using Voice Gateways to Connect a CUCM to a PBX
A second design methodology for using voice gateways is used when companies are making
the migration to a Cisco Unified Communications solution. The process of migrating
away from a traditional PBX typically involves interconnecting the CUCM and PBX for a
period of time. Some users may have new Cisco phones, while others might still have the
older phones that connect to the legacy PBX. End users on one system need to be able to
A Closer Look at Voice Gateways
83
communicate easily with users on the other. In this situation, a voice gateway can be used
to provide a common channel-signaling method. A digital trunk interface on the voice
gateway connects to a digital trunk interface on the legacy PBX. Figure 3.2 shows the
physical setup of the design.
FIGURE 3.2
Connecting CUCM to a PBX
Legacy PBX
Network
IP Network
IP Phone
Legacy Phone
PRI Link
V
IP Phone
CUCM
Express
Voice
Gateway
IP Phone
Legacy Phone
Legacy PBX
Legacy Phone
As you can see, the voice gateway represents the jumping- off point from the pure IP
switched network and the legacy PBX network. In this particular setup, the two networks
are interconnected by a PRI trunk. Depending on the type of PBX used, a different digital
trunk might be used. The key point is that the PBX and voice gateway must use identical
signaling on each side.
Voice Gateway Dial Peers
In order to route voice traffic properly from one point to another point using H.323 or SIP
voice gateways, we need to configure dial peers. A dial peer is a device that can make or
receive a call in a voice network. With VoIP networks, there are two types of dial peers:
POTS dial peers
VoIP dial peers
Let’s review both of these to see how they function in IP and PSTN networks.
POTS Dial Peers
POTS dial peers are considered to be traditional telephony devices such as analog phones,
cellular phones, and fax machines. From a voice gateway perspective, the POTS dial peer is
a simple dial-string-to -port mapping. Figure 3.3 illustrates a POTS dial-peer scenario.
84
Chapter 3
FIGURE 3.3
Voice over IP (VoIP)
POTS dial peers
PSTN
IP Network
V
Voice Gateway
Analog Phone
PSTN
V
Voice Gateway
Analog Phone
POTS Dial Peer
POTS Dial Peer
As you can see in this example, a single POTS dial peer runs from the analog phone
located on the PSTN to our local voice gateway.
VoIP Dial Peers
These dial peers include any VoIP- capable endpoint, router, and gateway within the IP
network. Just like POTS dial peers, VoIP dial peers use a dial string for mapping purposes.
The difference is that instead of mapping the dial string to a physical interface, the VoIP
dial peer maps the dial string to a remote IP network device. Figure 3.4 helps to explain
VoIP dial peers.
FIGURE 3.4
VoIP dial peers
PSTN
Analog Phone
IP Network
V
Voice Gateway
VoIP Dial Peer
PSTN
V
Voice Gateway
Analog Phone
VoIP Dial Peer
In this example, there is a VoIP dial peer for each side of the IP network. They are
needed because each voice gateway requires a dial-peer configuration in order to identify
the call source and destination endpoints.
Dial Peers and Call Legs
Call legs are logical connections between dial-peer origination and termination points on
IP networks. They associate with dial peers on a one -to - one basis. A call leg is considered
to be either a POTS or a VoIP leg, depending on which network the call leg represents. For
example, Figure 3.5 shows a VoIP and POTS call communication scenario where phone
A is making a call to phone B.
A Closer Look at Voice Gateways
FIGURE 3.5
85
Call legs
PSTN
IP Network
V
Voice Gateway
Analog Phone
A
Inbound POTS
Call Leg
Outbound VoIP
Call Leg
PSTN
V
Voice Gateway
Inbound VolP
Call Leg
Analog Phone
B
Outbound POTS
Call Leg
This example shows that every voice IP terminating device has a call leg associated
with it. Any voice gateway will have two associated call legs/dial peers for each logical
connection. By contrast, a POTS dial peer has only a single call leg/dial peer associated
with it. This is because once the call is placed out on the PSTN, we don’t really have any
control over how it is switched. Therefore, we have control over only one dial peer and,
ultimately, one call leg.
Comparing Voice Gateway Communication Protocols
As you have learned, voice gateways provide a bridge between the Cisco Unified
Communications Manager and either the PSTN or a legacy PBX. There are two types of
voice gateways in the CUCM Express system. The voice gateway might reside on the same
router as your CUCM Express software; this is known as an integrated voice gateway. If
the PSTN trunks reside on the same hardware as the CUCM Express software, calls can be
directly switched from the PSTN lines to the IP phones with little configuration. Figure 3.6
shows the integrated gateway setup, which requires no voice gateway protocol:
FIGURE 3.6
Integrated voice gateway
PSTN
IP Phone
CUCM Express
Analog Phone
The separated voice gateway sits on a different router than the CUCM Express
software. If the voice gateway is separate, you need a signaling protocol to transport
the calls across the IP network between the CUCM and the voice gateway. Figure 3.7
depicts the separated voice gateway architecture and the link that requires a voice gateway
signaling protocol.
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F I GU R E 3 .7
Voice over IP (VoIP)
Separated voice gateway
Voice Gateway
Signaling Protocol
PSTN
V
IP Phone
CUCM Express
Voice Gateway
Analog Phone
The Unified Communications system supports four methods of gateway communication.
Let’s briefly look at these four gateway protocols.
The H.323 Gateway Protocol
H.323 is an ITU standards-based peer-to -peer protocol. It is a bundle of multiple protocols
signaling and controlling voice and video data. Table 3.1 lists the core protocols used
within the H.323 suite and their functionality.
TA B L E 3 .1
H.323 core protocols
H.323 Subprotocol
Function
RAS
Messaging protocol used by the CUCM for gatekeeper discovery
and registration. Also used to pass database lookup and CAC
information.
H.245
Performs call control functions.
Voice Codecs (G.711,
G.729, etc.)
Performs encoding/decoding of voice streams.
H.225
Performs call setup and codec negotiation over TCP 1720.
As you can see from the table, H.323 uses the various protocols within the suite to
signal and control transport over IP. Transport and packetization are then performed
using the Real-time Transport Protocol (RTP). RTP is an IETF standard for transport of
time-sensitive packets such as voice. RTP will be described in detail later in this chapter.
H.323 has been around since the mid-1990s and is considered to be the most mature of all
voice gateway signaling protocols. The protocol suite is very versatile and can be utilized
in multiple ways, including signaling for endpoints, gatekeepers, and voice gateways. For
CUCM Express environments, which our book focuses on, you’ll typically see H.323 used
to communicate between CUCM Express and a voice gateway.
A Closer Look at Voice Gateways
87
The H.323 architecture uses a peer-to -peer model. This means that the voice gateway
is independent of the CUCM. Therefore, a more complex configuration is necessary on the
voice gateway. This is because the gateway is responsible for maintaining the dial plans and
route patterns.
You may run across a separate device in large legacy H.323 environments called an
H.323 gatekeeper. This device is essentially a database that contains H.323 mappings
(telephone numbers) to IP addresses. When a phone number is dialed, the gatekeeper
is queried to determine what location (IP address) the remote phone is located in. For
example, if a user dials 555-1212, the gatekeeper knows that this phone can be reached
at the IP address of 10.10.4.220. Our Cisco voice gateway needs a way to communicate
with the H.323 gatekeeper in order to facilitate this gatekeeper “lookup” functionality. It
uses the RAS protocol to discover and communicate with the gatekeeper. Figure 3.8 shows
the RAS communication between a voice gateway and an H.323 gatekeeper:
FIGURE 3.8
RAS between VG and H.323 gatekeeper
H.323
Gatekeeper
RAS
Protocol
LAN
PSTN
V
CUCM Express
Voice Gateway
The gatekeeper is also responsible for any call admission control (CAC) that the
administrator has configured. CAC is responsible for determining whether the caller has
the right to ring the requested number. In addition, it can also determine whether there is
enough bandwidth at the time of the call to make a successful connection.
The SIP Gateway Protocol
Session Initiation Protocol (SIP) is an IETF standard gateway communication method
that uses a peer-to -peer architecture. It can run over TCP or UDP. By default, SIP uses
UDP port 5060 when configured on a CUCM. Because it is a peer-to -peer system, the
intelligence, such as dial plans and route patterns, resides on the voice gateway sides. It
also means that the voice gateway configuration is more complex than with the clientserver gateway protocols. SIP is considered the successor to H.323 and is in increasingly
widespread use in new environments. But unlike H.323, SIP is merely responsible for call
setup and control. Other protocols outside of SIP such as UDP/RTP/RTCP are ultimately
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responsible for the transport of the voice stream. We’ll discuss these voice packet transport
protocols in the next section.
The MGCP Gateway Protocol
The Media Gateway Control Protocol (MGCP) is a client-server architecture, which
means all of the intelligence, such as dial peers and route plans, resides on the CUCM.
This centralized call control model means the gateway simply facilitates the voice routing
functions that the CUCM determines. It is an IETF standard protocol and is the newest
of the standards-based voice gateway signaling protocols. It is also one of the simplest
to configure on the voice gateway. Because of its client-server nature, the bulk of the
configuration is performed on the CUCM and very little is required on the router itself.
MGCP can run over either TCP or UDP. Signaling information between the CUCM and
gateway runs over port UDP/2427 and TCP/2428 by default.
The SCCP Gateway Protocol
The Skinny Client Control Protocol (SCCP) is a Cisco proprietary gateway protocol. It
uses a lightweight client-server architecture that allows the CUCM to maintain the dial
plans and route patterns centrally and uses the gateway as a method of transport, similar
to MGCP. Because SCCP is proprietary, your voice gateway must be a Cisco device. You
may want to look into using SCCP as your voice gateway protocol, because it provides
additional features that are not available with the standards-based protocols. The SCCP
protocol runs over TCP port 2000.
We’ll spend part of Chapter 7 configuring gateways and trunk communication between
the CUCM Express and a separate voice gateway.
An Overview of Voice and Video
Transport Protocols
Data payloads such as voice and video that our Cisco Unified Communications Systems
facilitate require specific protocols to be handled properly. In regard to the CCNA Voice
certification, you need to be aware of the protocols that the CUCM solutions use to
transport voice endpoint packets from one endpoint to another. This section details the
three protocols that are used in Cisco VoIP environments: Real-time Transport Protocol,
Compressed Real-time Transport Protocol, and Real-time Control Transport Protocol.
The Real-Time Transport Protocol
The foundation of all voice and video communication over an IP network begins with the
Real-time Transport Protocol (RTP). RTP is an IETF RFC 1889 and 3050 standard for
the delivery of unicast and multicast voice/video streams. RTP almost always uses UDP for
transport. UDP, unlike TCP, is an unreliable best- effort service. That may sound like a bad
An Overview of Voice and Video Transport Protocols
89
thing, but in reality it is the most efficient method for transport of streaming data. A besteffort service such as UDP does not attempt to retransmit or reorder packets as TCP does.
If you think about it, once a voice packet is lost in transit, there is no reason to attempt to
retransmit it, because once the packet reaches its destination, the sound wave contained
in the packet would not make sense to the end user if it is delivered out of order. UDP also
does not provide any flow control or error correction. This cuts down on the overhead of
each datagram and therefore is much more efficient.
The RTP header does offer some important information about its payload in each
encapsulated packet. Figure 3.9 lays out the RTP header information.
FIGURE 3.9
Bits:
RTP header details
2
1 1
4
1
7
16
V
P X
CC
M
PT
Sequence Number
Timestamp
Synchronization Source Identifier (SSRC)
Contributing Source (CSRC) - Optional
Let’s briefly look at each segment of the RTP header:
Version (V): 2 bits
This field specifies the version of RTP that is being used.
Padding (P): 1 bit If this bit is set, it indicates that this RTP packet has one or more
octets at the end that are not part of the payload. Padding is often used for encrypting
RTP payloads.
Extension (X): 1 bit If this bit is set, it indicates that the fi xed header is followed by a
single header extension.
Marker (M): 1 bit
the stream.
A field used to signify significant events to the application using
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Payload Type (PT): 7 bits This field identifies the type of RTP data that is inside the
payload (either voice or video data).
Sequence Number: 16 bits This field increments by one for each RTP packet in a
particular stream. This may be used by upper-layer protocols to detect any packet loss or
sequencing issues.
Timestamp: 32 bits The timestamp marks each packet with an encapsulation time. It is
most often used for jitter and synchronization calculations.
Synchronization Source Identifier (SSRC): 32 bits This field marks each RTP stream
differently so multiple RTP streams from the same source are kept separate.
Contributing Source (CSRC): 32 bits An optional field that enumerates contributing
sources to streams that come from multiple sources.
According to the RFC, RTP can utilize any UDP port as long as it is even numbered. It
is up to the application to determine which port is used, although voice traffic is typically
in the range of 16384 to 32767. The UDP port it uses for a practical RTP stream is chosen
at random. That same port is used the entire duration of the stream. Once one of the IP
phones hangs up, that RTP session is terminated and the port is released. RTCP, which
we’ll talk about in a moment, uses the next-incremented odd-numbered port, always
creating a pair of ports representing a call leg.
Compressed RTP
You may have noticed that the entire RTP header is a hefty 12 bytes in length. Combined
with a 20 -byte IP header and a UDP header of 8 bytes, there is a whopping 40 bytes in
header information alone. Because RTP data is very sensitive to any kind of delay, some
way to compress this header information was needed. Compressed RTP (cRTP) was
developed as a solution.
cRTP takes that massive 40 -byte header compilation and cuts it down to anywhere
between 2 and 5 bytes. Figure 3.10 shows the cRTP compression byte reduction rate.
F I G U R E 3 .1 0
RTP to cRTP
40 Bytes Uncompressed
UDP
Header
IP Header
cRTP:
2-5 Bytes
IP
U R
D T
P P
RTP Header
An Overview of Voice and Video Transport Protocols
91
Because much of the information contained in the IP/UDP and RTP headers is static,
cRTP essentially removes this information once it is known on both ends of the wire. By
not sending this static data, it conserves precious bandwidth. In this sense, cRTP caches
static information so it does not have to resend it across the same link. cRTP is most
effective on WAN links that are T1 speed and below. Anything above this bandwidth
renders the compression essentially useless.
If you implement cRTP in a production network, make sure you
closely monitor the CPU utilization. Unless you have specialized
compression hardware installed on your router, compression is
performed in software and utilizes the main CPU. To see the utilization
on a Cisco router, run a show processes cpu command while in
privileged exec mode.
Real-Time Transport Control Protocol
RTP has a partner when it comes to the transport of real-time data. Real-time Transport
Control Protocol (RTCP) works directly with RTP to provide out- of-band monitoring
for the streaming of the RTP- encapsulated data. RTCP packets are sent to participants
of a particular RTP stream. The main function of RTCP is to provide feedback about the
quality of the RTP transmissions. The real-time application can use this information to
adapt encoding settings if the protocol detects congestion. That means if congestion is
discovered on the remote end of the stream, the receiver can inform the sender to use a
lower-quality codec and therefore help with any bottlenecks. Following are some of the
most common RTP data that RTCP tracks:
Total packet count for the stream
Packet loss of the stream
Packet delay of the stream
Amount of jitter on the stream
As mentioned, RTP uses even-numbered UDP ports for transport. RTCP uses the nexthighest odd-numbered port for its transmission. For example, if RTP is running on UDP
port 22864, then the corresponding RTCP packets run on UDP port 22865.
Keep in mind that an RTP/RTCP stream for voice traffic is one -way only.
Thus for a single voice call, there are two RTP/RTCP streams. One instance
is for transmission of voice to the called party; the other is for voice traffic
being received.
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Comparing VoIP Endpoint
Signaling Protocols
You know that VoIP uses IP for routing decisions, UDP for packet delivery, and RTP/RTCP
for real-time transport. You also need to understand how the CUCM solution handles
the signaling responsibilities for voice endpoints. VoIP endpoint signaling protocols are
responsible for locating endpoints, negotiation of various functions, and the setup and
teardown of voice calls. You must be familiar with two endpoint signaling protocols: SCCP
and SIP. Each protocol differs in architecture, call control, and other services. Let’s look at
each of them.
SCCP
The Skinny Call Control Protocol (SCCP) is a Cisco proprietary voice signaling protocol
based on a client-server architecture. The clients can be any of the Cisco endpoint
phones such as the Cisco 7971 or Cisco IP Communicator softphone. The server in the
architecture is our Cisco Unified Communications Manager. All SCCP clients must
communicate with the CUCM in order to place a call. In this regard, the phone is
essentially a “dumb” device that fully relies on the CUCM to give it intelligence for call
setup and teardown. So while the CUCM handles the call setup control, the phone is
responsible only for the processing of RTP/RTCP packets. SCCP messages are transported
over TCP port 2000. Because SCCP uses TCP for transport, messages can utilize TCP
functionality such as error correction and guaranteed delivery of packets.
When an SCCP- enabled phone registers with it, the CUCM requests certain information
from the phone: IP address, station ID, device types, and the codecs the phone can
understand. The CUCM stores this information so it knows how to best handle the setup
of calls to a particular endpoint. The CUCM uses SCCP messages for keepalives to the
client phones. A keepalive is a message sent by one device to another at specific intervals
to verify that communication between the two is functioning. If the CUCM does not
receive a certain number of keepalive responses over a period of time, the stored endpoint
information is cleared from the CUCM memory until it reconnects, at which time the
phone will have to go through the information-gathering process again.
Once the phone capabilities are registered on the CUCM, SCCP is used again to send
out all the necessary information that the phone requires, including its phone number,
button templates, time/date synchronization, and any other configurable options and
displays.
Messages are constantly sent between the client phone and the CUCM for everything
that a user does on a phone. For example, when a user picks up the handset, an off-hook
notification message is sent from the phone via SCCP to the CUCM. The CUCM then
sends everything the phone should do in response to going off-hook. These include the
message the LCD is to display, the softkeys that are to be displayed, and, of course, a
Comparing VoIP Endpoint Signaling Protocols
93
dial-tone signal. So clearly, SCCP is very much a client-server technology, because control
of the phone is strictly maintained by the Communications Manager. It is important to
keep in mind that the client-server model between the endpoint and CUCM is only for
signaling; the actual voice packets encapsulated in RTP and the RTCP control data are
transported directly from one endpoint to the other. Figure 3.11 shows the traffic flow for
SCCP and RTP/RTCP:
F I G U R E 3 .11
SCCP and RTP/RTCP flow
SCCP Signaling
SCCP Signaling
RTP/RTCP
SIP
The Session Initiation Protocol (SIP) differs from SCCP in several ways. For one, it is an
IETF RFC 3261 standard instead of the Cisco proprietary signaling protocol. This means
that third-party phones and applications can be used on the CUCM system using SIP. The
RFC for SIP states that the protocol was designed for the creation and management of
multimedia sessions over the Internet. Its architecture is a peer-to -peer model in theory.
Figure 3.12 shows how both SIP and RTP/RTCP communicate directly between endpoints.
F I G U R E 3 .1 2
SIP and RTP/RTCP flow
SIP Signaling
RTP/RTCP
SIP phones working in a Cisco Unified Communications environment are set up in SIP
proxy mode. The CUCM SIP proxy is used for making requests on behalf of endpoints.
This helps to facilitate policy enforcement by the CUCM administrator. When the SIP
phone is set up to work in a Cisco UC environment, the SIP proxy IP address is configured
to be the IP address of the CUCM Express. Figure 3.13 shows how the CUCM is used as
an SIP proxy.
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F I G U R E 3 .1 3
Voice over IP (VoIP)
CUCM SIP proxy
CUCM Express
is SIP Proxy
SIP Signaling
SIP Signaling
RTP/RTCP
As soon as an SIP session is established between the two SIP endpoints, the actual voice
stream flows between the users. The voice stream is managed by RTP and RTCP. The RTP/
RTCP stream is identical whether you are using SIP or SCCP for signaling. This is because
the voice streams use the signaling protocols only for setup and teardown. As soon as SIP
sets up the call, RTC/RTCP proceeds independently from SIP.
Voice Signaling Protocols in Review
It is very important that you be able to successfully compare and contrast the different
types of gateway and endpoint signaling protocols. Table 3.2 displays each protocol and its
characteristics.
TA B L E 3 . 2
Comparison of voice signaling protocols
Protocol
Standard
Architecture
Call Control
CUCM Uses
SCCP
Cisco proprietary
Client-server
Centralized
Voice gw/trunk and
endpoint to CUCM
SIP
IETF
Peer-to-peer
Distributed
Voice gw/trunk and
endpoint to CUCM
H.323
ITU
Peer-to-peer
Distributed
Voice gw/trunk
MGCP
IETF
Client-server
Centralized
Voice gw/trunk
Comparing the Common Voice Codecs
95
Comparing the Common
Voice Codecs
As you learned previously, voice codecs are responsible for the encoding and decoding of
voice signals. They also can compress the digital signal so that more voice calls can be
sent across a fi nite amount of bandwidth. In this section, we’re going to compare the most
common types of voice codecs used in a Unified Communications solution.
G.711
The G.711 ITU standard codec is also known as pulse code modulation (PCM). This
codec samples voice signals at a frequency of 8,000 samples per second. It provides the
best quality among the most commonly used codecs, but that comes at a price. There
are two common types of G.711 binary conversion techniques on most voice systems.
The fi rst G.711 type is called u-law and is most commonly used in the United States,
Canada, and Japan. The second G.711 binary conversion type is called a - law and is
used just about everywhere else on the planet. It is important to note that the two
PCM techniques are not compatible with each other and need to be transcoded for
interoperability.
Each phone call on the network requires 64 Kbps on the wire. Let ’s do some quick
math to see how much compression is being used on the G.711 codec. Using Dr. Nyquist’s
formula, we will get 8,000 voice samples each second. Each of these samples is 8 bits in
length. So if we multiply 8,000 × 8, we get 64,000 bps, or 64 Kbps. This means that G.711
uses no compression when it encodes the voice stream! If you have plenty of bandwidth,
G.711 is the way to go. If you are running voice over a low-speed WAN link or are
planning to use wireless IP phones, then you may want to consider a codec that compresses
the call into a smaller data stream.
G.729
The G.729 ITU standard codec samples the voice signal at the same rate as G.711 of 8,000
samples per second per the Nyquist rate theorem. Also like G.711, the bit rate is fi xed at
8 per sample. The major difference between G.711 and the variations of G.729 has to do
with compression. Using what’s known as conjugative-structure algebraic- code- excided
liner prediction (CS -ACELP), the G.729 codecs use alternate sampling methods and
algebraic expressions as a codebook to predict the actual numeric representation. These
smaller algebraic expressions are then sent to the remote side, where they are decoded and
the audio is synthesized to mimic the original audio tones. This audio waveform prediction
and synthesization degrades the quality of the voice signal by making the speaker’s voice
sound robotic. But the G.729’s use of CS -ACELP allows the compressed voice signal to
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require only 8 Kbps per call as opposed to 64 Kbps required per stream of G.711. This
means that eight voice calls using the G.729 codec can be made in the space of just one
G.711 codec call. Not a bad tradeoff when you are attempting to run voice over low-speed
WAN links!
G.729a
The G.729a ITU standard codec is very similar to G.729. They use the same 8Kbps
bandwidth consumption per call. Where the two codecs differ is in the type of algorithm
used to encode the voice signal. G.729 is considered a high- complexity codec and G.729 a
medium- complexity codec. Table 3.3 displays the complexity level of several of the more
popular codecs available today.
TA B L E 3 . 3
Voice codec complexity
Medium Complexity
High Complexity
G.711
G.729
G.729a
G.728
G.726
iLBC
All codecs are classified as either medium or high complexity. These categories are
important when dealing with DSPs, as we discussed earlier. We know that DSPs are
required for translating digital signals to analog and for transcoding between different
digital codecs. High- complexity codecs use more DSP processing power than mediumcomplexity codecs. Because DSP resources are fi nite, you may need to move from G.729 to
G.729a to free up those DSP resources. The downside is that the voice quality of G.729a
compared to G.729 is marginally worse.
iLBC
The Internet Low Bandwidth Codec is fairly new to the voice world and still waiting
to fully catch on. The codec uses either 20ms or 30ms voice samples, and they end up
consuming 15.2 Kbps or 13.3 Kbps, respectively. One other major benefit the iLBC touts is
its ability to handle moderate amounts of packet loss. If some of your VoIP packets are lost
in transit, iLBC ’s built-in techniques help the call continue with little notice to the user.
Where did this codec come from? Unlike all the other codecs we’ve discussed, it wasn’t
defi ned by the ITU. In 2000 a group of VoIP industry leaders collaborated and came up
Comparing the Common Voice Codecs
97
with the new codec. Hopefully, the cooperation between all the major players will fi nally
get the industry to settle on a universal codec. iLBC is the fi rst codec to attempt this.
Fortunately, Cisco is now beginning to build their IP phones to be able to understand the
iLBC codec. Other third-party vendors are also including iLBC support on their endpoints.
So in addition to using the G.711 and G.729 codecs, the phones can utilize iLBC. As this
book is being written, the following Cisco 7900 series phones understand the iLBC codec:
7906G
7911G
7921G
7925G
7942G
7945G
7962G
7965G
7975G
Which Codec Is Right for You?
In some instances there is a right or wrong choice in using one codec over another.
But most of the time, it really depends on the business requirements. Here are some
considerations to help you in making a codec decision:
What do my endpoints support? As stated earlier, Cisco phones support G.711 and
G.729. Newer phones also support the iLBC codec. So in Cisco environments you are going
to use one of these three. Don’t forget, however, that some codecs have variants, such as the
G.729 and G.729a protocols.
How many DSP channels will I need? DSP chips contain multiple channels with which
to handle codec transcoding. If you fi nd yourself having to constantly transcode one codec
into another, you probably made a poor codec decision. DSP resources are not cheap. If you
can limit transcoding, you should.
How much voice quality do I need? Depending on your users and the environment
they are used in, voice quality may or may not be a major factor in your decision-making
process. A tried but true method of determining the quality of the audio of a voice codec is
called the Mean Opinion Score (MOS). Developed by the ITU, this quality-measurement
system rates the quality of voice streams for a particular codec. It is actually very
nonscientific, because the MOS is an average “opinion” of a group of people who listen to
the same sentence for each codec tested. These listeners then rank the voice quality on a
scale of 1 to 5. A score of 1 is the worst quality, and a score of 5 is excellent quality.
Table 3.4 lists the codecs we discussed and each one’s MOS score.
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TA B L E 3 . 4
MOS scores of common VoIP codecs
Codec
MOS
G.711
4.1
iLBC
4.1
G.729
3.92
G.729a
3.7
How much bandwidth do I have to play with? If you are going to run voice over lowbandwidth WAN links, then you should defi nitely look at codecs that use at least some
compression.
Which codec works best for my environment? If you have a high amount of packet loss
on your network, you may want to look into the codecs that offer the best recoverability
when VoIP packets are lost in transit.
Calculating IP Voice Packet Sizes
Continuing our topic of codec comparisons, let’s focus on how different codecs affect the size
of the IP packets that are sent across a network. Usually, choosing a codec depends on the type
of environment the phones will operate in. You must be careful when you wish to use VoIP in
low-bandwidth situations, such as running over low-speed WAN connections. If you need to
run voice over the WAN, it’s critical to know the size of the voice packets to determine how
your network will scale. You should be able to calculate how much bandwidth a particular IP
voice packet consumes so you can properly engineer you network for the approximate number
of simultaneous calls you are anticipating. In calculating packet sizes, there are some packet
components whose size is static and never changes. Other components are variable and can be
manipulated depending on the codec used to change their size. The following sections present
all of the static and variable services that change the size of the packets.
Voice Packet Payload
A voice packet payload consists of the following:
Sample times of audio streams: Typically, most codecs take sample sizes between 10ms
and 30ms.
Codec bandwidth used: We already know how to calculate this using the Nyquist
calculation.
Calculating IP Voice Packet Sizes
99
One factor we always use to calculate the size of a single voice packet payload is the fact
that we use an 8-bit sample. Therefore, to calculate how big our voice payload is in bytes,
we use the calculations shown in the following example.
Let’s say we’re using the G.711 codec with a sample size of 20 ms. Using the Nyquist
calculation, we already know that the codec bandwidth rate for G.711 is 64 Kbps.
Therefore, to determine the number of bits per packet, it’s a simple calculation of
multiplying the codec bandwidth rate by our sample size:
Audio_payload_bits = 20 × 64
Audio_payload_bits = 1280
Now that we know what our payload is in bits, we divide by 8 to determine the bytes of
audio contained in a single voice packet for the G.711 codec using a 20ms sample:
Audio_payload_bytes = 1280 / 8
Audio_payload_bytes = 160
Layer 2 Header Information
Layer 2 header information depends on the Layer 2 methods you are using on your
network. Typically LANs use Ethernet as the Layer 2 transport. Two common WAN
Layer 2 protocols are Frame Relay and PPP:
Ethernet header: 20 bytes
Frame Relay: 4 – 6 bytes
Point-to -Point Protocol (PPP): 6 bytes
Layer 3 Header Information
Since we’re dealing with VoIP, the Layer 3 numbers are static for our calculation. All VoIP
must use IP, UDP, and RTP/RTCP. Therefore, we need header information, which uses up
space! Here is the number of bytes each one uses:
IP: 20 bytes
UDP: 8 bytes
RTP/RTCP: 12 bytes
Special Case Packet Additions
You may need to tunnel or encrypt voice traffic over your network. If this is the case, you
must include the overhead for whatever protocol you are using. Here is a short breakdown
of the bytes added by some of the more popular encryption techniques being used today:
IPSec: 50 –57 bytes
GRE tunneling protocol: 4 -20 bytes (Cisco uses 8 bytes)
MPLS tagging: 4 bytes per tag (may be more than one tag present)
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Now we can add everything together to determine how fat or skinny our VoIP packets
are! This will give us the total packet size in bytes. This is great, but our LAN/WAN
links are calculated in either Kbps or Mbps. We need to do two more calculations; fi rst we
must figure out our bandwidth requirements in bytes per second and ultimately how many
bits per second.
Calculating Bytes per Second
In addition to the number of bytes your voice packet will consume, you’ll need to go back
and retrieve the sample size rate in ms that the codec uses. In our example, we are using
20ms samples. We’re looking to calculate the number of bytes per second that a voice
stream will generate. There are 1000 ms in 1 second. Therefore we can use this calculation
to come up with the number of packets per second (packets_per_second):
packets_per_second = 1000 / sample_size
packets_per_second = 1000 / 20
packets_per_second = 50
With this information, we want to see how many bytes are consumed every second.
Let ’s use the example voice packet size of 220 bytes (160 bytes in payload + 60 bytes for
Ethernet and IP overhead) and our calculated 50 packets per second to come up with
bytes per second. We can use the following equation to determine the number of bytes
per second:
bytes_per_second = voice_packet_size 3 packets_per_second
bytes_per_second = 220 × 50
bytes_per_second = 11000
Calculating Bits per Second
Our last step is to convert our bytes per second into bits per second. This is just a matter of
multiplying our bytes_per_second value by 8:
bits_per_second = bytes_per_second × 8
Using our 11,000 bytes_per_second number above, we can determine bits per second as
follows:
bits_per_second = 11000 × 8
bits_per_second = 88000
This means that each RTP stream consumes 88,000 bps, or 88 Kbps, using the
uncompressed G.711 codec over Ethernet.
You can use this information to help determine how many voice streams your network
can support depending on the codec and L2/L3 technologies that you wish to utilize.
Calculating IP Voice Packet Sizes
101
Size Calculation Examples
Let’s practice our IP voice packet size calculations with the following two examples:
Example 1 information The codec that we have chosen uses a sample time of 30 ms and
a codec bandwidth of 8 Kbps. Our packet will be traversing only Ethernet networks. The
packets will be tunneled using GRE.
Example 1 solution
First we need to calculate the voice packet payload size into bits:
audio_payload_bits = 30 × 8
audio_payload_bits = 240
Next we convert bits into bytes:
audio_payload_bytes = 240 / 8
audio_payload_bytes = 30
Then we need to add up our Layer 2 and Layer 3 header information:
Ethernet header: 20 bytes
IP: 20 bytes
UDP: 8 bytes
RTP/RTCP: 12 bytes
Total = 60 bytes
Because we’re tunneling using GRE, this adds an additional 8 bytes to the packet size.
Adding all three numbers together gives us our voice IP packet size:
Audio payload: 30 bytes
L2/L3 header: 60 bytes
GRE encapsulation: 8 bytes
Total voice packet: = 98 bytes
Now we can figure out the packets, bytes, and bits per second this stream uses:
Packets per second
packets_per_second = 1000 / sample_size
packets_per_second = 1000 / 30
packets_per_second = 33.33
Bytes per second
bytes_per_second = voice_packet_size 3 packets_per_second
bytes_per_second = 98 × 33.33
bytes_per_second = 3,266.34
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Bits per second
bits_per_second = bytes_per_second 3 8
bits_per_second = 3,266.34 × 8
bits_per_second = 26,130.72
So this particular stream uses approximately 26,100 bps, or 26.1 Kbps, per stream.
Example 2 information Our second example codec uses a sample time of 10 ms and a
codec bandwidth of 32 Kbps. Our packet will be going across an MPLS WAN link. MPLS
uses Ethernet for transport over the WAN, so it will use standard Ethernet headers in
addition to the MPLS tags. The voice packets will be encrypted over the MPLS network
using IPSec, which consumes 57 bytes for encapsulation.
Example 2 solution
First we need to calculate the voice packet payload size in bits:
audio_payload_bits = 10 × 32
audio_payload_bits = 320
Next we convert bits into bytes:
audio_payload_bytes = 320 / 8
audio_payload_bytes = 40
Then we need to add up our Layer 2 and Layer 3 header information:
Ethernet header: 20 bytes
MPLS header: 4 bytes
IP: 20 bytes
UDP: 8 bytes
RTP/RTCP: 12 bytes
Total = 64 bytes
We want to encrypt our sensitive voice traffic over the MPLS network using IPSec, which
adds 57 bytes to the overall packet size. Adding all three numbers together gives us our
voice IP packet size:
Audio payload: 40 bytes
L2/L3 header: 64 bytes
IPSec encryption: 57 bytes
Total voice packet = 161 bytes
Again, we can figure out the packets, bytes, and bits per second this stream uses:
Packets per second
packets_per_second = 1000 / sample_size
packets_per_second = 1000 / 10
packets_per_second = 100
Calculating IP Voice Packet Sizes
103
Bytes per second
bytes_per_second = voice_packet_size 3 packets_per_second
bytes_per_second = 161 × 100
bytes_per_second = 161,000
Bits per second
bits_per_second = bytes_per_second × 8
bits_per_second = 161,000 × 8
bits_per_second = 128,800
So this particular stream uses 128,800 bps, or 128.8 Kbps, per stream.
Now you should have a good understanding of how to calculate the approximate size of
a voice packet depending on the codec and additional services you require. You can then
divide this number into the number of bytes available on your network links to see the
maximum number of simultaneous calls your network could handle. Using Example 2,
let’s assume our MPLS network is 20 Mbps, or 20,000 Kbps, and 50 percent of the circuit
is already being utilized with data traffic. This leaves us with 10,000 Kbps on average for
voice. Given that each RTP stream consumes 128.8 Kbps, we can determine the number of
simultaneous calls that can be made on the link:
simultaneous_calls = voice_in_Kbps / available_bw_in_Kbps
simultaneous_calls = 128.8 / 10,000
simultaneous_calls = 77.64
Reducing Voice Packet Sizes
You can reclaim some bytes from your voice streams in additional ways. One method that
was already mentioned is the RTP header compression, or cRTP. As detailed earlier in this
chapter, you can use cRTP to cache the 40 -byte RTP/IP/UDP header information on one
side of a link and send only the remainder of the header information to the other side of the
link. This cuts the 40 -byte header information down to 2 –5 bytes.
Another great bandwidth-saving utility is called Voice Activity Detection (VAD). VAD
monitors the voice conversation that is taking place, and when it detects silence on the RTP
stream, it stops transmitting RTP packets across the wire. VAD is not enabled by default.
When you make a phone call, you might be surprised to learn that anywhere from 35 to
40 percent of the conversation is complete silence! When VAD is disabled, the RTP
stream will package up the silent voice slices and send them over as normal. VAD is
more intelligent and will send only RTP data that actually contains voice. VAD must
communicate with the other end on the connection so it will play a prerecorded “silence”
VoIP packet instead of the actual packet from the source.
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It is important to realize that the 35 – 40 percent is an average number
based on several studies. If you have a Chatty Cathy on the other end, this
number will obviously drop. Also, areas with a great deal of background
noise—such as a data center with constantly buzzing fans—will not
see much of an impact at all if VAD is enabled. When performing your
bandwidth savings calculations, it ’s better to play it more conservatively
and use a number between 10 and 20 percent.
Examples of When to Use Specific Codecs
At times it is highly recommended and sometimes required to use one type of codec over
another. These are Cisco recommendations based on best-practice situations for common
tasks. This section describes which codecs are recommended for voice and modem passthrough and when incorporating MoH over a low-speed WAN link.
ATA Fax and Fax Modem Pass-through
Using the Cisco ATA hardware, it is possible to connect analog phones and fax machines
to an IP network. A problem arises when you attempt to use a fax machine without the
correct codec. If you try to configure the port to use G.729 or G.729a, the compression
methods used are not enough to correctly interpret the analog signals that the fax machine
is transmitting. Therefore it is necessary to configure fax machines and fax modems to use
the G.711 codec. Because that data is uncompressed, the analog signals that fax machines
and modems use are more accurately digitized. When the digital signal is decoded on the
other side, to the digital sample can successfully be rebuilt into an analog wave that is
understood by the receiving fax machine or modem.
Music on Hold
The G.729 and G.729a codecs are designed to be optimized for human speech. This
allows the data to be compressed down to a tiny 8Kbps stream. These compressed codecs
are commonly used for transmission of voice over low-speed WAN links. But because the
codecs are optimized solely for speech, they often do not provide adequate quality for
Music on Hold (MoH) streams. If you determine that the MoH quality is unacceptable
using the G.729 and G.729a codecs, there is a way to force MoH to use the higher-quality
G.711 codec while voice communication still uses one of the lower-quality codecs using
CUCM regions. This technique is outside the scope of this book.
Summary
Chapter 3 examined VoIP in Cisco networks beginning with an overview of the four layers
of the Unified Communications Model— Infrastructure, Call Control, Applications, and
Endpoints — and the components in each layer.
Exam Essentials
105
Voice gateways perform crucial infrastructure functions, and this chapter examined
the four types of voice gateway protocols used in a Unified Communications environment.
RTP and RTSP are the transport protocols for streaming media over IP networks, and we
examined how they differ. The Cisco Unified Communications Manager uses two endpoint
signaling protocols, SCCP and SIP, and this chapter showed how they differ.
The CUCM can use four common voice codecs: G.711, G.729, G.729a, and iLBC. The
second half of the chapter described each one, how they differ, and the situations where
each codec is most often used. As an administrator, your choice of which codec to use
depends on calculating the voice packet size. The chapter concluded by discussing and
illustrating those calculations.
This information should provide you with a thorough understanding of exactly how
voice and data are transported on and between IP networks. Voice and data are two great
services that can work together when VoIP is properly configured.
Exam Essentials
Know the Cisco Unified Communications Model. The four layers of the communications
model are the Infrastructure, Call Control, Applications, and Endpoints layers. Each layer
plays a specific role in the Unified Communications system.
Understand how voice gateways connect to the PSTN and legacy PBX systems. Voice
gateways bridge an IP network to a non-IP legacy network, either of which may be public
or private.
Understand the role DSPs play in the Unified Communications system. DSPs provide
analog-to -digital translation, codec transcoding, echo cancellation, DTMF relay services,
and media termination points. DSP farms typically reside on voice gateway routers.
Understand the difference between POTS and VoIP dial peers. POTS dial peers are
traditional PSTN technology, while VoIP dial peers connect to endpoints using IP.
Know how dial peers and call legs are associated with each other and where they initiate
and terminate on the voice network. A call leg is always associated with a dial peer. It
marks a logical point along the path of a phone call.
Understand the protocols that are responsible for signaling between the CUCM and
a voice gateway. The H.323, SIP, MGCP, and SCCP protocols can be used for voice
gateway signaling. Each protocol has different setups that change the way the dial peers
and route plans are located.
Know which protocol is responsible for the transport of the actual voice payloads. RTP
is used for the transport of voice packets on a Unified Communications system. RTCP is
responsible for out- of-band monitoring of the RTP packets.
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Know which protocol is used for compression of voice packets. cRTP header compression
can be used to shrink the IP, UDP, and RTP headers to decrease bandwidth consumption.
Understand the two Cisco Unified Communications endpoint signaling protocols. SCCP
is a Cisco proprietary protocol that uses client-server architecture. SIP is an IETF
standards-based protocol that uses peer-to -peer architecture. SIP is used in a proxy server
mode by the CUCM.
Be able to compare and contrast all voice signaling protocols. Understand the differences
in standards, architecture, call control, and CUCM uses for SCCP, SIP, H.323, and MGCP.
Understand the most common voice codecs and when to use them. The three most
common codecs are G.711, G.729, and G.729a. Each codec has advantages and
disadvantages depending on the network speed it runs on.
Understand when specific codecs are either required or highly desirable. It is important
to know which codec is required in situations where fax machines and fax modems are
used. Also, when using voice over low-speed WAN links, you need to understand why
G.729 should be used for voice and G.711 for MoH.
Written Lab 3.1
Write the answers to the following questions:
1.
What are the four layers of the Unified Communications Model?
2.
What are the two signaling protocols used in the Endpoints layer?
3.
Name the layer where the Call Manager resides.
4.
What can you do at the Infrastructure layer of the UC Model to ensure that voice
packets have priority over data packets?
5.
What is the term used when DSPs translate between two different codecs?
6.
List the two types of voice gateway dial peers.
7.
Name the two signaling protocols that can be used for both voice gateway and
endpoint signaling.
8.
What is a possible side effect when using cRTP?
9.
What is the size of IP/UDP/RTP header information that is uncompressed?
10. What is G.711 also known as?
(The answers to Written Lab 3.1 can be found following the answers to the Review
Questions for this chapter.)
Review Questions
107
Review Questions
1.
At which layer of the UC model can ACLs be implemented to limit which IPs and ports can
access the CUCM?
A. Infrastructure layer
2.
B.
Data Link layer
C.
Call Control layer
D.
Applications layer
E.
Session layer
The CUCM is responsible for all of the following except:
A. User management
3.
B.
Call signaling
C.
Call processing
D.
QoS enforcement
E.
Media resource management
What can you use to help eliminate voice congestion on slow WAN links?
A. RTCP
4.
B.
RTP
C.
cRTP
D.
TCP/IP
What Unified Communications solution resides in the UC Model Applications layer and is
responsible for handling endpoint location information for emergency services?
A. Unity
B.
5.
Cisco Unified Communications Manager
C.
E911
D.
Cisco Emergency Responder
E.
SRST
Which Cisco UC application provides integration with Cisco Presence applications
such as voice, video, email, instant messaging, and web collaboration?
A. Unity
B.
Customer Contact Solution
C.
Call Control
D.
Cisco Unified Communications Manager
E.
Real-time Transport
Chapter 3
108
6.
Voice over IP (VoIP)
At which layer of the UC model are ATAs found?
A. Infrastructure layer
7.
B.
Endpoints layer
C.
Call Control layer
D.
Application layer
What is the centralized location for hardware that handles codec transcoding and voice
translation services?
A. Infrastructure layer
8.
B.
Transcoding farm
C.
DSP farm
D.
Applications layer
E.
Unity
What is the term for the reflection of sound waves that arrive to the listener a short time
after the direct sound is heard?
A. Echo
9.
B.
Refraction
C.
DTMF
D.
DSP
E.
Transcoding
Which method is not a valid hardware DSP option?
A. Installed on a network module (NM)
B.
Installed on a compact flash (CF) card
C.
Installed on a WIC module
D.
Installed on a VWIC module
E.
Directly plugged into the router motherboard
10. When RTP payloads are encrypted, which RTP header bit is set?
A. Extension
B.
Padding
C.
Payload Type
D.
SSRC
Review Questions
109
11. When a phone call is made between two IP endpoints, how many RTP and RTCP streams
are established?
A. Four RTP and four RTCP
B.
One RTP and one RTCP
C.
Two RTP and one RTCP
D.
One RTP and two RTCP
E.
Two RTP and two RTCP
12. What types of signaling protocols allow the endpoints to contain the intelligence to place
their own calls?
A. Client-server
B.
SCCP
C.
Peer-to -peer
D.
CDP
13. What is needed to convert G.729a to G.711?
A. PSTN resources
B.
Analog voice gateway
C.
Transcoding resources
D.
H.323 signaling
E.
SCCP signaling
14. What function is commonly present in a VoIP network but never found in a purely
traditional telephony network?
A. Call processing
B.
Call supervision
C.
Dial plans
D.
Transcoding
15. When two voice gateways are separated by a VoIP network, what type of dial peer is
required to complete calls between the two sites?
A. VoIP dial peer
B.
POTS dial peer
C.
IP dial peer
D.
PSTN dial peer
Chapter 3
110
Voice over IP (VoIP)
16. What is in charge of translating an analog voice signal to digital?
A. SCCP gateway
B.
H.323
C.
DSP
D.
Transcoder
E.
MGCP gateway
17. What are the names for logical hops along a voice network that are used to complete a call
from one phone to another?
A. Dial strings and dial plans
B.
SCCP and SIP
C.
VLAN and CDP
D.
FXS and FXO
E.
Dial peers and call legs
18. What situation would require a voice gateway?
A. Connecting calls over an IP WAN
B.
Connecting calls over a LAN
C.
Connecting calls over a high-speed MAN
D.
Connecting calls to the PSTN
E.
Connecting calls between two Ethernet switches
19. What type of voice gateway signaling protocol would you implement if you wish to
configure dial peers directly on the voice gateway router?
A. SCCP
B.
H.323
C.
RTP
D.
cRTP
20. What protocol is responsible for the sequencing of voice packets?
A. RTCP
B.
Jitter
C.
G.711
D.
UDP
E.
RTP
Answers to Review Questions
111
Answers to Review Questions
1.
A. The Infrastructure layer is where all route/switch, QoS, and security are performed.
2.
D. QoS enforcement is handled by network devices located in the Infrastructure layer of
the Unified Communications Model.
3.
C. cRTP compresses the IP/UDP/RTP headers to shrink the overall packet size of voice
packets.
4.
D. The Emergency Responder keeps a database listing the location of all endpoints on the
system. This information is relayed to emergency services when needed.
5.
B. The Customer Contact Solution integrates value-added communications features to
improve customer relations.
6.
B. ATAs are basically IP phones that have analog ports attached to them so analog signals
can be converted to digital and packetized for transport.
7.
C. A DSP farm is a router (commonly the voice gateway router) that contains one or more
DSP hardware chips.
8.
A. Echo is often experienced on voice calls and is amplified by codec compression. DSPs
are used to help reduce excess echo.
9.
B. DSPs are hardware modules. They are not installed on CF cards.
10. B. The Padding bit is set, which indicates that the RTP has one or more octets at the end
of the encapsulated packet that are not part of the payload. This additional information is
typically used for encryption purposes.
11. E. A single phone call requires one RTP and one RTCP stream for each phone. This means
that two RTP/RTCP streams are established for every IPT call.
12. C. Peer-to-peer signaling protocols give call-making intelligence directly to
the endpoint.
13. C. Transcoding is the process of translating between different digital
voice codecs.
14. D. Transcoding is the process of translating between digital voice codecs.
15. A. VoIP dial peers interconnect two voice gateways running VoIP.
16. C. A digital signal processor converts analog voice into digital and digital into analog.
17. E. Dial peers and call legs are logical hops required to connect an end-to-end call.
18. D. You need a voice gateway to connect to the PSTN.
19. B. H.323 is a peer-to-peer protocol, so all dial-peer configuration is decentralized from the
CUCM and is done on the voice gateway.
20. E. RTP includes a sequencing field in its header.
Chapter 3
112
Voice over IP (VoIP)
Answers to Written Lab 3.1
1.
Infrastructure layer, Call Control layer, Applications layer, Endpoints layer
2.
SCCP and SIP
3.
Call Control layer
4.
QoS
5.
Transcode
6.
POTS and voice network
7.
SCCP and SIP
8.
Increased CPU utilization
9.
40 bytes
10. Pulse code modulation (PCM)
Chapter
4
Configuring
the Network
Infrastructure
for Voice
THE CCNA VOICE EXAM TOPICS
COVERED IN THIS CHAPTER INCLUDE
THE FOLLOWING:
Describe and configure a Cisco network to support VoIP.
Describe the purpose of VLANs in a VoIP environment.
Describe the environmental considerations to support VoIP.
Configure switched infrastructure to support voice and
data VLANs.
Describe the purpose and operation of PoE.
Identify the factors that impact voice quality.
Describe how QoS addresses voice quality issues.
Identify where QoS is deployed in the UC infrastructure.
Implement Cisco Unified Communications Manager
Express to support endpoints using CLI.
Describe the requirements and correct settings for DHCP
and NTP.
Configure DHCP and NTP.
Anyone who loves Chinese food knows that rice is the
foundation of a typical Chinese meal, whether it’s Mongolian
beef, sweet and sour chicken, or mu shu pork. Even though
these three dishes are quite different in taste, they all have one thing in common. They are
always served with rice. You can think of the network infrastructure as your bed of rice.
The “main dish” in our network may be data or it may be voice; both utilize the same
network. In this chapter we’ll discuss how to utilize the network infrastructure to support
voice capabilities.
We’ll also explore the options available for powering your Cisco IP phones. Then
you’ll see how we configure our network for voice by configuring VLANs, trunks, and
inter-VLAN routing. Next we will explore the VLAN Trunking Protocol to see what
it is used for and how to configure it. We’ll cover some quality of service (QoS) basics,
and you’ll see how to configure auto - QoS for VoIP. We’ll then discuss how to eliminate
potential voice problems by configuring various link- efficiency techniques. Finally, we’ll
cover some network services that help support voice functionality, including DHCP and
NTP services.
Power Options for IP Phones
So you’ve decided to take the plunge and order some Cisco IP phones. When you receive
and unpack the phones, the fi rst question you might ask yourself is, “How the heck do I
power these things up?” There are three ways of providing power to your phones:
Power brick
Powered patch panel/power injector
Power over Ethernet (PoE) switch
Let’s review each power method in detail.
Power Brick
The power brick is an obvious choice. It connects to a power port on the back of the phone
and plugs into a standard 110V AC outlet on your wall. You then connect a Category 5 or
6 Ethernet cable into a switch to provide network connectivity.
Power Options for IP Phones
115
Power bricks do not come standard on any Cisco IP phone! Cisco assumes
that you will use some method of PoE. Many people forget about this
and are very disappointed when they have to place a second order for
power bricks.
The power brick option may be useful in situations when you will be using only a
handful of phones. Otherwise, you may want to investigate a PoE option, because it can
be more cost effective, and quite simply, it ’s nicer to combine power and Ethernet in
one cable to eliminate the need for a second connection to the phone. Figure 4.1 shows
a Cisco phone receiving power from the power brick with a separate connection to the
LAN using Ethernet.
F I G U R E 4 .1
The power brick option
Power Brick
Ethernet
Connection
Network
Non-PoE
Switch
Powered Patch Panel/Power Injector
A second power option is to have a device that sits in between your IP phone and a nonPoE – capable switch. This is known as a midspan method because the power sits in the
middle of the connection. A powered patch panel can terminate nonpowered Ethernet
on one end and a powered Ethernet termination point on the other. These patch panels
allow the power to be connected at the wiring closet, so no power brick is required,
and the phone receives both power and Ethernet over a single Category 5 or 6 Ethernet
cable. A standard Category 5/6 cable has a total of eight wires. 10BaseT and 100BaseT
Ethernet utilize only RJ- 45 pins 1, 2, 3, and 6. Pins 1 and 2 are for transmit and 3 and
6 are for receive. The other wires are essentially unused. Midspan switches will use the
unused wires 4, 5, 7, and 8 to power the endpoints. The problem with this midspan setup
is that 1000BaseT uses all four pairs of wires to transmit and receive. 1000BaseT must
use pins 1, 2, 3, and 6 for power. Therefore, the midspan option is applicable only for
10/100BaseT.
116
Chapter 4
Configuring the Network Infrastructure for Voice
Most patch panels typically come in either 24 - or 48-port configurations. This may be the
most cost-effective method if you have a significant investment in non-PoE – capable access
switches. It also allows for a centralized location where you can provide uninterruptible power
supply (UPS) power so the phones will remain functioning in the event of a power outage.
Figure 4.2 shows how a powered patch panel can provide PoE functionality.
FIGURE 4.2
The powered patch panel option
Power and
Ethernet
Ethernet
Only
Network
Provides power to
entire patch panel
Non-PoE
Switch
You can also purchase a Cisco power injector. These devices provide the same midspan
“sit-in-the-middle” power function as the powered patch panel but only for a single phone.
Power over Ethernet Switch
The most streamlined and efficient method to provide power to phones (and other PoE capable devices) is the Power over Ethernet (PoE) switch. The switch is responsible
for detecting and outputting the required power on each switchport. By adding PoE
functionality to the switch, fewer devices need UPS protection in the event of a power
outage. Figure 4.3 shows how inline power switches provide power to endpoints.
FIGURE 4.3
The PoE switch option
Power and
Ethernet
Network
PoE
Switch
Power Options for IP Phones
117
You need to be aware of a couple of “gotchas” when it comes to powering Cisco phones
with any PoE option. The fi rst deals with the type of inline power and quantity that the
phone supports. The second thing to watch out for is ensuring your switch can properly
handle the power load. Let’s fi rst look at the two inline power methods for Cisco switches,
and then we’ll look at switch power capacities.
Inline Power Method 1: Cisco Inline Power
Cisco began offering a proprietary Inline Power option to customers before a universal
standard was available. In early 2000, Cisco began selling Catalyst switches with
their proprietary Inline Power (ILP) functionality. Unlike the midspan switches, ILP
uses the same RJ - 45 pins 1, 2, 3, and 6 to provide power to the phones. Using the
same wiring that Ethernet uses to transmit and receive is called phantom power. In a
sense, ILP could have been used to power 1000Base -T Ethernet phones, but none were
available at the time.
Cisco’s proprietary Inline Power provides a fi xed 6.3 watts of power to any device
that supports the power method. ILP detects a capable device by sending a low-voltage
AC signal across the transmit pairs and expects to receive the same signal coming back
on the receive pairs. This is because the ILP capable phones have a low-pass fi lter that
bridges the specifi c voltage signal from TX to RX. Once the switch receives the voltage
back on the receive pair, it knows that the device requires power and initially sends 6.3
watts on that specifi c switchport. This provides enough power for the device to boot
into low power mode. When the device has fully booted into low-power mode, CDP
messages are exchanged between the switch and PoE device to negotiate the actual
power required by the device up to 15.4 watts.
Inline Power Method 2: Cisco IEEE 802.3af
In mid-2003 the IETF came out with the 802.3af PoE standard. This became the de facto
standard for powering Ethernet over 10/100/1000BaseT. The standard states that power
can be sent across the Cat5/6 cabling either on active transmit/receive pairs or over the
inactive pairs for 10/100Base-T. Because 1000Base-T requires pins 1, 2, 3, and 6 for power,
Cisco uses this standard on their 802.3af-supported PoE switches.
The 802.3af standard handles endpoint detection using a different method than ILP.
It uses a low-powered DC signal sent across a copper pair. Just like ILP, the voltage is
looped back to the switch by a slightly more advanced fi lter to signal that the end device
is capable of receiving power. Unlike ILP, 802.3af has five specific classes of power
that it can transmit. It knows the power level the end device requires by the voltage
strength that it receives during the detection phase. Table 4.1 lists the 802.3af power
classifications.
Chapter 4
118
TA B L E 4 .1
Configuring the Network Infrastructure for Voice
IEEE 802.3af classifications
Class
Usage
Minimum Power Level at
the Switch (in Watts)
Maximum Power Level at
the Device (in Watts)
0
Default
15.4
0.44–12.95
1
Optional
4.0
0.44–3.84
2
Optional
7.0
3.84–6.49
3
Optional
15.4
6.49–12.95
4
Reserved for future
N/A
N/A
Class 0 is the default class and allocates a full 15.4 W of power to any device that falls
into the category. This class is for devices whose vendor did not choose to implement a power
classification. You’ll commonly find this in inexpensive PoE products. Moving up, a device
that declares itself as class 1 will have a maximum power requirement level between 0.44 and
3.84 W. The switch allocates 4.0 W of power for these devices. Class 2 allocates 7.0 W for
devices requiring a maximum power level of 3.84 to 6.49 W. Class 3 is for any device that
requires 6.49 to 12.95 W, and the switch allocates 15.4 W of power. Class 4 is not currently
in use but was set aside so an additional power level can be added in the future.
Cisco Inline Power Switch Backward Compatibility
Because Cisco jumped the gun a few years early with their prestandard ILP, they now need
to support the newer 802.3af as well as their proprietary ILP standard on their Catalyst
line of PoE switches. The methods of power detection are somewhat different between ILP
and 802.3af, and Cisco has come up with a method that allows its switches to detect the
power requirements of Cisco phones. Here are the steps the PoE switch goes through for
powering Cisco IP phones:
1.
The switch uses continuous low-powered AC and DC signals to the PoE ports. If an AC
signal is looped back to the switch, the device is ILP capable. If DC power is looped
back to the switch, the device is 802.3af capable.
2.
If the phone is only capable of using the ILP proprietary inline-power method, the
phone boots into “low power mode” and negotiates the actual power required by the
device using CDP. The phone then boots with the correct power requirements and the
process ends.
3.
If the phone is only capable of using 802.3af or supports both ILP and 802.3af, the switch
and PoE device use additional low-voltage signaling to determine the power class. When
the negotiation process is complete, the switch provides the necessary power to fully
boot the IP phone.
Power Options for IP Phones
119
The Right PoE for the Job
Tiana, a network engineer, was tasked with rolling out new Cisco 7971G IP desk phones
with color displays to replace the older 7940 series phones already in place. Tiana
thought this would be a simple project involving a few configuration changes on the
CUCM and a matter of swapping out the old phones for new ones. However, this was
not the case, because when Tiana attempted to swap the old phones with the new, she
discovered that the new phones did not power up. Tiana was confused by this because
the phones were using a PoE module in a 6500 series switch. The PoE module worked
perfectly for the old phones, so why didn’t it work for the new ones?
It turns out that the PoE module that was installed in the switch used the older Cisco ILP
method. This worked fine for the 7940 phones because they supported both the ILP and
80.2.3af standard. The newer 7971G phones, however, are only capable of supporting
the 802.3af standard.
This discovery caused a short delay in the rollout of the new phones. Tiana purchased a
switch module offering 802.3af power, the 7965 phones powered up, and the project was
finally complete.
Cisco PoE Intelligent Power Management
Depending on the types of endpoints you deploy and the type of switch and power supply
used, you need to be aware that you can eventually exhaust the amount of power available
to the switch. If you add too many PoE phones to a switchport, the switch may have
already allocated all the available power, and therefore your device will not receive the
necessary electricity to power the phone. Also, the 802.3af classification system can often
set aside more power than is necessary, which can unnecessarily limit the number of PoE
devices that can be powered. To better understand this situation, we need to briefly discuss
how to determine how many watts a power supply can generate. While this will not give
you an exact number, it should get you fairly close. It’s really just a simple math equation:
watts = volts × amps
Let’s use a Cisco Catalyst 4506 -E with 2- 48 port 10/100/1000 PoE modules and a
Supervisor 4 module as our example. The standard AC power supply is 110 volts with a
15-amp circuit. Therefore
watts = 110 × 15
watts = 1650
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Cisco plays its power limits conservatively, however, and it rates the 4500 Series 110V
15-amp power supplies at 1300 watts. This is to prevent any circuit trips when power
reaches the limit of 1650 watts. The maximum amount of power to boot and run the
switch supervisor module and line cards is approximately 600 watts. So let’s subtract that
from our total:
1300 – 600 = 700 watts
That’s how much power we have to allocate to various PoE devices that we can run
off the switch. Now let’s say you want to add a number of Cisco 7965G IP phones to this
switch. The 7965G is an 802.3af class 3 device. According to the 802.3af specification,
each 7965G phone added to this switch consumes 15.4 watts of power. Therefore, how
many phones can we put on this switch before the power is maxed out?
700/15.4 = 45.45
So even though we have 96 PoE ports on the switch, we can power only 45 of the 7965
phones. Technically, we do have two power supplies. By default they run in redundancy
mode. This is basically an active/standby situation where you can utilize the power from
only one power supply. You can configure them to utilize both power supplies by issuing
the power redundancy-mode combined command.
I do not recommend you do this, however, unless you are really need the extra power.
But you should at least know it’s an option to have in your back pocket if needed.
Cisco switches also have the ability to throttle back allocated power resources so that
unused watts can be put back into the power allocation pool. Using our 7965G phone
example again, according to the 802.3af standard, 15.4 watts will be allocated to the
phone. In reality, the phone will use a maximum of 12 watts. During the Cisco PoE switchnegotiation process using CDP, the switch will negotiate the power allocation down to the
12-watt limit. This is what Cisco refers to as Intelligent Power Management (IPM). So
now that we require only 12 watts of power per phone, let’s recalculate how many phones
we can fully power on our switch:
700/12 = 58.33
Using a Cisco PoE solution with IPM, we can power 58 phones instead of 45.
Cisco PoE Management Modes
The power inline IOS commands allow you to change PoE settings on a port-by-port
basis. Let’s look at the PoE interface commands available to us:
4506-switch(config-if)#power inline ?
auto
Automatically detect and power inline devices
never
Never apply inline power
static
High priority inline power interface
Power Options for IP Phones
121
This is the default setting. If the endpoint is a Cisco device such as a Cisco 7960 IP
phone, the power settings will be negotiated automatically. To demonstrate this, let’s look
at a show power inline command output on the 4500 switch:
4506-switch#show power inline gigabitEthernet 3/2
Interface Admin Oper
Power(Watts)
Device
Class
From PS
To Device
--------- ------ ---------- ---------- ---------- ------------------- ----Gi3/2
auto
on
13.5
12.0
Cisco IP Phone 7965 3
Interface
AdminPowerMax
(Watts)
---------- --------------Gi3/2
15.4
We can see that the Admin setting is set to auto. Using CDP, the switch detected the
Cisco 7965 phone. The switch placed it into 802.3af power settings as a class 3 device
and thus allocated 15.4 watts of power to it. However, the switch went one step further and
dropped the power output to the device to 12.0 watts.
Now see what happens if we change port 3/2 so it will not send any power on the port
regardless of what the device is on the other end:
4506-switch(config)#interface gi3/2
4506-switch(config-if)#power inline never
After running this command, you see that the port is no longer powering the phone.
Note that both the Admin and Oper status are set to off:
4506-switch#show power inline gi3/2
Interface Admin Oper
Power(Watts)
Device
Class
From PS
To Device
--------- ------ ---------- ---------- ---------- ------------------- ----Gi3/2
off
Interface
off
AdminPowerMax
(Watts)
---------- --------------Gi3/2
15.4
0
0
n/a
n/a
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Finally, let’s see what the results are with our 7965 phone if we statically set the power
fi rst to 6 watts and then to 15.4 watts:
4506-switch(config)#interface gi3/2
4506-switch(config-if)#power inline static max 6000
4506-switch(config-if)#
1d22h: %ILPOWER-5-ILPOWER_POWER_DENY: Interface Gi3/2: inline power denied
When I set the max power to 6 watts, a log message is generated telling us “inline power
denied.” Because this is a class 3 device, we have to configure the minimum watts for the
802.3af standard. The show power inline instruction shows the Admin status as static
but the Oper status as POWER_DENY. Also note that the switch does not identify the device as
being a 7965, because that needs a minimum of 7 watts to exchange CDP information.
4506-switch#show power inline gi3/2
Interface Admin Oper
Power(Watts)
Device
Class
From PS
To Device
--------- ------ ---------- ---------- ---------- ------------------- ----Gi3/2
static power-deny 0
0
Ieee PD
3
Interface
AdminPowerMax
(Watts)
---------- --------------Gi3/2
6.0
Let’s bring the phone back online by configuring the static max to 15.4 watts:
4506-switch(config)#interface gi3/2
4506-switch(config-if)#power inline static max 15400
4506-switch(config-if)#
Now we see that the phone is recognized and boots. It is now utilizing more power (a
full 15.4 watts) than when it is configured to auto -negotiate the power:
4506-switch#sh power inline gi3/2
Interface Admin Oper
Power(Watts)
Device
Class
From PS
To Device
--------- ------ ---------- ---------- ---------- ------------------- ----Gi3/2
Interface
static on
AdminPowerMax
(Watts)
17.3
15.4
Cisco IP Phone 7965 3
Understanding and Configuring VLANs and Voice VLANs
123
---------- --------------Gi3/2
15.4
The power inline static command can be useful if you have non- Cisco phones that
you know use only 8 watts, for example. Because non- Cisco devices cannot negotiate using
CDP, each port will allocate the full 15.4 watts of power to the device. You can hard- code
the power settings to 8 watts to save 7.4 watts per port!
Understanding and Configuring
VLANs and Voice VLANs
I highly recommend that you logically separate voice and data devices on your network.
To see why this step is necessary, you need to understand the concept of virtual LANs.
We’ll see how to configure VLAN trunk links to span logical VLANs across multiple
physical Layer 2 switches. Then we’ll look at how to configure Layer 3 inter-VLAN routing
so that devices on separate VLANs can communicate with one another. I ’ll then introduce
you to the VLAN Trunking Protocol (VTP), explaining what it is used for and how to
configure it. Finally, we’ll look at voice VLANs to see why they are needed and how to
deploy them in your network. Let ’s get started:
An Overview of VLANs
A virtual LAN (VLAN) is a logical segmentation of the network that allows a group of
devices to act as if they were on the same physical network. Devices that reside on the same
VLAN share the same broadcast domain. Figure 4.4 shows how a single physical switch
can be logically broken up into two separate “logical” switches.
FIGURE 4.4
Logical VLAN separation
Switch
VLAN 10
VLAN 20
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In this example we have four PCs connected to a Layer 2 switch. The switch is
configured to have two PCs in VLAN 10 and the other two in VLAN 20. All devices
within a VLAN are typically configured in the same IP subnet. So using our example,
devices residing in VLAN 10 will be configured with an IP address within the 192.168.10.
X/24 range, whereas devices in VLAN 20 are configured with an IP address within the
192.168.20.X/24 IP space.
Breaking up broadcast domains is critical on larger networks. Broadcast traffic in
flat networks with hundreds or thousands of devices can lead to degraded performance.
VLANs allow an administrator to break up a single broadcast domain into multiple
domains to decrease broadcast traffic. Let’s look at how we would configure VLANs on a
Layer 2 Cisco switch.
Configuring VLANs
Using Figure 4.4 as our example, let’s configure a Layer 2 switch for VLANs 10 and 20.
We’re going to say that VLAN 10 is our Sales department VLAN and VLAN 20 is our
Marketing department. Before we configure anything, let’s see what our switch is currently
configured with by issuing a show vlan brief command:
Switch#show vlan brief
VLAN Name
Status
Ports
---- -------------------------------- --------- ------------------------------1
default
active
Fa0/1, Fa0/2, Fa0/3, Fa0/4
Fa0/5, Fa0/6, Fa0/7, Fa0/8
Fa0/9, Fa0/10, Fa0/11, Fa0/12
Fa0/13, Fa0/14, Fa0/15, Fa0/16
Fa0/17, Fa0/18, Fa0/19, Fa0/20
Fa0/21, Fa0/22, Fa0/23, Fa0/24
Fa0/25, Fa0/26, Fa0/27, Fa0/28
Fa0/29, Fa0/30, Fa0/31, Fa0/32
Fa0/33, Fa0/34, Fa0/35, Fa0/36
Fa0/37, Fa0/38, Fa0/39, Fa0/40
Fa0/41, Fa0/42, Fa0/43, Fa0/44
Fa0/45, Fa0/46, Fa0/47, Fa0/48
Gi0/1, Gi0/2
1002 fddi-default
act/unsup
1003 token-ring-default
act/unsup
1004 fddinet-default
act/unsup
1005 trnet-default
act/unsup
Understanding and Configuring VLANs and Voice VLANs
125
The default setup for a Cisco switch is that all ports reside in VLAN 1. Now let’s
configure our two new VLANs and give them proper names:
Switch#configure terminal
Switch(config)# vlan 10
Switch(config-vlan)#name Sales
Switch(config)#vlan 20
Switch(config-vlan)#name Marketing
Switch(config-vlan)#end
Switch#show vlan brief
To verify that we properly created our two new VLANs, we’ll run another show vlan
brief command:
Switch#show vlan brief
VLAN Name
Status
Ports
---- -------------------------------- --------- ------------------------------1
default
active
Fa0/1, Fa0/2, Fa0/3, Fa0/4
Fa0/5, Fa0/6, Fa0/7, Fa0/8
Fa0/9, Fa0/10, Fa0/11, Fa0/12
Fa0/13, Fa0/14, Fa0/15, Fa0/16
Fa0/17, Fa0/18, Fa0/19, Fa0/20
Fa0/21, Fa0/22, Fa0/23, Fa0/24
Fa0/25, Fa0/26, Fa0/27, Fa0/28
Fa0/29, Fa0/30, Fa0/31, Fa0/32
Fa0/33, Fa0/34, Fa0/35, Fa0/36
Fa0/37, Fa0/38, Fa0/39, Fa0/40
Fa0/41, Fa0/42, Fa0/43, Fa0/44
Fa0/45, Fa0/46, Fa0/47, Fa0/48
Gi0/1, Gi0/2
10
Sales
active
20
Marketing
active
1002 fddi-default
act/unsup
1003 token-ring-default
act/unsup
1004 fddinet-default
act/unsup
1005 trnet-default
act/unsup
Our two new VLANs have successfully been configured, but all the ports still reside in VLAN 1.
Let’s move ports Fa0/1–24 into the Sales VLAN and Fa0/25–48 into the Marketing
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VLAN by entering into config-if-range mode to configure multiple switchports and then issuing
the switchport access vlan command to set the ports to reside in the proper VLAN:
Switch#configure terminal
Switch(config)#interface range fastEthernet 0/1 - 24
Switch(config-if-range)#switchport access vlan 10
Switch(config-if-range)#exit
Switch(config)#interface range fastEthernet 0/25 - 48
Switch(config-if-range)#switchport access vlan 20
Switch(config-if-range)#end
To verify that our ports are now in the correct VLANs, let’s run the show vlan brief
command one last time:
Switch#show vlan brief
VLAN
---1
10
Name
-------------------------------default
Sales
Status
--------active
active
20
Marketing
active
1002
1003
1004
1005
fddi-default
token-ring-default
fddinet-default
trnet-default
act/unsup
act/unsup
act/unsup
act/unsup
Ports
------------------------------Gi0/1, Gi0/2
Fa0/1, Fa0/2, Fa0/3, Fa0/4
Fa0/5, Fa0/6, Fa0/7, Fa0/8
Fa0/9, Fa0/10, Fa0/11, Fa0/12
Fa0/13, Fa0/14, Fa0/15, Fa0/16
Fa0/17, Fa0/18, Fa0/19, Fa0/20
Fa0/21, Fa0/22, Fa0/23, Fa0/24
Fa0/25, Fa0/26, Fa0/27, Fa0/28
Fa0/29, Fa0/30, Fa0/31, Fa0/32
Fa0/33, Fa0/34, Fa0/35, Fa0/36
Fa0/37, Fa0/38, Fa0/39, Fa0/40
Fa0/41, Fa0/42, Fa0/43, Fa0/44
Fa0/45, Fa0/46, Fa0/47, Fa0/48
As you can see, ports Fa0/1–24 now reside in the Sales VLAN, and ports Fa0/25 – 48 are
in the Marketing VLAN.
Configuring VLAN Trunks
The previous example showed how to configure VLANs on a single switchport. But what if
you have a network that spans multiple floors and requires the interconnection of multiple
Understanding and Configuring VLANs and Voice VLANs
127
switches? Taking this scenario one step further, what if you have Sales and Marketing
employees connected to multiple switches, but you would like them to reside in the same
logical VLAN? The solution to this problem is interconnecting switches with a VLAN trunk
port. Figure 4.5 shows our new network topology with two switches that have VLANs 10
and 20 trunked between them.
FIGURE 4.5
A VLAN trunk
Fa0/1
Switch A
Trunk VLAN
10 and 20
Fa0/1
Switch B
VLAN 10
VLAN 20
A VLAN trunk port is a link between two Layer 2 switches that can transport traffic
from multiple VLANs. It keeps the traffic between the VLANs separate by tagging each
frame. VLAN tagging essentially places a VLAN identifier on each frame. In our example,
frames on VLAN 10 that need to go from one switch to the other are tagged as belonging
to VLAN 10.
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Detailed discussion of VLAN trunking is beyond the scope of this book, but it is an
important part of preparing the network infrastructure for voice applications. I’ll show you
the most common method for configuring VLAN trunk links using the 802.1Q trunking
protocol. 802.1Q is an IEEE standard that is supported by virtually every network vendor
in the world.
Using Figure 4.5 as our example, let’s configure a VLAN trunk link between Switches A
and B using the 802.1Q trunking protocol on port Fa0/1. For simplicity’s sake, let’s assume
that both Switch A and Switch B have been identically configured to switch VLAN 10 and
20. Configuring an 802.1Q trunk between the switches requires the following steps.
Step 1: Configure the VLAN Trunk Encapsulation Type
You can configure your VLAN trunk encapsulation for either the 802.1Q standard or
Cisco’s proprietary ISL trunking. I highly recommend that you configure your trunk
using 802.1Q. I haven’t seen anyone using ISL trunks for years. You can also configure
one side of the trunk to negotiate the encapsulation method, but since we know we want
to use 802.1Q, just keep it simple and specifically configure it as such. The command for
configuring VLAN trunk encapsulation is switchport trunk encapsulation.
Step 2: Configure the VLAN Trunk Mode
There are several options for the VLAN trunk’s operational mode, including dynamic
desirable and dynamic auto. But since we know we want to configure the port as a trunk,
we can simply hard- code the port using the switchport mode trunk command. Let’s
configure Switch A and Switch B for trunking on port Fa0/1.
Here’s the Switch A configuration command:
Switch-A#configure terminal
Switch-A(config)#interface fa0/1
Switch-A(config-if)#switchport trunk encapsulation dot1q
Switch-A(config-if)#switchport mode trunk
Switch-A(config-if)#end
Here’s the Switch B configuration command:
Switch-B#configure terminal
Switch-B(config)#interface fa0/1
Switch-B(config-if)#switchport trunk encapsulation dot1q
Switch-B(config-if)#switchport mode trunk
Switch-B(config-if)#end
Now let’s issue a show interfaces trunk command on Switch A to see what our trunk
looks like:
Switch-A#show interfaces trunk
Understanding and Configuring VLANs and Voice VLANs
Port
Fa0/1
Port
Fa0/1
Mode
on
Encapsulation
802.1q
Status
trunking
129
Native vlan
1
Vlans allowed on trunk
1-4094
Port
Fa0/1
Vlans allowed and active in management domain
1,10,20
Port
Fa0/1
Switch-A#
Vlans in spanning tree forwarding state and not pruned
1,10,20
At this point, our VLAN trunk is up and running and is successfully tagging frames
on VLANs 10 and 20 between the two switches. There is one last cleanup step that is
typically taken to keep the configuration clean. As you’ll notice from the output of the show
interfaces trunk command, the VLANs that are allowed to send traffic between the
two switches are 1– 4094. This basically means that any new VLANs created on the two
switches can be trunked. Just as a precautionary measure, I typically limit which VLANs
are allowed on the trunk by issuing a switchport trunk allowed command on both trunk
interfaces to limit trunking to VLANs 10 and 20. Let’s do this now.
Here’s the Switch A configuration command:
Switch-A#configure terminal
Switch-A(config)#interface fa0/1
Switch-A(config-if)#switchport trunk allowed vlan 10,20
Switch-A(config-if)#end
Here’s the Switch B configuration command:
Switch-B#configure terminal
Switch-B(config)#interface fa0/1
Switch-B(config-if)#switchport trunk allowed vlan 10,20
Switch-B(config-if)#end
Now when we run the show interfaces trunk command again, we see that only
VLANs 10 and 20 are allowed to traverse the trunk:
Switch-A#show interfaces trunk
Port
Fa0/1
Mode
on
Encapsulation
802.1Q
Status
trunking
Native vlan
1
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Port
Fa0/1
Vlans allowed on trunk
10,20
Port
Fa0/1
Vlans allowed and active in management domain
10,20
Port
Fa0/1
Switch-A#
Vlans in spanning tree forwarding state and not pruned
10,20
Now we have a two -switch network with two VLANs that are properly trunked
together. A Sales department user on a PC on Switch A, VLAN 10, can communicate with
another Sales department user attached to Switch B on VLAN 10. The same is true for
the Marketing department users on VLAN 20. But what happens if a Sales department
user PC needs to communicate with a Marketing department PC? Because VLANs break
up switches into multiple logical switches, the PCs on different VLANs currently have no
way of communicating with each other. To solve this dilemma, we need to configure interVLAN routing.
Implementing Inter-VLAN Routing
When you statically configure IP addressing on a PC, you are required to enter the
following information:
A unique IP address
A subnet mask
A default gateway
Every VLAN on your network is assigned its own IP subnet. For example, our Sales
VLAN (VLAN 10) has been given the following IP space:
IP subnet: 192.168.10.X/24
Default gateway: 192.168.10.1
Because the IP space is a /24, the IP space has a subnet mask of 255.255.255.0.
End devices such as PCs can be assigned to IP addresses between 192.168.10.2 and
192.168.10.254.
Our Marketing VLAN (VLAN 20) is in a separate IP space:
IP subnet: 192.168.20.X/24
Default gateway: 192.168.20.1
When a PC on VLAN 10 talks to another PC on the same VLAN, it uses broadcast
messages to initially find the end device. Because broadcasts are contained within a
Understanding and Configuring VLANs and Voice VLANs
131
specific VLAN, we need a different way for a PC on VLAN 10 to figure out how to reach a PC
on VLAN 20. This is what the default gateway is for. The default gateway is configured on a
Layer 3 device such as a router. It is the router’s job to intercept requests from devices on one
VLAN and send them to devices located on another VLAN. Thus the term gateway,
because it is the only way to escape the confines of a VLAN. Router gateways keep track
of other networks and store this information in a routing table. This is how inter-VLAN
routing works.
There are essentially three methods for configuring inter-VLAN routing:
Individual router links for each VLAN
VLAN trunked router link (router- on-a-stick)
Layer 3 switching
Let’s look at each of these to see how they differ and how they are configured.
For simplicity ’s sake, we assume that in each of the three configuration
scenarios we have the switch preconfigured with VLANs 10 and 20.
Individual Router Links
One way to set up inter-VLAN routing is to configure a separate router link for every
VLAN on your network. In our two-VLAN environment, the design would be set up like
Figure 4.6.
FIGURE 4.6
Individual router links
Fa0/0
Router
Fa0/1
VLAN 10
Fa0/1
VLAN 20
Fa0/2 Switch
VLAN 10
VLAN 20
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On the switch side, all that needs to be done is to ensure that the interface is configured
as an access port in the correct VLAN. On the router side, we need to enable each router
interface and configure it to be the default gateway. Let’s look at how we go about
configuring our router and switch to route VLANs 10 and 20 between themselves. Here is
the switch configuration command:
Switch#configure terminal
Switch(config)#interface fa0/1
Switch(config-if)#switchport access vlan 10
Switch(config-if)#interface fa0/2
Switch(config-if)#switchport access vlan 20
Switch(config-if)#end
Here is the router configuration command:
Router#configure terminal
Router(config)#interface fastEthernet 0/0
Router(config-if)#ip address 192.168.10.1 255.255.255.0
Router(config-if)#exit
Router(config)#interface fastEthernet 0/1
Router(config-if)#ip address 192.168.20.1 255.255.255.0
You can verify that the routes for your two networks are configured by issuing a show
ip route command on the router:
Router#show ip route
Codes: C - connected, S - static, I - IGRP, R - RIP, M - mobile, B - BGP
D - EIGRP, EX - EIGRP external, O - OSPF, IA - OSPF inter area
N1 - OSPF NSSA external type 1, N2 - OSPF NSSA external type 2
E1 - OSPF external type 1, E2 - OSPF external type 2, E - EGP
i - IS-IS, L1 - IS-IS level-1, L2 - IS-IS level-2, ia - IS-IS inter area
* - candidate default, U - per-user static route, o - ODR
P - periodic downloaded static route
Gateway of last resort is not set
C
C
192.168.10.0/24 is directly connected, FastEthernet0/0
192.168.20.0/24 is directly connected, FastEthernet0/1
Because both networks are directly connected, the router knows exactly how to reach the
networks. When a PC on VLAN 10 communicates with a PC on VLAN 20, traffic goes out
switchport Fa0/1 to router port Fa0/0. The router then looks up how to access the IP space
that VLAN occupies. The router determines that VLAN 20 traffic should be sent out router
interface Fa0/1, which sends it to switchport Fa0/2, where it ultimately reaches the PC.
Understanding and Configuring VLANs and Voice VLANs
133
If you have a very small number of VLANs on your network, the individual router link
design may work for you. However, you can see that every new VLAN requires both a
separate switchport and router port. Therefore, this type of inter-VLAN routing design
does not scale well.
Next, let’s look at how we can configure an 802.1Q trunk link between the router and
switch so we can send multiple VLANs across the same link. Using this setup, we eliminate
the need for separate ports per VLAN, and therefore this method scales much better.
VLAN Trunked Router Link
We’ve already discussed how to configure a VLAN trunk link between two Layer 2
switches. Configuring a trunked connection between a router and a switch is very similar. In
fact, the switch configuration is identical. Figure 4.7 shows how a single VLAN trunk link
can transport multiple VLANs from the switch to the router to provide inter-VLAN routing.
F I GU R E 4 .7
A VLAN trunked router link
Router
Fa0/0
Fa0/1
Trunk: VLAN 10
Switch
and VLAN 20
VLAN 10
VLAN 20
Looking at the figure, you can see how this design gets the nickname “router on a stick.”
A single link is responsible for handling inter-VLAN routing. Let’s start configuring our VLAN
trunked router link configuration by first setting up the 802.1Q trunk port on the switch:
Here’s the switch configuration:
Switch#configure terminal
Switch-A(config)#interface fa0/2
Switch-A(config-if)#switchport trunk encapsulation dot1q
Switch-A(config-if)#switchport mode trunk
Switch(config-if)#switchport trunk allowed vlan 10,20
Switch(config-if)#end
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Next, we’ll configure VLAN trunking on the router Fa0/0 interface. To accomplish this,
we need to ensure that the physical interface is enabled, by entering interface configuration
mode and issuing a no shutdown command. Then we need to create subinterfaces on
our Fa0/0 port. These subinterfaces handle traffic for a specific VLAN. You can think
of them as virtual ports off a single physical port. On these subinterfaces, you configure
the encapsulation as dot1Q (for 802.1Q trunking). This allows the router interface to
understand 802.1Q tagging. Finally, you apply your gateway IP address to the subinterface.
Here’s the router configuration:
Router#conf t
Enter configuration commands, one per line. End with CNTL/Z.
Router(config)#int fa0/0
Router(config-if)#no shutdown
Router(config-if)#exit
Router(config)#int fa0/0.10
Router(config-subif)#encapsulation dot1Q 10
Router(config-subif)#ip address 192.168.10.1 255.255.255.0
Router(config-subif)#exit
Router(config)#interface fastEthernet 0/0.20
Router(config-subif)#encapsulation dot1Q 20
Router(config-subif)#ip address 192.168.20.1 255.255.255.0
Router(config-subif)#end
To verify that our VLAN trunk is properly configured, we’ll go back to the switch we
previously configured and issue a show interface trunk command:
Switch#show interface trunk
Port
Fa0/2
Mode
on
Encapsulation
802.1Q
Status
trunking
Native vlan
1
Port
Fa0/2
Vlans allowed on trunk
10,20
Port
Fa0/2
Vlans allowed and active in management domain
10,20
Port
Fa0/2
Switch#
Vlans in spanning tree forwarding state and not pruned
10,20
Understanding and Configuring VLANs and Voice VLANs
135
The VLAN trunk is properly configured on the switch and is ready to send both VLAN
10 and 20 traffic destined to the default gateway on switchport Fa0/2.
The limit to the number of subinterfaces configured on a physical router port varies
from device to device, but this number is in the hundreds for newer Cisco routers. Keep
in mind that all of these subinterfaces share the bandwidth of the single link, so it is
important not to oversubscribe the link. For example, if we trunk VLAN 10 and 20 on
a 100Mbps FastEthernet connection, they both have to share that 100 Mbps. The more
VLANs you add, the more traffic is sent over the link, and ultimately the more congested
the trunk links become.
To verify our routing, we’ll connect to the router and issue the show ip route
command:
Router#show ip route
Codes: C - connected, S - static, I - IGRP, R - RIP, M - mobile, B - BGP
D - EIGRP, EX - EIGRP external, O - OSPF, IA - OSPF inter area
N1 - OSPF NSSA external type 1, N2 - OSPF NSSA external type 2
E1 - OSPF external type 1, E2 - OSPF external type 2, E - EGP
i - IS-IS, L1 - IS-IS level-1, L2 - IS-IS level-2, ia - IS-IS inter area
* - candidate default, U - per-user static route, o - ODR
P - periodic downloaded static route
Gateway of last resort is not set
C
192.168.10.0/24 is directly connected, FastEthernet0/0.10
C
192.168.20.0/24 is directly connected, FastEthernet0/0.20
Router#
You’ll notice that this routing table output looks very similar to the routing table for
the individual router link. The main difference is that the router sees the IP networks as
being on the same physical interface, Fa0/0, but it distinguishes the 192.168.10.0/24 subnet
as belonging to subinterface Fa0/0.10 and the 192.168.20.0/24 network as belonging to
Fa0/0.20. The router tags the packet with the proper destination 802.1Q VLAN identifier,
and the switch is responsible for removing the tag and sending it to the correct destination
switchport.
The VLAN trunked router link design is far more scalable than the individual router
link design. You will fi nd this setup in many small to medium-size businesses. If you have
a fairly large Ethernet LAN and need the ability to add many VLANs on your network
without being concerned about congestion on trunked router links, you can implement
Layer 3 switching using special Cisco multilayer switches.
Layer 3 Switching
Layer 3 switching essentially takes the concept of a VLAN trunked router link between
a Layer 2 switch and a router and combines the Layer 2 switch and router into a single
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device. A Layer 3 switch (also known as a multilayer switch) has the ability to create Layer
3 VLAN interfaces on the switch hardware. Now, instead of having to traverse a VLAN
trunked link to reach a default gateway, which is rerouted directly back out the same link
to the switch, this process is performed on the switch using dedicated hardware modules
called Application-Specific Integrated Circuits (ASICs). The traffic is then switched on a
Layer 3 backplane, which can typically handle several gigabits of bandwidth or more.
This allows the administrator to create many more VLANs on a network without the
worry of overutilizing a trunk link. It is also the preferred method for administrators
because they need only support one Layer 3 switch instead of a Layer 2 switch and a
router. Figure 4.8 shows how a Layer 3 switch is responsible for both the switching and
routing functions.
FIGURE 4.8
A Layer 3 switch
VLAN 10 GW:192.168.10.1
VLAN 20 GW:192.168.20.1
Si
Layer 3
Switch
VLAN 10
VLAN 20
To configure Layer 3 switching, you need to ensure that your switch is capable of
running in Layer 3 mode. By default, Layer 3 switching is disabled. To enable it, you must
issue the ip routing configuration command. Next, you configure VLAN interfaces for
each VLAN and assign the subnet gateway IP address to the virtual interface. Finally, you
issue a no shutdown command to enable the virtual interface. Let’s go ahead and start the
configuration:
L3-Switch#configure terminal
L3-Switch(config)#ip routing
L3-Switch(config)#interface vlan 10
L3-Switch(config-if)#ip address 192.168.10.1 255.255.255.0
L3-Switch(config-if)#no shutdown
Understanding and Configuring VLANs and Voice VLANs
137
L3-Switch(config-if)#exit
L3-Switch(config)#interface vlan 20
L3-Switch(config-if)#ip address 192.168.20.1 255.255.255.0
L3-Switch(config-if)#no shutdown
L3-Switch(config-if)#end
Now that we’ve configured our VLAN interfaces, we can view them just as if they were
physical Ethernet ports. Let ’s look at the VLAN 10 and VLAN 20 virtual interfaces by
issuing a show interface vlan command:
L3-Switch#show interface vlan 10
Vlan10 is up, line protocol is up
Hardware is EtherSVI, address is 000b.465e.5600 (bia 000b.465e.5600)
Internet address is 192.168.10.1/24
MTU 1500 bytes, BW 1000000 Kbit, DLY 10 usec,
reliability 255/255, txload 1/255, rxload 1/255
Encapsulation ARPA, loopback not set
ARP type: ARPA, ARP Timeout 04:00:00
Last input never, output never, output hang never
Last clearing of “show interface” counters never
Input queue: 0/75/0/0 (size/max/drops/flushes); Total output drops: 0
Queueing strategy: fifo
Output queue: 0/40 (size/max)
5 minute input rate 0 bits/sec, 0 packets/sec
5 minute output rate 0 bits/sec, 0 packets/sec
0 packets input, 0 bytes, 0 no buffer
Received 0 broadcasts (0 IP multicast)
0 runts, 0 giants, 0 throttles
0 input errors, 0 CRC, 0 frame, 0 overrun, 0 ignored
0 packets output, 0 bytes, 0 underruns
0 output errors, 0 interface resets
0 output buffer failures, 0 output buffers swapped out
L3-Switch#show interface vlan 20
Vlan10 is up, line protocol is up
Hardware is EtherSVI, address is 000b.465e.5800 (bia 000b.465e.5800)
Internet address is 192.168.20.1/24
MTU 1500 bytes, BW 1000000 Kbit, DLY 10 usec,
reliability 255/255, txload 1/255, rxload 1/255
Encapsulation ARPA, loopback not set
ARP type: ARPA, ARP Timeout 04:00:00
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Last input never, output never, output hang never
Last clearing of “show interface” counters never
Input queue: 0/75/0/0 (size/max/drops/flushes); Total output drops: 0
Queueing strategy: fifo
Output queue: 0/40 (size/max)
5 minute input rate 0 bits/sec, 0 packets/sec
5 minute output rate 0 bits/sec, 0 packets/sec
0 packets input, 0 bytes, 0 no buffer
Received 0 broadcasts (0 IP multicast)
0 runts, 0 giants, 0 throttles
0 input errors, 0 CRC, 0 frame, 0 overrun, 0 ignored
0 packets output, 0 bytes, 0 underruns
0 output errors, 0 interface resets
0 output buffer failures, 0 output buffers swapped out
You can see that the VLAN 10 and 20 interfaces are up and the hardware states that it is
EtherSVI. SVI stands for switch virtual interface. Essentially, it’s a virtual interface created
in software but switched in hardware using special ASICs.
Keep in mind that the VLAN interface will be in a “down” state until you
have a device on that VLAN. If you are having problems configuring the
VLAN and cannot get it to an “up” state, configure a switch access port
to belong to that particular VLAN and attach a PC to the switchport. The
VLAN will then move from being down to up. You may alternately have an
enabled trunk port carrying that VLAN between switches to achieve the
same up state.
You now have the skills to create VLANs and inter-VLAN routing. In larger networks
with many switches and VLANs, configuration of VLANs can become an administrative
nuisance. The next section will discuss VTP, a tool that you can use to help reduce the
VLAN administration burden on your network.
Using the VLAN Trunking Protocol
Every switch on your network maintains a VLAN database for every VLAN it knows
about. If your VLANs span multiple switches, it can become burdensome to configure
the VLAN so that it is visible in every VLAN database switch. The VLAN trunking
protocol (VTP) is a way to add, delete, and modify VLANs on a single switch and have
that VLAN information propagate into the VLAN database on other switches within
your network.
Understanding and Configuring VLANs and Voice VLANs
139
To completely delete previously configured VLANs that are in the VLAN
database of a switch, you can issue the following global command:
Switch#delete vlan.dat
The vlan.dat file is where the VLAN database is stored on the switch. You
can then reload the switch, and all the previously configured VLANs are no
longer there.
You need to understand several concepts in VTP before using the protocol. In fact, you
need to be very careful with VTP because it is possible to accidentally delete VLANs when
adding new switches to the network if you aren’t careful! I’ll fi rst tell you about the three
VTP modes and when they should be used. Next I’ll discuss VTP revision numbers and
what they are used for. Then I’ll review the most common VTP configuration options used
in production networks.
Choosing a VTP Mode
You can configure a switch to be in one of three VTP modes: server, client, or transparent.
Each mode serves a different purpose, and you should take care to configure each switch
accordingly if you are planning to use VTP on your network. Here’s a summary of the
three VTP modes you can choose from.
VTP Server
The VTP server mode allows an administrator to add, delete, and modify VLANs on the
network over trunked links. All changes are propagated to other switches within the VTP
domain. VTP server is the default VTP mode on all Cisco switches.
VTP Client
VTP client mode listens to the VTP server and copies its VLAN settings to its own VLAN
database. It also forwards the VTP update messages from the server to other switches
within the same VTP domain on its trunked links. This mode does not allow you to add,
delete, or modify VLANs.
VTP Transparent
VTP transparent mode basically disables VTP on the switch. You can add, delete, and
modify VLAN information on the switch, but it never propagates this information to
any other switch. Also, if the switch in this mode receives VTP update messages from
other switches it is trunked with, it ignores the updates locally but will pass them on to
connected neighbors.
Understanding VTP Revision Numbers
VTP uses the concept of revision numbers to help ensure that the most recent VLAN
database changes are propagated to all other switches within the same VTP domain. The
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higher the revision number, the more trusted the information, and therefore that is the
VLAN information that is updated to every other switch.
Each time a VLAN change is made on the switch (adding, deleting, or updating), the
revision number increments. This revision number is then incremented on all switches
within the VTP domain to ensure consistency.
You must be very careful when adding a switch to a network that is
configured for VTP server mode and uses the same VTP domain and/
or password. If you have a switch that you ’ve been tinkering around
with in the lab setting up various test VLANs, each VLAN add/delete
will increment the VLAN revision number. If the newly added switch’s
revision number ends up being higher than the revision number of the
current production switches, VTP will assume that the new switch has
more recent VLAN information and will overwrite all of your VLANs
with the information contained on the new switch. To ensure that this
does not happen, you should always reboot your switch before adding
it to the network. Changing the VTP mode to transparent and then to
either client or server will reset the revision number back to zero. One
other tip is to use different VTP domain names and passwords on your
lab equipment. Remember that the switches must be part of the same
domain and share the same password in order to add, delete, modify,
and forward VTP messages.
Configuring VTP on Your Network
As mentioned earlier, VTP is in server mode by default. If you plan to use VTP on your
network, there is no need to have every switch on the network configured for server mode.
Typically, your distribution block switches are set for server mode, and the access layer
switches are set to run in client mode. In our example, we’ll configure Switch A to be in
VTP server mode, and it will send updates to Server B configured as a VTP client.
Figure 4.9 depicts our network layout, which assumes that both switches have been
preconfigured for 802.1Q trunking and have VLAN 10 and 20 already set up.
FIGURE 4.9
An example of VTP configuration
VTP Server
802. 1Q Trunk
VTP Client
Switch-A
Switch-B
VTP Revision: 5
VTP Revision: 5
VLAN
Database:
VLAN
Database:
VLAN 10
VLAN 20
VLAN 10
VLAN 20
Understanding and Configuring VLANs and Voice VLANs
141
There are two versions of VTP. Both versions essentially perform the same functions on
Ethernet networks, so it really doesn’t matter which one you choose as long as everything
in your network is running the same version. Typically, I recommend running VTP
version 2 everywhere to ensure consistency.
The other vital configuration step is to configure the VTP domain. The VTP domain
is the VTP group that a switch belongs to. Once a switch is in a specific domain, it listens
only to updates within this group. All other VTP update messages are ignored. If the
switch has never been configured, the VTP domain will be blank. Because the switch is
in VTP server mode by default, it will join the fi rst VTP domain from which it receives a
VTP update message. The only way to change the VTP name once it is set is to change it
manually while directly connected to the switch.
A third VTP configuration parameter that is optional but highly recommended is the
VTP password. VTP passwords ensure that no unauthorized switches can be added to
the network and no VLAN information is changed.
Now that you understand the VTP configuration options, let’s configure our two
switches with the following VTP setup:
Switch A:
VTP Mode: Server
VTP Version: 2
VTP Domain: Sybex
VTP Password: CCNAVoice
Switch B:
VTP Mode: Client
VTP Version: 2
VTP Domain: Sybex
VTP Password: CCNAVoice
Make sure you remember that both the VTP domain and VTP password
you assign to the switches are case sensitive!
First let’s issue the show vtp status command on Switch A so you can see how VTP is
set up by default on all Cisco switches:
Switch-A#show vtp status
VTP Version
Configuration Revision
Maximum VLANs supported locally
Number of existing VLANs
VTP Operating Mode
:
:
:
:
:
2
5
1005
7
Server
Chapter 4
142
VTP
VTP
VTP
VTP
MD5
Configuring the Network Infrastructure for Voice
Domain Name
Pruning Mode
V2 Mode
Traps Generation
digest
:
:
:
:
:
Disabled
Disabled
Disabled
0xA7 0x2B 0x66 0xB2 0x7C 0x0A 0xC7 0x3C
Now we’ll configure VTP on Switch A:
Switch-A(config)#vtp
Switch-A(config)#vtp
Switch-A(config)#vtp
Switch-A(config)#vtp
mode server
domain Sybex
version 2
password CCNAVoice
Another show vtp status command will verify that the changes were made correctly.
We’ll also run a show vtp password command to ensure that we have our VTP password set:
Switch-A#show vtp status
VTP Version
Configuration Revision
Maximum VLANs supported locally
Number of existing VLANs
VTP Operating Mode
VTP Domain Name
VTP Pruning Mode
VTP V2 Mode
VTP Traps Generation
MD5 digest
:
:
:
:
:
:
:
:
:
:
2
5
1005
7
Server
Sybex
Disabled
Enabled
Disabled
0x26 0x3A 0xA0 0x8C 0x66 0x01 0xA1 0xF
Switch-A#show vtp password
VTP Password: CCNAVoice
Now that we have our VTP server properly set up on our network, let’s configure Switch
B as our VTP client:
Switch-B(config)#vtp
Switch-B(config)#vtp
Switch-B(config)#vtp
Switch-B(config)#vtp
mode client
domain Sybex
version 2
password CCNAVoice
Running show VTP status and show vtp password shows us that the switch was set to
VTP client and has all the proper configuration settings. Also notice that the Configuration
Understanding and Configuring VLANs and Voice VLANs
143
Revision number is 5, which is the same as the VTP server that it received VTP update
messages from:
Switch-B#show vtp status
VTP Version
Configuration Revision
Maximum VLANs supported locally
Number of existing VLANs
VTP Operating Mode
VTP Domain Name
VTP Pruning Mode
VTP V2 Mode
VTP Traps Generation
MD5 digest
:
:
:
:
:
:
:
:
:
:
2
5
1005
7
Client
Sybex
Disabled
Enabled
Disabled
0x26 0x3A 0xA0 0x8C 0x66 0x01 0xA1 0xF
Switch-B#show vtp password
VTP Password: CCNAVoice
Back on Switch A, we’ll configure a new VLAN for our Management group. The new
VLAN number is 30:
Switch-A#conf t
Switch-A(config)#vlan 30
Switch-A(config-vlan)#name Management
Switch-A(config-vlan)#end
This information should propagate over to Switch B. Figure 4.10 diagrams the process
by which the VTP Server updates the VTP Client switch.
F I G U R E 4 .1 0
The VTP update process
VTP Server
802.1Q Trunk
Switch-A
VTP Revision: 6
VTP Client
Switch-B
VTP Revision: 6
VTP Revision: 5
VLAN
Database:
VLAN
Database:
VLAN 10
VLAN 20
VLAN 10
VLAN 20
VLAN 30
VLAN 30
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Configuring the Network Infrastructure for Voice
Let’s take a look at the process by running the show vtp status and show vlan brief
commands:
Switch-B#show vtp status
VTP Version
Configuration Revision
Maximum VLANs supported locally
Number of existing VLANs
VTP Operating Mode
VTP Domain Name
VTP Pruning Mode
VTP V2 Mode
VTP Traps Generation
MD5 digest
:
:
:
:
:
:
:
:
:
:
2
6
1005
8
Client
Sybex
Disabled
Enabled
Disabled
0xAA 0x10 0xFC 0x76 0xFD 0xD5 0xB3 0x4C
Notice that the Configuration Revision number has incremented, and now the number
of existing VLANs is 8 instead of 7.
Switch-B#show vlan brief
VLAN
---1
10
Name
-------------------------------default
Sales
Status
--------active
active
20
Marketing
active
30
1002
1003
1004
1005
Management
fddi-default
token-ring-default
fddinet-default
trnet-default
active
act/unsup
act/unsup
act/unsup
act/unsup
Ports
------------------------------Gi0/1, Gi0/2
Fa0/1, Fa0/2, Fa0/3, Fa0/4
Fa0/5, Fa0/6, Fa0/7, Fa0/8
Fa0/9, Fa0/10, Fa0/11, Fa0/12
Fa0/13, Fa0/14, Fa0/15, Fa0/16
Fa0/17, Fa0/18, Fa0/19, Fa0/20
Fa0/21, Fa0/22, Fa0/23, Fa0/24
Fa0/25, Fa0/26, Fa0/27, Fa0/28
Fa0/29, Fa0/30, Fa0/31, Fa0/32
Fa0/33, Fa0/34, Fa0/35, Fa0/36
Fa0/37, Fa0/38, Fa0/39, Fa0/40
Fa0/41, Fa0/42, Fa0/43, Fa0/44
Fa0/45, Fa0/46, Fa0/47, Fa0/48
Sure enough, Switch B has the new VLAN 30 configured in its VLAN database!
Understanding and Configuring VLANs and Voice VLANs
145
Configuring and Verifying Voice VLANs
When it comes to configuring a separate VLAN for voice traffic, the process of
configuration and VLAN creation is exactly the same whether traffic is going router-to switch or switch-to -switch. The voice VLAN configuration differs when you want to use
Cisco IP phones that incorporate a data port on the phone for PC connections. Many Cisco
mid- and high-range phones such as the 7945G give users the ability to plug a PC into an
Ethernet port on the phone to provide network connectivity. The phone essentially becomes
a three-port switch at that point. One port connects the phone to the access-layer switch,
the second (virtual) port is for voice traffic to the phone, and the third port is to connect to
a PC for standard data transport. Figure 4.11 shows how a PC is plugged directly into the
phone, which is essentially trunked with both a voice and data VLAN.
F I G U R E 4 .11
A Cisco IP phone switch
Cisco
Phone
Switch
Fa0/5
PC
Trunk Link
Voice VLAN
Data VLAN
As you can see, the connection between the switch and the Cisco phone is an 802.1Q trunk
link. It is necessary to have a VLAN trunk because we have our voice and data separated on
two different VLANs. When configuring the VLAN trunk on the switchport that connects
to the phone, we use a slightly different method. The Cisco IOS has a unique command to
identify a VLAN as a voice VLAN. The command is switchport voice vlan. Even though
the switchport command doesn’t specifically reference 802.1a, in all actuality, this trunk link
between our switch and the Cisco phone is not a full-fledged 802.1Q trunk like those we have
practiced configuring between two switches and a switch and router. Instead, the Cisco switch
and Cisco IP phone use CDP to implement this quasi-trunk. The VLAN that is configured as
the voice VLAN is marked with an 802.1Q tag, while the data VLAN is considered to be
the native VLAN and is left unmarked. This VLAN trunk is capable of handling only two
VLANs— one tagged VLAN for voice and one untagged VLAN for data.
It used to be that the VLAN trunk link between the access switch and Cisco
IPT phone was indeed a full-blown 802.1Q trunk. Unfortunately, it was easy
to fool this setup, and PCs could easily join the voice VLAN and use sniffers
to collect and re - create voice calls. Because the new quasi-trunk setup uses
CDP to identify which devices can join the voice VLAN, the new method is
much more secure.
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With that understanding of the voice VLAN, let’s add a new voice VLAN (VLAN 100)
and configure port Fa0/5 to trunk the voice VLAN (using the switchport voice vlan
command) and the Sales VLAN (data VLAN that is already configured on the switch) to
the Cisco phone according to Figure 4.11. First we’ll configure the voice VLAN:
Switch#configure terminal
Switch(config)#vlan 100
Switch(config-vlan)#name Voice
Switch(config-vlan)#end
Next let’s configure the switchport to quasi-trunk VLAN 100 for voice and VLAN 10
for or data transport:
Switch#configure terminal
Switch(config)#interface fa0/5
Switch(config-if)#switchport voice vlan 100
Switch(config-if)#switchport access vlan 10
Switch(config-if)#end
Great! Now we’ll run show vlan brief to verify that our port Fa0/5 is in both VLAN
10 and VLAN 100:
Switch#sh vlan brief
VLAN
---1
10
Name
-------------------------------default
Sales
Status
--------active
active
Ports
------------------------------Gi0/1, Gi0/2
Fa0/1, Fa0/2, Fa0/3, Fa0/4
Fa0/5, Fa0/6, Fa0/7, Fa0/8
Fa0/9, Fa0/10, Fa0/11, Fa0/12
Fa0/13, Fa0/14, Fa0/15, Fa0/16
Fa0/17, Fa0/18, Fa0/19, Fa0/20
Fa0/21, Fa0/22, Fa0/23, Fa0/24
Fa0/25, Fa0/26, Fa0/27, Fa0/28
Fa0/29, Fa0/30, Fa0/31, Fa0/32
Fa0/33, Fa0/34, Fa0/35, Fa0/36
Fa0/37, Fa0/38, Fa0/39, Fa0/40
Fa0/41, Fa0/42, Fa0/43, Fa0/44
Fa0/45, Fa0/46, Fa0/47, Fa0/48
20
Marketing
active
30
100
1002
1003
Management
Voice
fddi-default
trcrf-default
active
active
Fa0/5
act/unsup
act/unsup
Introduction to Quality of Service (QoS)
1004 fddinet-default
1005 trbrf-default
Switch#
147
act/unsup
act/unsup
Sure enough, port Fa0/5 belongs to both the Sales (VLAN 10) and voice (VLAN 100)
VLANs. Next we’ll discuss how we can mark and prioritize the voice traffic using the most
common QoS techniques.
Introduction to Quality of Service (QoS)
Quality of Service (QoS) is such an enormous topic in the world of networking that I could
fi ll an entire book dedicated solely to it. In fact, the Cisco CCVP certification dedicates the
642- 642 exam to QoS topics. Not only is there a great deal to discuss about QoS, many of
the details of its mechanisms are very complex and difficult to grasp when you fi rst start
learning about them. The CCNA Voice track requires you to know the very basics of QoS
in order to get some exposure to the topic. Fortunately for us, Cisco is making the actual
implementation of QoS easier all the time with the concept of auto - QoS. We will start by
discussing why we need QoS and the network requirements for voice packets. Then I ’ll
explain the concept of QoS trust boundaries and where they should be located. Finally,
I’ll introduce you to auto - QoS and how it simplifies much of the complexity for proper
implementation on a network.
Quality of Service (QoS) is the ability to identify time-sensitive traffic and give it priority
over other forms of traffic. IP networks have restrictions on both the amount of bandwidth
between two points and the latency, or time it takes for a packet to be moved between two
points. From a pure bandwidth point of view, the goal is to eliminate or at least reduce
bottlenecks on your network. Bottlenecks are network links interconnecting endpoints where
the amount of data sent out an interface exceeds the physical capabilities of the interface.
Figure 4.12 shows the network location where a bottleneck is most likely to occur.
F I G U R E 4 .1 2
A network bottleneck
1-000 Mbps
1-000 Mbps
Possible
Bottleneck
100 Mbps
1-000 Mbps
1-000 Mbps
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When we talk about network latency, we’re mainly talking about network delay and
network jitter. Some amount of network delay is always going to be present on even the
fastest networks. This is called fi xed delay, which means the time it takes to send an
electrical or optical signal across a certain distance. This type of delay is very slight and
does not affect time-sensitive traffic such as voice.
Variable delay, on the other hand, is what QoS attempts to eliminate or at least reduce.
Variable delay refers to those bottleneck situations when your time-sensitive traffic has to
sit in a queue and wait for other packets to be sent out of the interface before your voice
packet can be sent. By incorporating QoS, we can give our voice packets priority over
traffic that is not time-sensitive so that we won’t be waiting in the queue and will thereby
reduce variable delay.
Jitter refers to the variations in time of arrival for time-sensitive packets. If it takes
40 ms (milliseconds) for the fi rst voice packet to arrive and 90 ms for the second packet to
arrive, the jitter level is 50 ms.
One last network issue to watch out for with time-sensitive data is packet loss. If
bottlenecks get to the point where the queues start fi lling up, packet loss occurs. If this
happens, you can implement QoS to begin discarding less- critical data, which can be
identified using QoS classification and marking methods.
Specific network requirements for QoS for voice and video have been established for you
to use as guidelines. As long as you meet or beat the following criteria, your voice/video
applications should not experience any problems.
End-to - end delay: 150 ms or less
Jitter: 30 ms or less
Packet loss: 1 percent loss or less
So the goal for us is to implement QoS in order to provide a much more consistent and
steady transport mechanism for our voice packets. While our best- effort design may work
well for data, voice traffic requires a bit more care to function optimally. Now that we
know what we’re trying to accomplish with QoS, let’s turn our attention to how it works.
The QoS function has three stages, which we’ll look at in turn:
1.
Traffic classification
2.
Traffic marking
3.
Traffic queuing
Traffic Classification
Traffic classification is the process of identifying time-sensitive packets. The identification
process must be performed fi rst because the equipment must be able to clearly identify
certain traffic. Creating voice VLANS makes it easy to identify voice traffic because we can
assume that any packets on the voice VLAN should be classified as such.
Introduction to Quality of Service (QoS)
149
Traffic Marking
Traffic marking is the process of flagging critical packets so the rest of the network can
properly identify them and give them priority over all other traffic. Cisco phones have the
ability to mark voice packets with a Class of Service (CoS) value. The CoS is a field within
the Layer 2 Ethernet frame header that marks traffic as being one of eight (0 to 7) classes.
The higher the CoS value, the more priority is given. By default, voice traffic is marked
with a classification of 5. If data is not marked with a CoS, it is given a value of 0. The CoS
is used by Layer 2 switches for proper queuing.
The Cisco phone also marks the IP packet with a Type of Service (ToS) identifier. The
ToS essentially does the same thing as the CoS but is intended to be used by Layer 3 devices
such as routers and switches.
Traffic Queuing
Traffic queuing is the process of ordering certain types of traffic for transport over
LAN/WAN interfaces. Many different queuing techniques are available, which can be
overwhelming. Fortunately, one queuing technique is considered optimal for voice traffic,
Low Latency Queuing (LLQ). LLQ does the best job of eliminating variable delay, jitter,
and packet loss on a network. LLQ on a switch creates a strict-priority queue for voice
traffic. The auto - QoS configuration method utilizes LLQ as its default mechanism.
Now let’s look at how and where we can classify and mark traffic using the high-level
design concept of QoS trust boundaries.
Identifying QoS Trust Boundaries
We can classify, mark, and begin enforcing queuing strategies for IP traffic at several
points along a network. But where should this process begin? The simplest answer is
to push your trust boundary out as far to the endpoint as possible. But depending on
the type of network, you may have to pull the boundary in a bit depending on how
much you trust the end devices (that ’s why it ’s called a “trust ” boundary!). If you
have full control of endpoints, then you control the CoS and ToS markings that are
generated, and you can push the trust boundary out to the phone and even PC level.
If you do not have as much control over your network, it might be better to begin
marking CoS/ToS values as soon as the traffic hits your switch. Also, you may run into
a situation where your access -layer switches cannot be configured for QoS. Because
of this, you have no choice but to configure the trust boundary at the distribution
layer. Figure 4.13 displays where trust boundaries can be implemented within a typical
network.
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Trust boundaries
Trust Any
Endpoint
Trust Cisco
Phones
Trust at
Access Layer
Trust at
Distribution Layer
Si
Most organizations will trust CoS/ToS markings from the Cisco phones but will not
trust the markings from devices attached to the phone such as a PC. When network data
from the PC reaches the Cisco phone, the switch will ignore the CoS/ToS markings and
consider all data packets to have a value of 0.
Now that you understand the basics of QoS and trust boundaries, let ’s see how we can
easily implement them on our network using auto - QoS.
Auto-QoS Implementation Options
There are only three options for configuring auto - QoS on an interface using the auto
qos voip command. Once you understand these three options, configuring QoS on your
network will be a snap! Here is the output of the switch when configuring auto - QoS:
Switch(config-if)#auto qos voip ?
cisco-phone
Trust the QoS marking of Cisco IP Phone
cisco-softphone Trust the QoS marking of Cisco IP SoftPhone
trust
Trust the DSCP/CoS marking
Let’s look at the options to understand when each one should be used.
You should use this option when you want to trust the QoS markings from
your Cisco phone. Note that I said “Cisco” phone and not “IP” phone. Cisco uses CDP
between the switch and phone to ensure that the device is indeed a phone and not some
other device attempting to get a better classification for its traffic. Because CDP is Cisco
proprietary, it works only when Cisco phones are connected or when other companies
license CDP technology (such as Mitel IP Phones).
cisco-phone
This option is very similar to the cisco-phone option except it trusts
the CoS/ToS markings on PCs that are running the Cisco IP Communicator software.
The IP Communicator software runs CDP once again to ensure that the device is properly
identified as a Cisco phone.
cisco-softphone
Introduction to Quality of Service (QoS)
151
The trust option basically means that the switch will trust any CoS/ToS value
received and treat the traffic accordingly. Be cautious when configuring this on access
ports, because people “in-the-know” can manipulate the classification markings of data
traffic on their PCs and have their data sent as priority traffic when it should be treated as
normal traffic. But where the trust option should absolutely be used is between all of the
switch and router interfaces that interconnect your network equipment. As soon as you set
a location for your trust boundary, all other devices within that boundary can safely trust
the CoS/ToS markings they receive.
trust
That’s all there is to auto- QoS. Let’s use Figure 4.14 as our network example
for configuring QoS on a production network. Assume that the Sales, Marketing,
Management, and Voice VLANs are preconfigured on the network. Switchport Fa0/5 is
configured to use VLAN 10 for data and VLAN 100 for voice traffic. Also assume that
802.1Q trunking is configured between the switch and the CME router.
F I G U R E 4 .1 4
An example of QoS
Trust Boundary
Fa0/5
Fa0/1
Fa0/1
Fa0/5
Trunk Link
Switch-A
Switch-B
CME
First, we need to set our trust boundary. Let’s assume that we’ll trust the Cisco phones
but not trust ordinary PCs. Therefore, our trust boundary is set at the phone, using the
auto qos voip cicso-phone command:
Switch-A#confure terminal
Switch-A(config)#interface fastEthernet 0/5
Switch-A(config-if)#auto qos voip cisco-phone
Switch-A(config-if)#end
Let’s see exactly what auto - QoS is configured on our port, using the show run
interface command:
Switch-A#sh run int fa0/5
Building configuration...
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Current configuration : 487 bytes
!
interface FastEthernet0/5
switchport access vlan 10
switchport mode dynamic desirable
switchport voice vlan 100
mls qos trust device cisco-phone
mls qos trust cos
auto qos voip cisco-phone
wrr-queue bandwidth 10 20 70 1
wrr-queue min-reserve 1 5
wrr-queue min-reserve 2 6
wrr-queue min-reserve 3 7
wrr-queue min-reserve 4 8
wrr-queue cos-map 1 0 1
wrr-queue cos-map 2 2 4
wrr-queue cos-map 3 3 6 7
wrr-queue cos-map 4 5
priority-queue out
spanning-tree portfast
We can see that the auto qos voip command actually configured all kinds of things
on the interface! The important thing we need to identify is that we’re trusting the Cisco
phone with the auto qos voip cisco-phone entry.
Once the trust boundary is set, we know that the interfaces connecting our Layer 2
switch to the CME router should be configured using the auto qos voip trust command.
Here’s the switch trunk port configuration for Fa0/1:
Switch-A#confure terminal
Switch-A(config)#interface fastEthernet 0/1
Switch-A(config-if)#auto qos voip trust
Switch-A(config-if)#end
Let’s look at our running configuration for our switch uplink to see the differences
between the auto qos voip trust configurations and the auto qos voip cisco-phone
output:
Switch-A#show run interface fa0/1
Building configuration...
Current configuration : 436 bytes
!
interface FastEthernet0/1
Configuring Other Link Efficiency Techniques
153
switchport trunk encapsulation dot1q
switchport trunk allowed vlan 10,20,100
switchport mode trunk
mls qos trust cos
auto qos voip trust
wrr-queue bandwidth 10 20 70 1
wrr-queue min-reserve 1 5
wrr-queue min-reserve 2 6
wrr-queue min-reserve 3 7
wrr-queue min-reserve 4 8
wrr-queue cos-map 1 0 1
wrr-queue cos-map 2 2 4
wrr-queue cos-map 3 3 6 7
wrr-queue cos-map 4 5
priority-queue out
Notice that from a QoS configuration standpoint, the only difference between the trust
and cisco-phone configuration is the auto qos voip trust command.
The configuration of the opposite- end switch is identical. Once you’ve completed
configuring all the interfaces, congratulations; you’ve successfully implemented QoS for
voice on your network!
Configuring Other Link
Efficiency Techniques
In addition to configuring QoS on your network for voice support, you can use two other
link efficiency techniques to help with the consistent transport of VoIP. These techniques
are compression and link fragmentation and interleaving (LFI).
Compression Techniques
Compression can come in many forms. I’ve already touched on how to use different voice
codecs to compress the audio payload. I will cover how to implement various codecs on
your CME in the next chapter. I ’ve also mentioned the concept of using RTP compression
(cRTP) across WAN links to help ease any congestion issues as the result of the WAN link
being a bottleneck. Here is how to configure cRTP on a T1 PPP serial interface, which is
a common place to implement this type of compression. Figure 4.15 shows the network
layout for our example.
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An example of compression
T1 PPP
S1/0
S1/0
Router-A
Router-B
cRTP Compression
Configuration Steps
Configuring basic cRTP requires only one step, defi ning the ip rtp header-compression
ietf-format command on the link. Of course, the link also needs a common serial link
encapsulation type and IP addressing scheme for IP transport to work properly. Here are
the basic steps for configuring your serial links for cRTP.
The Router-A configuration looks like this:
Router-A(config)# interface serial 1/0
Router-A(config-if)# encapsulation ppp
Router-A(config-if)#ip add 10.1.1.1 255.255.255.252
Router-A(config-if)# ip rtp header-compression ietf-format
Router-A(config-if)# end
The Router-B configuration looks like this:
Router-B(config)# interface serial 1/0
Router-B(config-if)# encapsulation ppp
Router-B(config-if)#ip add 10.1.1.1 255.255.255.252
Router-B(config-if)# ip rtp header-compression ietf-format
Router-B(config-if)# end
We can now verify that cRTP is enabled and compressing RTP headers by issuing
the sho ip rtp header-compression command. Here’s the output of this command on
Router-A:
Router-A#show ip rtp header-compression
RTP/UDP/IP header compression statistics:
Interface Serial1/0 (compression on, IETF)
Rcvd:
1473 total, 1452 compressed, 0 errors, 0 status msgs
0 dropped, 0 buffer copies, 0 buffer failures
Sent:
1234 total, 1216 compressed, 0 status msgs, 379 not predicted
41995 bytes saved, 24755 bytes sent
2.69 efficiency improvement factor
Network Infrastructure Services for VoIP support
155
Connect: 16 rx slots, 16 tx slots,
6 misses, 0 collisions, 0 negative cache hits, 13 free contexts
99% hit ratio, five minute miss rate 0 misses/sec, 0 max
Link Fragmentation Interleaving (LFI)
A second link- efficiency technique that is commonly used on PPP multilink circuits is called
link fragmentation interleaving (LFI). This process takes much larger data packets and
fragments them into smaller, more manageable sizes. It then is able to send voice packets in
between the newly fragmented data packets. This process ensures that voice packets have
a more consistent variable delay and significantly cuts down on voice jitter. You configure
LFI over a PPP multilink by using the ppp multilink command to enable multilink PPP
and then the ppp multilink interleave command to enable LFI.
Network Infrastructure Services
for VoIP support
The network infrastructure equipment can also provide supplementary services to assist in
the support of a Cisco Unified Communications solution. Routers or Layer 3 switches can
provide Dynamic Host Control Protocol (DHCP) services to your phones to dynamically
assign IP addresses and other network information to the phones. The infrastructure can
also serve as a centralized point for synchronizing your UC equipment clocks by being the
Network Time Protocol point of reference. Let ’s look at how we configure both of these
network services for our VoIP solution.
Configuring DHCP for Voice Functionality
DHCP allows an endpoint device (such as a Cisco IP phone) to boot up on the network
and request network information, which it dynamically receives from a DHCP server. This
section shows how to configure DHCP on your CME router for your end devices.
DHCP server functionality is considered a service on your IOS router. It is disabled by
default. To enable the DHCP service you use the following configuration command:
Router(config)# service dhcp
The next step in your DHCP server- configuration process is to ensure that specific
IP addresses on your network are never handed out to endpoints. You must specifically
exclude IPs such as default gateways and other static interfaces that are already in use. If
you skip this step, you run the risk of having an endpoint assigned an IP address that is
already in use, which causes an IP confl ict. IP confl icts are a very bad thing because they
disrupt proper IP transport on your network for the devices with the confl ict. You can
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Configuring the Network Infrastructure for Voice
configure a range of IP addresses to exclude with a single command. Typically, I configure
all of my network interface and other critical IP address assignments on lower IP space.
Therefore, if I’m creating a DHCP pool using a 255.255.255.0 subnet mask, I will exclude
the fi rst 20 IP addresses within that range. For example, suppose I want to configure a
DHCP pool for the 192.168.100.0/24 range. My excluded command would then look
like this:
Router(config)# ip dhcp excluded-address 192.168.100.1 192.168.100.20
Next, you want to create the DHCP pool for your network space. The fi rst part in
creating the pool is to name it. Because we want to create a pool for our IP phones, we’ll
name the pool voip-pool. Here is the command to complete this step:
Router(config)# ip dhcp pool voip-pool
As soon you name your DHCP pool, you are placed into dhcp - config mode. This is
where you actually create your IP scope with the network command and any additional
DHCP information you want to give to the endpoints. Following are the common
parameters for endpoints.
Default-router This parameter is mandatory for any endpoints. It tells the endpoint
what IP address to use for its default-gateway.
Domain-name
This parameter specifies the domain name you want your endpoints to use.
DNS-server This parameter informs the endpoints about the IP addresses of their
DNS servers for name resolution. You can specify up to eight DNS servers with a
single command.
This command allows you to specify how long an endpoint is to maintain the
dynamically assigned IP address. You can specify the number of days, hours, or minutes or
even tell it to maintain the address infi nitely.
Another critical parameter that you will want to configure when setting up DHCP
for your Cisco IP phones is the IP address of the TFTP server where the Cisco phone
configuration fi les are located. All Cisco phones (SIP and SCCP) must download a
configuration fi le when they fi rst boot. This configuration fi le contains important
information for the phone to properly function with the CUCM. The IP phones must know
the location of the TFTP server so they can request the configuration fi le. The DHCP
option 150 parameter is used to provide the IP address of the server.
For example, to implement the following information
Lease
Network: 192.168.100.0/24
Default router: 192.168.100.1
Domain name: ccnavoice1.com
DNS server: 192.168.10.5
TFTP server: 192.168.100.10
Lease time: 6 hours
Network Infrastructure Services for VoIP support
157
we would use these DHCP configuration parameters:
Router(dhcp-config)#network 192.168.100.0 255.255.255.0
Router(dhcp-config)#default-router 192.168.100.1
Router(dhcp-config)#domain-name ccnavoice1.com
Router(dhcp-config)#dns-server 192.168.10.5
Router(dhcp-config)#option 150 ip 192.168.100.10
Router(dhcp-config)#lease 0 6 0
Router(dhcp-config)#end
Once those configuration steps are complete, when your IP phones are on the voice
VLAN, they will dynamically receive.
Monitoring and Troubleshooting the DHCP Service
You can monitor your DHCP service with the following useful show commands:
show ip dhcp binding Use this command to display the dynamic IP to MAC address
mappings. It also lets you know when a specific lease will expire. The following example
shows the binding for the DHCP leased IP address 192.168.100.101:
Router# show ip dhcp binding 192.168.100.101
IP address
Hardware address
192.168.100.101 00a0.9802.32de
Lease expiration
Mar 01 2009 12:00 AM
Type
Automatic
This command lists any IP address confl icts and the time the
detection occurred. It also indicates the method of confl ict detection. The example shows a
confl ict for the IP address 192.168.100.101:
show ip dhcp conflict
Router# show ip dhcp conflict
IP address
192.168.100.101
Detection Method
Ping
Detection time
Mar 01 2009 12:28 PM
Configuring the Network Time Protocol
The Network Time Protocol (NTP) should be configured on every single piece of network
equipment in a production network. It is very important to have synchronized times for all
of your logging information. It is also important in the voice world to have your phones
properly synchronized for time. Issues can arise when clocks are mismatched. For instance,
your Unity voice mail needs to know the proper time so it can correctly inform a user when
a person called and left a message.
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Usually, I will specify two devices on a network that have access to a public time source
from the National Institute of Standards and Technology (NIST). Keep in mind that NTP
runs over UDP port 123, so make sure you have this port open on your fi rewall ruleset to
allow access. Configuring a time source on a Cisco IOS device is quite simple. First, you
specify the time zone that your equipment resides in, using the clock timezone command.
Next, you issue the command ntp server and specify an IP address of one of the public
time servers. Then you can configure all of your other network devices to peer with the
device receiving an external clock. Let’s configure a router for an external NTP server. This
example will use the external time source IP address of 192.5.41.41.
Router#configure terminal
Router(config)#clock timezone CHICAGO -6
Router(config)#ntp server 192.5.41.41
Router(config)#end
You may live in a location that adheres to daylight savings time. If
that is the case, you will need to let the Cisco device know by using a
clock summer-time command option. We can issue a show ntp status
command to verify that Router-A is synchronized. Keep in mind that it may
take several minutes for the synchronization process to complete.
Router#show ntp associations
address
ref clock
st when poll reach delay offset
disp
*~192.5.41.41
.USNO.
1
285
512 377
33.9
1.23
1.0
* master (synced), # master (unsynced), + selected, - candidate, ~ configured
Our NTP server is listed and synchronized. We know it is properly synchronized because
of the asterisk (*) to the left of the IP address.
Summary
In this chapter we began with the three options available for powering your Cisco
IP phones. You then learned about network configuration basics such as VLAN
configuration and how to route properly between VLANs. We also spent some time
examining the proper way to implement VTP on the network for ease of managing
VLANs. We then went over quality of service basics and how to configure auto - QoS for
VoIP. We discussed additional link- efficiency techniques to help you to eliminate any
bottlenecks on the network. Lastly, we covered how to configure network services that
help support voice functionality.
Exam Essentials
159
As you can see, you need to configure some modifications on your network
infrastructure route/switch gear to ready it for voice traffic. But at its heart, the
infrastructure is basically the same in terms of moving your packets from point A to
point B. Just like with Chinese food, you may have different main dishes (voice and data
traffic), but they’re always served with a foundation of rice (network infrastructure).
Exam Essentials
Know the three different power options for IP phones. The power brick is attached to the
phone and plugs directly into the wall outlet. A power patch panel or power injector sits
between an IP phone and a standard non-PoE switch; power is sent to the phone over the
same cable that voice traffic resides on. Finally, the PoE switch offers power directly from
the switch to the phone over an Ethernet cable.
Understand the different PoE proprietary and IETF standards of IP phones and PoE
switches. As long as the power source can meet the requirements of the IP phone, the
Cisco PoE switch uses CDP to negotiate the best power option.
Know how to calculate and manipulate power requirements for your IP phone deployment
for PoE switches. It is important to determine your power requirements for any PoE
devices on the network. You should also understand how to change the PoE management
modes on a per-switch and per-switchport basis.
Understand the purpose of VLANs. VLANs segment broadcast domains. A logical
VLAN acts as a physically separate network.
Know what VLAN trunks are used for. VLAN trunks transport multiple VLANs across
the same physical link while keeping the traffic separate using VLAN tags.
Understand the need of inter-VLAN routing. When you have two separate VLANs, you
need a way for the two VLANs to communicate with each other. Inter-VLAN routing is a
Layer 3 feature that routes Layer 2 traffic for inter-VLAN communication.
Understand the VLAN Trunking Protocol (VTP). VTP is a service to help assist with the
adds/changes/deletions of VLANs on a network. The three VTP modes are server, client,
and transparent. VTP uses revision numbers to keep track of the latest updates on the
network.
Understand the difference between data and voice VLANs. Cisco switches use CDP to
identify Cisco IP phones on the network. Voice VLANs are configured differently at the
switchport level. Finally, voice VLANs are tagged on the switchport, while any PC that is
connected to a switch or Cisco IP phone is untagged.
Know the network requirements for voice. End-to - end delay should be 150 ms or less.
Jitter should be 30 ms or less. Packet loss should be at 1 percent or less.
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Know the three QoS classification steps.
then marked, and fi nally queued.
Understand QoS traffic trust boundaries.
better the QoS will ultimately be.
QoS requires the traffic to be fi rst classified,
The closer to the source your boundary is, the
Know how to implement auto - QoS on a network. Auto - QoS is designed to simplify
the QoS implementation for voice traffic. You should be familiar with a few auto - QoS
commands to properly configure your infrastructure for QoS of voice.
Understand basic link- efficiency techniques. Techniques such as compression and LFI can
help transport your voice traffic as efficiently as possible across the infrastructure.
Know how to configure network services for VoIP support. Cisco IP phones rely heavily
on DHCP servers for information such as IP address, default-router, DNS, and the location
of IP phone configuration fi les by defi ning the option 150 parameter for a TFTP server.
Understand the purpose of NTP and how to configure it. NTP is used to synchronize
time for all of your phone equipment on the network. Synchronization of time helps to
ensure proper operation and support of your VoIP network.
Written Lab 4.1
Write the answers to the following questions:
1.
What is the command to display the power settings on a PoE switchport?
2.
What is the command to see what VLAN all of your switchports belong to?
3.
What two commands are used to configure a new VLAN 100 with the name of Voice?
4.
What interface command assigns a switchport to a voice VLAN of 55?
5.
What interface command assigns a switchport to a VLAN of 105?
6.
What two interface commands configure the port to be a VLAN trunk that uses
802.1Q tagging?
7.
What interface command sets the VLAN trunk to allow only VLANs 10, 20, and 30
to be transported over the link?
8.
What command lets you view the VTP information currently configured on a switch?
9.
What command lets you view the VTP password configured on the switch?
10. What command enables DHCP on the router?
(The answers to Written Lab 4.1 can be found following the answers to the review
questions for this chapter.)
Hands-on Labs
161
Hands-on Labs
Here is a list of the labs in this chapter:
Lab 4.1: Setting Power Options on PoE Ethernet Interfaces
Lab 4.2: Configuring Voice and Data VLANs and Switchport Assignment
Lab 4.3: Setting up VTP
Lab 4.4: Configuring Auto - QoS
Lab 4.5: Setting up a DHCP Server
To complete these labs, you need the following equipment:
Lab 4.1: one Cisco PoE switch
Lab 4.2: one Cisco switch
Lab 4.3: one Cisco switch
Lab 4.4: one Cisco switch
Lab 4.5: one Cisco router
Hands-on Lab 4.1: Setting Power Options on PoE
Ethernet Interfaces
1.
Log into your PoE switch and go into privileged exec mode by typing enable.
2.
View the current power settings by typing show power inline fa0/1. You should
see that the switchport is set for an Admin state of auto. This is the default PoE
configuration.
3.
Enter into interface configuration mode for PoE port Fa0/1 by typing configuration
terminal and interface fa0/1.
4.
View your power options by typing power inline ?. You should see the following
options:
PoE-switch(config-if)#power inline ?
auto
Automatically detect and power inline devices
never
Never apply inline power
static
High priority inline power interface
5.
Configure the switchport so that it never supplies power to end devices by typing power
inline never. Exit configuration mode by typing end.
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Verify your configuration changes by again viewing the power settings by typing show
power inline fa0/1. You should now see that the Admin state is off and no power is
allocated to the switchport.
Hands-on Lab 4.2: Configuring Voice
and Data VLANs and Switchport Assignment
1.
Log into your switch and go into privileged exec mode by typing enable.
2.
Enter configuration mode by typing configuration terminal.
3.
Configure two new VLANs on the switch. VLAN 10 is named Data and VLAN
20 is named Voice. To accomplish this, type vlan 10, and you will enter VLAN
configuration mode. Type name Data. Then configure VLAN 20 by typing vlan 20.
Once in VLAN 20 configuration mode you can label the VLAN by typing name Voice.
4.
Exit configuration mode by typing end.
5.
Verify your configuration changes by typing show vlan brief. You should see
something similar to the following:
PoE-switch#show vlan brief
VLAN Name
Status
Ports
---- -------------------------------- --------- ------------------------------1
default
active
Fa0/1, Fa0/2, Fa0/3, Fa0/4
Fa0/5, Fa0/6, Fa0/7, Fa0/8
Fa0/9, Fa0/10, Fa0/11, Fa0/12
Fa0/13, Fa0/14, Fa0/15, Fa0/16
Fa0/17, Fa0/18, Fa0/19, Fa0/20
Fa0/21, Fa0/22, Fa0/23, Fa0/24
Fa0/25, Fa0/26, Fa0/27, Fa0/28
Fa0/29, Fa0/30, Fa0/31, Fa0/32
Fa0/33, Fa0/34, Fa0/35, Fa0/36
Fa0/37, Fa0/38, Fa0/39, Fa0/40
Fa0/41, Fa0/42, Fa0/43, Fa0/44
Fa0/45, Fa0/46, Fa0/47, Fa0/48
Gi0/1, Gi0/2
10
Data
active
20
Voice
active
1002 fddi-default
act/unsup
Hands-on Labs
1003 token-ring-default
1004 fddinet-default
1005 trnet-default
163
act/unsup
act/unsup
act/unsup
6.
Next, we want to configure port Fa0/1 to belong to VLAN 10 if the end device is a PC
and VLAN 20 if the end device is a Cisco IP phone. To set this up, we must enter interface
configuration mode by typing configuration terminal and interface fa0/1.
7.
Assign VLAN 10 to the port by typing switchport access vlan 10. Then configure
the voice VLAN to 20 by typing switchport voice vlan 20.
8.
Exit configuration mode by typing end.
9.
Verify your configuration changes by typing show vlan brief. You should now see
that port Fa0/1 belongs to both VLAN 10 and 20:
PoE-switch#show vlan brief
VLAN Name
Status
Ports
---- -------------------------------- --------- ------------------------------1
default
active
Fa0/2, Fa0/3, Fa0/4, Fa0/5
Fa0/6, Fa0/7, Fa0/8, Fa0/9
Fa0/10, Fa0/11, Fa0/12, Fa0/13
Fa0/14, Fa0/15, Fa0/16, Fa0/17
Fa0/18, Fa0/19, Fa0/20, Fa0/21
Fa0/22, Fa0/23, Fa0/24, Fa0/25
Fa0/26, Fa0/27, Fa0/28, Fa0/29
Fa0/30, Fa0/31, Fa0/32, Fa0/33
Fa0/34, Fa0/35, Fa0/36, Fa0/37
Fa0/38, Fa0/39, Fa0/40, Fa0/41
Fa0/42, Fa0/43, Fa0/44, Fa0/45
Fa0/46, Fa0/47, Fa0/48, Gi0/1
Gi0/2
10
Data
active
Fa0/1
20
Voice
active
Fa0/1
1002 fddi-default
act/unsup
1003 token-ring-default
act/unsup
1004 fddinet-default
act/unsup
1005 trnet-default
act/unsup
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Hands-on Lab 4.3: Setting Up VTP
1.
Log into your switch and go into privileged exec mode by typing enable.
2.
View the VTP settings by typing show vtp status. You should see that the default
VTP mode is server.
3.
Enter configuration mode by typing configuration terminal.
4.
We want this switch to be able to receive and forward VTP messages but not make any
additions, changes, or deletions. Therefore, we type vtp mode client.
5.
We want the switch to belong to the VTP domain called lab -domain. Type vtp domain
lab-domain.
6.
Our VTP domain uses VTP version 2. We then need to type vtp version 2.
7.
Finally, all of our switches within the VTP domain lab-domain use a password of
mypassword. To configure this, type vtp password mypassword.
8.
Exit configuration mode by typing end.
9.
Verify your configuration changes by typing show vtp status and show vtp
password. You should see the following configuration:
PoE-switch#show vtp status
VTP Version
Configuration Revision
Maximum VLANs supported locally
Number of existing VLANs
VTP Operating Mode
VTP Domain Name
VTP Pruning Mode
VTP V2 Mode
VTP Traps Generation
MD5 digest
:
:
:
:
:
:
:
:
:
:
2
5
1005
8
Client
lab-domain
Disabled
Enabled
Disabled
0x26 0x3A 0xA0 0x8C 0x66 0x01 0xA1 0xF
Switch-A#show vtp password
VTP Password: mypassword
Hands-on Lab 4.4: Configuring Auto-QoS
1.
Log into your switch and go into privileged exec mode by typing enable.
2.
Enter interface configuration mode for port Fa0/1 by typing configuration terminal
and interface fa0/1.
Hands-on Labs
3.
165
Check your Auto - QoS options by typing auto qos voip ?. You should see the
following output:
PoE-switch(config-if)#auto qos voip ?
cisco-phone
Trust the QoS marking of Cisco IP Phone
cisco-softphone Trust the QoS marking of Cisco IP SoftPhone
trust
Trust the DSCP/CoS marking
4.
Configure the switchport to trust the QoS marking of a Cisco IP phone. Type auto qos
voip cisco-phone.
5.
Exit interface configuration mode by typing end.
Hands-on Lab 4.5: Setting Up a DHCP Server
1.
Log into your router and go into privileged exec mode by typing enable.
2.
Enter configuration mode by typing configuration terminal.
3.
Enable the DHCP service by typing service dhcp.
4.
We’ll use the 172.16.1.0/24 network as our DHCP pool. We also want to exclude
the first five IP addresses from being handed out. To do this, type ip dhcp excluded
address 172.16.1.1 172.16.1.5.
5.
Enter DHCP configuration mode and name the pool lab -pool by typing ip dhcp
pool lab-pool.
6.
Now that we’re in DHCP configuration mode, configure the IP pool space by typing
network 172.16.1.0 255.255.255.0.
7.
8.
Set the default-gateway for the DHCP devices to 172.16.1.1 by typing
default-router 172.16.1.1.
Set the domain name for the DHCP devices to lab -domain.com by typing domain-name
lab-domain.com.
9.
Set the DNS server for the DHCP devices to 4.2.2.2 by typing dns-server 4.2.2.2.
10. Set the TFTP server for the DHCP devices to 192.168.100.100 by typing
option 150 ip 192.168.100.100.
11. Exit DHCP configuration mode by typing end.
Chapter 4
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Configuring the Network Infrastructure for Voice
Review Questions
1.
What Cisco IP phone power options power the phones using the Ethernet connection? To
accomplish this, they sit between the IP phone and non-PoE switch. Choose all that apply.
A. Power brick
2.
B.
802.3af switch
C.
ILP switch
D.
Powered patch panel
E.
Power injector
An ILP PoE switch can power devices of up to how many watts?
A. 6.0 W
3.
B.
15.4 W
C.
6.3 W
D.
7.0 W
What protocol does a Cisco IP phone use to tell the PoE switch how much power it requires
for the phone?
A. PoE protocol
4.
B.
iLBC
C.
VTP
D.
STP
E.
CDP
What Cisco power saving method helps to negotiate the exact power requirements of a
Cisco IP phone?
A. IPM
5.
B.
802.3af
C.
ILP
D.
CDP
What protocol is used to notify the Cisco IP phone of its voice VLAN number?
A. PoE
B.
Spanning Tree Protocol
C.
Cisco Discovery Protocol
D.
VLAN Trunking Protocol
Review Questions
6.
167
What command syntax properly configures a Cisco switchport for a Cisco IP phone on
VLAN 50 while in interface configuration mode?
A. switchport mode access 50
7.
B.
switchport access vlan 50
C.
switchport voice vlan 50
D.
switchport access voice 50
E.
switchport access voice vlan 50
Devices that reside in the same VLAN share what?
A. The same collision domain
8.
B.
The same VTP domain
C.
The same voice VLAN domain
D.
The same broadcast domain
What IOS command lets you view which interfaces are configured as VLAN trunk links?
A. show trunk
9.
B.
show interfaces trunk
C.
show switchport trunk
D.
show trunk interfaces
What do you need to configure to allow devices on one VLAN to communicate with
devices on a different VLAN?
A. Inter-VTP routing
B.
Inter-VLAN routing
C.
Broadcast bridging
D.
VLAN bridging
E.
Spanning Tree Protocol
10. Which VTP mode allows administrators to add, delete, and modify VLAN information on
the switch without propagating that information to any other switch?
A. VTP server
B.
VTP client
C.
VTP cluster
D.
VTP transparent
E.
VTP access
11. How are the voice and native data VLANs treated differently on the link between the Cisco
switch and the Cisco IP phone?
A. The voice VLAN is tagged using 802.1Q and the data VLAN is not tagged.
B.
The voice VLAN is tagged using ISL and the data VLAN is tagged using 802.1Q.
C.
The voice VLAN is not tagged and the data VLAN is tagged using ISL.
D.
The voice VLAN is not tagged and the data VLAN is tagged using 802.1Q.
Chapter 4
168
Configuring the Network Infrastructure for Voice
12. What is the cause of jitter, a form of variable delay on a network?
A. CODEC processing
B.
Compression techniques
C.
Transcoding delay
D.
Queuing delay
13. What is the maximum end-to - end delay for voice packets on a network according to Cisco?
A. 200 ms
B.
100 ms
C.
150 ms
D.
80 ms
E.
250 ms
14. What is the first step in the QoS process?
A. Traffic marking
B.
Traffic classification
C.
Traffic queuing
D.
Traffic forwarding
15. Ideally, where should the QoS trust boundary be located?
A. At the distribution layer
B.
At the core layer
C.
As close to the endpoint as possible
D.
As far away from the endpoint as possible
E.
At the L3 gateway
16. When you configure auto - QoS on your Cisco switches using the auto qos voip command,
what queuing technique is used?
A. PQ
B.
LLQ
C.
Fast Queuing
D.
Custom Queuing
17. Which of the following can be used to eliminate delay and jitter of time-sensitive traffic
such as voice? Choose all that apply.
A. Interface buffering
B.
LFI
C.
VTP
D.
Increasing the bandwidth
E.
QoS
Review Questions
169
18. Why is it important to configure DHCP option 150 for Cisco voice networks?
A. It defines the default gateway for the phone.
B.
It defines the IP address of the TFTP server.
C.
It defines the IP address of the communications manager.
D.
It defines the IP address for CDP.
19. What IOS configuration mode syntax can be used to remove the first 10 IP addresses from
DHCP scope using the 192.168.1.1/24 subnet?
A. ip excluded-address 192.168.1.1 192.168.1.10
B.
ip excluded-address dhcp 192.168.1.1 192.168.1.10
C.
dhcp excluded-address 192.168.1.1 192.168.1.10
D.
ip dhcp excluded-address 192.168.1.1 192.168.1.10
20. A Cisco 7985 IP phone is an 802.3af class 3 device that requires 15.4 W of power to
operate. You plug the phone into a Cisco PoE switch and the phone does not properly
power up. What could be the problem? Choose all that apply.
A. The PoE switch supports both ILP and 802.3af but only up to class 2 devices.
B.
The PoE switch supports only ILP.
C.
The PoE switch is overutilized and cannot power any additional devices.
D.
The PoE switch supports 802.3af devices up to class 3.
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Chapter 4
Configuring the Network Infrastructure for Voice
Answers to Review Questions
1.
D, E . Both the powered patch panel and the power injector sit between the IP phone and
switch and power the phone on the Ethernet cable.
2.
B. An ILP PoE switch provides up to 15.4 watts of power to capable devices.
3.
E. The Cisco Discovery Protocol (CDP) is used to discover IP phones and negotiate power
options.
4.
A . Cisco Intelligent Power Management (IPM) works between Cisco PoE switches and
Cisco IP phones to negotiate and allocate the exact amount of power needed by the phone.
5.
C . CDP is used between a Cisco switch and Cisco IP phone to inform the phone of its voice
VLAN number.
6.
C . The switchport voice vlan 50 command is the correct syntax.
7.
D. Devices within the same VLAN share a broadcast domain.
8.
Answer: B . The show interfaces trunk command is the proper syntax to see which
interfaces are configured as VLAN trunk links.
9.
B . You must configure some type of inter-VLAN routing for devices located on separate
VLANs to communicate with each other.
10. D. VTP transparent mode does not propagate any changes to any other connected switches.
11. A . Voice VLANs are tagged with 802.1Q and the native data VLAN is left untagged.
12. D. Variable delay occurs in bottleneck situations where voice traffic has to sit in a queue
and wait for other packets to be sent out of the interface before a voice packet can be sent.
This delay is the cause of jitter.
13. C. A maximum of 150 ms end-to-end can be handled and still maintain high-quality voice calls.
14. B . Traffic is always classified fi rst in the QoS process.
15. C . QoS performs best when the trust boundaries are as close to the endpoints as possible.
16. Answer: B . Low Latency Queuing (LLQ) is the queuing mechanism used when you configure auto - QoS.
17. B, D, E . You can implement LFI and QoS techniques and/or increase bandwidth to help
eliminate bottlenecks on the wire that can cause delay and jitter.
18. B . Option 150 defi nes the IP address of the TFTP server, where the phone can download
configuration fi les.
19. D. The ip dhcp excluded-address syntax is the proper way to exclude IP addresses from the
DHCP scope.
20. B, C. Some older Cisco PoE switches support only the Cisco proprietary ILP option, which
can power devices requiring up to 6.3 W. A second possibility is that the switch is powering
many other PoE devices and has simply run out of power to allocate to the newly added phone.
Answers to Written Lab 4.1
Answers to Written Lab 4.1
1.
show power inline
2.
show vlan brief
3.
vlan 100, name Voice
4.
switchport voice vlan 55
5.
switchport access vlan 105
6.
switchport mode trunk and switchport trunk encapsulation dot1q
7.
switchport trunk allowed vlan 10,20,30
8.
show vtp status
9.
show vtp password
10. service dhcp
171
Chapter
5
CUCM Express
Installation and Basic
Configuration
THE CCNA VOICE EXAM TOPICS
COVERED IN THIS CHAPTER INCLUDE
THE FOLLOWING:
Implement Cisco Unified Communications Manager
Express to support endpoints using CLI.
Describe the appropriate software components needed to
support endpoints.
Describe the requirements and correct settings for TFTP.
Configure TFTP.
Describe the differences between key- system and
PBX modes.
Describe the differences between the different types of
ephones and ephone - DNs.
Configure Cisco Unified Communications Manager Express
endpoints.
Perform basic maintenance and operations tasks to
support the VoIP solution.
Describe basic troubleshooting methods for Cisco Unified
Communications Manager Express.
In Chapter 5 we’re going to start configuring the CUCM
Express. Up until now, you’ve been reading all the theory and
rhetoric behind PSTN and VoIP technology as well as how to
design and build your network to support VoIP. Now you’re fi nally going to be able to dig
in and install software to support voice.
This chapter starts off discussing the CUCM licensing options available to you. Then
we’ll move on to installing the Cisco IOS that supports voice as well as show how to
download and install the CUCM Express software, which interacts with the Cisco IOS
software to function. Then you will learn the essential command-line configuration
steps required on a CUCM Express system for call processing to function, and you will
configure phones and extensions to the point where you can make phone calls. Once you
can properly configure basic phone capabilities using the command line, you will learn how
to enable and use the CUCM Express graphical web interface to configure and manage
your voice environment. Lastly, we’ll go over a troubleshooting methodology that helps
you quickly identify and resolve common CUCM Express problems. I will also detail some
handy show and debug commands to assist you with the troubleshooting process.
Understanding CUCM
Express Licensing
One of the more complex tasks required when ordering Cisco voice gear is the way Cisco
handles licensing structures. You need three Cisco licenses to run your CUCM Express
system and Cisco phones on your network:
Cisco IOS license for voice capabilities
CCME Express feature license
Individual user licenses for the total number of Cisco phones
In this section we’ll review each of these so you can properly license and run a CUCM
Express system and Cisco IP phones.
IOS Licenses for Voice
The fi rst license allows you to download and operate a version of Cisco IOS that has
CUCM Express functionality. When you purchase a router, it comes with an IOS feature
Understanding CUCM Express Licensing
175
set with which you can run the router. The license also allows you to download and install
new versions of this IOS feature set when they become available.
CUCM Express Feature Licenses
Just because you own the license to run the voice- capable IOS image doesn’t mean you can
start adding Cisco IP phones. The second license you need is the CUCM Express feature
license. This license determines how many phones you can run on the CUCM. They are
sold in bundles; the smallest bundle is 25 Cisco IP phones. Table 5.1 shows the current
CUCM Express feature license bundles available.
TA B L E 5 .1
CUCM Express 7965 feature license bundles
License
Description
FL-CCME-250
CUCM Express support for up to 250 IP phones
FL-CCME-175
CUCM Express support for up to 175 IP phones
FL-CCME-100
CUCM Express support for up to 100 IP phones
FL-CCME-50
CUCM Express support for up to 50 IP phones
FL-CCME-35
CUCM Express support for up to 35 IP phones
FL-CCME-25
CUCM Express support for up to 25 IP phones
Let’s say you have an environment that on day one requires 150 IP phones. For this
number, you should purchase the FL - CCME -175 license. Then, as the business grows, the
next installment of phones on this network is 30, bringing the total number to 180. Instead
of purchasing all new licenses, you can simply add to your 175-license total by purchasing
the FL - CCME -25 license. Now you have CUCM Express license support for up to
200 phones.
Cisco Phone User Licenses
Finally, you need the Cisco phone user license. When you place an order for Cisco phones,
you are given three different license options for each Cisco phone. For example, Table 5.2
lists the part numbers and descriptions for the 7965G phone:
176
Chapter 5
CUCM Express Installation and Basic Configuration
TA B L E 5 . 2
Cisco IP phone part numbers
Part Number
Description
CP-7965G=
Spare phone w/o license
CP-7965G-CH1
Phone w/ CUCM user license
CP-7965G-CCME
Phone w/ CUCM Express user license
As you can see, you are given several ordering choices for a single phone! The CP7965G= is simply a spare phone. It does not come with a license. These are most commonly
purchased to serve as “cold spares” at businesses. If a licensed phone on the network
were to break, it could be replaced with the unlicensed spare as a one-to - one trade. These
unlicensed phones are less expensive but they can be used only as replacements.
The other two license options are for either the CUCM/CUCMBE or the CUCM
Express call-processing systems. The pricing is slightly different for these two parts. The
CH1 licenses are more expensive than the CCME licenses, but the CH1 licenses can
legitimately be used by the larger CUCM system. By contrast, the CCME licenses cannot
be used for the CUCM/CUCMBE systems. So if you think you may upgrade from a CUCM
Express system to one of the bigger CUCM systems, you may want to go ahead and
purchase the CH1 licenses so you won’t have to purchase phone user licenses twice.
Cisco CUCM Express License Bundles
Cisco is attempting to make licensing for voice capabilities easier on the purchaser by
bundling Cisco ISR hardware with both the IOS and CUCM Express feature licenses.
These CUCM Express license bundles are basically ready to go. All you need to do after
buying one is to choose which Cisco phones you want to install and make sure each phone
purchased comes with either the CH1 or CCME license. Table 5.3 shows a few examples of
the types of ISR voice bundles that Cisco currently offers.
TA B L E 5 . 3
ISR bundle examples
Bundle Part Number
Description
CUCM Express Licenses
CISCO3825-CCME/K9
Cisco 3825 ISR with IOS SP and
voice services
Up to 168 Cisco phones
CISCO2851-CCME/K9
Cisco 2851 ISR with IOS SP and
voice services
Up to 48 Cisco phones
CISCO2801-CCME/K9
Cisco 2801 ISR with IOS SP and
voice services
Up to 8 Cisco phones
Cisco Voice IOS and CUCM Express Software Installation
177
Cisco Voice IOS and CUCM
Express Software Installation
The CUCM Express upgrade/installation process requires the installation of two separate
but dependent pieces of software. One is the Cisco IOS with voice services, and the other is
the CUCM Express software. The versions of these two software items must be compatible
with each other to avoid any operability issues. Cisco has a handy website that gives you a
very clear IOS compatibility matrix for the version of IOS/CUCM Express software you
wish to run. This matrix is frequently updated and can be found at http://www.cisco
.com/en/US/docs/voice_ip_comm/cucme/requirements/guide/33matrix.htm
Once you’ve decided on the IOS and CUCM Express software you wish to run on your
supported router hardware, you can download the software from the http://www.cisco
.com/go/software website. Downloading the IOS software from the Cisco website is quite
simple. The IOS is a single fi le with a .bin extension. Once it is downloaded, you can put
the software on a TFTP server and transfer it over to the router.
In order to download software from Cisco, you must register at
www.cisco.com and have a valid service contract for the software you
wish to acquire.
The process of uploading an IOS image to your router is fairly simple. For example,
suppose you’ve just downloaded the c3825-ipvoicek9-mz.124-15.XZ2.bin IOS image and
have it sitting on your TFTP server at 192.168.1.11. On your Cisco router, you issue the
copy tftp: flash: command to upload the image to the router compact flash (CF) drive.
The router then asks you for additional information such as the IP address of your TFTP
server and the fi lename of the image you wish to TFTP. You can also rename the fi le if you
want it to be named something different on the router. Here is the TFTP upload process
in action:
Router#copy tftp: flash:
Address or name of remote host [192.168.1.11]?
Source filename [c3825-ipvoicek9-mz.124-15.XZ2.bin]?
Destination filename []? c3825-ipvoicek9-mz.124-15.XZ2.bin
Accessing tftp://192.168.1.11/c3825-ipvoicek9-mz.124-15.XZ2.bin...
Erase flash: before copying? [confirm]n
Loading c3825-ipvoicek9-mz.124-15.XZ2.bin from 192.168.1.11
(via GigabitEthernet0/0): !!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!
!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!
!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!
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Chapter 5
CUCM Express Installation and Basic Configuration
[OK - 47576204 bytes]
Verifying checksum... CCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCC
CCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCC OK (0xB420)
47576204 bytes copied in 859.716 secs (55339 bytes/sec)
Router#
Make sure you have sufficient storage space on your router prior to
uploading any files. To verify the amount of space on the flash drive of
a router, issue a show flash privileged exec command to see the total
number of bytes free. If you don’t have enough space, you’ll have to delete
other files/images using the delete flash: <file name> command.
Using a USB Thumb Drive as an Alternate Method for Uploading Software
If you have a router with USB ports such as any of the newer Cisco ISRs, you might find
that uploading IOS and CUCM Express software just got a little easier and faster! You
could go through the typical process of setting up a TFTP server and transferring the files
across the network. This can take a great deal of time and effort to accomplish. If you
happen to have physical access to the router, you can simply load the image files onto
your trusty USB thumb drive and insert it into the router. The router will mount your USB
thumb drive as usbflash1 (or usbflash2 if you have two USB ports), and you can run copy
usbflash1: flash: to move your images to the router CF. For example:
TechRepublic-Router# copy usbflash1:/c3825-ipvoicek9-mz.124-15.XZ2.bin flash:
Destination filename [c3825-ipvoicek9-mz.124-15.XZ2.bin]?
Copy in progress...CCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCC
CCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCC
CCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCC
CCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCC
CCCCCCCCCCCCCCCCC
[output omitted]
Verifying checksum... CCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCC
CCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCCC OK (0xB420)
47576204 bytes copied in 68.756 secs (1193025 bytes/sec)
This method also works for uploading the CME software using the archive tar /xtract
command, which you will learn how to use next. Not only does this save time setting up a
TFTP server, but copying from a USB flash drive is much faster than TFTP!
Cisco Voice IOS and CUCM Express Software Installation
179
Installing CUCM Express software on your IOS router can either be a pain or a real
snap depending on the method you choose. The CUCM Express software consists of over
100 separate fi les that can be downloaded individually or as a Zip fi le from the
cisco.com/software website. If you download the Zip package, you must unzip the
fi les prior to uploading them to the router. That leaves you with quite a task, because
you need to TFTP every fi le to the router flash drive.
An alternative method is to download and archive a prepackaged CUCM
Express system that comes as a single .tar fi le. The .tar packages will be labeled
cme-full.X.X.X.tar or cme-basic.X.X.X.tar, where X.X.X is the CUCM Express version
number. The cme - basic package contains the more common Cisco phone load fi les,
whereas the cme -full package includes all the load fi les. In addition, the cme-basic package
includes the necessary web GUI fi les. The cme-full version includes all of the same web
GUI fi les as well but also contains additional ring tone fi les, desktop backgrounds, and
basic automatic call distribution (B -ACD) fi les.
Once you’ve selected the .tar package that is compatible with the IOS image you’ve
already uploaded, you can upload the CUCM Express software via TFTP, using the
archive tar /xtract command. This command automatically extracts the individual fi les
and places them in an orderly directory structure for your convenience.
If you download and install a .tar package, you can always download and
install additional software functionality that you need but was not included
in the package. To do so, find the software files you require and TFTP them
onto the router flash drive where the prepackaged software has been
extracted.
The next example shows us again using 192.168.1.11 as our TFTP server, and we are
uploading and extracting fi les from the single cme-full-4.3.0.0.tar fi le:
Router#archive tar /xtract tftp://192.168.1.11/cme-full-4.3.0.0.tar flash:
Loading cme-full-4.3.0.0.tar from 192.168.1.11 (via GigabitEthernet0/0): !
extracting bacdprompts/app-b-acd-2.1.2.2-ReadMe.txt (18836 bytes)
extracting bacdprompts/app-b-acd-2.1.2.2.tcl (24985 bytes)
extracting bacdprompts/app-b-acd-aa-2.1.2.2.tcl (35485 bytes)
extracting bacdprompts/en_bacd_allagentsbusy.au (75650 bytes)
extracting bacdprompts/en_bacd_disconnect.au (83291 bytes)
extracting bacdprompts/en_bacd_enter_dest.au (63055 bytes)!
extracting bacdprompts/en_bacd_invalidoption.au (37952 bytes)
extracting bacdprompts/en_bacd_music_on_hold.au (496521 bytes)!!
extracting bacdprompts/en_bacd_options_menu.au (123446 bytes)!
extracting bacdprompts/en_bacd_welcome.au (42978 bytes)
extracting bacdprompts/en_bacd_xferto_operator.au (34794 bytes)!
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180
CUCM Express Installation and Basic Configuration
extracting CME43-full-readme-v.2.0.txt (22224 bytes)
extracting Desktops/320x212x12/CampusNight.png (131470 bytes)
extracting Desktops/320x212x12/CiscoFountain.png (80565 bytes)!
extracting Desktops/320x212x12/List.xml (628 bytes)
extracting Desktops/320x212x12/MorroRock.png (109076 bytes)
extracting Desktops/320x212x12/NantucketFlowers.png (108087 bytes)
extracting Desktops/320x212x12/TN-CampusNight.png (10820 bytes)
extracting Desktops/320x212x12/TN-CiscoFountain.png (9657 bytes)
extracting Desktops/320x212x12/TN-Fountain.png (7953 bytes)
extracting Desktops/320x212x12/TN-MorroRock.png (7274 bytes)!
extracting Desktops/320x212x12/TN-NantucketFlowers.png (9933 bytes)
extracting Desktops/320x212x12/Fountain.png (138278 bytes)
extracting gui/Delete.gif (953 bytes)
extracting gui/admin_user.html (3845 bytes)
extracting gui/admin_user.js (647358 bytes)!!!
extracting gui/CiscoLogo.gif (1029 bytes)!
[output cut]
This process takes a few minutes. Once it’s complete, we can issue a dir flash:
command to see the contents and structure of our CUCM Express system on the compact
flash drive:
Router#dir flash:
Directory of flash:/
1
13
drw-rw-
0
22224
14 drw27 drw45 -rw46 drw127 drw161 -rw.XZ2.bin
0
0
496521
0
0
47576204
Apr 7 2009 18:17:56 +00:00
Apr 7 2009 18:25:56 +00:00
bacdprompts
CME43-full-readme-v.2.0.t
Apr
Apr
Apr
Apr
Apr
Apr
Desktops
gui
music-on-hold.au
phone
ringtones
c3825-ipvoicek9-mz.124-15
xt
7
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18:18:06
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18:26:22
18:18:28
18:31:02
18:37:22
+00:00
+00:00
+00:00
+00:00
+00:00
+00:00
511664128 bytes total (395001856 bytes free)
From the output of the dir flash:command, you can see that our IOS image is on the
CF along with several CUCM Express software directories. You can look into the various
Cisco Voice IOS and CUCM Express Software Installation
181
directories as well. For instance, if you want to view the contents of the gui directory, you
would do the following:
Router#dir flash:/gui
Directory of flash:/gui/
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
-rw-rw-rw-rw-rw-rw-rw-rw-rw-rw-rw-rw-rw-rw-rw-rw-rw-
953
3845
647358
1029
174
16344
864
6328
4558
3724
76699
843
1347
2399
870
9968
3412
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Apr
Apr
Apr
Apr
Apr
Apr
Apr
Apr
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18:26:06
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Delete.gif
admin_user.html
admin_user.js
CiscoLogo.gif
Tab.gif
dom.js
downarrow.gif
ephone_admin.html
logohome.gif
normal_user.html
normal_user.js
sxiconad.gif
Plus.gif
telephony_service.html
uparrow.gif
xml-test.html
xml.template
511664128 bytes total (395001856 bytes free)
The gui directory lists all of the fi les needed for utilizing the CUCM Express web
graphical user interface (GUI). Another important directory is the phone directory. It
contains the fi les requested by any Cisco phones on the CUCM Express. Let’s drill into the
phone directory and its subdirectory, labeled 7945-7965:
Router#dir flash:phone/7945-7965
Directory of flash:phone/7945-7965/
48
49
50
51
52
53
-rw-rw-rw-rw-rw-rw-
2496963
585536
2453202
326315
555406
638
Apr
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Apr
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2009
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18:26:30
18:26:34
18:26:44
18:26:46
18:26:48
18:26:50
+00:00
+00:00
+00:00
+00:00
+00:00
+00:00
apps45.8-3-2-27.sbn
cnu45.8-3-2-27.sbn
cvm45sccp.8-3-2-27.sbn
dsp45.8-3-2-27.sbn
jar45sccp.8-3-2-27.sbn
SCCP45.8-3-3S.loads
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CUCM Express Installation and Basic Configuration
642
642
Apr 7 2009 18:26:50 +00:00
Apr 7 2009 18:26:52 +00:00
term45.default.loads
term65.default.loads
511664128 bytes total (395001856 bytes free)
This directory stores all of the fi les that a Cisco 7945 through 7965 series IP phone will
require. These fi les are requested by the Cisco phones and are downloaded using TFTP.
We’ll configure CUCM Express telephony services later in this chapter.
Initial CUCM Express Configuration
Cisco IP phones using the SCCP signaling protocol rely on servers to receive information
such as the firmware and configuration files. This section details the files that the phones
require and shows how to configure them on the CUCM Express router. First, you’ll see how
to turn the CUCM Express into a TFTP server to offer up specific Cisco phone firmware
files. Then we’ll move on to the four mandatory CUCM Express system configurations
needed to support IP phones. Finally, I’ll show you how to configure and generate individual
phone configuration files to allow each Cisco phone to have unique functionality within the
voice system. After all these steps are completed, your Cisco phone will be able to connect
successfully to its host CUCM Express and use the information gathered to function as a
VoIP phone!
Configuring CUCM Express as a TFTP Server
When a Cisco IP phone successfully powers up, it will use CDP to determine the voice
VLAN it should belong to and then request and receive, at a minimum, an IP address/
subnet mask and gateway IP address via DHCP. It also must have the all-important
option 150 IP address, which is the location of the TFTP server. As you’ve already
learned, for voice the TFTP server is responsible for delivering Cisco phone firmware
and configuration files to the phones when requested. The TFTP server can be located
anywhere on your network, but in smaller environments, the CUCM Express router is
configured for TFTP. This is the first server the IP phone gets its information from. One
group of files that our Cisco IP phone will request is its firmware, which is specifically
tailored to the type of Cisco phone hardware. If you are using your CUCM Express router
to handle TFTP server functionality, you must configure the IOS to serve up the firmware
that your phones will request. Because we’ve downloaded and extracted the .tar CUCM
Express software, the extraction process neatly placed all the necessary firmware files
needed by most phones into an easy-to-understand directory structure. All you need to
do is figure out which Cisco phones you will want to allow on your network and then
Initial CUCM Express Configuration
183
configure the router to serve the appropriate files. You can see all of the firmware file
directories by issuing the dir flash:/phone command:
Directory of flash:/phone/
47 drw56 drw58 drw60 drw69 drw71 drw79 drw88 drw96 drw101 drw110 drw118 drw511664128 bytes total
0
Apr 7 2009 18:18:28
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Apr 7 2009 18:19:58
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Apr 7 2009 18:28:46
0
Apr 7 2009 18:29:30
0
Apr 7 2009 18:29:38
0
Apr 7 2009 18:30:06
0
Apr 7 2009 18:30:34
(395001856 bytes free)
+00:00
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+00:00
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+00:00
+00:00
+00:00
+00:00
+00:00
7945-7965
7937
7914
7906-7911
7920
7931
7942-7962
7921
7940-7960
7970-7971
7975
7941-7961
Let’s assume that we are going to be configuring Cisco 7945, 7965, and 7970 phones
in our environment. Therefore, we need to configure our TFTP server to offer all of the
fi les within the flash:/phone/7945-7965 and flash:/phone/7970-7971 directories. Note
that some of the fi rmware fi les work for multiple phones. For example, the fi rmware fi les
required by the Cisco 7945 are the same as those required by the 7965. This is because the
phones are essentially identical except for the number of extension buttons they have.
The 7945 has four extension buttons, whereas the 7965 has six.
Configuring the Cisco CUCM Express router to serve as a TFTP server for the
fi rmware fi les is quite simple. Each fi rmware fi le needs to have its own tftp-server
flash:/phone/<firmware_file> command. Also note that because our CUCM Express
fi les are organized with a directory structure, we must provide a directory alias for the
Cisco phones. Remember that Cisco phones are unintelligent devices for the most part.
They know only the name of the fi rmware fi les and not where they are located. Because
we’ve organized our CUCM Express software into directories, we must create aliases so
that when the Cisco phone asks for a fi le, it knows which subdirectory the fi le is located
in. Let ’s use the 7945 -7965 phone fi rmware fi les as an example. We’ll fi rst run the dir
flash:/phone/7945-7965 command to see what fi rmware fi les those specific phones
will require:
Router#dir flash:phone/7945-7965
Directory of flash:phone/7945-7965/
48
49
-rw-rw-
2496963
585536
Apr 7 2009 18:26:30 +00:00
Apr 7 2009 18:26:34 +00:00
apps45.8-3-2-27.sbn
cnu45.8-3-2-27.sbn
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50
51
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-rw-rw-rw-rw-rw-rw-
CUCM Express Installation and Basic Configuration
2453202
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Apr
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Apr
Apr
Apr
7
7
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18:26:44
18:26:46
18:26:48
18:26:50
18:26:50
18:26:52
+00:00
+00:00
+00:00
+00:00
+00:00
+00:00
cvm45sccp.8-3-2-27.sbn
dsp45.8-3-2-27.sbn
jar45sccp.8-3-2-27.sbn
SCCP45.8-3-3S.loads
term45.default.loads
term65.default.loads
These phones will need all eight fi les to fully function properly. To offer up these fi les for
downloading to the phones, we need to configure the following:
Router#configure terminal
Enter configuration commands, one per line. End with CNTL/Z.
Router(config)#tftp-server flash:/phone/7945-7965/apps45.8-3-2-27.sbn alias
apps45.8-3-2-27.sbn
Router(config)#tftp-server flash:/phone/7945-7965/cnu45.8-3-2-27.sbn alias
cnu45.8-3-2-27.sbn
Router(config)#tftp-server flash:/phone/7945-7965/cvm45sccp.8-3-2-27.sbn alias
cvm45sccp.8-3-2-27.sbn
Router(config)#tftp-server flash:/phone/7945-7965/dsp45.8-3-2-27.sbn alias
dsp45.8-3-2-27.sbn
Router(config)#tftp-server flash:/phone/7945-7965/jar45sccp.8-3-2-27.sbn alias
jar45sccp.8-3-2-27.sbn
Router(config)#tftp-server flash:/phone/7945-7965/SCCP45.8-3-3S.loads alias
SCCP45.8-3-3S.loads
Router(config)#tftp-server flash:/phone/7945-7965/term45.default.loads alias
term45.default.loads
Router(config)#tftp-server flash:/phone/7945-7965/term65.default.loads alias
term65.default.loads
Router(config)#
We’ve now successfully configured our CUCM Express router to serve up fi rmware fi les
for the Cisco 7945 and 7965 phones using TFTP. Let’s go ahead and fi nish off this example
by configuring the router to serve up fi rmware fi les for the Cisco 7971 phones. First we
look in the phone directory for the 7970 and 7971 phones:
Router#dir flash:/phone/7970-7971
Directory of flash:/phone/7970-7971/
102
103
104
105
106
107
-rw-rw-rw-rw-rw-rw-
2494499
547706
2456051
530601
538527
638
Apr
Apr
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Apr
7
7
7
7
7
7
2009
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18:29:46
18:29:48
18:29:58
18:30:00
18:30:04
18:30:06
+00:00
+00:00
+00:00
+00:00
+00:00
+00:00
apps70.8-3-2-27.sbn
cnu70.8-3-2-27.sbn
cvm70sccp.8-3-2-27.sbn
dsp70.8-3-2-27.sbn
jar70sccp.8-3-2-27.sbn
SCCP70.8-3-3S.loads
Initial CUCM Express Configuration
108
109
-rw-rw-
642
642
Apr 7 2009 18:30:06 +00:00
Apr 7 2009 18:30:06 +00:00
185
term70.default.loads
term71.default.loads
511664128 bytes total (395001856 bytes free)
Now we configure IOS to begin serving up these fi les using TFTP:
Router#configure terminal
Enter configuration commands, one per line. End with CNTL/Z.
Router(config)#tftp-server flash:/phone/7970-7971/apps70.8-3-2-27.sbn alias
apps70.8-3-2-27.sbn
Router(config)#tftp-server flash:/phone/7970-7971/cnu70.8-3-2-27.sbn alias
cnu70.8-3-2-27.sbn
Router(config)#tftp-server flash:/phone/7970-7971/cvm70sccp.8-3-2-27.sbn alias
cvm70sccp.8-3-2-27.sbn
Router(config)#tftp-server flash:/phone/7970-7971/dsp70.8-3-2-27.sbn alias
dsp70.8-3-2-27.sbn Router(config)#tftp-server flash:/phone/7970-7971/
jar70sccp.8-3-2-27.sbn alias jar70sccp.8-3-2-27.sbn
Router(config)#tftp-server flash:/phone/7970-7971/SCCP70.8-3-3S.loads alias
SCCP70.8-3-3S.loads
Router(config)#tftp-server flash:/phone/7970-7971/term70.default.loads alias
term70.default.loads
Router(config)#tftp-server flash:/phone/7970-7971/term71.default.loads alias
term71.default.loads
Router(config)#
That’s all there is to it! At this point, if you were to add one of these phones to your
network, it would receive all the necessary IP information and download the phone
fi rmware fi les from the TFTP server. The phone will not register to the CUCM Express,
however. It is still missing vital configurations that must be set up on the CUCM Express
for the registration process to occur. The next section of this chapter shows how to
configure the CUCM Express to allow Cisco phones to work with the call processor and
how to identify and serve up default configuration fi les to your Cisco IP phones.
Configuring the Mandatory CUCM
Express System Settings
The majority of CUCM Express configuration tuning happens while in config-telephony
mode. You must accomplish four configuration steps to get the system to properly register
phones for call processing. These steps are:
1.
Configure the source IP address for the CUCM Express.
2.
Configure the maximum number of ephones and ephone-DNs (directory numbers)
allowed on the CUCM Express.
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CUCM Express Installation and Basic Configuration
3.
Identify and set the firmware load files that Cisco IP phones should request based on
the Cisco phone model.
4.
Generate and serve up default phone configuration files via TFTP to the Cisco IP phones.
The next four sections detail each of these steps.
Step 1: Configure the Source CUCM Express IP Address
The source IP address defi nes the location of the CUCM Express call-processing unit.
All of the Cisco IP phones on the network will use this address for all communications
with the CUCM Express hardware. After a Cisco phone downloads the correct fi rmware
used via TFTP, it requests and receives generic information about the CUCM Express.
One item is the source IP address where the CUCM Express can be found. In the example
shown in Figure 5.1, we’ll assume that all of our IP phones reside on the voice VLAN of
192.168.10.0/24.
F I G U R E 5 .1
A sample CUCM Express network
Telephony Source IP:
CUCM Express
192.168.10.1
Trunk VLAN
1 and 10
Switch
Cisco Phone
Voice
VLAN 10
192.168.10.0/24
Cisco Phone
Cisco Phone
Data
VLAN 1
192.168.1.0/24
We’re going to use the 192.168.10.1 IP address as our source IP for the Call Manager.
The configuration of the CUCM Express source IP address is as follows:
Router#configure terminal
Router(config)#telephony-service
Router(config-telephony)#ip source-address 192.168.10.1
Router(config-telephony)#end
Router#
Initial CUCM Express Configuration
187
You’ll see later how this information is eventually packaged within a default
configuration fi le and sent to all Cisco IP phones on the network.
Step 2: Configure Max Ephones and DNs
Step 2 of our CUCM Express system configuration involves setting the maximum number of
ephones and ephone-DNs. Ephones represent physical phones. They are how you identify a
particular device within the IOS. Ephone-DNs, on the other hand, are the number extensions
configured on each phone. Figure 5.2 shows a Cisco phone with buttons for multiple ephoneDNs. This particular Cisco phone has buttons to handle up to six ephone-DNs.
FIGURE 5.2
Cisco IP phone extension buttons
By default, the maximum number of both ephones and ephone - DNs is 0. You might
ask, what ’s the point of Cisco setting the defaults to 0 if I have to set them to 1 or more
to get a single phone to work? The answer has to do with memory allocation. When a
maximum ephone and ephone - DN are set, the router sets aside memory for each one. For
example, if you set max ephones to 10 and max-dn to 50, the router allocates memory for
each of the 10 ephones and all 50 ephone - DNs regardless of whether you actually use
them or not! You should keep in mind to not set the maximums too high, because you
could overtax your router. In our example, we’re going to set our max ephones to 8 and
our max-dn to 20:
Router#configure terminal
Router(config)#telephony-service
Router(config-telephony)#max-ephones 8
Router(config-telephony)#max-dn 20
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The maximum number of ephones and ephone-DNs that can be configured depends
on the hardware, because different devices have different amounts of memory installed in
them. Also, let’s say that our max-ephones is 8 and we attempt to add a ninth phone to the
CUCM Express. When this occurs, the phone will not be allowed to register and we will see
a “Registration Rejected” message on the phone display, as shown in Figure 5.3.
FIGURE 5.3
“Registration Rejected” message
Also, if you exceed the max-dn number, you will receive an error when you attempt to
configure the maximum +1 ephone-DN. The following example has max-dn set to 20, so on
the 21st ephone-DN configuration, we’ll see this log message on the CUCM Express console:
Router(config)#ephone-dn 21
dn 21 exceeds max-dn 20
Router(config)#
Step 3: Identify and Set Firmware Load Files
Step 3 of the CUCM Express system configuration process deals with how we
handle the distribution of fi rmware for our Cisco phones. As I ’ve already shown
you, we’ve identified the fi les that our Cisco phones need and have confi gured our
router to serve them using TFTP. The CUCM Express telephony processes must also
be configured to set the fi rmware fi les we choose to defi ne for each phone hardware
type. As mentioned earlier, when the phones fi rst communicate, they have very little
information and must be told virtually everything. One piece of information a phone
does possess is its hardware type (Cisco phone model). The CUCM Express uses this
Initial CUCM Express Configuration
189
information to determine which fi rmware load fi le the phone should request. The
fi rmware load fi le basically tells the CUCM Express what fi rmware to instruct the Cisco
phone to download. It can be a bit diffi cult to figure out which fi rmware load fi le you
need to confi gure for each phone. The best way to fi nd out which load fi les you need is
to do a search on the cisco.com website. Search for “CME X.X fi rmware,” where X.X
is the version of the CUCM Express software you are running. For example, Figure 5.4
shows a search for CME 4.3 fi rmware on Cisco’s website.
FIGURE 5.4
Searching for Cisco phone firmware
Now that we have a listing for the fi rmware fi les each phone requires, we can
determine the single load fi le for each phone that needs to be configured within confi gtelephony mode. In Figure 5.5, we see that we need to configure SCCP45.8-3-3S.loads
as our load fi le.
FIGURE 5.5
Cisco phone load file table
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We know that we need to use SCCP45.8-3-3S.loads as our key load fi le because
1.
We are using SCCP as our signaling protocol.
2.
The file marked with an asterisk (*) is the load file.
Let’s configure our Call Manager to tell our Cisco 7945, 7965, and 7970 phones which
fi rmware load fi les they should request:
Router#configure terminal
Router(config)#telephony-service
Router(config-telephony)#load 7945 SCCP45.8-3-3S.loads
Updating CNF files
CNF files update complete for phonetype(7945)
Router(config-telephony)#load 7965 SCCP45.8-3-3S.loads
Updating CNF files
CNF files update complete for phonetype(7965)
Router(config-telephony)#load 7970 SCCP70.8-3-3S.loads
Updating CNF files
CNF files update complete for phonetype(7970)
If you are using CUCM Express software that is earlier than version
4.3, do not include the .sbin or .loads extension at the end of the
load command. For versions 4.3 and above, use the complete filename
including the .sbin or .loads suffix.
Step 4: Generate and Serve Default Phone Configuration Files
The default phone configuration is the fi le that informs a Cisco phone of all the general
information it needs to communicate with the CUCM Express system. Included in this
default phone configuration is the source IP address and port with which the phones can
communicate to the Call Manager. It also includes the load configuration fi lenames we just
fi nished setting up.
At this point, I’m referring to the phone configuration fi les as “default” because there
is nothing unique about the configurations yet. Once we begin configuring phone extensions
and other settings unique to the phones, this information will be compiled and stored as a
single phone configuration file. But since none of that information is configured at this time,
the configuration files have only the default information that all Cisco phones share.
The phone configuration file is automatically updated every time a change is made that
affects the configuration. For example, if you need to add additional load files for a Cisco phone,
as soon as an addition occurs in the telephony-service configuration prompt, the configuration
file updates itself. You can also manually update the phone configuration file by issuing the
create cnf-files config-telephony command. Here is an example of this command:
Initial CUCM Express Configuration
191
Router(config-telephony)#create ?
cnf-files create XML cnf for ethernet phone
Router(config-telephony)#create cnf-files
Creating CNF files
Once these four steps have been completed, we can back out of config-telephony mode
and fi nish our basic configuration by configuring ephones and ephone-DNs.
Troubleshooting IP Registration Problems
A brand-new Cisco IPT deployment was taking place at a manufacturing facility. After
configuring the CUCM Express for proper deployment, Matt, the engineer responsible
for implementation of the project, began installing 7945, 7965, and 7971 phones on desks
within the office. While powering up the phones, Matt discovered that not all of the IP
phones were properly registering. The 7945 and 7965 phones registered properly, but
the 7971 phones did not. To determine the source of the problem, Matt went through the
following troubleshooting steps:
1.
Matt’s first thought was that the maximum number of ephones and ephone-DNs had
been reached. This was not possible, however, because the phones were implemented
at random, so some of the 7945 and 7965 phones registered properly after a 7971 phone
failed. Also, there were no “Registration Rejected” messages on the phone display.
2.
Another possibility was that there might be an incorrect configuration with the DHCP
server. Matt verified that the 7971 phones were indeed receiving an IP address and
that the option 150 IP address was correct. Also, all of the Cisco phones were on the
same voice VLAN and received the same DHCP pool information, so if there was a
problem with DHCP, it would affect all phones and not just the 7971s.
3.
Matt then verified that the firmware load files were properly set up on the CUCM
Express. Indeed, the configuration showed the proper load commands within the
telephony-service section.
4.
Finally, it dawned on Matt that even though the option 150 was properly configured
within the DHCP settings, he may have forgotten to actually configure the TFTP
server to hand out the 7971 firmware files. Checking this within the CUCM Express,
Matt found that he had correctly configured the TFTP service to serve up 7945 and
7965 load files but not those for the 7971 phones.
To remedy this, Matt used the tftp-server flash: command to serve up all the
necessary .loads files. Once this was completed, Matt rebooted the 7971 phones, and
they were then able to register with the CUCM Express properly.
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Configuring Ephone and Ephone-DNs
Up until now, all of the configuration information that the Cisco IP phones receive from
the CUCM Express has been generic information that all of the phones share. Ephone and
ephone-DN configurations are the way the administrator can control the unique features that
belong to each phone. We’ll first look at what an ephone-DN is and how to configure the most
basic type. Then we’ll learn about ephones and how to apply an ephone-DN to an ephone.
Configuring an Ephone-Directory Number
An ephone-DN is what you would think of as a telephone number. This is the extension
that a user dials when they wish to ring your phone. On the CUCM Express, we can use
many different ephone-DN configuration settings to add functionality, but for now, all we
want to do is add a single ephone-DN to a phone. We’ll get fancier with the configurations
in the next chapter. From a directory number standpoint, you need to fi rst create an
ephone-DN logical tag. Then, once you are in config- ephonedn configuration mode, you
give the ephone-DN an extension number. Let’s configure our fi rst ephone-DN with an
extension of 4001 and a second ephone-DN with the extension 4002:
Router#configure terminal
Router(config)#ephone-dn 1
Router(config-ephone-dn)#number 4001
Router(config-ephone-dn)#exit
Router(config)#ephone-dn 2
Router(config-ephone-dn)#number 4002
Router(config-ephone-dn)#end
Router#
Now that we have two directory numbers configured, let’s apply them to two Cisco
phones using the ephone configuration command.
Configuring an Ephone
An ephone configuration is the logical representation of a physical IP phone. This is where
you apply all the unique ephone-DNs and other settings that are ultimately pushed down
to the phone hardware. Every phone on the CUCM Express has a unique ephone tag in
which all of the phone configurations are applied. The CUCM Express maps the ephone
configuration to the unique MAC address of the phone. By using the MAC address,
the phone can physically move around the network and continue to maintain the same
configuration settings wherever it goes. The MAC address of each Cisco phone can be
found in four locations, for your convenience:
On the box the Cisco phone ships in. The MAC address is also stored on a UPC bar
code located below the MAC address on the box. You can use a bar code scanner to scan
the MAC address into a spreadsheet and then export this spreadsheet to your CUCM,
which can be a great convenience if you are adding hundreds of phones or more.
Initial CUCM Express Configuration
193
On the back of the Cisco phone. A UPC bar code is also located here.
Within the Settings menu of a powered-up Cisco phone.
On the console connection or VTY interface by issuing a show cdp neighbors detail
command when a phone is connected to a Cisco switch.
Since we’re using the most basic phone configuration, the only information we’ll need to
configure ephones is the MAC address of each phone and the ephone-DN tag we wish to apply.
Let’s configure two Cisco phones with our ephone-DN extension numbers. Ephone 1 will be
configured to use extension 4001 and ephone 2 will be configured with extension 4002:
Router#configure terminal
Router(config)#ephone 1
Router(config-ephone)#mac-address 0014.1c4d.2589
Router(config-ephone)#button 1:1
Router(config-ephone)#exit
Router(config)#ephone 2
Router(config-ephone)#mac-address 0014.4c7f.a49b
Router(config-ephone)#button 1:2
Router(config-ephone)#end
Router#
The CUCM can automatically assign extensions to brand -new phones that
do not have a specific ephone -to - MAC-address mapping configured. Using
the auto assign command in config-telephony mode, you can specify the
hardware types eligible for auto -assign as well as the ephone - DNs to be
assigned. As soon as you power up an eligible phone and it registers to
the CUCM Express, auto-assign kicks in and builds an ephone configuration.
It pulls in the MAC address of the phone and configures the lowest unused
tagged ephone -DN from the range specified. This option is perfect for new
environments where it doesn’t matter who receives a particular extension
number or for fast deployments where editing can come later.
The mac-address configuration is self- explanatory, but the button config- ephone mode
configuration needs some explanation. The fi rst number of the button command indicates
the Cisco IP phone button that is being configured. For example, on a Cisco 7960, six
extension buttons are available, so this number could be 1– 6. On the other hand, a 7940
series has only two buttons, so this number could only be 1 or 2. The colon (:) indicates
that you want a standard ring for this extension. We’ll sort out the many different types of
audible and silent rings later on, but for now we just want a standard ring for our phone.
The last number in the configuration specifies the ephone-DN to apply to the physical
phone. Since we specified that ephone 1 uses ephone-DN 1, the extension on button 1 of
ephone 1 will be 4001. And therefore ephone 2 will be configured to use ephone-DN 2 or
extension 4002.
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Making Your First Call Powered by CUCM Express
This is the moment you’ve all been waiting for— the opportunity to actually pick up a
phone handset, hear a dial tone, and dial the second extension to hear it ring! If you
haven’t done so already, you can connect your phones to the network and power them up
so they properly register. Alternatively, if the phones have been connected throughout the
configuration processes, chances are that they have registered to your CUCM Express but
did not get an updated configuration fi le that includes the phone extension. If the phone is
registered and communicating with the CUCM Express, you can either restart or reset the
phones via the command line. As you’ll see next, these two methods are slightly different.
Restart
A restart is a quick reset of the phone. The phones connect to the TFTP server and update
any changes to the configuration fi le. The restart command will update the following
information:
Directory numbers (DNs)
Phone buttons
Speed dial
You have the ability to restart either all of the connected phones at once or one at a time.
If you wish to restart all of the phones, you must be in config-telephony mode and issue a
restart all command. Here is an example of the output of this command:
Router(config)#telephony-service
Router(config-telephony)#restart all
Reset 2 phones: at 5 second interval
per phone
Starting with 7960 phones
- this could take several minutes
Router(config-telephony)#
Reset/Restart-all looking for phones registered as type 30008 7902
Reset/Restart-all looking for phones registered as type 20000 7905
[output omitted]
Reset/Restart-all looking for phones registered as type 436 7965
Reset-All: Requesting Restart for phone SEP0021A086D04D at 192.168.10.12
deviceType 436 Idle [count=1]
May 2 07:28:51.878: %IPPHONE-6-UNREGISTER_NORMAL: ephone-1:SEP0021A086D04D
IP:192.168.10.12 Socket:1 DeviceType:Phone has unregistered normally.
Reset/Restart-all looking for phones registered as type 30006 7970
[output omitted]
Reset/Restart-all looking for phones registered as type 30016 CIPC
Reset-All: Requesting Restart for phone SEP001E68E1AFE9 at 192.168.1.15
Initial CUCM Express Configuration
195
deviceType 30016 Idle [count=2]
May 2 07:29:04.858: %IPPHONE-6-UNREGISTER_NORMAL: ephone-2:SEP001E68E1AFE9
IP:192.168.1.15 Socket:3 DeviceType:Phone has unregistered normally.
May 2 07:29:05.250: %IPPHONE-6-REG_ALARM: 23: Name=SEP001E68E1AFE9 Load=
7.0.1.0 Last=Reset-Restart
May 2 07:29:06.122: %IPPHONE-6-REGISTER: ephone-2:SEP001E68E1AFE9
IP:192.168.1.15 Socket:3 DeviceType:Phone has registered.
Reset/Restart-all looking for phones registered as type 39999 none
[output omitted]
Reset/Restart-all looking for phones registered as type -1 Unknown Ephone type
Restart-All issued for 2 phones
To restart a single phone, you navigate into config- ephone configuration mode of the
specific phone you wish to restart and issue a restart command. Here’s an example:
Router(config)#ephone 1
Router(config-ephone)#restart
restarting 0021.A086.D04D
Router(config-ephone)#
May 2 07:55:12.377: %IPPHONE-6-UNREGISTER_NORMAL: ephone-1:SEP0021A086D04D
IP:192.168.10.12 Socket:1 DeviceType:Phone has unregistered normally.
Reset
The reset command performs a full boot of the phone. This process requires the phone to
go through both the TFTP download and DHCP renewal processes, so it takes more time
for the phone to become fully operational within the CUCM Express system. In addition
to handling the same three configuration updates that the restart command can perform,
the reset command updates the phone if any of the following were added, deleted, or
modified:
Date/time
Phone firmware
CUCME source IP address
TFTP download path
Voice mail access number
Just like the restart command, reset can be performed on all phones or a
single phone. To reset all phones, you must be in config-telephony mode and issue a reset
all command. For a single ephone, navigate to the ephone you desire and enter a reset
command. Here is the command-line output when we reset all the phones on the system:
Router(config)#telephony-service
Router(config-telephony)#reset all
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ITS configuration has been changed, selecting sequence-all reset
Reset 2 phones: sequentially with 240 second per-phone timeout to guarantee
TFTP access
- this could take several minutes per phone
you may abort this process using ‘reset cancel’
Starting reset sequence with 7960 phones
Router(config-telephony)#
Reset/Restart-all looking for phones registered as type 30008 7902
Reset/Restart-all looking for phones registered as type 20000 7905
[output omitted]
Reset/Restart-all looking for phones registered as type 436 7965
Reset-All: Requesting Reset for phone SEP0021A086D04D at 192.168.10.12
deviceType 436 7965 Idle [count=1]
Reset-All received Unregister from ephone-1 SEP0021A086D04D
May 2 07:56:31.941: %IPPHONE-6-UNREGISTER_NORMAL: ephone-1:SEP0021A086D04D
IP:192.168.10.12 Socket:6 DeviceType:Phone has unregistered normally.
May 2 07:57:08.905: %MGCP-3-INTERNAL_ERROR: mgcp_cfg_commands: nvgen lawfulintercept: should not happen
May 2 07:57:33.149: %IPPHONE-6-REG_ALARM: 25: Name=SEP0021A086D04D Load=
SCCP45.8-3-2S Last=Initialized
May 2 07:57:33.165: %IPPHONE-6-REGISTER: ephone-1:SEP0021A086D04D
IP:192.168.10.12 Socket:1 DeviceType:Phone has registered.
Reset sequence-all, Ready to reset next phone (last 61 sec)
Reset sequence-all, Ready to reset next phone (last 61 sec)
Reset/Restart-all looking for phones registered as type 30006 7970
[output omitted]
Reset/Restart-all looking for phones registered as type 30016 CIPC
Reset-All: Requesting Reset for phone SEP001E68E1AFE9 at 192.168.1.15
deviceType 30016 CIPC Idle [count=2]
Reset-All received Unregister from ephone-2 SEP001E68E1AFE9
May 2 07:57:41.885: %IPPHONE-6-UNREGISTER_NORMAL: ephone-2:SEP001E68E1AFE9
IP:192.168.1.15 Socket:3 DeviceType:Phone has unregistered normally.
May 2 07:57:48.545: %IPPHONE-6-REG_ALARM: 22: Name=SEP001E68E1AFE9 Load=
7.0.1.0 Last=Reset-Reset
May 2 07:57:50.269: %IPPHONE-6-REGISTER: ephone-2:SEP001E68E1AFE9
IP:192.168.1.15 Socket:3 DeviceType:Phone has registered.
Reset sequence-all, Ready to reset next phone (last 8 sec)
[output omitted]
Initial CUCM Express Configuration
Reset/Restart-all looking for phones registered as type -1 Unknown Ephone type
Reset-All issued for 2 phones
And here’s an example of resetting the single ephone 1:
Router(config)#ephone 1
Router(config-ephone)#reset
reseting 0021.A086.D04D
Router(config-ephone)#
May 2 07:53:49.937: %IPPHONE-6-UNREGISTER_NORMAL: ephone-1:SEP0021A086D04D
IP:192.168.10.12
You can also reset a Cisco phone using the handset unit by pressing the
Settings button followed by **#** on the keypad.
Once your phones restart or reset, they will receive the updated configuration fi le
containing their individual extension configured for button 1. Figure 5.6 shows what
ephone 2 looks like with its configured extension:
FIGURE 5.6
Configured Cisco phone extension
Go ahead and use a phone to dial the other phone extension. Congratulations, you’ve
officially configured basic call processing on a Cisco CUCM Express!
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Basic Configuration Using the
Telephony Service Setup Script
One configuration method that is often brought up is the telephony-service setup script
command. The telephony service setup script is a command-line script that walks an
administrator through a series of DHCP and voice questions to automatically configure
ephones and ephone-DN settings. The CUCM Express is then set up for auto-assign so
that it hands out extension numbers automatically when you begin adding phones to your
network. At the same time, it grabs the phone’s MAC address and puts it into the ephone
configuration so it will continue to receive the same extension from that point on. To
demonstrate this functionality, we’ll configure two phones with single lines that begin with
extension 8001. We’ll fi rst be asked if we want to configure DHCP. The single subnet that
our two phones will be on is 192.168.10.0/24. The CUCM Express, gateway, and TFTP
server IP address will be 192.168.10.1. To use the script, we enter configuration mode and
type the following:
Router#configure terminal
Router(config)#telephony-service ?
setup Start setup for Cisco Unified Communications Manager Express. Please
refer to
http://www.cisco.com/en/US/products/sw/voicesw/ps4625/tsd_products_
support_
series_home.html
for full documentation.
<cr>
Router(config)#telephony-service setup
--- Cisco IOS Telephony Services Setup --Do you want to setup DHCP service for your IP Phones? [yes/no]: yes
Configuring DHCP Pool for Cisco IOS Telephony Services :
IP network for telephony-service DHCP Pool:192.168.10.0
Subnet mask for DHCP network :255.255.255.0
TFTP Server IP address (Option 150) :192.168.10.1
Default Router for DHCP Pool :192.168.10.1
Do you want to start telephony-service setup? [yes/no]: yes
Configuring Cisco IOS Telephony Services :
Enter the IP source address for Cisco IOS Telephony Services :192.168.10.1
Basic Configuration Using the Telephony Service Setup Script
199
Enter the Skinny Port for Cisco IOS Telephony Services : [2000]:
How many IP phones do you want to configure : [0]: 2
Do you want dual-line extensions assigned to phones? [yes/no]: no
What Language do you want on IP phones :
0 English
1 French
2 German
3 Russian
4 Spanish
5 Italian
6 Dutch
7 Norwegian
8 Portuguese
9 Danish
10 Swedish
11 Japanese
[0]: 0
Which Call Progress tone set do you want on IP phones :
0 United States
1 France
2 Germany
3 Russia
4 Spain
5 Italy
6 Netherlands
7 Norway
8 Portugal
9 UK
10 Denmark
11 Switzerland
12 Sweden
13 Austria
14 Canada
15 Japan
[0]: 0
What is the first extension number you want to configure (maximum 32 digits):
8001
Do you have Direct-Inward-Dial service for all your phones? [yes/no]: no
Do you want to forward calls to a voice message service? [yes/no]: no
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Do you wish to change any of the above information? [yes/no]: no
---- Setup completed config --Router(config)#
*May 19 18:45:32.007: %LINK-3-UPDOWN: Interface ephone_dsp DN 1.1, changed
state to up
*May 19 18:45:32.007: %LINK-3-UPDOWN: Interface ephone_dsp DN 2.1, changed
state to up
That’s all there is to the script; now we can issue a show telephony-service command
to see what this script has configured on our CUCM Express:
Router#show telephony-service
CONFIG (Version=7.0(0))
=====================
Version 7.0(0)
Cisco Unified Communications Manager Express
For on-line documentation please see:
http://www.cisco.com/en/US/products/sw/voicesw/ps4625/tsd_products_support_
series_home.html
ip source-address 192.168.10.1 port 2000
max-ephones 2
max-dn 2
max-conferences 12 gain -6
dspfarm units 0
dspfarm transcode sessions 0
conference software
privacy
no privacy-on-hold
hunt-group report delay 1 hours
hunt-group logout DND
max-redirect 5
cnf-file location: system:
cnf-file option: PER-PHONE-TYPE
network-locale[0] US
(This is the default network locale for this box)
network-locale[1] US
network-locale[2] US
network-locale[3] US
network-locale[4] US
user-locale[0] US
(This is the default user locale for this box)
Basic Configuration Using the Telephony Service Setup Script
201
user-locale[1] US
user-locale[2] US
user-locale[3] US
user-locale[4] US
srst mode auto-provision is OFF
srst ephone template is 0
srst dn template is 0
srst dn line-mode single
time-format 12
date-format mm-dd-yy
timezone 0 Greenwich Standard Time
no transfer-pattern is configured, transfer is restricted to local SCCP phones
only.
keepalive 30 auxiliary 30
timeout interdigit 10
timeout busy 10
timeout ringing 180
timeout transfer-recall 0
timeout ringin-callerid 8
timeout night-service-bell 12
caller-id name-only: enable
edit DN through Web: disabled.
edit TIME through web: disabled.
Log (table parameters):
max-size: 150
retain-timer: 15
create cnf-files version-stamp 7960 May 19 2009 14:08:11
transfer-system full-consult
transfer-digit-collect new-call
auto assign 1 to 2
local directory service: enabled.
Extension-assigner tag-type ephone-tag.
Router#
You can now boot two phones on the network. Once they have fully booted and
received the phone numbers 8001 and 8002, you can run show telephony-service ephone
and show telephony-service ephone-dn to verify their configurations:
Router#show telephony-service ephone
Number of Configured ephones 2 (Registered 1)
ephone 1
Device Security Mode: Non-Secure
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mac-address 001E.68E1.AFE9
type CIPC
button 1:1
keepalive 30 auxiliary 0
max-calls-per-button 8
busy-trigger-per-button 0
Always send media packets to this router: No
Preferred codec: g729r8 pre-ietf
conference drop-mode never
conference add-mode all
conference admin: No
privacy: Yes
privacy button: No
user-locale US
network-locale US
!
ephone 2
Device Security Mode: Non-Secure
mac-address 0021.A086.D04D
type 7965
button 1:2
keepalive 30 auxiliary 0
max-calls-per-button 8
busy-trigger-per-button 0
Always send media packets to this router: No
Preferred codec: g729r8 pre-ietf
conference drop-mode never
conference add-mode all
conference admin: No
privacy: Yes
privacy button: No
user-locale US
network-locale US
!
Router#show telephony-service ephone-dn
ephone-dn 1
number 8001
preference 0 secondary 9
huntstop
call-waiting beep
Basic Configuration Using the GUI
203
ephone-dn 2
number 8002
preference 0 secondary 9
huntstop
call-waiting beep
This script gives you a basic phone setup, but many engineers new to the CUCM
Express system fi nd it provides a good way to get started using the command line to set
up phone lines. As you become more comfortable setting up phones manually, you’ll fi nd
that to be the better option simply because the setup script is very limited in what it can
configure.
Basic Configuration Using the GUI
The Cisco Communications Manager Express has two configuration methods. You’ve just
seen the fi rst— using the command line. This method is what engineers need to be most
familiar with, because it allows you to configure 100 percent of the voice features and is
much faster to use over time. If you are transitioning support of the CUCM Express to
a less technically skilled administrator, they may fi nd that the graphical user interface
(GUI) is more user friendly and intuitive for basic tasks such as setting up new phones and
extensions. The web GUI allows an administrator to use a web browser to connect to the
CUCM Express to configure many of the telephony features available on the system. This
section will cover enabling the GUI interface and show you some of the basics of navigating
the interface.
Enabling the GUI Interface
By default, the CUCM Express GUI interface is disabled. You must work through four
steps using the command-line interface to enable the web GUI. You fi rst must enable either
the HTTP or the HTTPS server on the router. You also need to tell the router the location
of the root web directory. Third, you need to allow for HTTP authentication from locally
configured usernames and passwords. Lastly, you must configure a telephony service web
administrator username and password. Following is an example of how to accomplish all
of these tasks on the IOS router.
Enabling the HTTP(S) Server Process
Cisco routers have the ability to run web server processes for basic router IOS monitoring
and configuration. Because our IOS router is also a CUCM Express router, you must
enable the web processes to use the Call Manager web GUI as well. You can enable HTTP,
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HTTPS, or both forms of access. The following commands configure both HTTP and
HTTPS server processes on the router:
Router#configure terminal
Router(config)#ip http server
Router(config)#ip http secure-server
% Generating 1024 bit RSA keys, keys will be non-exportable...[OK]
Router(config)#exit
Router#
Setting the Root Web Directory
Even though you’ve enabled web services on your router, the CUCM GUI still will not
work until you tell the router where it can fi nd the web pages that are stored somewhere
on the router. Typically, the web page fi les are found in just two locations. If you extracted
all the CUCM Express fi les directly onto the router flash drive without a directory
structure, you can simply issue the following command:
Router#configure terminal
Router(config)#ip http path flash:/
Router(config)#end
Router#
If your CUCM Express fi les are organized in a directory structure, you will need to let
the CUCM Express router know where your web page fi les are in a subdirectory referred to
as a root web directory. If you have your fi les in a directory structure, the web fi les are in
the flash:/gui directory. Here is the output of the show dir flash:/gui command:
Directory of flash:/gui/
28
29
30
31
32
33
34
35
36
37
38
39
40
41
-rw-rw-rw-rw-rw-rw-rw-rw-rw-rw-rw-rw-rw-rw-
953
3845
647358
1029
174
16344
864
6328
4558
3724
76699
843
1347
2399
Apr
Apr
Apr
Apr
Apr
Apr
Apr
Apr
Apr
Apr
Apr
Apr
Apr
Apr
7
7
7
7
7
7
7
7
7
7
7
7
7
7
2009
2009
2009
2009
2009
2009
2009
2009
2009
2009
2009
2009
2009
2009
18:26:06
18:26:06
18:26:10
18:26:12
18:26:12
18:26:12
18:26:12
18:26:14
18:26:14
18:26:14
18:26:16
18:26:16
18:26:16
18:26:18
+00:00
+00:00
+00:00
+00:00
+00:00
+00:00
+00:00
+00:00
+00:00
+00:00
+00:00
+00:00
+00:00
+00:00
Delete.gif
admin_user.html
admin_user.js
CiscoLogo.gif
Tab.gif
dom.js
downarrow.gif
ephone_admin.html
logohome.gif
normal_user.html
normal_user.js
sxiconad.gif
Plus.gif
telephony_service.html
Basic Configuration Using the GUI
42
43
44
-rw-rw-rw-
870
9968
3412
Apr 7 2009 18:26:18 +00:00
Apr 7 2009 18:26:18 +00:00
Apr 7 2009 18:26:20 +00:00
205
uparrow.gif
xml-test.html
xml.template
To inform your CUCM Express router that it should look inside the flash:/gui
directory for the web fi les, run the following command:
Router#configure terminal
Router(config)#ip http path flash:/gui
Router(config)#end
Router#
Now the CUCM Express will look in the gui subdirectory for all web fi les.
Enabling Local Web Authentication
Next on the task list is to set your CUCM Express router so that it allows for authentication
from locally configured usernames and passwords. To do this, use the ip http
authentication local command. Here’s the configuration output that accomplishes this task:
Router#configure terminal
Router(config)#ip http authentication local
Router(config-telephony)#end
Router#
Creating a CUCM Express Administrator Account
The CUCM Express administrator account is separate from any user accounts that are made
on the router side of the configuration. These special administrators have access only to the
CUCM Express configuration capabilities, so you can create a separation of duties between
route/switch administrators and voice administrators. There are several methods for creating
these users, but the most basic is to create a system administrator using a local username and
password. The following example shows how I created an administrator named WebAdmin
with a secret password of cisco. The secret 0 portion of the command specifies that we will
be entering our password in plaintext form (0) and that we want the secret password to be
encrypted when the command is applied to the running configuration.
Router#configure terminal
Router(config)#telephony-service
Router(config-telephony)#web admin system name WebAdmin secret 0 cisco
Router(config-telephony)#end
Router#
At this point, you can connect to the CUCM Express web GUI by pointing your web
browser at https://<CUCM Express source IP address>/ccme.html
Figure 5.7 shows the SSL warning message that our web browser displays.
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SSL warning message
We’re receiving this message because the SSL certificate that the router created when we
enabled HTTPS was self-signed. In a production environment, you may want to purchase
a certificate from a trusted certificate authority such as VeriSign to avoid receiving this
warning message from your browser. Setting up a trusted certificate is beyond the scope of
this book. For now, just click to continue on to the website.
The next screen we are presented with is an access popup asking us for a username and
password. These are the CUCM Express administrator username and password that we
just created. Figure 5.8 shows the access window with our credentials being entered.
FIGURE 5.8
CUCM Express GUI login
Basic Configuration Using the GUI
207
Now that we’re logged in, we can see the CUCM Express GUI for the fi rst time, as
indicated in Figure 5.9.
FIGURE 5.9
CUCM Express GUI main page
Along the top of the screen are drop -down menus for Configure, Voice Mail,
Administration, Reports, and Help. Next, we’ll take a quick look at some of the most
frequently used web GUI options.
CUCM Express Web GUI Basics
The CUCM Express Web GUI is divided into five tabs. The Voice Mail tab is strictly for the
configuration of a Unity system. Because we haven’t yet covered Unity Express, we’ll save
that section for later on in the book. The Help tab is a basic help tool to assist new users
with the “where and why” of the configuration tool. This section will focus on the other
three tabs, and you’ll see where to look to configure and verify the most important options
the system has available to it.
Using the Configure Menu
The Configure tab is where administrators will spend most of their time. Here, you can
add, change, and delete phones and phone extensions. Figure 5.10 shows the phones
currently configured on the system.
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Configured phones
To configure a new phone, you click the Add button to set up the new phones.
To add a new extension, you click the Extensions option within the Configure tab. This
lists all currently configured extensions. To configure a new extension, you click the Add
button here. If you do that now, however, you will receive a pop -up message indicating that
this configuration option is disabled on your system, as shown in Figure 5.11.
F I G U R E 5 .11
Add Extension Number alert pop-up message
This is one of the few web GUI features that are disabled by default on the CUCM
Express system. To enable the web administrator to add or change extensions, you must
issue the dn-webedit config-telephony command using the command line:
Router#configure terminal
Router(config)#telephony-service
Router(config-telephony)#dn-webedit
Router(config-telephony)#end
Router#
Now when you click the Add button to add an extension on your system, instead of
receiving the disabled message, you will see the display shown in Figure 5.12.
Basic Configuration Using the GUI
F I G U R E 5 .1 2
The Add An Extension Number display
A third important Configure menu option is labeled System Parameters. This is where
you configure all of the global voice parameters, including dial plans, hunt groups, and
system messages. Figure 5.13 shows this section’s GUI interface.
F I G U R E 5 .1 3
Configuring system parameters
209
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CUCM Express Installation and Basic Configuration
Using the Administration Menu
When you make any changes to the voice system, the modifications are automatically made.
Some changes require you to either restart or reset the phone, but the changes themselves
are automatically put into the IOS running configuration. As you’ve learned from your
CCNA studies, however, if the router were to reboot, the changes that are in the running
configuration are erased and the router loads the last-saved startup configuration. To ensure
that our changes are maintained after a router reboot, we need to move to the Administration
menu and choose the Save Router Config option. This essentially performs a write memory
on the router to save the running configuration into the startup configuration.
If your router administrator is not also the CUCM Express web GUI
administrator, you need to coordinate this action because it will save not
only the CUCM Express configuration changes but also any changes the
router administrator has made using the command line.
Figure 5.14 shows this process being performed on the web GUI.
F I G U R E 5 .1 4
Saving the router configuration
Using the Reports Menu
The Reports menu within the web GUI maintains data and displays it in the
form of basic reports. The most widely used report is Call History, which shows
call origin and destination along with start and end dates/times. This report can
Using CUCM Express Verification and Troubleshooting Commands
211
be useful for troubleshooting or other administrative tasks that management may
need to perform. Figure 5.15 shows the output of a Call History report on the
web GUI.
F I G U R E 5 .1 5
A Call History report
Using CUCM Express Verification
and Troubleshooting Commands
When setting up a CUCM Express for the fi rst time, you may need some basic
troubleshooting skills. This section goes through some of the more common troubleshooting
steps, including how to figure out why a Cisco phone won’t register and how to determine
the state of an ephone on your network.
Troubleshooting Cisco Phone Registrations
There will come a time when you add a new Cisco phone to your CUCM Express
environment and it will not register. Because we understand the boot process, we have a
methodical way of troubleshooting the problem. Here is the order in which troubleshooting
should be performed:
1.
Troubleshoot DHCP issues.
2.
Troubleshoot TFTP issues.
3.
Troubleshoot ephone registration issues.
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Troubleshooting these three issues in order will help you to fi nd and fi x the vast majority
of phone registration problems you’ll encounter.
Troubleshooting DHCP Issues
When the phone boots up, one of the fi rst things it displays is a “Configuring IP” message.
This tells you that the phone is attempting to fi nd the DHCP servers so it can receive the
IP address and TFTP information needed to download the fi rmware and configuration
fi les. You can verify that your phone is receiving DHCP information by using the debug ip
dhcp server events command. Here’s an example of the output you will receive when a
device successfully receives an IP address from the DHCP server that is configured on your
CUCM Express router:
Router#debug ip dhcp server events
DHCP server event debugging is on.
May 17 18:18:54.303: DHCPD: Sending notification of ASSIGNMENT:
May 17 18:18:54.303: DHCPD: address 192.168.10.2 mask 255.255.255.0
May 17 18:18:54.303:
DHCPD: htype 1 chaddr 0021.a086.d04d
May 17 18:18:54.303:
DHCPD: lease time remaining (secs) = 86400
On a Cisco IP phone, you can verify that your phone received DHCP information
by pressing the Settings button and navigating to the Network Configuration area. If your
phone is not receiving an IP address, you should begin troubleshooting this as a DHCP
problem and not a VoIP fi rmware or configuration problem.
If your Cisco IP phone is not receiving an IP address, it might be because the DHCP
broadcast message is not reaching the DHCP server. If your DHCP server is set up on a
subnet other than the subnet where the IP phone resides, then the DHCP broadcast message
will never reach the server because, as you learned in chapter 4, broadcasts are contained
within a single VLAN. You can either set up a DHCP server on every single VLAN on your
network or use the ip helper-address X.X.X.X command on your router or VLAN Layer
3 interfaces. X.X.X.X is the IP address of your DHCP server. Maintaining multiple DHCP
servers can be a cumbersome task, so you are much better off using the helper-address
command to forward the DHCP requests on to your DHCP server that is located on a
different subnet. What does this command do when configured? When the Layer 3 gateway
interface hears a broadcast message from a DHCP client, the broadcast request is turned
into a unicast message and forwarded to the IP address of the DHCP server. The DHCP
server can then receive the request and hand out the appropriate IP address based on the
source IP address, which will be the gateway IP of the Layer 3 interface.
Troubleshooting TFTP Issues
If your phone is receiving DHCP information, the next thing it attempts to do is
download the fi rmware and configuration fi les required to operate. If your phone is stuck
with the “Registering” notification on the screen, you can try to run the debug tftp
events command to see if your phone is requesting fi les that are not on your TFTP server.
Keep in mind that this command is useful only if your router is acting as the TFTP
Using CUCM Express Verification and Troubleshooting Commands
213
server. Here is an example of the output of this command for a phone that successfully
receives some but not all of the requested fi rmware and configuration fi les:
Router#debug tftp events
TFTP Event debugging is on
Router#
May 17 18:51:36.855: TFTP: Looking for CTLSEP001E68E1AFE9.tlv
May 17 18:51:37.887: TFTP: Looking for SEP001E68E1AFE9.cnf.xml
May 17 18:51:37.887: TFTP: Opened system:/its/XMLDefaultCIPC.cnf.xml, fd 9,
size 1056 for process 248
May 17 18:51:37.891: TFTP: Finished system:/its/XMLDefaultCIPC.cnf.xml, time
00:00:00 for process 248
May 17 18:51:42.315: TFTP: Looking for Communicator/LdapDirectories.xml
May 17 18:51:43.423: TFTP: Looking for Communicator/LdapDialingRules.xml
May 17 18:51:49.823: TFTP: Looking for SEP001E68E1AFE9.cnf.xml
May 17 18:51:49.823: TFTP: Opened system:/its/XMLDefaultCIPC.cnf.xml, fd 9,
size 1056 for process 248
May 17 18:51:49.827: TFTP: Finished system:/its/XMLDefaultCIPC.cnf.xml, time
00:00:00 for process 248
May 17 18:51:50.035: TFTP: Looking for CTLSEP001E68E1AFE9.tlv
May 17 18:51:50.043: TFTP: Looking for English_United_States/ipc-sccp.jar
May 17 18:51:50.059: TFTP: Looking for CTLSEP001E68E1AFE9.tlv
May 17 18:51:50.063: TFTP: Looking for United_States/g3-tones.xml
May 17 18:51:50.315: %IPPHONE-6-REG_ALARM: 25: Name=SEP001E68E1AFE9 Load=
7.0.1.0 Last=Initialized
May 17 18:51:51.791: %IPPHONE-6-REGISTER: ephone-1:SEP001E68E1AFE9 IP:
192.168.10.4 Socket:1 DeviceType:Phone has registered.
Router#
Any line that begins with Looking means that the Cisco phone is requesting the
fi le. If the TFTP server knows where a fi le is located, it will process the fi le, giving you
the Opening statement. Finally, once the fi le is transferred you will receive a Finished
message.
As you can see in the sample output, this phone registered to the CUCM
Express even though it did not receive all of the files it requested. Some of
the files, such as LdapDirectories.xml, are supplementary services that
do not affect phone registration. The TFTP server did manage to serve up
the required files for the phone to register on the system.
If your phones are not receiving the necessary fi rmware or configuration fi les, you
should make sure that your TFTP server is configured to serve up the fi les your phone is
214
Chapter 5
CUCM Express Installation and Basic Configuration
requesting. To do so, you can issue a show telephony-service tftp-bindings command.
Here’s a sample of typical output from this command:
Router#show telephony-service tftp-bindings
tftp-server system:/its/united_states/7960-tones.xml alias United_States/7960tones.xml
tftp-server system:/its/united_states/7960-font.xml alias English_United_States
/7960-font.xml
tftp-server system:/its/united_states/7960-font.xml alias English_United_States
/7920-font.xml
tftp-server system:/its/united_states/7960-dictionary.xml alias English_United_
States/7960-dictionary.xml
tftp-server system:/its/united_states/7960-kate.xml alias English_United_States
/7960-kate.xml
tftp-server system:/its/united_states/7960-kate.xml alias English_United_States
/7920-kate.xml
tftp-server system:/its/united_states/SCCP-dictionary.xml alias English_United_
States/SCCP-dictionary.xml
tftp-server system:/its/SEPDEFAULT.cnf alias SEPDefault.cnf
tftp-server system:/its/XMLDefault.cnf.xml alias XMLDefault.cnf.xml
tftp-server system:/its/ATADefault.cnf.xml alias ATADefault.cnf.xml
tftp-server system:/its/XMLDefaultCIPC.cnf.xml alias SEP001E68E1AFE9.cnf.xml
tftp-server system:/its/XMLDefault7965.cnf.xml alias SEP0021A086D04D.cnf.xml
If there are any fi les that are being requested and not listed by this command, you should
locate them on your flash drive and serve them up using the tftp-server configuration
command.
Troubleshooting Ephone Registration Issues
If you believe your TFTP server is serving up all the necessary fi les to your phone, the fi nal
step is to look at the ephone registration process itself. To do so, you can issue the debug
ephone register command. Following is an example of a successful registration of a 7965
phone. There’s a great deal of information here, but as you sift through the output, you
can see how the phone at IP 192.168.10.3 initiates the registration process. The unique
phone configuration fi le of this phone, SEP0021A086D04D, is then used to configure all of the
necessary information into the phone, including the following steps:
Sets IP address of the CUCM Express system
Sets the date/time format
Sets softkeys
Verifies voice codec capabilities
Configures extensions
Using CUCM Express Verification and Troubleshooting Commands
215
Router#debug ephone register
EPHONE registration debugging is enabled
Router#
May 17 19:23:48.243: New Skinny socket accepted [1] (1 active)
May 17 19:23:48.243: sin_family 2, sin_port 51244, in_addr 192.168.10.3
May 17 19:23:48.243: skinny_add_socket 1 192.168.10.3 51244
May 17 19:23:48.307: %IPPHONE-6-REG_ALARM: 25: Name=SEP0021A086D04D Load=
SCCP45.8-3-2S Last=Initialized
May 17 19:23:48.307:
Skinny StationAlarmMessage on socket [2] 192.168.10.3 SEP0021A086D04D
May 17 19:23:48.307: severityInformational p1=0 [0x0] p2=0 [0x0]
May 17 19:23:48.307: 25: Name=SEP0021A086D04D Load= SCCP45.8-3-2S Last=
Initialized
May 17 19:23:48.335: ephone-(2)[2] StationRegisterMessage (1/2/21) from
192.168.10.3
May 17 19:23:48.335: ephone-(2)[2] Register StationIdentifier DeviceName
SEP0021A086D04D
May 17 19:23:48.335: ephone-(2)[2] StationIdentifier Instance 0
deviceType
436
May 17 19:23:48.335: ephone-2[1/-1]:stationIpAddr 192.168.10.3
May 17 19:23:48.335: ephone-2[1/-1][SEP0021A086D04D]:maxStreams 5
May 17 19:23:48.335: ephone-2[1/-1][SEP0021A086D04D]:From Phone raw protocol
Ver 0x8570000C
May 17 19:23:48.335: ephone-2[1/-1][SEP0021A086D04D]:protocol Ver 0x8570000C
May 17 19:23:48.335: ephone-2[1/-1][SEP0021A086D04D]:phone-size 13200 dn-size
784
May 17 19:23:48.335: ephone-(2) Allow any Skinny Server IP address 192.168.10.1
May 17 19:23:48.335: ephone-2[1/-1][SEP0021A086D04D]:Found entry 1 for
0021A086D04D
May 17 19:23:48.335: %IPPHONE-6-REGISTER: ephone-2:SEP0021A086D04D IP:
192.168.10.3 Socket:2 DeviceType:Phone has registered.
May 17 19:23:48.335: Phone 1 socket 2
May 17 19:23:48.335: Skinny Local IP address = 192.168.10.1 on port 2000
May 17 19:23:48.335: Skinny Phone IP address = 192.168.10.3 51244
May 17 19:23:48.339: ephone-2[1/2][SEP0021A086D04D]:Signal protocol
ver 9 to phone with ver 12
May 17 19:23:48.339: ephone-2[1/2][SEP0021A086D04D]:Date Format M/D/Y
May 17 19:23:48.339: ephone-2[1/2]:RegisterAck sent to sockettype ephone
socket 2: keepalive period 30 use sccp-version 9
May 17 19:23:48.339: ephone-2[1/2]:CapabilitiesReq sent
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May 17 19:23:48.355: ephone-2[1/2][SEP0021A086D04D]:Skinny IP port 3500
set for socket [2]
May 17 19:23:48.355: ephone-2[1/2]:ButtonTemplateReqMessage
May 17 19:23:48.355: ephone-2[1/2]:ButtonTemplateReqMessage waiting for
Caps
May 17 19:23:48.355: ephone-2[1/2]:StationSoftKeyTemplateReqMessage
May 17 19:23:48.355: ephone-2[1/2]:StationSoftKeyTemplateResMessage
May 17 19:23:48.355: ephone-2[1/2]:StationSoftKeySetReqMessage
May 17 19:23:48.355: ephone-2[1/2]:StationSoftKeySetResMessage
May 17 19:23:48.359: ephone-2[1/2]:StationConfigStatReqMessage
May 17 19:23:48.359: ephone-2[1/2][SEP0021A086D04D]:
StationConfigStatMessage sent for device SEP0021A086D04D (40/280)
May 17 19:23:48.363: ephone-2[1/2]:CapabilitiesRes received
May 17 19:23:48.363: ephone-2[1/2][SEP0021A086D04D]:Caps list 9
WideBand_256K 40 ms, is_mtp 0
G711Ulaw64k 40 ms, is_mtp 0
G711Alaw64k 40 ms, is_mtp 0
ILBC 60 ms, is_mtp 0
G729AnnexB 60 ms, is_mtp 0
G729AnnexAwAnnexB 60 ms, is_mtp 0
G729 60 ms, is_mtp 0
G729AnnexA 60 ms, is_mtp 0
Unrecognized Media Type 257 1 ms, is_mtp 0
May 17 19:23:48.363: ephone-2[1/2]:Process pending button template
May 17 19:23:48.363: ephone-2[1/2]:ButtonTemplateReqMessage
May 17 19:23:48.363: ephone-2[1/2][SEP0021A086D04D]:
StationButtonTemplateReqMessage set max presentation to 6
May 17 19:23:48.363: ephone-2[1/2]:CheckAutoReg
May 17 19:23:48.363: ephone-2[1/2]:AutoReg is disabled
May 17 19:23:48.363: ephone-2[1/2][SEP0021A086D04D]:Setting 6 lines 0
speed-dials on phone (max_line 6)
May 17 19:23:48.363: ephone-2[1/2][SEP0021A086D04D]:First Speed Dial
Button location is 0 (0)
May 17 19:23:48.363: ephone-2[1/2]:ButtonTemplate lines=6 speed=0
buttons=6 offset=0
May 17 19:23:48.363: ephone-2[1/2][SEP0021A086D04D]:ButtonTemplate
buttonCount=6 totalButtonCount=6 buttonOffset=0
May 17 19:23:48.363: ephone-2[1/2][SEP0021A086D04D]:Configured 0 speed
dial buttons
May 17 19:23:48.423: ephone-2[1/2][SEP0021A086D04D]:StationLineStatReqMessage
from ephone line 1
Using CUCM Express Verification and Troubleshooting Commands
May 17 19:23:48.423: ephone-2[1/2]:StationLineStatReqMessage ephone line 1
DN 2 = 8002 desc = 8002 label =
May 17 19:23:48.423: ephone-2[1/2][SEP0021A086D04D]:
StationLineStatResMessage sent to ephone (1 of 6)
May 17 19:23:48.435: ephone-2[1/2][SEP0021A086D04D]:
StationForwardStatReqMessage line 1 from ephone
May 17 19:23:48.435: Skinny StationForwardStatMessage line 1 sent on ephone
socket [2] for ephone-2
May 17 19:23:48.435: activeForward 0 AllActive 0 BusyActive 0
NoAnswerActive 0
May 17 19:23:48.435: ephone-2[1/2][SEP0021A086D04D]:
StationLineStatReqMessage from ephone line 2
May 17 19:23:48.435: ephone-2[1/2][SEP0021A086D04D]:
StationLineStatReqMessage from ephone line 2 Invalid DN -1
May 17 19:23:48.435: ephone-2[1/2][SEP0021A086D04D]:
StationLineStatResMessage sent to ephone (2 of 6)
May 17 19:23:48.435: ephone-2[1/2][SEP0021A086D04D]:
StationForwardStatReqMessage line 2 from ephone
May 17 19:23:48.435: Skinny StationForwardStatMessage line 2 sent
on ephone socket [2] for ephone-2
May 17 19:23:48.435: activeForward 0 AllActive 0 BusyActive 0
NoAnswerActive 0
May 17 19:23:48.435: ephone-2[1/2][SEP0021A086D04D]:
StationLineStatReqMessage from ephone line 3
May 17 19:23:48.435: ephone-2[1/2][SEP0021A086D04D]:
StationLineStatReqMessage from ephone line 3 Invalid DN -1
May 17 19:23:48.435: ephone-2[1/2][SEP0021A086D04D]:
StationLineStatResMessage sent to ephone (3 of 6)
May 17 19:23:48.435: ephone-2[1/2][SEP0021A086D04D]:
StationForwardStatReqMessage line 3 from ephone
May 17 19:23:48.435: Skinny StationForwardStatMessage line 3 sent on
ephone socket [2] for ephone-2
May 17 19:23:48.435: activeForward 0 AllActive 0 BusyActive 0 NoAnswerActive 0
May 17 19:23:48.435: ephone-2[1/2][SEP0021A086D04D]:
StationLineStatReqMessage from ephone line 4
May 17 19:23:48.435: ephone-2[1/2][SEP0021A086D04D]:
StationLineStatReqMessage from ephone line 4 Invalid DN -1
May 17 19:23:48.435: ephone-2[1/2][SEP0021A086D04D]:
StationLineStatResMessage sent to ephone (4 of 6)
May 17 19:23:48.435: ephone-2[1/2][SEP0021A086D04D]:
StationForwardStatReqMessage line 4 from ephone
May 17 19:23:48.435: Skinny StationForwardStatMessage line 4 sent on
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ephone socket [2] for ephone-2
May 17 19:23:48.435: activeForward 0 AllActive 0 BusyActive 0 NoAnswerActive 0
May 17 19:23:48.435: ephone-2[1/2][SEP0021A086D04D]:StationLineStatReqMessage
from ephone line 5
May 17 19:23:48.435: ephone-2[1/2][SEP0021A086D04D]:StationLineStatReqMessage
from ephone line 5 Invalid DN -1
May 17 19:23:48.435: ephone-2[1/2][SEP0021A086D04D]:StationLineStatResMessage
sent to ephone (5 of 6)
May 17 19:23:48.435: ephone-2[1/2][SEP0021A086D04D]:StationForwardStatReqMessage
line 5 from ephone
May 17 19:23:48.435: Skinny StationForwardStatMessage line 5 sent on ephone
socket [2] for ephone-2
May 17 19:23:48.435: activeForward 0 AllActive 0 BusyActive 0 NoAnswerActive 0
May 17 19:23:48.435: ephone-2[1/2][SEP0021A086D04D]:StationLineStatReqMessage
from ephone line 6
May 17 19:23:48.435: ephone-2[1/2][SEP0021A086D04D]:StationLineStatReqMessage
from ephone line 6 Invalid DN -1
May 17 19:23:48.435: ephone-2[1/2][SEP0021A086D04D]:StationLineStatResMessage
sent to ephone (6 of 6)
May 17 19:23:48.435: ephone-2[1/2]:SkinnyCompleteRegistration
May 17 19:23:48.435: ephone-2[1/2][SEP0021A086D04D]:StationForwardStatReqMessage
line 6 from ephone
May 17 19:23:48.435: Skinny StationForwardStatMessage line 6 sent on ephone
socket [2] for ephone-2
May 17 19:23:48.435: activeForward 0 AllActive 0 BusyActive 0 NoAnswerActive 0
May 17 19:23:48.667: ephone-2[1/2]:MediaPathEventMessage
May 17 19:23:48.667: ephone-2[1/2]:MediaPathEventMessage
May 17 19:23:48.739: ephone-2[1/2][SEP0021A086D04D]:Skinny Available Lines
6 set for socket [2]
May 17 19:23:48.739: ephone-2[1/2]:Already done SkinnyCompleteRegistration
In the above debug output, there are several Invalid messages that state the following:
StationLineStatReqMessage from ephone line 2 Invalid
This same log message is repeated for lines 2 – 6 on a 7965 phone. This is because
the 7965 phone can have up to six line buttons, but only button 1 is configured for an
extension. The Invalid log essentially states that line buttons 2 – 6 are not usable.
Determining the State of an Ephone
Once your phones are configured and registered on your CUCM Express system, you’ll
want to familiarize yourself with the show ephone command, because it provides a wealth
of information that can prove very useful when troubleshooting. We’ll be going back
Using CUCM Express Verification and Troubleshooting Commands
to this show command throughout the book, but this is a good time to show how you
can determine the state of an Ephone registration process and the state of a configured
extension. First, let’s look at the different registration states you will see.
Ephone Registration States
An Ephone can be in three different states. Table 5.4 lists the states and what each
state means.
TA B L E 5 . 4
Ephone registration states
State
Meaning
REGISTERED
Indicates the phone is registered to the CUCM Express and is active.
UNREGISTERED
Indicates the phone unregistered normally form the CUCM Express
and is not active.
DECEASED
Indicates the phone unregistered abnormally because of a keepalive
timeout.
Let’s look at all three of these states by issuing the show ephone command:
Router#show ephone
ephone-1[0] Mac:0021.A086.D04D TCP socket:[-1] activeLine:0 DECEASED
mediaActive:0 offhook:0 ringing:0 ringRate: 0 reset:0 reset_sent:0 paging 0
debug:0 caps:9
IP:192.168.10.12 51055 7965 keepalive 8 max_line 6
button 1: dn 1 number 4001 CH1
DOWN
Preferred Codec: g711ulaw
ephone-2[1] Mac:0021.A02E.7D9A TCP socket:[5] activeLine:0 REGISTERED in SCCP
ver 12/9
mediaActive:0 offhook:0 ringing:0 ringRate: 0 reset:0 reset_sent:0 paging 0
debug:0 caps:9
IP:192.168.10.13 50271 7965 keepalive 6 max_line 6
button 1: dn 2 number 4002 CH1
IDLE
Preferred Codec: g711ulaw
ephone-3[2] Mac:001E.68E1.AFE9 TCP socket:[1] activeLine:0 UNREGISTERED
mediaActive:0 offhook:0 ringing:0 ringRate: 0 reset:0 reset_sent:0 paging 0
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debug:1 caps:8
IP:192.168.10.14 1556 CIPC keepalive 127 max_line 8
button 1: dn 1 number 4003 CH1
DOWN
Preferred Codec: g711ulaw
Three ephones are configured on this CUCM Express. Ephone-1 is in a DECEASED state,
which means that the CUCM Express has lost contact with the switch. The CUCM Express
uses keepalives to monitor the state of the phones. After six missed keepalive messages,
the phone is placed into a DECEASED state. This typically happens when a phone loses
power. Ephone-2 is in a REGISTERED state. This means that this phone is operational on
the network and is ready to make and receive calls. Lastly, ephone-3 is in an UNREGISTERED
state. This state means that the phone gracefully unregistered from the CUCM Express. We
can see what type of phone this is on line 3 where it says the phone hardware is CIPC. Given
that ephone-2 is a Cisco IP Communicator, the phone likely unregistered when the user
exited the application.
Ephone Extension States
A second piece of information that we can gain from the show ephone command is the state
of a phone extension. An ephone extension can have six ephone extension states. Table 5.5
provides a description of each of these states.
TA B L E 5 . 5
Ephone extension states
Ephone Registration
State
Description
N/A
UNREGISTRED /
DECEASED
Ephone registration is not
registered to the CUCM Express.
IDLE
On-hook
REGISTERED
Ephone is ready to make and
receive calls.
SEIZE
Off-hook
REGISTERED
Ephone handset has been picked
up, but no call has been made.
RINGING
Off-hook
REGISTERED
Ephone is calling another
extension.
ALERTING
On-hook
REGISTERED
Ephone is receiving a call from
another extension.
CONNECTED
Off-hook
REGISTERED
An active call is in progress
between two or more extensions.
State
On- or Off-hook
DOWN
Using CUCM Express Verification and Troubleshooting Commands
221
Let’s look at the show ephone command to see what each of the ephone extension states
looks like while we go through the process of ephone registration and call processing.
Ephone Extension DOWN State
The two following examples of ephone extensions show that the ephone registration
process is in either a DECEASED or an UNREGISTRED state for the extensions to be in a
DOWN state:
ephone-1[0] Mac:0021.A086.D04D TCP socket:[-1] activeLine:0 DECEASED
mediaActive:0 offhook:0 ringing:0 ringRate: 0 reset:0 reset_sent:0 paging 0
debug:0 caps:9
IP:192.168.10.12 51055 7965 keepalive 8 max_line 6
button 1: dn 1 number 4001 CH1
DOWN
Preferred Codec: g711ulaw
ephone-3[2] Mac:001E.68E1.AFE9 TCP socket:[1] activeLine:0 UNREGISTERED
mediaActive:0 offhook:0 ringing:0 ringRate: 0 reset:0 reset_sent:0 paging 0
debug:1 caps:8
IP:192.168.10.14 1556 CIPC keepalive 127 max_line 8
button 1: dn 1 number 4003 CH1
DOWN
Preferred Codec: g711ulaw
Ephone Extension IDLE State
A phone is ready to either make or receive calls when the extension is in an IDLE state. In
order for this to happen, the ephone must be properly REGISTERED to the CUCM Express,
as shown here:
ephone-1[0] Mac:0021.A086.D04D TCP socket:[1] activeLine:0 REGISTERED in SCCP
ver 12/9
mediaActive:0 offhook:0 ringing:0 ringRate: 0 reset:0 reset_sent:0 paging 0
debug:0 caps:9
IP:192.168.10.12 52084 7965 keepalive 0 max_line 6
button 1: dn 1 number 4001 CH1
IDLE
Preferred Codec: g711ulaw
Ephone Extension SEIZE State
When an end user on ephone-2 wishes to make a call, they pick up the handset of the
phone. As we know, this action changes the phone from an on-hook state to an off-hook
state. This is called a line seizure. When this happens, the show ephone command has the
ephone extension in an IDLE state, as shown here:
ephone-2[1] Mac:0021.A02E.7D9A TCP socket:[5] activeLine:1 REGISTERED in SCCP
ver 12/9
222
Chapter 5
CUCM Express Installation and Basic Configuration
mediaActive:0 offhook:1 ringing:0 ringRate: 0 reset:0 reset_sent:0 paging 0
debug:0 caps:9
IP:192.168.10.13 50271 7965 keepalive 16 max_line 6
button 1: dn 2 number 4002 CH1
SEIZE
Preferred Codec: g711ulaw
Active Call on DN 2 chan 1 :4002 0.0.0.0 0 to 0.0.0.0 0 via 0.0.0.0
G711Ulaw64k 160 bytes no vad
Tx Pkts 0 bytes 0 Rx Pkts 0 bytes 0 Lost 0
Jitter 0 Latency 0 callingDn -1 calledDn -1
Ephone Extension RINGING and ALERTING States
Let’s say that a user picks up a phone and dials an extension. Once that process reaches
the CUCM, the phone where the user called from is put into a RINGING state. At this
point the CUCM sends back the audible ringing tone through the phone handset to indicate
that the call is being processed, and the user is waiting for the handset of the called phone
to be picked up to complete the call. At the same time, the called phone goes into an
ALERTING state. In this state the called phone is on-hook but ringing to alert the end user
that someone is attempting to speak with them. The show ephone output looks like this:
ephone-1[0] Mac:0021.A086.D04D TCP socket:[1] activeLine:0 REGISTERED in SCCP
ver 12/9
mediaActive:0 offhook:0 ringing:1 ringRate: 0 reset:0 reset_sent:0 paging 0
debug:0 caps:9
IP:192.168.10.12 52084 7965 keepalive 1 max_line 6
button 1: dn 1 number 4001 CH1
RINGING
Preferred Codec: g711ulaw
call ringing on line 1
ephone-2[1] Mac:0021.A02E.7D9A TCP socket:[5] activeLine:1 REGISTERED in SCCP
ver 12/9
mediaActive:0 offhook:1 ringing:0 ringRate: 0 reset:0 reset_sent:0 paging 0
debug:0 caps:9
IP:192.168.10.13 50271 7965 keepalive 17 max_line 6
button 1: dn 2 number 4002 CH1
ALERTING
Preferred Codec: g711ulaw
Active Call on DN 2 chan 1 :4002 0.0.0.0 0 to 0.0.0.0 0 via 0.0.0.0
G711Ulaw64k 160 bytes no vad
Tx Pkts 0 bytes 0 Rx Pkts 0 bytes 0 Lost 0
Jitter 0 Latency 0 callingDn -1 calledDn 1
Summary
223
Ephone Extension CONNECTED State
The remote phone rings and the end user picks up the phone to answer it. At this point, the
CUCM places both phones into a CONNECTED state. You can also see in the show ephone
command that it lists the source and destination IP addresses.
ephone-1[0] Mac:0021.A086.D04D TCP socket:[1] activeLine:1 REGISTERED in SCCP
ver 12/9
mediaActive:1 offhook:1 ringing:0 ringRate: 0 reset:0 reset_sent:0 paging 0
debug:0 caps:9
IP:192.168.10.12 52084 7965 keepalive 2 max_line 6
button 1: dn 1 number 4001 CH1
CONNECTED
Preferred Codec: g711ulaw
Active Call on DN 1 chan 1 :4001 192.168.10.12 25848 to 192.168.10.13 23436 via
192.168.10.12
G711Ulaw64k 160 bytes no vad
Tx Pkts 219 bytes 37668 Rx Pkts 219 bytes 37668 Lost 0
Jitter 0 Latency 0 callingDn 2 calledDn -1
ephone-2[1] Mac:0021.A02E.7D9A TCP socket:[5] activeLine:1 REGISTERED in SCCP
ver 12/9
mediaActive:1 offhook:1 ringing:0 ringRate: 0 reset:0 reset_sent:0 paging 0
debug:0 caps:9
IP:192.168.10.13 50271 7965 keepalive 18 max_line 6
button 1: dn 2 number 4002 CH1
CONNECTED
Preferred Codec: g711ulaw
Active Call on DN 2 chan 1 :4002 192.168.10.13 23436 to 192.168.10.12 25848 via
192.168.10.13
G711Ulaw64k 160 bytes no vad
Tx Pkts 470 bytes 80840 Rx Pkts 468 bytes 80496 Lost 0
Jitter 0 Latency 0 callingDn -1 calledDn 1
Summary
Chapter 5 got us to the nuts and bolts of configuring the CUCM Express, which is a key
goal in the process of learning how to configure CUCM Express hardware and ultimately
passing the 640 - 460 exam. We began with Cisco’s licensing options to ensure that you
properly license your hardware and software. I then showed you how to download and
install both the Cisco IOS and CUCM Express software onto a Cisco voice- capable router.
224
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CUCM Express Installation and Basic Configuration
Once the software was up and running, we moved into the configuration phase of the
chapter. Here I gave you the four mandatory configuration steps required for the CUCM
Express to operate: configure the source IP address, configure the maximum number of
ephones and ephone-DNs allowed, identify and set the fi rmware load fi les that Cisco IP
phones should request, and generate and serve up default phone configuration fi les via
TFPT. From there, we set up basic ephones and ephone-DNs to the point where a call
could be made between two Cisco IP phones. You then saw how to set up and use the GUI
interface for administration of the voice system. Although the GUI is less efficient than
working from the command line, it allows you to delegate some tasks to less- experienced
administrators. Finally, you learned some troubleshooting steps and techniques to help you
resolve any CUCM configuration or registration issues you may see in your studies and in
day-to -day work.
Exam Essentials
Understand the three CUCM Express licenses. Cisco has three different licenses for
CUCM environments. One license is for the voice- capable IOS. The second is the CCME
Express feature license. The third is the individual user license.
Know how to download and install the voice - capable IOS and CUCM Express
software. On Cisco- capable router hardware, you must know how to download and
properly install the software. To do so, you must download compatible software from
cisco.com and either TFTP or use an external flash drive to transfer the software to the
router’s internal flash drive.
Know how to find CUCM Express software files on the router flash drive. When you
install the CUCM Express software using the archive tar /xtract command, the software is set up in a directory structure. It is important to know how to fi nd phone fi rmware
and other fi les by using the dir flash: commands.
Know how to configure your CUCM Express router as a TFTP server. TFTP is required
in a Cisco voice environment because Cisco phones request fi rmware and configuration
fi les from TFTP servers. You should know how to configure your CUCM Express router to
serve up the necessary fi les that are stored in flash.
Know how to configure the four mandatory CUCM Express system configuration settings
using the command line. The four mandatory configuration settings to get a CUCM
Express router ready for operation are source IP address, max ephones and ephone-DN,
fi rmware load fi les, and default configuration fi les.
Understand what the auto -assign configuration command does. Auto -assign allows you
to set up a pool of ephone-DNs. When Cisco IP phones connect to the CUCM Express
for the fi rst time, the auto -assign function registers the ephone. It maps an ephone-DN
taken from the pool to the MAC address of the phone. This functionality is a great way to
partially automate a phone rollout.
Written Lab 5.1
225
Understand the difference between the telephony-service restart and reset commands.
Restart is a quick reset of the phone. It is good to use when you make changes to the
configuration fi le, including changes to DNs, phone buttons, and speed dial. Reset is a
full boot of the phone. This command causes the phone to go through a DHCP renewal
process. It is also required when you change global parameters such as date/time, CUCME
source IP, and TFTP download path.
Understand how to configure basic CUCM Express parameters using the telephony-service
startup script. This script helps you to configure basic parameters by asking a series of
questions regarding the needs of the phone system. Once you have answered the questions,
the script automatically configures the telephony-service parameters.
Know how to navigate and administrate the CUCM Express system using the GUI
interface. The CUCM Express has a GUI interface that can be enabled to allow administrators to manage and monitor the voice system.
Know how to troubleshoot CUCM Express registration and extension states. Understand
how to best troubleshoot registration and extension problems using command line debug
and show commands.
Written Lab 5.1
Write the answers to the following questions:
1.
What privileged exec command do you use when you want to copy a file from TFTP to
the router’s internal flash drive?
2.
What privileged exec command do you use when you want to copy and extract CUCM
Express software that is compressed as a .tar file?
3.
What configuration command tells the CUCM Express to serve up the flash:/
phone/7945-7965/SCCP45.8-3-2-27.sbn file via TFTP?
4.
What config-telephony command sets the source IP address for the CUCM Express
system to 172.16.55.100?
5.
What config-telephony command sets the maximum number of ephones to 30?
6.
What config-telephony command sets the maximum number of ephone-DNs to 50?
7.
What config-telephony command tells the CUCM Express that when a 7945 phone
requests a firmware load file, the file offered is SCCP45.8-3-3S.loads?
8.
What configuration command begins the setup script process for the CUCM Express?
9.
What configuration command enables the HTTPS server?
10. What debug command can you use to troubleshoot TFTP server issues?
(The answers to Written Lab 5.1 can be found following the answers to the review
questions for this chapter.)
Chapter 5
226
CUCM Express Installation and Basic Configuration
Hands-on Labs
To complete the labs in this section, you need a CUCM Express router and two Cisco
IP phones. The phones used in this example are 7940s, but you can use any phone or IP
Communicator you wish. The labs will follow the logical network design shown in
Figure 5.16.
F I G U R E 5 .1 6
CUCM Express lab diagram
Ext. 444
Telephony Source IP:
192.168.10.1
CUCM Express
Ext. 555
These labs build on each other, so it is best to perform them in the order listed:
Lab 5.1: Configuring the CUCM Express as a TFTP Server
Lab 5.2: Configuring the CUCM Express for Basic Phone Operation
Lab 5.3: Enabling HTTP/HTTPS GUI Administration on the CUCM Express
Hands-on Lab 5.1: Configuring the CUCM
Express as a TFTP Server
In this lab, we are going to add 7940 phones to our voice network. In order for them to
work properly, we need to configure the CUCM Express router as a TFTP server to serve
up the fi rmware fi les that the 7940 phones require.
1.
Log into your CUCM Express router and go into privileged exec mode by
typing enable.
2.
Check to see which firmware files the 7940 phones need by viewing the files on the
flash drive. To do this, type dir flash:/phone/7940-7960. You should see something
similar to the following output:
Directory of flash:/phone/7940-7960/
97
98
99
100
-rw-rw-rw-rw-
129824
458
705536
130228
Apr
Apr
Apr
Apr
7
7
7
7
2009
2009
2009
2009
18:29:32
18:29:32
18:29:36
18:29:36
+00:00
+00:00
+00:00
+00:00
P00308000500.bin
P00308000500.loads
P00308000500.sb2
P00308000500.sbn
Hands-on Labs
227
3.
Enter into configuration mode by typing configuration terminal.
4.
Configure the CUCM Express router to serve up the 7940 firmware files. Note that
because the files are organized in a directory structure, you need to include the alias
command:
tftp-server
tftp-server
tftp-server
tftp-server
5.
flash:/phone/7940-7960/P00308000500.bin alias P00308000500.bin
flash:/phone/7940-7960/P00308000500.loads alias P00308000500.bin
flash:/phone/7940-7960/P00308000500.sb2 alias P00308000500.bin
flash:/phone/7940-7960/P00308000500.sbn alias P00308000500.bin
Exit configuration mode by typing end.
Hands-on Lab 5.2: Configuring the CUCM
Express for Basic Phone Operation
1.
Log into your CUCM Express router and go into privileged exec mode by typing
enable.
2.
Enter into config-telephony mode by typing configuration terminal and then
telephony-service.
3.
Configure the IP source address to the address given in the diagram by typing
ip source-address 192.168.10.1.
4.
Configure the maximum ephones to 5 and maximum ephone-DNs to 10 by typing
max-ephones 5 and then max-dn 10.
5.
Set the firmware load files for the 7940 phones by typing load 7960-7940 PPPPPPPP
.loads.
6.
Exit config-telephony mode by typing exit.
7.
Configure ephone-DN 1 to have the number 444 by typing ephone-dn 1 and then
number 444.
8.
Configure ephone-DN 2 to have the number 555 by typing ephone-dn 2 and then
number 555.
9.
Configure the MAC address of ephone 1 by typing ephone 1 and then mac-address
XXXX.XXXX.XXXX. Your MAC address will be unique.
10. Configure the MAC address of ephone 2 by typing ephone 2 and then mac-address
XXXX.XXXX.XXXX. Your MAC address will be unique.
11. Configure button 1 of ephone 1 to use ephone-DN 1 by typing ephone 1 and then
button 1:1.
12. Configure button 1 of ephone 2 to use ephone-DN 2 by typing ephone 2 and then
button 1:2.
13. Exit config- ephone mode by typing end.
Chapter 5
228
CUCM Express Installation and Basic Configuration
Hands-on Lab 5.3: Enabling HTTP/HTTPS GUI
Administration on the CUCM Express
1.
Log into your CUCM Express router and go into privileged exec mode by typing
enable.
2.
Enter into configuration mode by typing configuration terminal.
3.
Enable both HTTP and HTTPS services by typing ip http server and ip http
secure-server.
4.
Verify that your web files are in a directory structure by typing dir flash:/gui. The
output will look like this:
Directory of flash:/gui/
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
5.
-rw-rw-rw-rw-rw-rw-rw-rw-rw-rw-rw-rw-rw-rw-rw-rw-rw-
953
3845
647358
1029
174
16344
864
6328
4558
3724
76699
843
1347
2399
870
9968
3412
Apr
Apr
Apr
Apr
Apr
Apr
Apr
Apr
Apr
Apr
Apr
Apr
Apr
Apr
Apr
Apr
Apr
7
7
7
7
7
7
7
7
7
7
7
7
7
7
7
7
7
2009
2009
2009
2009
2009
2009
2009
2009
2009
2009
2009
2009
2009
2009
2009
2009
2009
18:26:06
18:26:06
18:26:10
18:26:12
18:26:12
18:26:12
18:26:12
18:26:14
18:26:14
18:26:14
18:26:16
18:26:16
18:26:16
18:26:18
18:26:18
18:26:18
18:26:20
+00:00
+00:00
+00:00
+00:00
+00:00
+00:00
+00:00
+00:00
+00:00
+00:00
+00:00
+00:00
+00:00
+00:00
+00:00
+00:00
+00:00
Delete.gif
admin_user.html
admin_user.js
CiscoLogo.gif
Tab.gif
dom.js
downarrow.gif
ephone_admin.html
logohome.gif
normal_user.html
normal_user.js
sxiconad.gif
Plus.gif
telephony_service.html
uparrow.gif
xml-test.html
xml.template
Set the CUCM Express router to look for web files inside the flash:gui/ directory by
typing ip http path flash:/gui. Or if your web files are not in a subdirectory, type
ip http path flash:/.
6.
Allow for local HTTP authentication by typing ip http authentication local.
7.
Enter into config-telephony mode by typing telephony-service.
8.
Create an admin account with the username of Adam and a secret password of
cisco123 by typing web admin system name Adam secret 0 cisco123.
9.
Exit config-telephony mode by typing end.
Review Questions
229
Review Questions
1.
Which of the following is not a CUCM Express license?
A. Cisco SCCP license
2.
B.
Cisco IOS license for voice capabilities
C.
CCME Express feature licenses
D.
Individual user license
When ordering new Cisco 7985 IP phones, which part number should you order if you
want the phones to be properly licensed for both CUCM Express and the bigger CUCM
solutions?
A. CP-7985- CH1
3.
B.
CP-7985- SCCP
C.
CP-7985=
D.
CP-7985- CCME
What is the correct Cisco IOS command-line syntax to TFTP a file to a Cisco voice router’s
internal flash?
A. put tftp: flash:
4.
B.
copy flash: tftp
C.
put flash: tftp:
D.
copy tftp: flash
Which IOS command is used when you want to uncompress CUCM Express software that
is stored as a .tar file?
A. uncompress tar /xtract
5.
B.
unzip tar /xtract
C.
archive tar /xtract
D.
uncompress file /xtract
What is the tftp-server IOS command used for?
A. To identify the IP address of the TFTP server
B.
To set option 150 for DHCP clients
C.
To identify files the router serves via TFTP
D.
To enable Secure FTP
Chapter 5
230
6.
CUCM Express Installation and Basic Configuration
Which of the following is not a mandatory CUCM Express setting?
A. Configure source IP address.
7.
B.
Configure auto -assign.
C.
Configure max ephones and ephone-DNs.
D.
Set the firmware load files for the phones.
E.
Generate and serve default phone configuration files.
What command is used to manually update the phone configuration load files on a CUCM
Express?
A. create firmware
8.
B.
create cnf-files
C.
load cnf-files
D.
load firmware
Which of the following places can the Cisco IP phone MAC address be found? Choose all
that apply.
A. On the box the Cisco phone shipped in
9.
B.
In the Settings menu of the Cisco IP phone
C.
On the back of the phone
D.
On the phone handset
Which command-line operation does a quick reset of all phones currently registered on a
CUCM Express system using a single command?
A. Router(config-telephony)#restart reset
B.
Router(config-ephone)#restart all
C.
Router(config-telephony)#restart all
D.
Router(config-ephone)#restart reset
E.
Router(config-ephone)#reset all
F.
Router(config-telephony)#reset
10. Besides power cycling a Cisco IP phone, how can you reset it using the keypad buttons?
A. Press **#**.
B.
Press the Help button and then **#**.
C.
Press the Directory button and then **#**.
D.
Press the Settings button and then **#**.
Review Questions
231
11. Why would you not want to set max-ephones and max-DN to the highest allowable number
for your hardware unless necessary?
A. The router CPU has to run additional processes, which can slow down the voice
system.
B.
The router allocates memory for each ephone and ephone-DN allocated. The router’s
memory can quickly fill up and cause performance problems.
C.
The router allocates flash storage for each ephone and ephone-DN allocated. This
space can fill up and cause performance problems.
D.
The router will log error messages until all ephones and ephone-DN allocations
are used.
12. Which four configuration steps are required using the IOS command line to enable the web
GUI interface?
A. Configure the enable password.
B.
Configure the enable secret password.
C.
Enable the HTTP/HTTPS server.
D.
Configure http local authentication.
E.
Configure the web administrator username and password.
F.
Configure TACACS+ authentication.
G. Set the web root directory.
13. In which GUI configuration menu would you add an ephone-DN?
A. Administration
B.
Configure
C.
Reports
D.
Voice Mail
14. In what tab can you perform the GUI equivalent to a “write memory” in the command line?
A. Administration
B.
Configure
C.
Reports
D.
Reset
15. When troubleshooting a Cisco phone that powers up and connects to the network but will
not register, what is the first logical thing to check?
A. Ensure that the proper firmware and configuration files are accessible to the phone.
B.
Ensure that the ephone is properly configured in the CUCM Express configuration.
C.
Make sure that the phone is receiving the correct IP address and other network
parameters through DHCP.
D.
Check to see if the clock is properly synchronized with NTP.
Chapter 5
232
CUCM Express Installation and Basic Configuration
16. On a Cisco phone, where can you verify that you received all the proper network parameters through DHCP?
A. Press the * button; then select Network Configuration.
B.
Press the Help button; then select Network Configuration.
C.
Press the Directory button; then select Network Configuration.
D.
Press the Settings button; then select Network Configuration.
17. What two commands are best used to troubleshoot when you believe your Cisco IP phone
is not getting the proper firmware file?
A. debug ephone register
B.
debug tftp events
C.
debug ip packet
D.
show telephony-service tftp-bindings
E.
debug tftp-bindings
18. When viewing show ephone output like the following, what does SEIZE mean on the
extension?
ephone-2[1] Mac:0021.A02E.7D9A TCP socket:[5] activeLine:1 REGISTERED in SCCP
ver 12/9
mediaActive:0 offhook:1 ringing:0 ringRate: 0 reset:0 reset_sent:0 paging 0
debug:0 caps:9
IP:192.168.10.13 50271 7965 keepalive 16 max_line 6
button 1: dn 2 number 4002 CH1
SIEZE
Preferred Codec: g711ulaw
Active Call on DN 2 chan 1 :4002 0.0.0.0 0 to 0.0.0.0 0 via 0.0.0.0
G711Ulaw64k 160 bytes no vad
Tx Pkts 0 bytes 0 Rx Pkts 0 bytes 0 Lost 0
Jitter 0 Latency 0 callingDn -1 calledDn -1
A. The phone is currently in a call.
B.
The phone is on-hook.
C.
The phone is off-hook and unregistered.
D.
The phone is off-hook.
E.
The phone is receiving a call.
Review Questions
233
19. When viewing show ephone output like the following, what does ALERTING mean on the
extension?
ephone-2[1] Mac:0021.A02E.7D9A TCP socket:[5] activeLine:1 REGISTERED in SCCP
ver 12/9
mediaActive:0 offhook:1 ringing:0 ringRate: 0 reset:0 reset_sent:0 paging 0
debug:0 caps:9
IP:192.168.10.13 50271 7965 keepalive 17 max_line 6
button 1: dn 2 number 4002 CH1
ALERTING
Preferred Codec: g711ulaw
Active Call on DN 2 chan 1 :4002 0.0.0.0 0 to 0.0.0.0 0 via 0.0.0.0
G711Ulaw64k 160 bytes no vad
Tx Pkts 0 bytes 0 Rx Pkts 0 bytes 0 Lost 0
Jitter 0 Latency 0 callingDn -1 calledDn 1
A. The phone is currently in a call.
B.
The phone is on-hook.
C.
The phone is calling another extension.
D.
The phone is receiving a call.
20. If your CUCM Express is configured with max-ephone 10, what happens when the 11th
phone is added to the network?
A. The new phone will register but will only be able to receive calls.
B.
The new phone will register, but the oldest registered phone will be forced to
unregister.
C.
The new phone will not be able to register.
D.
The new phone will register and a log message will be generated notifying the administrator that the max-ephone limit has been exceeded.
234
Chapter 5
CUCM Express Installation and Basic Configuration
Answers to Review Questions
1.
A. You need a license for the voice IOS, the CUCME software for a specific number of
endpoints, and the individual user licenses for endpoints.
2.
A. If you want to be properly licensed for both the CUCM Express and the bigger CUCM
voice solutions, you should order the CH1 part. This is a great idea if you think you might
upgrade from an Express solution to one of the bigger Cisco Communications Managers in
the future.
3.
D. The correct syntax to TFTP a fi le from a TFTP server to a voice router flash is copy
tftp: flash.
4.
C. The correct syntax to TFTP a file from a TFTP server to a voice router flash is
archive tar /xtract.
5.
C. The command is used to specify fi les that the router can serve to clients
using TFTP.
6.
B. The auto-assign configuration is not necessary for the operation of a CUCM Express system.
7.
B. You manually update configuration load fi les using the create cnf-files command.
8.
A, B, C . The MAC address can be found on the box, on the back of the phone, and within
the Settings menu of a powered-up phone.
9.
C . The restart all command within config-telephony mode performs a quick reset of all
registered phones.
10. D. To reset a phone locally, press the Settings button and then **#**.
11. B . The CUCM Express allocates a specified amount of memory for each
ephone and ephone-DN. This memory is essentially wasted if it is not needed. It
could possibly cause performance problems if your router requires that memory for
other uses.
12. C, D, E, G. You must enable the HTTP server, HTTPS server, or both and then set the
root directory. Also, you need to create a web administrator and password. This is the username/password used to authenticate to the GUI interface. Lastly, you must allow users to
log in using locally configured usernames and passwords.
13. B. You add, delete, and change ephone-DN configurations in the Configure menu tab.
14. A . You can save configuration changes on the GUI under the Administration menu by
selecting Save Router Config.
15. C . When a phone powers up and connects to the network, its fi rst task is to receive network parameters such as an IP address, gateway, subnet mask, and the option 150 parameter. If your phone is not receiving one or more of these, it will fail to properly register.
16. D. You can fi nd the Network Settings information by pressing the Settings button on a
Cisco IP phone.
Answers to Review Questions
235
17. B, D. You can use debug tftp events to see which fi les a Cisco phone is asking for. With
this information, you can then view which fi les are configured to be served via TFTP by
issuing a show telephony-service tftp-bindings command.
18. D. When a user picks up the phone handset, the phone goes into an off-hook state. This is
referred to as an extension seizure.
19. D. Alerting means that someone is trying to call that ephone-DN but the user has not yet
picked up the handset.
20. C. The phone will not register until another phone unregisters or max-ephone is changed to 11.
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CUCM Express Installation and Basic Configuration
Answers to Written Lab 1
1.
copy tftp: flash
2.
archive tar /xtract
3.
tftp-server flash:/phone/7945-7965/SCCP45.8-3-2-27.sbn alias
SCCP45.8-3-2-27.sbn
4.
ip source-ip 172.16.55.100
5.
max-ephones 30
6.
max-dn 50
7.
load 7945 SCCP45.8-3-3S.loads
8.
telephony-service setup
9.
ip http secure-server
10. debug tftp events
Chapter
6
CUCM Express
Advanced
Configuration
THE CCNA VOICE EXAM TOPICS
COVERED IN THIS CHAPTER INCLUDE
THE FOLLOWING:
Implement Cisco Unified Communications Manager
Express to support endpoints using CLI.
Configure call-transfer per design specifications.
Configure voice productivity features including hunt groups,
call park, call pickup, paging groups, and paging/intercom.
Configure Music on Hold.
Perform basic maintenance and operations tasks to
support the VoIP solution.
Describe basic troubleshooting methods for Cisco Unified
Communications Manager Express.
Whenever I go to buy a new car, the first things I look
at are the basics. How reliable is the car? How much legroom is
there? Does it get decent gas mileage? Does the value
hold up for resale? Answering these questions will give me the basic functionality that I
desire in my next automobile. This is similar to what we’ve done up to this point in
configuring our CUCM Express. We have the basics nailed down, so we can make and
receive calls. Once that’s done, we need to start thinking about the bells and whistles. Call
parking, hunt groups, intercom, Music on Hold—all these phone features are similar to
added features of a car such as a six-disc CD changer, satellite radio, and built-in Bluetooth.
While they may not be absolutely necessary, they’re great to have! In Chapter 6 we cover
many of the most popular configuration features on the CUCM Express.
Chapter 6 will begin by showing the difference in ephone-DN and ephone configuration
between key-system and PBX models. We’ll also cover the various phone button separator
options. Next, you’ll learn how to configure telephony service features that assist in tailoring
the phone system to meet the geographical and business environment where the system and
its users reside. Then you’ll learn how to configure advanced voice productivity features that
provide additional functionality to your users. After that, we’ll look at how to configure
some of the access and accounting features available on the CUCM Express. Finally, you’ll
learn how to configure Music on Hold in both unicast and multicast environments.
Configuring Key System
and PBX DNs and Ephones
A key system is commonly used in small businesses where the vast majority of calls come
from the PSTN. The key-system model is based on one extension being shared among many
phones. Alternatively, a PBX system uses a model of one extension per phone. This section
will show typical DN and ephone configurations for each setup so you can not only configure
these models yourself but also identify the configuration differences between the two.
Configuring Key Systems
In a key-system environment, you historically see the entire PSTN extension configured on
the line instead of a truncated four- or five-digit extension. Furthermore, all phones must
be capable of answering any call. This means that all the ephone-DNs will be configured
as buttons on every phone. This is known as a shared line. One way of creating this
shared-line setup is to configure a single ephone-DN and apply it on multiple ephones.
Configuring Key System and PBX DNs and Ephones
239
The following key system example configuration shows two phone DNs that represent two
separate external PSTN phone numbers. The DNs are assigned to both phones, and both
will ring when the number is dialed. The fi rst phone to answer gets the call.
Router#configure terminal
Router(config)#ephone-dn 1
Router(config-dn)#number 5555552121
Router(config-dn)#ephone-dn 2
Router(config-dn)#number 5555559191
Router(config-dn)#exit
Router(config)#ephone 1
Router(config-ephone)#button 1:1 2:2
Router(config)#ephone 2
Router(config-ephone)#button 1:1 2:2
Router(config-ephone)#end
Router#
Figure 6.1 shows what ephone 1 looks like after these configurations are made and the
phone is reset.
F I G U R E 6 .1
One DN, multiple ephones
Let’s say that a phone call is placed to 555-555-2121. Both ephone 1 and ephone 2
would ring. If ephone 2 answers the call fi rst, line 1 of ephone 1 shows this line as in use
by lighting the extension button red and showing the double-handset icon next to the line
number. Figure 6.2 shows line 1 of ephone 1 in use.
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FIGURE 6.2
CUCM Express Advanced Configuration
DN 1 in use
Because line 1 is in use, if ephone 1 needs to make a call, it must use line 2, which is
currently not in use.
An alternative shared-line method is to configure multiple ephone DNs with the same
extension number. You can then configure the ephones to use the separate ephone-DN
configurations. You set a preference on the ephone-DN configuration so one particular
phone will always ring fi rst. If the preferred ephone is busy, then the ephone with the next
preference will ring instead. In the CUCM, preferences range from 0 to 9, with the lowest
number being the first preference. We can accomplish this configuration, multiple ephoneDNs with a shared line, using the preference configuration command. The following
configuration example shows how to configure two ephone-DNs with a single phone number.
You can see that ephone-DN 1 has a preference of 0, which means that when a call is made
to this extension, it will choose to ring the phone that is configured to use ephone-DN 1 first:
Router#configure terminal
Router(config)#ephone-dn 1
Router(config-ephone-dn)#number 5555557777
Router(config-ephone-dn)#preference 0
Router(config-ephone-dn)#ephone-dn 2
Router(config-ephone-dn)#number 5555557777
Router(config-ephone-dn)#preference 1
Router(config-ephone-dn)#exit
Configuring Key System and PBX DNs and Ephones
241
Router(config)#ephone 1
Router(config-ephone)#button 1:1
Router(config-ephone)#ephone 2
Router(config-ephone)#button 1:2
Router(config-ephone)#end
Router#
If you did not configure a preference on the ephone - DNs or you set them
to be the same, the CUCM would round -robin the calls between the two
ephones. The preference command gives you control of where the CUCM
Express routes calls.
If ephone 1 is in use, any new call will also be sent to ephone 1 because it is the lowest
preferred DN regardless of whether the phone is busy or not. So a second call placed to our
extension would receive a busy signal, and ephone 2 would never receive any calls. To get
around this problem, we configure ephone-DN 1 with the no huntstop command. The
no huntstop command tells the CUCM Express that it should look for the next preferred
ephone-DN if the most preferred phone is busy. Now when ephone 1 is busy, a second call
placed on the shared extension will roll over and ring ephone 2:
Router#configure terminal
Router(config)#ephone-dn 1
Router(config-ephone-dn)#number 5555557777
Router(config-ephone-dn)#preference 0
Router(config-ephone-dn)#no huntstop
Router(config-ephone-dn)#ephone-dn 2
Router(config-ephone-dn)#number 5555557777
Router(config-ephone-dn)#preference 1
Router(config-ephone-dn)#exit
Router(config)#ephone 1
Router(config-ephone)#button 1:1
Router(config-ephone)#ephone 2
Router(config-ephone)#button 1:2
Router(config-ephone)#end
Router#
Another shared-line key system configuration we need to look at is when the phone
extensions are configured as dual-line and octo -line DNs. So far, we’ve configured only
single-line phones. A single-line phone can make or receive only one call at a time. So if
the line is already in use, you cannot place the call on hold to make a second call. Likewise,
if line 1 is in use, a second phone call to the extension will receive a busy signal.
Dual - line phones, on the other hand, allow the phone to place calls on hold or receive
a second call when in use. And octo - line phones are capable of handling up to eight
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simultaneous calls on a single phone button extension. Dual- and octo -line phones are
configured within the ephone-DN as shown here:
Router(config)#ephone-dn 1 ?
dual-line dual-line DN (2 calls per line/button)
octo-line octo-line DN (8 calls per line/button)
Configuring ephone-DNs with dual lines is extremely beneficial because it allows
additional functionality when your phone is in use. For now, let’s assume that our small
business has a single PSTN line that is to be shared between two phones configured with
dual-line ephone-DNs. Just as in the previous configuration example, we want to ensure
that the fi rst call made to the extension is received on ephone 1 and that a second call rolls
over to ephone 2 if ephone 1 is already in a call. Let’s say we configure the following:
Router#configure terminal
Router(config)#ephone-dn 1 dual-line
Router(config-ephone-dn)#number 5555557777
Router(config-ephone-dn)#preference 0
Router(config-ephone-dn)#no huntstop
Router(config-ephone-dn)#ephone-dn 2 dual-line
Router(config-ephone-dn)#number 5555557777
Router(config-ephone-dn)#preference 1
Router(config-ephone-dn)#exit
Router(config)#ephone 1
Router(config-ephone)#button 1:1
Router(config-ephone)#ephone 2
Router(config-ephone)#button 1:2
Router(config-ephone)#end
Router#
In this situation, the first call will always go to ephone 1. But because the ephone-DN is
configured as a dual line, a second call will also go to ephone 1. Only a third simultaneous
call will make it to ephone 2. To get around this dual-line problem, we can use the huntstop
channel command on ephone-DN 1. The huntstop command prevents calls from hunting
for the second channel of the ephone-DN. So if we combine the no huntstop command with
the huntstop channel command, the first call always goes to ephone 1, and if channel 1 of
ephone 1 is busy, the second call is sent to ephone 2. Here is the full configuration example
to accomplish our goal:
Router#configure terminal
Router(config)#ephone-dn 1 dual-line
Router(config-ephone-dn)#number 5555557777
Router(config-ephone-dn)#preference 0
Router(config-ephone-dn)#no huntstop
Router(config-ephone-dn)#huntstop channel
Configuring Key System and PBX DNs and Ephones
243
Router(config-ephone-dn)#ephone-dn 2 dual-line
Router(config-ephone-dn)#number 5555557777
Router(config-ephone-dn)#preference 1
Router(config-ephone-dn)#exit
Router(config)#ephone 1
Router(config-ephone)#button 1:1
Router(config-ephone)#ephone 2
Router(config-ephone)#button 1:2
Router(config-ephone)#end
Router#
Additional phone button options also expand the shared-line experience of key systems.
In particular, you’ll learn about overlay buttons in the “Configuring Ephone Button
Options” section of this chapter.
Configuring PBX Systems
PBX systems are more commonly found in larger office environments and assign a unique
phone extension to every phone. This allows a caller to reach a specific person within an
organization directly. Also, because of the size of the environment, a large percentage of
phone calls are within the network. To help make life easier for the phone users, phone
extensions are used instead of the full phone number. Typical extensions are four or five
digits in length. These digits may correspond to the last digits of the full PSTN direct
inward dial (DID) if that is how their PSTN circuits are configured. Also, you will find that
the phones almost always are configured as dual-line ephone-DNs. This is because you
need a second line to enable the additional functionality the PBX system offers. In the last
section, you learned how to configure the most common key-system methods of sharing a
single phone number with multiple phones. Here is a very basic and common method of
configuring two PBX system phones with separate extension numbers:
Router#configure terminal
Router(config)#ephone-dn 1 dual-line
Router(config-ephone-dn)#number 8001
Router(config-ephone-dn)#ephone-dn 2 dual-line
Router(config-ephone-dn)#number 8002
Router(config-ephone-dn)#exit
Router(config)#ephone 1
Router(config-ephone)#button 1:1
Router(config-ephone)#ephone 2
Router(config-ephone)#button 1:2
Router(config-ephone)#end
Router#
Now you have two phones with separate extensions. You can then configure the CUCM
Express system for additional features to tailor your system to your environment. The
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remainder of this chapter details how to configure the most common advanced voice
features and options you’ll fi nd in use today.
Configuring Ephone Button Options
When it’s time to assign ephone-DNs to specific ephones, we use the button ephone- config
command. The separator between the line button you wish to configure and the ephoneDN identifier is an ephone button separator. Many button separator options are available.
Let’s look at all the options:
Router(config-ephone)#button ?
LINE button-index:dn-index pairs example 1:2 2:5
Configuration line:button with separator feature options:
: normal phone lines
example
button 1:2 2:5
s silent ring, ringer muted, call waiting beep muted
example
button 1s2 2s5
b silent ring, ringer muted, call waiting beep not muted
example
button 1b2 2b5
f feature ring
example
button 1f2 2f5
see also ‘no dnd feature-ring’
m monitor line, silent ring, call waiting display suppressed
example
button 1m2 2m5
see also ‘transfer-system full-consult dss’
w watch line (BLF), watch the phone offhook status via the phone’s primary
ephone-dn
example
button 1w2 2w5
o overlay lines, combine multiple lines per physical button
example
button 1o2,3,4,5
c overlay call-waiting, combine multiple lines per physical button
example
button 1c2,3,4,5
see also ‘huntstop channel’ for ephone-dn dual-line
x expansion/overflow, define additional expansion lines that are
used when the primary line for an overlay button is
occupied by an active call
Expansion works with ‘button o’ and not with ‘button c’
example
button 4o21,22,23,24,25
button 5x4
button 6x4
Different separator options may be use for each button
example
button 1:2 2s5 3b7 4f9 5m22 6w10
Table 6.1 details the function of each button separator.
Configuring Key System and PBX DNs and Ephones
TA B L E 6 .1
245
Button separator options
Separator
Option Name
Function
:
Normal ring
Phone rings normally with default ring tone. Also uses
flashing lights on line button and headset lamp to indicate
ring.
s
Silent ring
No audible ring when calls come into the phone. Uses
flashing lights on line button and headset lamp to indicate
ring. No audible call-waiting beep.
b
Silent with beep
No audible ring when calls come into the phone. Uses
flashing lights on line button and headset lamp to indicate
ring. Call-waiting beep is audible.
f
Feature ring
Phone rings using an alternate ring tone from the default.
m
Monitor line
Used to monitor status (on- or off-hook) of a line.
Commonly used on receptionist phones to see if an
employee is currently using the phone. No audible ring
when calls come into the phone, and the line cannot be
used to make or take calls.
w
Watch phone
Similar to the monitor mode except that it allows the user
to monitor all ephone-DNs on a phone instead of a single
ephone-DN. This mode presents a more accurate picture
of user availability than using the m separator.
o
Overlay line
Associates multiple ephone-DNs with a single line button.
No call-waiting functionality.
c
Overlay with call
waiting
Same as the overlay line but with call-waiting functionality
added.
x
Expansion line
Another overlay line option. The difference is that if the
line button extension is in use, new calls are allowed to
overflow to additional line buttons to help prevent a busy
signal.
The ring phone button options (:, s, b, and s) are fairly straightforward and need no
more explanation. We’ll focus on when you would want to use the monitor and overlay
button options.
Monitor Line
Let’s say you have an administrative assistant who is tasked with taking your calls and
transferring them to your phone when you are not busy with other calls. The monitor
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CUCM Express Advanced Configuration
line (m) button separator option allows the assistant’s phone to monitor your ephone-DN
and see if you are currently on a call using that ephone-DN. If you are already busy on the
line, the assistant can take a message for you. The line configured in monitor mode cannot
make or receive any calls. Instead, it is a visual aid to see if another line is being used. In
this example, my phone is assigned the number 4040. My administrative assistant has
their own number, 4041, assigned to button 1. Also configured is button 2 to monitor my
ephone-DN.
Router(config)#ephone-dn 1
Router(config-ephone-dn)#number 4040
Router(config-ephone-dn)#ephone-dn 2
Router(config-ephone-dn)#number 4041
Router(config-ephone-dn)#exit
Router(config)#ephone 1
Router(config-ephone)#button 1:1
Router(config-ephone)#ephone 2
Router(config-ephone)#button 1:2 2m1
Router(config-ephone)#end
Router#
Now when I pick up my phone to make a call, my administrative assistant can see that
I’m busy on that ephone-DN. Figure 6.3 shows the administrative assistant ’s phone when
the 4040 line is in use.
FIGURE 6.3
A phone configured to monitor extension 4040
Configuring Key System and PBX DNs and Ephones
247
One of the drawbacks to this setup would become apparent if my phone were to be
configured with multiple ephone-DNs. I would then need to create multiple monitor button
operators for each extension. A way around the monitor line limitation is to use the watch
phone button separator.
Watch Phone
The watch phone (w) button option does exactly the same thing as the monitor line option
with the exception that it monitors all of the ephone-DNs of an ephone instead of just one.
You configure the button to watch the primary line of a phone, and it monitors all lines on
the phone. This is far more useful than the monitor line option because you can see if any
of the lines on a phone are in use instead of just a single ephone-DN. As with the monitor
line option, a line configured with the watch phone option cannot make or receive any
calls. We’ll use the boss-and-assistant scenario again for this example. The boss has two
extensions (5111 and 5112) configured on lines 1 and 2. The administrative assistant’s
phone is configured to use extension 5113 on button 1. Button 2 is then configured to
watch the primary ephone-DN 1, as shown here:
Router(config)#ephone-dn 1
Router(config-ephone-dn)#number 5111
Router(config-ephone-dn)#ephone-dn 2
Router(config-ephone-dn)#number 5112
Router(config-ephone-dn)#ephone-dn 3
Router(config-ephone-dn)#number 5113
Router(config-ephone-dn)#exit
Router(config)#ephone 1
Router(config-ephone)#button 1:1 2:2
Router(config-ephone)#ephone 2
Router(config-ephone)#button 1:3 2w1
Router(config-ephone)#end
Router#
With this command, when the boss uses either ephone -DN 1 or 2, the administrative
assistant’s button 2 will show the line as in use. The watching phone’s display button shows
the phone in use when the following conditions occur on the watched phone:
Off-hook and/or in use
Unregistered or deceased phone
In DnD (do not disturb) mode
Overlay Line
Overlay (o) lines allow you to configure multiple ephone-DNs to a single phone button
on a Cisco phone. Cisco phones have a fi nite number of phone buttons. You can use the
overlay button option to assign multiple ephone-DNs to a single physical phone button.
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CUCM Express Advanced Configuration
Ephone-DNs that are configured on a particular ephone with the overlay option must
all be single-line or dual/octo -line phones. There cannot be a mix of single- and
multiline phones.
An overlay line is commonly used when a main line is answered by anyone in a specific
department. This overlay shared-line configuration is best paired with the preference and
no huntstop commands. In the next example, we have a department with two employees.
Each employee has a unique extension for their phone. There is also a shared-line number
(5454) that is configured as an overlay line on button 1. When we configure the ephoneDN, we make sure to configure the unique extension fi rst. The fi rst ephone configured
is the number that is displayed on the phone display LCD panel. The overlay line is
configured, but that number is never seen on the phone button display. The shared line is
configured on ephone-DN 3 and ephone-DN 4. Ephone-DN 3 has the lower preference and
will handle the fi rst call. It also is configured to use the no huntstop command to
look for the next-preferred ephone-DN with the same extension if the most-preferred phone
is busy.
Router#configure terminal
Router(config)#ephone-dn 1
Router(config-ephone-dn)#number 6001
Router(config-ephone-dn)#ephone-dn 2
Router(config-ephone-dn)#number 6002
Router(config-ephone-dn)#ephone-dn 3
Router(config-ephone-dn)#number 5454
Router(config-ephone-dn)#preference 0
Router(config-ephone-dn)#no huntstop
Router(config-ephone-dn)#ephone-dn 4
Router(config-ephone-dn)#number 5454
Router(config-ephone-dn)#preference 1
Router(config-ephone-dn)#exit
Router(config)#ephone 1
Router(config-ephone)#button 1o1,3,4
Router(config-ephone)#ephone 2
Router(config-ephone)#button 1o2,3,4
Router(config-ephone)#end
Router#
Each phone’s button 1 is configured with its unique number as well as the shared-line
number for the department. Calls placed to 6001 go only to ephone 1. Calls placed to 6002
go only to ephone 2. But calls placed to 5454 are sent to both phones. The configuration
uses only one phone button on each phone. Now other buttons are available to be
configured for additional lines or speed-dial capabilities if desired. Following is the output
of show ephone for our two configured ephones. As you can see, the fi rst number assigned
in the overlay configuration is bound to the phone and idle. The shared number is visible
but not the primary number.
Configuring Key System and PBX DNs and Ephones
249
Router#sh ephone
ephone-1 Mac:0021.A084.4F0C TCP socket:[3] activeLine:0 REGISTERED in SCCP ver
12/8
mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0 caps:9
IP:192.168.10.2 49242 7965
keepalive 11 max_line 6
button 1: dn 1 number 6001 CH1
IDLE
overlay
overlay 1: 1(6001) 3(5454) 4(5454)
ephone-2 Mac:0021.A02E.7D9A TCP socket:[4] activeLine:0 REGISTERED in SCCP ver
12/8
mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0 caps:9
IP:192.168.10.3 49219 7965
keepalive 11 max_line 6
button 1: dn 2 number 6002 CH1
IDLE
overlay
overlay 1: 2(6002) 3(5454) 4(5454)
Let’s say a call is placed to extension 5454, and ephone 2 answers the call. Now a show
ephone looks like this:
Router#sh ephone
ephone-1 Mac:0021.A084.4F0C TCP socket:[3] activeLine:0 REGISTERED in SCCP ver
12/8
mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0 caps:9
IP:192.168.10.2 49242 7965
keepalive 14 max_line 6
button 1: dn 1 number 6001 CH1
IDLE
overlay
overlay 1: 1(6001) 3(5454) 4(5454)
ephone-2 Mac:0021.A02E.7D9A TCP socket:[4] activeLine:1 REGISTERED in SCCP ver
12/8
mediaActive:1 offhook:1 ringing:0 reset:0 reset_sent:0 paging 0 debug:0 caps:9
IP:192.168.10.3 49219 7965
keepalive 14 max_line 6
button 1: dn 3 number 5454 CH1
CONNECTED
overlay shared
overlay 1: 2(6002) 3(5454) 4(5454)
Active Call on DN 3 chan 1 :5454 192.168.10.3 27418 to 192.168.1.100 24646 via
192.168.10.3
G711Ulaw64k 160 bytes no vad
Tx Pkts 196 bytes 33712 Rx Pkts 192 bytes 33024 Lost 0
Jitter 7 Latency 0 callingDn 5 calledDn -1
At this point, ephone-DN 3, which is number 5454, is owned and controlled by
ephone 2. A second call is made to 5454; this time, ephone-DN 3 is in use, so the call rolls
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over to the next ephone-DN, which is 4. Because ephone 2 is configured with an overlay
with both ephone-DNs 3 and 4, the phone rings on ephone 2. A show ephone with both
ephone-DNs 3 and 4 in use looks like this:
ephone-1 Mac:0021.A084.4F0C TCP socket:[3] activeLine:1 REGISTERED in SCCP ver
12/8
mediaActive:1 offhook:1 ringing:0 reset:0 reset_sent:0 paging 0 debug:0 caps:9
IP:192.168.10.2 49242 7965
keepalive 19 max_line 6
button 1: dn 4 number 5454 CH1
CONNECTED
overlay shared
overlay 1: 1(6001) 3(5454) 4(5454)
Active Call on DN 4 chan 1 :5454 192.168.10.2 27274 to 192.168.1.101 24648 via
192.168.10.2
G711Ulaw64k 160 bytes no vad
Tx Pkts 0 bytes 0 Rx Pkts 0 bytes 0 Lost 0
Jitter 0 Latency 0 callingDn 5 calledDn -1
ephone-2 Mac:0021.A02E.7D9A TCP socket:[4] activeLine:1 REGISTERED in SCCP ver
12/8
mediaActive:1 offhook:1 ringing:0 reset:0 reset_sent:0 paging 0 debug:0 caps:9
IP:192.168.10.3 49219 7965
keepalive 19 max_line 6
button 1: dn 3 number 5454 CH1
CONNECTED
overlay shared
overlay 1: 2(6002) 3(5454) 4(5454)
Active Call on DN 3 chan 1 :5454 192.168.10.3 26148 to 192.168.1.100 24640 via
192.168.10.3
G711Ulaw64k 160 bytes no vad
Tx Pkts 738 bytes 126936 Rx Pkts 736 bytes 126592 Lost 0
Jitter 2 Latency 0 callingDn 6 calledDn -1
As you can see, this shared-line overlay configuration is a very good option in many
office environments. It also highlights a combination of PBX and key-system capabilities on
the CUCM Express. Installations that combine both PBX and key-system functionality are
commonly called hybrid systems.
Overlay with Call Waiting
This button separator option, c, is the same as the overlay, except that it adds call-waiting
functionality. Call waiting is the ability for a phone to receive two or more simultaneous
calls at the same time. The user can place a currently engaged call on hold to answer the
second call. To see this difference, we will configure our CUCM Express router with
the same configuration as the overlay example except we will use the call-waiting button
separator option. We’ll also have to configure ephone-DN 3 as a dual-line phone so it can
utilize call waiting:
Configuring Key System and PBX DNs and Ephones
251
Router#configure terminal
Router(config)#ephone-dn 1
Router(config-ephone-dn)#number 6001
Router(config-ephone-dn)#ephone-dn 2
Router(config-ephone-dn)#number 6002
Router(config-ephone-dn)#ephone-dn 3 dual-line
Router(config-ephone-dn)#number 5454
Router(config-ephone-dn)#preference 0
Router(config-ephone-dn)#no huntstop
Router(config-ephone-dn)#ephone-dn 4
Router(config-ephone-dn)#number 5454
Router(config-ephone-dn)#preference 1
Router(config-ephone-dn)#exit
Router(config)#ephone 1
Router(config-ephone)#button 1c1,3
Router(config-ephone)#ephone 2
Router(config-ephone)#button 1c2,3
Router(config-ephone)#end
Router#
So what are the results of this configuration? The fi rst call to extension 5454 is
handled by ephone-DN 3 because of its lower preference. A second call rolls over to
ephone-DN 4, because the no huntstop option was set. Ephone-DN 4 rings ephone 1, but
it also sends the call-waiting beep to ephone 2, which is currently in a call. This way, the
user on ephone 2 is notified of a second call. He or she can place the fi rst call on hold and
answer the second.
Expansion Line
The expansion button separator, x, is used to expand line coverage for an overlay
button (o). It does not work when the overlay separator button is configured for call
waiting (c). When the extensions configured as overlay lines are in use, the expansion lines
begin taking calls. In this example, we have ephone 1 configured to overlay ephone-DNs 1
and 2, which are both 7001. Ephone-DN 1 is also a dual-line phone. We also have button 2
configured as an overlay for line 1 on the phone:
Router#configure terminal
Router(config)#ephone-dn 1 dual-line
Router(config-ephone-dn)#number 7001
Router(config)#ephone-dn 2
Router(config-ephone-dn)#number 7001
Router(config-ephone-dn)#exit
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Router(config)#ephone 1
Router(config-ephone)#button 1o1,2 2x1
Router(config-ephone)#end
Router#
In this example, the fi rst call to 7001 goes to button 1. The second call also goes to
button 1, because it is a dual line and channel 2 is free. The third call will overflow
to button 2 because both lines are busy on button 1. Always remember that overflow lines
will be used only when all other lines are occupied.
Configuring Telephony Service Features
You’ll configure most telephony service features while in config-telephony mode. These
features provide multiple ways to tailor your voice environment to better fit the needs of
your end users. This section will show you how to configure several of the most important
telephony service features. We’ll look at how to change the language and ring tone
settings to match the location where your endpoints will reside. I’ll also show you how
to modify the date and time formats and modify the phone handset system message to
personalize your voice system. Lastly, you’ll learn how to create a local directory to assist
users in looking up phone extensions.
How to Configure User Locale and Network Locale
By default, the Cisco CUCM Express is set for the English (US) language for its location.
What happens if you need to deploy this system in say, Colombia, where Spanish is the
native language? To modify the language used on the Cisco phone handsets, including
softkeys, help, and other buttons, we can use the user-locale command. Let’s see what
language options are currently available:
Router(config-telephony)#user-locale ?
<0-4> user locale index 0 to 4 (0 is default)
DE
Germany
DK
Denmark
ES
Spain
FR
France
IT
Italy
JP
Japan
NL
Netherlands
NO
Norway
PT
Portugal
RU
Russian Federation
Configuring Telephony Service Features
SE
US
253
Sweden
United States
Using our Colombian deployment example, we’ll choose ES for our locale, so Spanish
will be displayed on our handsets:
Router#configure terminal
Router(config)#telephony-service
Router(config-telephony)#user-locale ES
Updating CNF files
CNF files update complete
Please issue ‘create cnf’ command after the locale change
Router(config-telephony)#create cnf-files
CNF file creation is already On
Updating CNF files
CNF files update complete
Whenever we make changes to the configuration of a telephone, we will need to reset the
phone in order to obtain all of the updated configuration and settings as manipulated.
Figure 6.4 shows a screenshot of the settings menu now that we’ve changed the language
and reset our phones.
FIGURE 6.4
Spanish phone settings
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The network-locale command modifies tones and cadence differences between
geographic regions. Unlike the user-locale, which changes language functions of the
phones, the network-locale settings are based on regional telephone standards in terms of
telephone signaling. Using our Colombia deployment example, we can use ES for the userlocale because Colombians speak the same language as Spaniards. The network-locale
settings differ, however, because each region has different tones within its geographic
regions:
Router(config-telephony)#network-locale ?
<0-4> network locale index 0 to 4 (0 is default)
AT
Austria
CA
Canada
CH
Switzerland
CO
Colombia
DE
Germany
DK
Denmark
ES
Spain
FR
France
GB
United Kingdom
IT
Italy
JP
Japan
NL
Netherlands
NO
Norway
PT
Portugal
RU
Russian Federation
SE
Sweden
US
United States
Router(config-telephony)#network-locale CO
Updating CNF files
CNF files update complete
Please issue ‘create cnf’ command after the locale change
Router(config-telephony)#create cnf-files
CNF file creation is already On
Updating CNF files
CNF files update complete
If you need to configure a user-locale and network-locale that are not
currently in your CUCM Express software, you can download the individual
user-locale .tar files here:
http://www.cisco.com/cgi-bin/tablebuild.pl/CME-Locale
Configuring Telephony Service Features
Can You Translate This for Me?
Jeff was an IT consultant who recently began installing CUCM Express solutions in
businesses. All of his implementations up to this point had been for local businesses in
the United States, where English is the dominant language. A recent client, however,
called for a Canadian deployment. Some employees had English as their primary
language and others had French. In addition, the company regularly had visits from
consultants from Spain, which required a third language. Since Jeff was new to the
language-localization features of the CUCM Express, he had to do a bit of research to
figure out the best configuration method to provide the three different language options
to users. He learned that if the CUCM Express is going to be in a mixed-language
environment, his best option was to configure user-locale and network-locale ephone
templates. This is an example of how the ephone templates were used to remedy this
situation:
Router#configure terminal
Router(config)#telephony-service
Router(config-telephony)# user-locale 1 ES
Router(config-telephony)# user-locale 2 FR
Router(config-telephony)# network-locale 1 ES
Router(config-telephony)# network-locale 2 FR
Router(config-telephony)#ephone-template 1
Router(config-ephone-template)# user-locale 1
Router(config-ephone-template)# network-locale 1
Router(config-ephone-template)#ephone-template 2
Router(config-ephone-template)# user-locale 2
Router(config-ephone-template)# network-locale 2
Router(config-ephone-template)#ephone 1
Router(config-ephone)# button 1:1
Router(config-ephone)#ephone 2
Router(config-ephone)# button 1:2
Router(config-ephone)# ephone-template 1
Router(config-ephone)#ephone 3
Router(config-ephone)# button 1:3
Router(config-ephone)# ephone-template 2
Router(config-ephone)#exit
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Router(config)#telephony-service
Router(config-telephony)#create cnf-files
CNF file creation is already On
Updating CNF files
CNF files update complete
Router(config-telephony)#restart all
This method sets up a very simple and streamlined way to configure ephones that fits
the needs of the local user. Note that by default, the English (US) locale is configured if
you do not specify a template. So, for example, ephone 1 is for English-speaking users
because there is no ephone template 1 or 2 specified.
Configuring the Date and Time Format
Similar to user-locale is the date and time format. Different countries display the date
and time differently. In the United States, the date is displayed as mm/dd/yy. In other
regions, such as Europe, the date is displayed as dd/mm/yy. The default format is mm/
dd/yy. If you wish to change the format on your Cisco IP phones, you use the date-format
command. You can specify the following formats:
Router(config-telephony)#date-format ?
dd-mm-yy Set date to dd-mm-yy format
mm-dd-yy Set date to mm-dd-yy format
yy-dd-mm Set date to yy-dd-mm format
yy-mm-dd Set date to yy-mm-dd format
Let’s change the date format to the European dd/mm/yy:
Router#configure terminal
Router(config)#telephony-service
Router(config-telephony)#date-format dd-mm-yy
Router(config-telephony)#end
Router#
Now when we reset our phones, we get the date to display with the day fi rst.
See Figure 6.5.
Configuring Telephony Service Features
FIGURE 6.5
257
A modified date format
Configuring the System Message
The CUCM Express system message is a custom-display text message that appears on
certain Cisco IP phones with large displays, such as the 7940 and 7960 grayscale displays
or the 7945 and 7965 color displays. The system message defaults to “Cisco Unified CME”
by default.
We can modify the default message with the system message config-telephony
command. The next example shows how to change the message to “ACME Incorporated”:
Router#configure terminal
Router(config)#telephony-service
Router(config-telephony)#system message ACME Incorporated
Router(config-telephony)#end
Router#
The system message does not need a reset or restart of the phones because this is
updated every time the phone receives a keepalive message from the CUCM Express.
Figure 6.6 shows the modified message on our phone.
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FIGURE 6.6
CUCM Express Advanced Configuration
A customized system message
The maximum length of a system message on a CUCM Express is
34 characters.
Configuring a Local Directory
The CUCM Express local directory is like a built-in phone book. The directory can be
used to search for and locate extensions based on caller-ID information that is configured
on the system. Configuration of the local directory takes place in config- ephone-DN and
config-telephony modes. Let’s look at our configuration options to see how all the pieces fit
together to make a custom-tailored directory to meet all your needs.
Configuring Caller-ID Ephone-DN Entries
When you configure an ephone, the name configuration command configures caller ID for
this extension. For example, let’s configure ephone-DN 1 with the extension 4001 and the
caller ID John Smith:
Configuring Telephony Service Features
259
Router#configure terminal
Router(config)#ephone-dn 1 dual-line
Router(config-ephone-dn)#number 4001
Router(config-ephone-dn)#name John Smith
Router(config-ephone)#end
Router#
When we assign this ephone-DN to an ephone and make a call from that ephone using
the 4001 extension, we see that the caller-ID information comes through on the called
phone, as shown in Figure 6.7.
F I GU R E 6.7
A caller-ID display
The name config- ephone-DN mode command also automatically enters this name into
the local directory. You can access the local directory using the Directory button located
on the Cisco phone to see the display shown in Figure 6.8.
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FIGURE 6.8
CUCM Express Advanced Configuration
The local directory in default first-name-first order
As you can see, by default, the directory displays names with fi rst name showing fi rst.
This may not be the way you want to have your directory listed. You can change this by
going into config-telephony mode and using the directory last-name-first command to
switch the listings around. Now all of your directory names will be listed alphabetically by
last name, as shown in Figure 6.9.
Configuring Manual Local Directory Entries
and System-Level Speed Dial
You can also add directory listings directly using the directory entry config-telephony
mode command. For example, let’s add a directory entry for Branch Chicago using this
method:
Router#configure terminal
Router(config)#telephony-service
Router(config-telephony)#directory entry 1 3125552777 name Branch Chicago
Router(config-telephony)#end
Router#
Configuring Voice Productivity Features
FIGURE 6.9
The local directory in last-name-first order
You can configure up to 100 manual directory entries (numbered 1–100). Directory
entries listed within the 34 –99 range are eligible to be configured as systemwide speeddial entries if desired.
Configuring Voice
Productivity Features
Voice productivity features are phone features that enhance the user’s calling experience.
In this section, you will learn how to configure the following productivity features:
Call forward
Call transfer
Call pickup
Call parking
Hunt groups
Intercom
Paging
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Call Forwarding
If you want to be able to answer your phone extension but are located somewhere else,
you can use the call forward voice productivity feature to direct all of your calls to a
different phone. There are two different types of call forwarding on the CUCM Express.
One method is to forward all calls on the Cisco phone handset itself. We’ll refer to this as
dynamic call forwarding. The second method is to configure forwarding within the Cisco
IOS. This is referred to as static call forwarding. Let’s look at how to set up both methods.
Setting Up Dynamic Call Forwarding
To forward all incoming calls using the handset, you can press the CFwdAll softkey on the
phone, enter the phone number you want to forward calls to, and then press either the End
softkey or the # button. Figure 6.10 shows that extension 204 has been forwarded to the
phone with the extension 201.
F I G U R E 6 .1 0
Call forwarding implemented on an IP phone
You can see that the icon in the upper-right corner next to the extension has changed
to indicate that this phone has been forwarded. To stop the phone from forwarding calls,
press the CFwdAll softkey again.
Setting Up Static Call Forwarding
Unlike dynamic call forwarding, static call forwarding allows for more options than to simply
forward all incoming calls. It is important to note that static forwarding can be overridden
Configuring Voice Productivity Features
263
using the CFwdAll softkey on Cisco phones. There are five ways to configure static call
forwarding on an ephone-DN. Table 6.2 lists each configuration option and its function.
TA B L E 6 . 2
Static call-forwarding options
Option
Description
All
Forward all incoming calls
Busy
Forward calls only when phone is busy
night-service
Forward calls only when CUCM Express is in night-service active
time mode
Noan
Forward calls after a specified amount of time when the phone has
not been answered
As an example, suppose we want our extension at 204 to be forwarded to extension 201
when we do not answer it after five rings. In the United States, a ring consists of a 4 -second
audible tone followed by 4 seconds of silence. Using this as our guide, we can configure our
extension to forward calls after 30 seconds without answer:
Router#configure terminal
Router(config)#ephone-dn 4
call-forward noan 201 timeout ?
<3-60000> Ringing no answer timeout duration in seconds
Router(config-ephone-dn)#call-forward noan 201 timeout 30
Router(config-ephone-dn)#end
Router#
The call-forward all and call-forward busy options are fairly self- explanatory.
The call-forward night-service option is useful only when the after-hours night service
functionality is configured on your CUCM Express. Night-service is a way of having calls
routed directly to voice mail during specified periods of time. It is typically used in keysystem configurations. Once it is properly configured, you can configure call forwarding to
take advantage of the night-service configuration.
One additional static call-forwarding configuration needs to be addressed.
The call-forward max-length option sets a maximum number of digits
that end users can enter for forwarding. This option helps to prevent
employees from forwarding calls to numbers that would ultimately incur
long- distance charges.
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Call Transfer
Call transfer is the process of moving an active phone conversation from one phone number
to another. On Cisco phones, you perform this process by pressing the Trnsfer softkey and
dialing the number where you wish to forward the call. Figure 6.11 shows a phone call in
progress with the Trnsfer key option available for use.
F I G U R E 6 .11
The result of pressing the Transfer (call transfer) softkey
There are three ways to configure transfer on the CUCM Express within configtelephony mode. Table 6.3 lists the three call transfer options available.
TA B L E 6 . 3
Call transfer options
Option
Description
full-blind
Transfers the call immediately after entering a forward number.
Available on single-line ephone-DNs.
full-consult
Allows you to speak to the transfer number party prior to forwarding
the phone conversation. This setup requires dual-line ephone-DNs.
local-consult
Similar in functionality to the full-consult option, but the handling of
voice traffic flow is inefficient. This is a Cisco proprietary method and
should be used only for backward compatibility with older Cisco phones.
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Most of your implementations will be configured for either full-blind or full-consult.
For single-line ephone-DNs, the only option is full-blind because consult transfers
require the use of a second line. On all dual- and octo-line ephone-DNs, the default
transfer method is full-consult. If the phone that is transferring is configured as a dualline ephone-DN but the second line is not available, the transfer method falls back to
full-blind.
You can also configure either a blind or consult transfer on each individual ephone-DN.
This example shows ephone-DN 1 being configured for blind transfers:
Router#configure terminal
Router(config)#ephone-dn 1
Router(config-ephone-dn)#transfer-mode ?
blind
Perform blind call transfers (without consultation) using single
phone line
consult Perform call transfers with consultation using second phone line if
available
Router(config-ephone-dn)#transfer-mode blind
Router(config-ephone-dn)#end
Router#
The CUCM Express has a built-in toll-fraud protection that is enabled by default. Unless
otherwise configured, calls can be transferred only to local on-network numbers on the
system. If you want to allow users to transfer calls off network, you must use the configtelephone transfer-pattern command to specify which dial strings are acceptable. The
permitted numbers can be specific configurations that must exactly match. For example, let’s
configure a transfer pattern for the CEO to be able to transfer calls to his home number at
555-332-3112:
Router#configure terminal
Router(config)#telephony-service
Router(config-telephony)#transfer-pattern 5553323112
Router(config-telephony)#end
Router#
This number is now the only off-network number allowed to be forwarded.
You can also configure off-network numbers using standard CUCM wildcard
characters. Wildcards let you configure multiple allowable off-network numbers without
having to specify every allowable number. These wildcards are used in multiple different
configuration options so it is very important that you properly identify each option.
Table 6.4 lists the wildcard options and their use.
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TA B L E 6 . 4
CUCM Express Advanced Configuration
Telephone number wildcard options
Option
Description
.
Matches any number 0–9 or the * key.
[ ]
A range of single number digits within the brackets. For example, [1-3]
means the number can be 1, 2, or 3. Commas can also be used to indicate
nonconsecutive numbers. For example, [1,3] means the number can
either be 1 or 3. Finally, the carat (^) is used to indicate that the digit is not
a number listed in the brackets. For example, [^1-3] means the number
is anything other than 1, 2, or 3.
( )
Indicates a pattern. This can be used in conjunction with the ?, %, or +
wildcard option.
?
The preceding digit occurs 0 or 1 time.
%
The preceding digit occurs either 0 or more times.
+
The preceding digit occurs either 1 or more times.
T
An inter-digit timeout. This essentially tells the router to pause to allow
time to collect additional digits. By default, the system will wait 10
seconds or until the # key is pressed.
Table 6.5 lists telephone number wildcard examples to demonstrate how each option
is used.
TA B L E 6 . 5
Telephone number wildcard examples
Number Pattern
Description
555…….
Matches a number that must begin with 555. The last 7 digits can be
any number.
555[3-5]……
Matches a number that must begin with 555. The next digit can be 3, 4,
or 5. The final 6 digits can be any number.
312(555)?….
Matches a number that must begin with 312. The 555 may be used
either 0 or 1 time. The final 4 digits can be any number.
011T
Matches a number that must begin with 011. The router than waits
for up to 32 digits to be entered before moving on. If the user stops
dialing, a timer (10 seconds by default) must complete before the router
processes the digits.
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Getting back to our transfer pattern example, let’s assume that we want our users to be
able to transfer calls to any number within the North American Numbering Plan. Here is
the best way to set up the transfer pattern using wildcards:
Router#configure terminal
Router(config)#telephony-service
Router(config-telephony)#transfer-pattern [2-9][0-8].[2-9]......
Router(config-telephony)#end
Router#
As you learned in Chapter 2, the NANP has a three-digit area code. Every area code must
start with a digit between 2 and 9. The second digit can be within 0 and 8. The last digit of
the area code can be any number. The first digit of the office code can be any number between
2 and 9, while all the rest of the office codes and station codes can be any number.
No More Missed Calls
Steve is the data center manager for a midsize ISP. He spends the vast majority of
his time inside the data center monitoring equipment to ensure stability. Steve often
spends time on the phone there talking to server and network engineers. One of his biggest
complaints about the phone is that he often cannot hear the call-waiting beep on the line
over the noise of the constantly running fans that keep network and server gear cool.
By default, call waiting is enabled on SCCP phones that are configured as dual/octo-line
phones or with multiple ephone-DNs configured. When the user is on the phone and a second
call comes through, the user will hear a subtle beep through the handset speaker indicating
the second call. Users nearly always want this setup, so no configuration changes are required.
In a data center or other noisy environment, a call-waiting beep may not be loud enough
to be heard. In this case, you may want to alter the call-waiting tone so it will actually ring
like a second call coming in. To accomplish this, you can configure specific ephones to
alert using an external ring rather than a beep through the handset. Here is an example of
how to configure ephone-DN 10 to use a call-waiting ring:
Router#configure terminal
Router(config)#ephone-dn 10
Router(config-ephone-dn)# call-waiting ring
Router(config-ephone-dn)#end
Router#
After this configuration change, when a second call comes into Steve’s phone in the data
center, the phone will ring, which is much more likely to be heard.
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Call Pickup
Call pickup is the process of answering a remote extension on your local phone. For
example, suppose ephone 10 is configured with extension 210 and it begins to ring. You
happen to be sitting at ephone 9, which is configured for extension 209. Using the
PickUp softkey button, you can answer extension 210 on ephone 9. You simply press
the PickUp key on your Cisco phone and enter the extension you want to answer (210 in
our example). The call would be rerouted to ephone 9 and use the 209 extension line. All
of this functionality is enabled by default on the CUCM Express. If you choose to disable
this feature, you can issue a no service directed-pickup command within
config-telephony mode.
You can also get fancy with call pickup by creating pickup groups to enhance the user
experience for users who commonly answer each other’s phones. To do so, we use the
pickup-group command in config- ephone-DN mode. This example shows how to configure
two different pickup groups with the IDs of 5110 and 5111:
Router#configure terminal
Router(config-ephone-dn)#ephone-dn 10
Router(config-ephone-dn)#pickup-group
Router(config-ephone-dn)#ephone-dn 11
Router(config-ephone-dn)#pickup-group
Router(config-ephone-dn)#ephone-dn 12
Router(config-ephone-dn)#pickup-group
Router(config-ephone-dn)#ephone-dn 13
Router(config-ephone-dn)#pickup-group
Router(config-ephone-dn)#end
Router#
5110
5110
5111
5111
Now, instead of using the PickUp softkey, you can use the GPickUp group softkey. After
pressing the softkey, you enter the pickup -group identification number.
Call Parking
Call parking lets you place a phone call in “parked” state using an unassigned ephone-DN
and then resume the call from any Cisco phone on the CUCM Express. Call park is essentially
is the same functionality as hold except that with hold, the call can be resumed only from the
local phone. Parking the call gives you more flexibility as to where the call is resumed from.
As stated previously, an unused ephone-DN must be configured and specifically
designated as a call parking space. In this example, we configure ephone-DNs 30 and 31 as
call parking spaces:
Router#configure terminal
Router(config)#ephone-dn 30
Router(config-ephone-dn)#number 3030
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Router(config-ephone-dn)#name Parking 1
Router(config-ephone-dn)#park-slot
Router(config-ephone-dn)#ephone-dn 31
Router(config-ephone-dn)#number 3031
Router(config-ephone-dn)#name Parking 2
Router(config-ephone-dn)#park-slot
Router(config-ephone-dn)#end
Router#
Figure 6.12 shows a Cisco phone that placed a call into parking slot with the extension
of 3030. Now that the call has been placed into call park, the recipient can resume the call
from any CUCM Express phone by pressing the PickUp softkey and then entering the 3030
extension number.
F I G U R E 6 .1 2
Call-parked IP phone
There are several other park-slot options that we should look at. Here are the options
available to us:
Router(config-ephone-dn)#park-slot ?
reserved-for Reserve this park slot for the exclusive use of the phone with
the extension indicated by the transfer target extension number
timeout
Set call park timeout
<cr>
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Use the reserved-for option if you want to have a parking space reserved for a specific
extension. No other extensions will be able to use this parking slot.
The timeout option allows you to configure parking slot timer values and notification
options when timers expire. These options are to help prevent people from parking calls
and then forgetting about them. That’s not the best way to treat your callers!
Hunt Groups
Hunt groups refer to the setup of a single phone number (the pilot number) that is
answered by one of several extensions that take turns ringing fi rst. The extension that
receives the next call to the pilot number depends on the hunt group selection algorithm
that is configured. Let’s fi rst look at the components of a hunt group, and then we’ll
learn how to configure them on the CUCM Express. Table 6.6 describes the hunt group
components.
TA B L E 6 . 6
Hunt group components
Component
Description
Hunt group tag
The tag to differentiate multiple hunt groups on the CUCM Express.
Pilot number
The ephone-DN that is dialed to reach a hunt group. A secondary DN
can also be configured using the secondary keyword.
Algorithm type
The algorithm method used to select which phone in the hunt group
should ring next. The options are longest-idle, peer, and sequential.
Member list
The ephone-DNs that belong to the hunt group.
Hops
The number of extensions that the algorithm will try to ring before going
to the final number. This is a valid command for longest idle and peer
algorithms.
Timeout
The number of seconds an extension in the hunt group will ring before
moving to the next extension in the group.
Final number
The number that is tried last after the number of hops has been
exceeded.
Hunt groups are configured by using the ephone-hunt configuration command. In
addition to the key command, you also specify the hunt group tag and algorithm type.
The tag can be any number between 1 and 100. The three different algorithm types are
described in Table 6.7.
Configuring Voice Productivity Features
TA B L E 6 . 7
271
Hunt group algorithms
Algorithm
Descriptions
Longest idle
Rings the phone in the hunt group that has been idle the longest.
Sequential
Rings the extensions in the exact order they were configured. Once it
gets to the end of the list, it dials the configured final number.
Peer
A circular algorithm where the first number tried is configured directly
to the right of the last number attempted.
As an example, we’ll configure hunt group 10 for pilot number 5001. This pilot has four
members with the extensions 201, 202, 203, and 204. We want to use the circular peer
method of choosing extensions to ring. An extension should ring for 20 seconds before
the call moves to the next extension in the list. Finally, we want to try up to three group
members before contacting the manager at extension 205:
Router#configure terminal
Router(config)#ephone-hunt 10 peer
Router(config-ephone-hunt)#pilot 5001
Router(config-ephone-hunt)#list 201 202 203 204
Router(config-ephone-hunt)#hops 3
Router(config-ephone-hunt)#timeout 20
Router(config-ephone-hunt)#final 205
Router(config-ephone-hunt)#end
Router#
Using this example, we’ll pretend that a call is placed into 5001, and it happens that
the fi rst number to ring is 203. That extension will ring for 20 seconds, and then the call
will try extension 204. That phone will ring for 20 seconds, and then the call will wrap
around the list and ring 201. Finally, if 201 does not answer the call, the maximum number
of hops is reached and the call will be forwarded one last time to extension 205. This
extension will continue to ring until someone answers it.
Intercom
Intercoms are mostly found in key-system CUCM Express systems, but they can also be
used in key-system/PBX hybrid setups. The intercom feature is essentially a speed dial with
automatic answer on the speakerphone. The speakerphone auto -answer on the called party
is in a mute state. The user must unmute the phone to be heard on the other end.
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Intercoms must have at least two phones configured for the service to work. Two new
ephone-DNs need to be configured. To ensure that these extensions are not accidentally
dialed by other phones, it is highly recommended that the number use a digit that cannot
be dialed using a phone handset. The letters a, b, c, and d can be used here. These letters
are DTMF priority and override tones that come in handy when you don’t want anyone
accidentally dialing this extension.
Once the ephone-DNs are configured using the intercom command and have extension
numbers, they need to be assigned to buttons on the ephones you want to have intercom
capability. In our example, we’ll configure ephone-DNs 15 and 16. One ephone-DN will
have the extension A900 and the other will be for A901. The intercom feature will be
configured on each ephone-DN to be able to use the intercom between the two extensions.
We’ll then assign the intercom extensions to button 2 of ephones 1 and 2:
Router#configure terminal
Router(config)#ephone-dn 15
Router(config-ephone-dn)#number A900
Router(config-ephone-dn)#intercom A901
Router(config-ephone-dn)#ephone-dn 16
Router(config-ephone-dn)#number A901
Router(config-ephone-dn)#intercom A900
Router(config-ephone-dn)#exit
Router(config)#ephone 1
Router(config-ephone)#button 2:15
Router(config-ephone)#ephone 2
Router(config-ephone)#button 2:15
Router(config-ephone)#end
Router#
We need to discuss a few key intercom configuration options because they might be
useful depending on how your users want to use the intercom voice-productivity feature.
Let’s look at the options:
Router(config-ephone-dn)#intercom A900 ?
barge-in
Allow intercom calls received on this DN to force other calls
into the call HOLD state to allow the incoming intercom call
to immediately connect without waiting
label
Define a text label for the intercom
no-auto-answer Disable intercom auto-answer
no-mute
Disable intercom mute-on-answer
<cr>
The barge-in option is useful when the intercom extension on the remote end is
already in progress. When you use this command, the current call will be placed on
hold and the new intercom call will connect.
Configuring Voice Productivity Features
The label option allows you to configure a name for the intercom to identify the
calling party.
The no-auto-answer option forces the called user to answer the intercom call
manually.
As stated earlier, the default behavior for intercom calls is to auto-answer the call
on the remote end and put it into mute state. The no-mute option performs the auto answer without putting the call into the muted state.
273
Paging
Paging is a one-way intercom. The best way to think about its use is at the grocery store.
When you accidentally drop a jar of spaghetti sauce, your action will quickly be followed
by an audible page from a store employee: “Cleanup on aisle 3.” Whereas the intercom is
typically to one phone, the page is a broadcast to multiple or possibly all phones. You can
configure paging groups to determine which ephone-DNs will receive a specific page.
A phone can be assigned to only one paging ephone - DN. However, you
can assign paging groups to include multiple ephone - DNs.
Paging can be configured as either unicast or multicast. Unicast configurations are
limited to a maximum of 10 receiving devices because of the inefficiencies and high
overhead of transmitting 10 separate streams containing the same information. A better
method is to configure multicast paging, which is much more efficient and scales well
beyond 10 receiving devices. You must also configure a UDP port number for your
multicast stream. The default UDP port for paging is 2000.
Multicast addressing IP space is designated in the range of 224.0.0.0
to 239.255.255.255. However, all CUCM Express capabilities including
multicast paging and Music on Hold exclude the 224.x.x.x range from use.
You must also ensure that your network is properly configured for multicast
routing prior to implementing any CUCM Express multicast features.
Configuration of multicast routing is outside the scope of this book.
Our fi rst paging example will configure a single ephone-DN to be used as our paging
extension of 5555. Pages will be multicast on 239.1.1.100 to ephones 1, 2, and 3:
Router#confure terminal
Router(config)#ephone-dn 1
Router(config-ephone-dn)#number 5555
Router(config-ephone-dn)#paging ip 239.1.1.100 port 2000
Router(config-ephone-dn)#exit
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Router(config)#ephone 1
Router(config-ephone)#paging-dn 1 multicast
Router(config-ephone)#ephone 2
Router(config-ephone)#paging-dn 1 multicast
Router(config-ephone)#ephone 3
Router(config-ephone)#paging-dn 1 multicast
Router(config-ephone)#end
Router#
Now that our paging group is configured, a user simply picks up the phone and dials
extension 5555 to automatically have a connection to ephones 1, 2, and 3 over their
speakerphones!
With our second paging configuration example, we’ll configure three different paging
ephone-DNs, labeled 1, 2, and 3. The numbers will be 5555, 5556, and 5557. We’ll use
different multicast IPs to ensure there is no overlap. Ephones 1 and 2 will be part of paging
group DN 1. Ephones 3 and 4 will be part of paging group DN 2. Finally, all four ephones
will be part of paging group DN 3: Here’s how we configure this setup:
Router#confure terminal
Router(config)#ephone-dn 1
Router(config-ephone-dn)#number
Router(config-ephone-dn)#paging
Router(config)#ephone-dn 2
Router(config-ephone-dn)#number
Router(config-ephone-dn)#paging
Router(config)#ephone-dn 3
Router(config-ephone-dn)#number
Router(config-ephone-dn)#paging
Router(config-ephone-dn)#exit
Router(config)#ephone 1
Router(config-ephone)#paging-dn
Router(config-ephone)#ephone 2
Router(config-ephone)#paging-dn
Router(config-ephone)#ephone 3
Router(config-ephone)#paging-dn
Router(config-ephone)#ephone 4
Router(config-ephone)#paging-dn
Router(config-ephone)#end
Router#
5555
ip 239.1.1.100 port 2000
5556
ip 239.1.1.101 port 2000
5557
group 1 2
1 multicast
1 multicast
2 multicast
2 multicast
Make note of the way we use the paging group ephone-DN configuration command
to add DNs 1 and 2 to DN 3 so that all four ephones will be paged. This allows us to get
around the limitation that an ephone can be configured with only a single paging-dn tag.
Configuring Voice Access and Accounting Features on the CUCM Express
275
Configuring Voice Access
and Accounting Features
on the CUCM Express
This section will cover how to configure voice access and accounting features on the
CUCM Express. First you’ll see how to defi ne the voice access that specific phones will
have to specific phone numbers. Different environments require different levels of voice
access restrictions. You’ll learn how to use different configuration techniques to limit access
to virtually any phone number based on time, date, and dial string.
The accounting portion of this section will show how you can log calls directly on the
CUCM Express and how to offload this information to a server for long-term storage. It
is very important to learn how to keep historical records of calls coming to and from your
phone system. This section shows you how to accomplish this task.
Call Blocking
There’s no doubt that certain phones should be configured to block specific destination
phone numbers. Not everyone should be allowed to dial long-distance and international
numbers. And it is likely that you’ll want to restrict all phones from being able to connect
to 900 and 976 numbers. On the Cisco CUCM Express, we can use the after-hours
config-telephony mode commands for call blocking based on time and dialing number.
There are also methods to exempt phones and users from call-blocking rules that are in
place. This way, you can have a great deal of flexibility in your calling plan to restrict calls
that should not be made in the fi rst place from office phones. Let’s fi rst look at the different
call-blocking options available, and then we’ll discuss various ways to exempt phones and
users from these call-block rules.
Blocking Calls by Date and Time (Toll Bar)
A toll bar allows you to block calls from being made on specific dates and/or during certain
hours of the day. To accomplish this, you must fi rst set the dates and times when the toll
bar is in place. Second, you need to configure dial-string patterns of the numbers you
choose to disallow. If a user dials a number that matches both the date/time and the
dial-string pattern, the call is not processed. Instead, the user hears a fast busy tone for
10 seconds and then the phone will go on-hook.
The syntax required to configure call blocking is to use the after-hours configtelephony mode commands. To give you an idea of how call blocking works, here is an
example of how we can block all long-distance and international calls on the weekend
beginning Friday at 6:00 pm. and on Christmas day:
Router#confure terminal
Router(config)#telephony-service
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Router(config-telephony)# after-hours
Router(config-telephony)# after-hours
Router(config-telephony)# after-hours
Router(config-telephony)# after-hours
Router(config-telephony)# after-hours
Router(config-telephony)# after-hours
Router(config-telephony)#end
Router#
day Fri 18:00 23:59
day Sat 00:00 23:59
day Sun 00:00 23:59
date Dec 25 00:00 23:59
block pattern 1 91..........
block pattern 2 9011T
As you can see, you can set the date/time by using either the after-hours day or afterhours date command. The time 00:00 23:59 is considered to be one entire day. Even
though 23:59 is listed, the seconds are automatically included, so it’s read by the router as
0:00:00 to 23:59:59. The after-hours block pattern command has a tag associated with
it. The tag is a numerical number to differentiate the different block patterns. These
tags can be between 1 and 100. When you create the block patterns, you can use
wildcards to help cover larger blocks of numbers with the fewest commands. These are
the same telephone number wildcards shown in Table 6.5 earlier in this chapter.
Configuring a Global Override Code
If you want certain people to be able to override any call-blocking functions from any
phone on the network, you can configure an override code using the after-hours
override-code config-telephony mode command. Here’s an example of the command
where a user can use softkeys to enter a PIN of 1234 to override the call blocks configured
previously:
Router#confure terminal
Router(config)#telephony-service
Router(config-telephony)#after-hours override-code 1234
Router(config-telephony)#end
Router#
Now a user who knows the override code can use any phone to dial either long-distance
or international destinations and use the Override softkey to fi rst enter the 1234 PIN
(personal identification number) that allows the override to occur. Once successfully logged
in using the override code, the user can make calls that would previously be blocked.
Configuring Override PIN per Ephone
Override PINs can also be created on an individual ephone. To do so, you use the pin
string config- ephone command. This example shows that the PIN of 4321 is used to
override the after-hours configurations we created earlier:
Router#confure terminal
Router(config)#ephone 1
Configuring Voice Access and Accounting Features on the CUCM Express
277
Router(config-ephone)# pin 4321
Router(config-telephony)#end
Router#
Now, any user who knows the PIN to this ephone can override the after-hours call
blocking by fi rst pressing the Login soft key and then entering the PIN. The user can then
dial any number that would previously have been blocked.
Configuring Auto Exempt Ephone
A third method is to completely exempt an ephone from the call-blocking setups. Now
users who have access to an ephone do not have to enter any sort of PIN to be able to make
long-distance and international calls. To accomplish this, we use the config- ephone mode
after-hours exempt command. Here’s how to configure this on ephone 1:
Router#confure terminal
Router(config)#ephone 1
Router(config-ephone)# after-hour exempt
Router(config-ephone)#end
Router#
Configuring Global Call Block
You will want to be able to block some numbers 24/7 to everyone, including phones with
override and exempt status. For these cases we can use the 7-24 option when configuring
the after-hours block command. In this example, we’re going to block absolutely
everyone from ever dialing 900 and 976 numbers on the phone system.
Router#confure terminal
Router(config)#telephony-service
Router(config-telephony)#after-hours block pattern 3 91900 7-24
Router(config-telephony)#after-hours block pattern 4 91976 7-24
Router(config-telephony)#after-hours block pattern 5 91...976 7-24
Router(config-telephony)#end
Router#
Call Detail Records
One accounting feature that most businesses wish to implement is the call detail record
(CDR). This feature keeps track of all calls made on the CUCM Express system. That way,
a business can keep track of who is calling whom, not only for budgeting but also to track
fraud and for emergency reasons. The CDR logs phone source and destination numbers
for both on- and off-network calls. This information can be stored in the router’s internal
logging buffer memory or on an external logging server. It can also be configured to be
sent to both. At a minimum, you should have this information sent to a syslog server. This
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is because the log buffer on the router is not a permanent solution. Eventually it will run
out of memory, and the oldest log data will be overwritten. Also, if the voice router loses
power, all logs in the buffer are lost. The next two examples will show how to configure
the CDR to use the router’s internal buffer memory and then an external syslog server.
Configuring CDR to the Internal Buffer Memory
If we want to enable logging of call history to the buffer, we fi rst must enable the buffer.
Logging to memory is disabled by default. To enable this feature, we use the logging
buffered command to turn on logging, using a portion of the router’s memory. You can
allocate various amounts of memory depending on how many logs you wish to keep. By
default, logs can use 4 KB of memory. You likely will want to set this to something higher.
The following example shows how to enable logging to the buffer and allocates 32 KB of
memory to this task.
Router#configure terminal
Router(config)#logging buffered 32768
Router(config)#end
Router#
At this point, the log buffer is set. The only problem is that the default number of call
records and the time that the records are stored in the buffer are very low. To change these
settings, you use the dial-control-mib command. The mib portion of the command stands
for Management Information Base. An MIB is an International Standards Organization
(ISO) standard format for databases used to manage communications devices. Because our
CUCM Express is a communications device, Cisco decided to use this standards-based
format for logging to the buffer. Here are the size and timer options as shown on the
CUCM Express system:
Router#configure terminal
Router(config)#dial-control-mib ?
max-size
Specify the maximum size of the dial control history table
retain-timer Specify timer for entries in dial control history table
Router(config)#dial-control-mib max-size ?
<0-500> Number of entries in the dial control history table
Router(config)#dial-control-mib retain-timer ?
<0-35791> Time (in minutes) for removing an entry
Because the CDR is a database, we must set a max-size that the database can be. The
maximum number of dial control history entries that can be saved depends on hardware.
By default, it is only 50 entries. The retain-timer option sets the maximum amount of
time the database will be kept on the router’s memory. This time is specified in minutes and
again is hardware dependent because it depends on the amount of memory the router has.
Configuring Voice Access and Accounting Features on the CUCM Express
279
By default, the timer is set for only 15 minutes. In our example, we will change the defaults
to more reasonable numbers. We’ll set the maximum number of entries to 400 and store
the database for three days (4320 minutes).
Router#confure terminal
Router(config)#dial-control-mib max-size 400
Router(config)#dial-control-mib retain-timer 4320
Router(config)#end
Router#
Finally, you must configure the call detail records to be logged. To accomplish this,
you use the gw-accounting syslog command. After you enter this command, go ahead
and place a call from one IP phone to another. Once you hang up the phone, issue a show
logging command on the CUCM Express where you are logging CDR information to the
internal buffer memory. Here’s an example of the output you will see:
Router#show logging
Syslog logging: enabled (1 messages dropped, 0 messages rate-limited,
0 flushes, 0 overruns, xml disabled, filtering disabled)
[output cut]
Console logging: disabled
Monitor logging: level debugging, 0 messages logged, xml disabled,
filtering disabled
Buffer logging: level debugging, 5 messages logged, xml disabled,
filtering disabled
Logging Exception size (4096 bytes)
Count and timestamp logging messages: disabled
Persistent logging: disabled
[output cut]
Log Buffer (32768 bytes):
*May 26 02:58:04.307: %SYS-5-CONFIG_I: Configured from console by console
*May 26 02:58:21.819: %VOIPAAA-5-VOIP_CALL_HISTORY: CallLegType 1,
ConnectionId E3E3F81B48D711DE82A887E7220CC055, SetupTime
*02:58:14.629 UTC Tue May 26 2009, PeerAddress 5002, PeerSubAddress ,
DisconnectCause 10 , DisconnectText normal call clearing (16), ConnectTime
*02:58:19.139 UTC Tue May 26 2009, DisconnectTime *02:58:21.819 UTC
Tue May 26 2009, CallOrigin 2, ChargedUnits 0, InfoType 2, TransmitPackets
0, TransmitBytes 0, ReceivePackets 0, ReceiveBytes 0
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*May 26 02:58:21.819: %VOIPAAA-5-VOIP_FEAT_HISTORY: FEAT_VSA=fn:TWC,ft:05/26/2009
02:58:14.623,cgn:5002,cdn:,frs:0,fid:250,fcid:E3E3F81B48D711DE82A887E7220CC055
,legID:F6,bguid:E3E3F81B48D711DE82A887E7220CC055
*May 26 02:58:21.819: %VOIPAAA-5-VOIP_CALL_HISTORY: CallLegType 1, ConnectionId
E3E3F81B48D711DE82A887E7220CC055, SetupTime *02:58:17.239 UTC
Tue May 26 2009, PeerAddress 5001, PeerSubAddress , DisconnectCause 10 ,
DisconnectText normal call clearing (16), ConnectTime *02:58:19.139 UTC
Tue May 26 2009, DisconnectTime *02:58:21.819 UTC Tue May 26 2009, CallOrigin
1, ChargedUnits 0, InfoType 2, TransmitPackets 0, TransmitBytes 0,
ReceivePackets 0, ReceiveBytes 0
*May 26 02:58:21.819: %VOIPAAA-5-VOIP_FEAT_HISTORY: FEAT_VSA=fn:TWC,
ft:05/26/2009 02:58:17.239,cgn:5002,cdn:5001,frs:0,fid:251,fcid:E3E3
F81B48D711DE82A887E7220CC055,legID:F7,bguid:E3E3F81B48D711DE82A887E7220CC055
It may seem cryptic, but all the information is there. You can obtain valuable
information from the CDR, as detailed in Table 6.8.
TA B L E 6 . 8
CDR information
Label
Description
PeerAddress
Phone number
DisconnectText
Method of disconnect
ConnectTime
Start call time/date
DisconnectTime
End call time/date
Look through the log entries shown previously to see if you can pick out each of these
details for yourself.
Configuring CDR to an External Syslog Server
As you’ve learned, logging this information to the router’s memory is not a permanent
solution. Syslog servers are used to offload and store computer and network hardware
log information for long periods of time. This is a great way to keep CDR information
for historical purposes. Syslog servers are easy to set up and maintain. Dozens of
different types of syslog servers are available today. Once you set up a server to store log
information, you simply point the CUCM Express router at the syslog server. To do this,
you use the logging ip-address command, where ip-address is the address of your syslog
server. Here is an example of how we can configure logging to a syslog server at 172.16.8.5:
Configuring Music on Hold (MoH)
281
Router#confure terminal
Router(config)#logging 172.168.8.5
Router(config)#end
Router#
That’s all there is to it. As stated earlier, it is possible and advisable to log CDR
information to both a syslog server and the local router buffer. That way, if you need to
look up recent CDR information, you can log in to the CUCM Express and check the logs.
And if you need to check historical CDR information, you can go to your syslog server,
which stores historical logs for a much longer time.
Configuring Music on Hold (MoH)
“Can you hold please?” Everyone hates to wait on hold. This is partly why Music- on Hold (MoH) was invented. So now, at least you can listen to music while you wait. Besides
making the wait less bothersome, it also helps the end user to know that they have not been
disconnected from the call. This section will detail how to use audio fi les stored on the
router flash for MoH.
On the CUCM Express system, a phone that uses either the g.711 or g.729 codec can
utilize MoH. The music fi le is presented in a g.711 format by default, so keep in mind that
any g.729 phones will require transcoding resources to use MoH. The sound quality will
also suffer because of the downgrade in fidelity from g.711 to g.729. You can store audio
fi les directly on the flash of the CUCM Express. The audio fi les that are compatible for use
on a CUCM Express system must have the following requirements:
File format of .au or .wav
G.711 codec format
8-bit rate at 8 kHz
Cisco includes a copyright-free MoH fi le when you download the CUCM Express
software. There also are audio - editing applications that can convert your MP3 fi les to the
proper format for use on the CUCM Express system. Just make sure you are fully aware of
any laws that prohibit the use of copyrighted music on your phone system.
There are two methods of providing audio fi le MoH on the CME. You can provide
MoH as a unicast stream or as a multicast stream. You configure unicast MoH streams
by using the moh command followed by the fi lename of the audio fi le. This command is
performed within config-telephony mode. Here is an example of how to configure mymusic
.wav as a unicast stream for our CUCM Express:
Router#confure terminal
Router(config)#telephony-service
Router(config-telephony)#moh mymusic.wav
Router(config-telephony)#end
Router#
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Keep in mind that this specific MoH configuration setup requires a separate audio
stream to be sent over the network for each phone that requests it. Every phone that is on
hold is another RTP stream on the network. Clearly this does not scale well, because it
adds more bandwidth overhead and increases the CPU usage on the CUCM Express. On
the other hand, if your Cisco IPT implementation spans multiple subnets, and these subnets
are not enabled for multicast routing, the unicast MoH method is your only choice.
An alternative method to the unicast MoH streams is to configure multicast MoH.
If your network is enabled for multicast routing, then you should utilize the multicast
functionality in the CUCM Express. Think of multicast as a radio station. If users tune
into that radio station, they all listen to the same stream. This is exactly how multicast
MoH works. You configure your “radio station” channel in the form of a multicast IP
address. When a user places someone on hold, they tap into that audio stream on the
network for the audio fi le. Additional users who are placed on hold also tap into this
same audio stream. This way, there is a single MoH RTP stream on the network, and it
is broadcast to any users who require it. To configure multicast MoH, you need to fi rst
identify the audio fi le just like in the unicast MoH configuration. In addition, you need
to configure the multicast IP address and port. Multicast addresses fall into the range of
224.0.0.0 to 239.255.255.255. However, multicast for MoH will not function within the
224.x.x.x range. The phones specifically do not support multicast in this range. A good
multicast IP range to configure multicast MoH with is the 239.x.x.x range. This IP block is
specifically set aside for private use within an organization.
By default, the MoH stream uses UDP port 2000. You can optionally change this port
if necessary, but it is not recommended, because the phones and CUCM are already set to
listen for RTP media transmissions on port 2000. Following is an example that enables
multicast MoH and specifies a multicast address for the audio stream. Just as in multicast
paging, the phones are set to listen to multicast MoH on UDP port 2000.
Router#confure terminal
Router(config)#telephony-service
Router(config-telephony)#moh mymusic.wav
Router(config-telephony)# multicast moh 239.23.4.10 port 2000
Router(config-telephony)#end
Router#
Using the Multicast MoH Route Command
One last optional multicast MoH command we need to address is the route command,
which allows you to configure up to four IP addresses to use as the source IP of the
multicast stream. The IP addresses must be either physical router IP addresses on
the CUCM Express or a loopback address. If the route command is not used, multicast MoH
will be sourced from the IP address assigned as the ip source-address command within
config-telephony mode. Let’s look at an example code snippet of a CUCM Express router:
Summary
283
Router#confure terminal
Router(config)#telephony-service
Router(config-telephony)#ip source-address 10.1.1.1
Router(config-telephony)#moh mymusic.wav
Router(config-telephony)#multicast moh 239.10.16.16 port 2000
Router(config-telephony)#end
Router#
Here you can see that we are multicasting our MoH music on the IP of 239.10.16.16.
The IP address that will be used as the source of the multicast stream is 10.1.1.1. Let’s say
we want to change that source IP address to 192.168.10.100, which is a loopback address
on the CUCM Express. It is always the best practice to source services using loopback
interfaces because they never go down. To do so we configure the following:
Router#confure terminal
Router(config)#telephony-service
Router(config-telephony)#multicast moh 239.10.16.16 port 2000 route
192.168.10.100
Router(config-telephony)#end
Router#
Disabling Multicast MoH on a Per-Ephone Basis
If part of your network is configured for multicast and another part is not, you can disable
multicast MoH on a per- ephone basis. If multicast MoH is disabled on an ephone, it will
use unicast for the stream instead. That way, you utilize multicast when you can and fall
back to unicasting the streams where needed. Multicast MoH is enabled by default on
all ephones. To disable it, you can use the no multicast-moh command within configephone mode. Here’s an example of disabling multicast MoH on ephone 12:
Router#confure terminal
Router(config)#ephone 12
Router(config-ephone)# no multicast-moh
Router(config-ephone)#end
Router#
Summary
Chapter 6 began with the configuration differences between key systems and PBX systems.
Then you learned how to configure telephony service features such as user-locale and
network-locale to tailor your CUCM Express to any business and/or geographical region.
You also learned advanced productivity configurations such as hunt groups and paging
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that give users added functionality. Access and accounting configuration options were
also covered so you can limit user access and log calls on the system properly. Lastly, you
learned how to set up Music on Hold within a CUCM Express environment.
Even though having a great car stereo or Xenon headlights might not be absolutely
necessary, the add- on features of a car can be very nice to have. The same goes with
the added features of a CUCM Express system. The added features we configured here
help to add value to a business by streamlining and simplifying the way we handle
calls. Without these added features, we still have a phone system —but it’s not nearly as
interesting or useful without the bells and whistles!
The next chapter looks at configuring voice gateways.
Exam Essentials
Know how to configure key systems and PBX system DNs. The CUCM can be configured
to act as a key system or a PBX depending on the needs of the environment. Key-system
phones are typically configured identically and share DNs. PBX systems are configured
with unique DNs on each phone and are individually tailored to meet the needs of the user.
Understand the different types of ephone button options. Using the button separator
when configuring extensions lets you set various ring options, phone monitoring, and
overlay features.
Know how to configure your CUCM Express to meet the needs of your users.
Depending on where you set up your CUCM Express, you may need to modify user options
to match the native language. In addition, the network options can be modified to match
the PSTN tone and cadence that are familiar to the area, and the CUCM Express can be
modified to display the date and time in a familiar format. The system message display can
be changed to customize the system for your environment.
Know how to configure local directory services. The local directory contains the phone
extension to name mapping on the CUCM Express. You can create these entries when
configuring the ephone-DN. In addition, you can manually enter listings that are not
directly configured as ephone-DNs on the system.
Understand the concept of static and dynamic call forwarding. Dynamic call forwarding
is performed by the user at the phone level. Static call forwarding is configured on the
CUCM Express.
Know the three call transfer option types and when to use them. Understand the
difference between full-blind, full-consult, and local-consult and the circumstances
of when you would want to use each of them.
Know how to configure call pickup groups. These groups allow a user to answer another
extension remotely. This functionality is important in call centers where people call into a
main number around the clock.
Written Lab 6.1
285
Understand the purpose of and how to configure call parking lines. Call parking spaces
allow a user to “park” a call in a designated ephone-DN and resume the call from any
other phone on the CUCM Express.
Understand the purpose of and how to configure hunt groups. Hunt groups allow
multiple phones to alternatively answer a shared ephone-DN. Different algorithms
determine which phone receives the next call.
Know how to configure the CUCM Express for intercom functionality. The intercom is a
great way for quick, two-way communication between predefi ned ephones.
Understand the concept of paging and how to configure It. Paging is one-way
communication to multiple predefi ned phones. The paging communication can be
configured either as multiple unicast streams or as a single multicast stream.
Know how to configure call blocking. Call blocking can be performed by date, time, or
the 7-24 option. There are also techniques to exempt a specific phone or user from the callblocking rules.
Know how to configure call detail records to log to the router buffer and to an external
syslog server. CDR information keeps track of phone calls on the CUCM Express. This
information can be useful for billing and fraud purposes.
Know how to configure Music on Hold. Much like paging, Music on Hold can be
configured to be sent as either a unicast stream to each phone on hold or as a multicast
stream. Understand the various options for each method.
Written Lab 6.1
Write the answers to the following questions:
1.
What is the config- ephone command to configure DN 1 on button 2 and DN 2 on
button 1?
2.
What is the config- ephone-DN command to make DN 10 have dual lines?
3.
What is the config- ephone command to assign button 2 to DN 8 and have it use an
alternate ring?
4.
What is the config- ephone-DN command to set a DN to be more preferred than a DN
that has its preference set to 2?
5.
What show command lists all ephones, buttons, overlay lines, status, and call source
and destinations?
6.
What config-telephony command changes the date layout to show yy-mm-dd?
7.
How would you configure a manual directory entry 1 for John Smith at extension
1001 while in config-telephony mode?
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8.
While in config-telephony mode, how would you configure users to be able to transfer
calls to any off-network 10 -digit number that begins with 555-777?
9.
How would you configure call blocking on Saturdays after 6:00 p.m. while in configtelephony mode?
10. What is the config-telephony command to configure multicast MoH using
239.100.100.240 and port 2001?
(The answers to Written Lab 6.1 can be found following the answers to the review
questions for this chapter.)
Hands-on Labs
To complete the labs in this section, you need a CUCM Express router and two Cisco
IP phones. The CUCM Express should be properly set up and ready for configuring IP
phones. Each lab in this section builds on the last and will follow the logical CUCM
Express PBX model design shown in Figure 6.13.
F I G U R E 6 .1 3
CUCM Express lab diagram
5001
5002
172.16.20.1/24
CUCME
5003
5004
Here is a list of the labs in this chapter:
Lab 6.1: Configuring a Hunt Group
Lab 6.2: Configuring a Call Parking Slot
Lab 6.3: Configuring Multicast Paging
Lab 6.4: Configuring Multicast MoH
Hunt Group
Pilot: 5000
Hands-on Labs
287
Hands-on Lab 6.1: Configuring a Hunt Group
In this lab, we’re going to configure a PBX-modeled CUCM Express with four phones.
Each phone will be configured with its individual extension on phone button 1. In addition,
three of the phones should be configured for a hunt group. Table 6.9 provides additional
phone information.
TA B L E 6 . 9
Lab 6 phone setup details
Phone
MAC Address
Primary Number
Ephone 1
XXXX.XXXX.XXXX
DN 1: 5001: dual-line
Ephone 2
XXXX.XXXX.XXXX
DN 2: 5002: dual-line
Ephone 3
XXXX.XXXX.XXXX
DN 3: 5003: dual-line
Ephone 4
XXXX.XXXX.XXXX
DN 4: 5004
The following information will help you configure the hunt group:
Hunt group tag: 1
Pilot number: 5000
Member extensions: 5001, 5002, 5003
Number of rings before moving to next member: 5
Algorithm: sequential
Final number: 5004
1.
Log into your CUCM Express router and go into configuration mode by typing enable
and then configuration terminal.
2.
Configure ephone-DN 1 to have the number 5001 by typing ephone-dn 1 dual-line
and then number 5001.
3.
Configure ephone-DN 2 to have the number 5002 by typing ephone-dn 2 dual-line
and then number 5002.
4.
Configure ephone-DN 3 to have the number 5003 by typing ephone-dn 3 dual-line
and then number 5003.
5.
Configure ephone-DN 4 to have the number 5004 by typing ephone-dn 4 and then
number 5004.
6.
Configure the MAC address of ephone 1 by typing ephone 1 and then mac-address
XXXX.XXXX.XXXX. Your MAC address will be unique.
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7. Configure button 1 of ephone 1 to use ephone-DN 1 by typing button 1:1.
8. Configure the MAC address of ephone 2 by typing ephone 2 and then mac-address
XXXX.XXXX.XXXX. Your MAC address will be unique.
9. Configure button 1 of ephone 2 to use ephone-DN 2 by typing button 1:2.
10. Configure the MAC address of ephone 3 by typing ephone 3 and then mac-address XX
XX.XXXX.XXXX. Your MAC address will be unique.
11. Configure button 1 of ephone 3 to use ephone-DN 3 by typing button 1:3.
12. Configure the MAC address of ephone 4 by typing ephone 4 and then mac-address
XXXX.XXXX.XXXX. Your MAC address will be unique.
13. Configure button 1 of ephone 4 to use ephone-DN 4 by typing button 1:4.
14. Exit config- ephone mode by typing exit.
15. Enter into config- ephone-hunt mode by typing ephone-hunt 1 sequential.
16. Enter the pilot number by typing pilot 5000.
17. Configure the members by typing list 5001 5002 5003.
18. Configure the number of rings allowed before moving to the next phone by typing
timeout 20.
19. Enter the final number by typing final 5004.
20. Exit config- ephone-hunt mode by typing end.
Hands-on Lab 6.2: Configuring a Call Parking Slot
1.
Log into your CUCM Express router and go into configuration mode by typing enable
and then configuration terminal.
2.
We want to create a single parking slot. For our lab parking slot, we’ll use ephone-DN
10. To do this type ephone-dn 10.
3.
Assign the extension 3000 to the ephone-DN by typing number 3000.
4.
Configure the ephone-DN to be a call park extension by typing park-slot.
5.
Name the ephone-DN Parking 1 by issuing name Parking 1.
6.
Exit config- ephone-DN mode by typing end.
Hands-on Lab 6.3: Configuring Multicast Paging
1.
Log into your CUCM Express router and go into configuration mode by typing enable
and then configuration terminal.
2.
We want to create a single paging ephone-DN. For our paging extension, we’ll use
ephone-DN 11. To do this, type ephone-dn 11.
Hands-on Labs
3.
4.
289
Assign the extension 7777 to the ephone-DN by typing number 7777.
Configure multicast paging on address 239.254.254.254 by typing paging ip
239.254.254.254 port 2000.
5.
Assign ephones 1, 2, and 3 to be alerted when someone dials 7777. To do this, type the
following:
ephone 1
paging-dn 11 multicast
ephone 2
paging-dn 11 multicast
ephone 3
paging-dn 11 multicast
6.
Exit config- ephone mode by typing end.
Hands-on Lab 6.4: Configuring Multicast MoH
1.
Log into your CUCM Express router and go into configuration mode by typing enable
and then configuration terminal.
2.
Enter into config-telephony mode by typing telephony service.
3.
We will use music-on-hold.au as our music file to play. To configure this, type moh
music-on-hold.au.
4.
Configure MoH for multicast at 239.1.1.250 by typing multicast moh 239.1.1.250.
5.
Exit config-telephony mode by typing end.
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Review Questions
1.
With multiple ephone-DNs sharing a single number, what command can you use to
prioritize which ephone-DN will always receive the incoming call if it is not in use?
A. priority
2.
B.
preference
C.
state
D.
no huntstop
With multiple ephone-DNs sharing a single number, when the phone preference for each
ephone-DN is the same, how is call routing handled for incoming calls?
A. Calls will be received on the ephone-DN with the lowest tag.
3.
B.
This configuration will not work. The ephone-DNs must be configured with different
priorities.
C.
Calls will be received on the ephone with the lowest tag.
D.
Calls will be handled round-robin style.
What config- ephone-DN command prevents calls from hunting to the second channel on
dual-line ephone-DNs?
A. huntstop channel
4.
B.
no huntstop
C.
no huntstop channel
D.
huntstop
What ephone overlay button separator would you use if you want calls to come in on this
extension only when all other lines are busy?
A. o
5.
B.
c
C.
w
D.
x
E.
m
What is the term used to describe the configuration of multiple ephone -DNs on a single
physical phone button?
A. Ephone
B.
Ephone-DN
C.
Dual-line
D.
Call waiting
E.
Overlay line
Review Questions
6.
291
What configuration option can you change so that Cisco phones will display information
on the screen in a different language?
A. network-locale
7.
B.
language-locale
C.
user-locale
D.
telephony-service-locale
What configuration mode do you need to be in to configure the CUCM Express system
message that displays on Cisco IP phones?
A. config t
8.
B.
config- ephone
C.
config-telephony
D.
config- ephone-DN
E.
config-voiceport
How are directory names and numbers entered into the local directory on a CUCM
Express? Choose all that apply.
A. By configuring a name and number on an ephone-DN and assigning it to an ephone
B.
By entering the name and number into the system using the config-telephony mode’s
directory entry command
C.
By uploading a spreadsheet onto the CUCM Express router that has an .xls extension
D.
By manually entering the name and number using the Cisco IP phone handset.
9. What is the proper syntax to statically configure call forwarding to extension 2020 on a
phone when the phone is currently in use?
A. Router(config-ephone)#call-forward noan 2020
B.
Router(config-ephone)#call-forward busy 2020
C.
Router(config-ephone-DN)#call-forward busy 2020
D.
Router(config-ephone-DN)#call-forward noan 2020
10. What are the three call-transfer configuration methods?
A. Blind
B.
Full-blind
C.
Partial- consult
D.
Full- consult
E.
Full-local
F.
Local- consult
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11. By default, the CUCM Express allows calls to be transferred only to extensions that are
on-network. If you want users to be able to transfer calls off-network to any number within
the 555-555-XXXX range, how do you configure this?
A. Router(config-telephony)#transfer-pattern 555555....
B.
Router(config-telephony)#transfer-pattern 555555????
C.
Router(config-telephony)#transfer-pattern 555555T
D.
Router(config-telephony)#transfer-pattern 5555554%
12. What Cisco telephony feature allows a user to answer, on their local phone, an extension
that is configured on a different phone?
A. Call transfer
B.
Call waiting
C.
Hunt group
D.
Call pickup
13. What is the main difference between call hold and call park?
A. Call hold allows the user to resume the call from any phone on the CUCM Express
system.
B.
Call hold disconnects the call after 120 seconds.
C.
Call park allows the user to resume the call from any phone on the CUCM Express
system.
D.
Call park disconnects the call after 120 seconds.
14. When referring to hunt groups, what is the name of the ephone-DN that is used to call a
hunt group?
A. Hunt group tag
B.
Member list
C.
Pilot number
D.
Final number
15. On a CUCM Express, how would you configure extension 4001 as the last number that a
hunt group should attempt?
A. Router(config-ephone-hunt)#list 4001
B.
Router(config-ephone-hunt)#final 4001
C.
Router(config-ephone-hunt)#last 4001
D.
Router(config-ephone-hunt)#timeout 4001
16. When configuring multicast broadcasting for features such as paging and Music on Hold,
what multicast IP range is not usable on the CUCM Express?
A. 239.0.0.0 to 239.255.255.255
B.
224.0.0.0 to 228.255.255.255
C.
224.0.0.0 to 224.255.255.255
D.
223.0.0.0 to 224.255.255.255
Review Questions
293
17. Given the following configuration, which ephones will be paged when you dial the 6003
paging extension?
Router(config)#ephone-dn 1
Router(config-ephone-dn)#number
Router(config-ephone-dn)#paging
Router(config)#ephone-dn 2
Router(config-ephone-dn)#number
Router(config-ephone-dn)#paging
Router(config)#ephone-dn 3
Router(config-ephone-dn)#number
Router(config-ephone-dn)#paging
Router(config-ephone-dn)#exit
Router(config)#ephone 1
Router(config-ephone)#paging-dn
Router(config-ephone)#ephone 2
Router(config-ephone)#paging-dn
Router(config-ephone)#ephone 3
Router(config-ephone)#paging-dn
Router(config-ephone)#ephone 4
Router(config-ephone)#paging-dn
Router(config-ephone)#end
6001
ip 239.1.80.100 port 2000
6002
ip 239.1.80.101 port 2000
6003
group 1 2
1 multicast
1 multicast
2 multicast
2 multicast
A. Ephones 1 and 2
B.
Ephone 1
C.
Ephones 1, 2, 3, and 4
D.
Ephones 3 and 4
18. Changes were made to add call-pickup groups to phones on your CUCM Express. The
problem is, the call-pickup group function is not available on the phones. What is likely the
problem?
A. Call pickup is only available on high- end phones such as the 7975.
B.
The CUCM Express configuration has not been written to memory.
C.
The Cisco IP phones need to be restarted.
D.
You must configure multicast on your network for call-pickup groups to function.
19. What is the proper syntax to block all users from being able to dial 900 numbers out the
PSTN on the CUCM Express? Assume that you must dial 9 to access the PSTN.
A. Router(config-telephony)#after-hours block pattern 91900 7-24
B.
Router(config-telephony)#after-hours block pattern 91900 Mo-Su
C.
Router(config-telephony)#after-hours block pattern 91900 exempt
D.
Router(config-telephony)#after-hours block pattern 91900 all
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20. Why is it recommended that you configure CDR to log to a syslog server and not just the
internal router memory? Choose all that apply.
A. The router has a limited amount of space, and the oldest logs will be overwritten over
time.
B.
The syslog server will provide additional call details that the buffer logs will not have.
C.
The router will lose all logs if it is restarted or loses power.
D.
The syslog server will lose all logs if it is restarted or loses power.
Answers to Review Questions
295
Answers to Review Questions
1.
B. The preference command allows you to set which ephone-DN will receive all calls when
not in use. The lower number is the more preferred ephone-DN.
2.
D. If preferences are the same, then calls will be handled in a round-robin manner.
3.
A. The huntstop channel command informs the CUCM Express that it should not route
the call to the second channel of a dual-line phone. Instead, the call will be routed to the
ephone-DN that is next in line according to the algorithm being used.
4.
D. The expansion (x) line button separator helps prevent a caller from receiving a busy signal. The calls will go to this line only when all other lines are busy.
5.
E . An overlay line is a phone button separator configuration option that allows you to configure multiple ephone -DNs on a single phone button.
6.
C . The user-locale option allows you to change the language displayed on the LCD screens
of Cisco IP phones.
7.
C . The system message displays any message you wish on all Cisco IP phones that are registered on the CUCM Express. The command can be configured while in config-telephony
mode.
8.
A, B . You can add names and numbers by adding a name and number to the ephone-DN
and assigning it to an ephone or by manually entering a number using
the directory config-telephony command.
9.
C . The correct syntax is call-forward busy 2020. This command is run while in configephone-DN mode.
10. B, D, F. The three call transfer configuration methods on a CUCM Express are full-blind,
full- consult, and local- consult.
11. A. Answer A is the correct wildcard sequence to allow calls to be transferred to any number in the 555-555-XXXX range. Answer C would also work, but the phone will have to
wait 10 seconds to time out before it processes the digits. It is incorrect, however, because a
user could dial other numbers outside the 555-555-XXXX range.
12. D. Call pickup is the process of answering a remote extension on a local phone. To do so,
you can use the PickUp or GPickUp softkey depending on how your call pickup is configured.
13. C . With call park, you configure a separate ephone-DN where you place calls that are waiting. You can then answer the parked calls from any phone on the CUCM Express system as
long as you know the ephone-DN where the call is parked.
14. C . The pilot number is the main DN that is used to call into a hunt group.
15. B . The fi nal command is a config- ephone-hunt configuration that lists the last number that
should be attempted after all members in the hunt group have not answered.
16. C . The entire 224.X.X.X multicast IP range cannot be used.
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17. C . The ephone-DN 6003 is configured with paging group 1 and 2. Ephones 1– 4 are configured with paging-DN 1 or 2. This means that when 6003 is dialed, it pages all four phones
simultaneously.
18. C . Don’t forget that many of the phone features require the phones to be restarted after
configuring them on the CUCM Express.
19. A . The 7–24 option restricts the pattern at all times with no exemptions.
20. A, C . The router has a limited amount of memory. Also, if the router is rebooted or loses
power, all logging is erased. The syslog server stores the logs to a storage drive, so it is better for maintaining historical logs over time.
Answers to Written Lab 6.1
Answers to Written Lab 6.1
1.
button 1:2 2:1
2.
ephone-dn 10 dual-line
3.
button 2f8
4.
preference 1 (or preference 0, which is the default)
5.
show ephone
6.
date-format yy-mm-dd
7.
directory entry 1 1001 name John Smith
8.
transfer-pattern 5557777.... (or 5557777T)
9.
after-hours day Sat 18:00 23:59
10. multicast moh 239.100.100.240 port 2001
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Chapter
7
Configuring Voice
Gateways for POTS
and VoIP
THE CCNA VOICE EXAM TOPICS
COVERED IN THIS CHAPTER INCLUDE
THE FOLLOWING:
Configure gateways, voice ports and dial peers to connect
to the PSTN and service provider networks.
Describe the function and application of a dial plan.
Describe the function and application of voice gateways.
Describe the function and application of voice ports in a
gateway.
Describe the function and operation of call legs.
Configure voice dial peers.
Describe the differences between PSTN and Internet
Telephony Service provider circuits.
When I think of voice gateways, I like to use the Star Trek
teleporter as a visual analogy. The teleporter on the starship
Enterprise is responsible for taking a human (or Vulcan),
dematerializing them into energy, and then rematerializing that person to a different
location. A voice gateway does something similar; it takes a voice stream that works on one
network and repackages it so the stream can be understood and sent over a different type
of network. Ultimately the voice stream reaches its intended target destination on the other
end of the connection. This chapter will discuss how we configure our voice gateways to
properly package and direct voice streams to nonnative (alien) networks.
In Chapter 7, you’ll first learn how to configure FXS and FXO voice ports. Then you’ll
see how to direct calls over these analog voice ports using POTS dial peers. We’ll move on
to how to configure the physical characteristics of digital T1 CAS and PRI interfaces and
how to use POTS dial peers for call routing. Next we’ll use VoIP dial peers to route offnetwork calls over the IP WAN connections. Once the analog and digital and basic IP WAN
configurations are covered, we’ll discuss the importance and process of coming up with a
dial-plan strategy. We’ll also go over the dial-peer decision-making process to learn how the
CUCM Express makes decisions when routing calls. Next, we’ll cover how to manipulate
phone numbers so that calls can be properly routed both on and off network. Finally, we’ll
go over how to configure trunks between voice gateways using both H.323 and SIP.
Configuring Analog FXS and FXO
Ports with Basic Dial Peers
When you insert an analog voice card into a Cisco router installed with a supported
version of IOS voice software, the router will automatically detect and add the new voice
interfaces to the configuration. You can view the configuration status of these interfaces
by performing a show run. Here is an example of show run output from a CUCM Express
router with a four-port FXS and a four-port FXO installed:
Router#sh run | begin voice-port
voice-port 0/0/0
!
voice-port 0/0/1
!
Configuring Analog FXS and FXO Ports with Basic Dial Peers
voice-port
!
voice-port
!
voice-port
!
voice-port
!
voice-port
!
voice-port
301
0/0/2
0/0/3
0/1/0
0/1/1
0/1/2
0/1/3
Another command that’s useful for viewing the voice ports your voice gateway has
installed is show voice port summary. Here is the output of this command. Notice
how you can easily distinguish an FXS from an FXO port with this command as well as
determine the signaling type (-ls for loopStart, -gs for groundStart) that is currently set:
Router#show voice port summary
PORT
==============
0/0/0
0/0/1
0/0/2
0/0/3
0/1/0
0/1/1
0/1/2
0/1/3
CH
SIG-TYPE
ADMIN
== ============ =====
-- fxs-ls
up
-- fxs-ls
up
-- fxs-ls
up
-- fxs-ls
up
-- fxo-ls
up
-- fxo-ls
up
-- fxo-ls
up
-- fxo-ls
up
OPER
====
dorm
dorm
dorm
dorm
dorm
dorm
dorm
dorm
IN
STATUS
========
on-hook
on-hook
on-hook
on-hook
idle
idle
idle
idle
OUT
STATUS
========
idle
idle
idle
idle
on-hook
on-hook
on-hook
on-hook
EC
==
y
y
y
y
y
y
y
y
Router#
I will begin by showing you how to configure FXS ports for attached analog telephones.
You will see how a phone number dial peer is properly applied to the FXS port, so anytime
a phone calls that number, the CUCM Express knows that the desired party is physically
attached to a specific analog interface.
We’ll then move on to see how FXO ports are configured to connect to the PSTN. You’ll
see that the configuration of FXS and FXO ports is similar except for the dial-peer options.
With FXS ports, they are connected to a physical end device such as an analog phone or
fax machine, which implies a simple dial-peer mapping to an extension. In the case of FXO
port connectivity to the PSTN, we need to set up our dial peers to be able to handle any
off-network phone numbers that we want to be able to reach. Because of this, dial peers
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Configuring Voice Gateways for POTS and VoIP
for FXO ports can become much more complex. You’ll learn how to accomplish this setup
more easily by using wildcards in your dial-peer statements.
Configuring FXS Ports
As you learned back in Chapter 2, FXS ports are responsible for connecting analog
endpoints such as analog phones, fax machines, and modems. Figure 7.1 shows two phones
connected to FXS ports on the CUCM Express.
F I G U R E 7.1
Two phones connected via FXS ports as dial peers
Voice-Port 0/0/0
Vo
ic
e-P
o
rt 0
/0
/1
Ext:
6001
Ext:
6002
You can see in the diagram that extension 6001 should be configured for the analog
phone on FXS voice-port 0/0/0, and the phone connected to voice-port 0/0/1 should
be configured with extension 6002. The following example shows how we can use the
station-id number command to set our caller ID number to the port. The fi rst task is to
enable caller ID on the FXS port because it is disabled by default. The caller-id enable
command enables caller ID on a per-port basis.
Because analog devices are not intelligent, the phone has no idea what number is
assigned to it. This way, you can swap analog phones at any time and not have to worry
about any fi rmware/configuration fi le issues that you may run into when swapping out IP
phones. A second command demonstrated here is station-id name. This configures the
caller ID information to the analog FXS port. Here is the output for configuring a name
and number to two FXS ports, as detailed in Figure 7.1:
Router#configure terminal
Router#voice-port 0/0/0
Router(config-voiceport)#caller-id enable
Router(config-voiceport)#station-id ?
name
A string describing station name
number A full E.164 telephone number
Configuring Analog FXS and FXO Ports with Basic Dial Peers
303
Router(config-voiceport)#station-id number 6001
Router(config-voiceport)#station-id name Adriana Castro
Router(config-voiceport)#exit
Router#voice-port 0/0/1
Router(config-voiceport)#caller-id enable
Router(config-voiceport)#station-id number 6002
Router(config-voiceport)#station-id name Brett Cowan
Router(config-voiceport)#end
Router#
In addition to extension numbers, we need to specify the type of signaling we wish to
use. The choices are ground start and loop start signaling, which we discussed in Chapter 2.
Table 7.1 details the configuration differences between the two signaling options.
TA B L E 7.1
FXS ground and loop start configuration choices
Signaling Type
Description
Default on FXS
Common Uses
Loop start
Closes loop
immediately after
off-hook is detected.
Default setting
When connecting analog end
devices. Not recommended
for high-volume analog trunks
because of glare.
Ground start
Requires ground
detection prior to the
loop closing.
Not the default
setting
When connecting to the PSTN
or another PBX.
You have already seen how you can configure the CUCM Express router to match
either the user or network locale for Cisco IP phones. Along those same lines, we can
configure our FXS analog port to match the location standards for analog phones. To
accomplish this, we can use the cptone config-voiceport command. Let’s look at the cptone
location options available to us:
Router#configure terminal
Router(config)#voice-port 0/0/0
Router(config-voiceport)#cptone ?
locale
2 letter ISO-3166 country code
AR
AU
AT
BE
BR
Argentina
Australia
Austria
Belgium
Brazil
IN
ID
IE
IL
IT
India
Indonesia
Ireland
Israel
Italy
PE
PH
PL
PT
RU
Peru
Philippines
Poland
Portugal
Russian Federation
304
CA
CN
CO
C1
C2
CY
CZ
DK
EG
FI
FR
DE
GH
GR
HK
HU
IS
Chapter 7
Canada
China
Colombia
Custom1
Custom2
Cyprus
Czech Republic
Denmark
Egypt
Finland
France
Germany
Ghana
Greece
Hong Kong
Hungary
Iceland
Configuring Voice Gateways for POTS and VoIP
JP
JO
KE
KR
KW
LB
LU
MY
MX
NP
NL
NZ
NG
NO
OM
PK
PA
Japan
Jordan
Kenya
Korea Republic
Kuwait
Lebanon
Luxembourg
Malaysia
Mexico
Nepal
Netherlands
New Zealand
Nigeria
Norway
Oman
Pakistan
Panama
SA
SG
SK
SI
ZA
ES
SE
CH
TW
TH
TR
AE
GB
US
VE
ZW
Saudi Arabia
Singapore
Slovakia
Slovenia
South Africa
Spain
Sweden
Switzerland
Taiwan
Thailand
Turkey
United Arab Emirates
United Kingdom
United States
Venezuela
Zimbabwe
Router(config-voiceport)#cptone
The cptone command stands for call progress tone. By default, cptone is configured for
the US locale. Let’s change the default in our example to match the tones found in Thailand
for both of the analog phones in our example:
Router#configure terminal
Router(config)#voice-port 0/0/0
Router(config-voiceport)#cptone TH
Router(config-voiceport)# voice-port 0/0/1
Router(config-voiceport)#cptone TH
Router(config-voiceport)#end
Router#
One fi nal configuration option that you should be familiar with is the ring frequency
command. Like the cptone command, ring frequency may need to be modified depending
on the location of the deployment. Telephone endpoints such as analog phones may be
country dependent. Depending on the phone in question, the ring frequency may need to be
adjusted for it to ring properly. This command sets the AC power output over the wires to
make the phone ring. The frequencies are measured in Hertz, and only a handful of options
are available, as shown here:
Router#configure terminal
Router(config)#voice-port 0/0/0
Router(config-voiceport)#ring frequency ?
Configuring Analog FXS and FXO Ports with Basic Dial Peers
20
25
30
50
ring
ring
ring
ring
frequency
frequency
frequency
frequency
20
25
30
50
305
Hertz
Hertz
Hertz
Hertz
Router(config-voiceport)#ring frequency
If you’re in a situation where you are setting up an analog phone and it just won’t ring
for some reason, you may need to adjust the frequency to match the standards of the
country’s PSTN. By default, Cisco sets the ring frequency to 25 Hz, which is the standard
for North America. In some European countries such as France, the maximum ring
frequency is 50 Hz. Here’s how to modify the frequency to match Europe’s default power
limit for analog phones:
Router#configure terminal
Router(config)#voice-port 0/0/0
Router(config-voiceport)#ring frequency 50
Router(config-voiceport)#exit
Router(config)#voice-port 0/0/1
Router(config-voiceport)#ring frequency 50
Router(config-voiceport)#end
Router#
Reviewing FXS Port Configuration and Status
A great way to review your FXS port configuration and line status is to use the show voice
port privileged exec command. Here we can check the configuration parameters and
port status of our voice ports. Following is an example of the information this command
displays about FXS port 0/0/0, which happens to be in use (off-hook):
Router#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 VIC3-4FXS/DID
Operation State is UP
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
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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 1000 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 TH
Analog Info Follows:
Currently processing Voice
Maintenance Mode Set to None (not in mtc mode)
Number of signaling protocol errors are 0
Impedance is set to 600r Ohm
Station name Adriana Castro, Station number 6001
Translation profile (Incoming):
Translation profile (Outgoing):
Voice card specific Info Follows:
Signal Type is loopStart
Ring Frequency is 25 Hz
Hook Status is Off 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
Configuring Analog FXS and FXO Ports with Basic Dial Peers
307
Hookflash-out Timing is set to 400 ms
No disconnect acknowledge
Ring Cadence is defined by CPTone Selection
Ring Cadence are [10 40] * 100 msec
Ringer Equivalence Number is set to 1
As you can see, we can learn a ton of information by issuing this command. We know
that the port is physically working because the Administrative State is UP; the Operation
State is also UP, which means the phone is off-hook. You can also see the configuration
options set, including Station name, Station number, and Region Tone. Now let’s look at
FXS port 0/0/1:
Router#sh voice port 0/0/1
Foreign Exchange Station 0/0/1 Slot is 0, Sub-unit is 0, Port is 1
Type of VoicePort is FXS VIC3-4FXS/DID
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 1000 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
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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 TH
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 Brett Cowan, Station number 6002
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 active
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 [10 40] * 100 msec
Ringer Equivalence Number is set to 1
Router#
This port looks very similar except for the difference in Station name and Station
number. Also note that the Operation State is DORMANT, which means that the phone
attached to FXS port 0/0/1 is on-hook. Next up, you’ll learn how to set POTS dial peers to
point to our FXS ports.
Configuring POTS Dial Peers for the FXS Ports
Now that the physical FXS port is configured and ready to go, we need to create a POTS
dial peer to hard- code a phone number to that port. Because both of our ports are
analog endpoints, the destination-pattern is limited to creating a destination pattern
number and assigning that to our analog FXS port. In this example, we assign the numbers
6001 and 6002 to our previously configured FXS ports.
Configuring Analog FXS and FXO Ports with Basic Dial Peers
309
Router#configure terminal
Router(config)#dial-peer voice ?
<1-2147483647> Voice dial-peer tag
Router(config)#dial-peer voice 6001 pots
Router(config-dial-peer)#destination-pattern 6001
Router(config-dial-peer)#port 0/0/0
Router(config-dial-peer)#dial-peer voice 6002 pots
Router(config-dial-peer)#destination-pattern 6002
Router(config-dial-peer)#port 0/0/1
Router(config-dial-peer)#end
Router#
At this stage, any analog phone can be attached to these ports. The attached analog
phones can then be reached by dialing the destination pattern. Keep in mind that the
extension number is statically assigned to the FXS port and not the analog phone. The
dial-peer voice tag number does not need to be equivalent to the extension number, but
this is often done for ease of remembering the configuration.
FXS PLAR Configuration
A special analog FXS configuration type that you’ll likely run across is the private line
automatic ringdown (PLAR) port. This specialized port will automatically ring any
number that is configured on the port as soon as the phone goes off-hook. A good way
to think about this type of port is as a “hotline” phone for emergency or informational
purposes such as an emergency phone inside an elevator. PLAR is also often used on
phones in public places outside locked doors. When someone wishes to gain access through
the locked doors, they pick up the phone, which automatically rings the security desk. Let ’s
use the security desk PLAR as our example. Figure 7.2 shows our PLAR- configured phone
that resides outside the locked doorway.
F I G U R E 7. 2
An FXS PLAR secure door scenario
Voice-Port
0/0/0
Lobby
Phone
Voice-Port
0/0/1
Security
Phone
Ext: 2001
310
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To accomplish this goal using the minimum number of steps, we fi rst must configure
voice-port 0/0/0 to function as a PLAR port. We can use the connection plar command in
config-voiceport mode to do this. Here are the configuration steps to configure voice-port
0/0/0 as a PLAR to ring extension 2001 automatically when the receiver is picked up.
Router#configure terminal
Router(config)#voice-port 0/0/0
Router(config-voiceport)#connection plar 2001
Router(config-voiceport)#end
Router#
Next, we’ll configure a POTS dial peer for extension 2001 and apply it to voice-port
0/0/1.
Router#configure terminal
Router(config)#dial-peer voice 2001 pots
Router(config-dial-peer)#destination-pattern 2001
Router(config-dial-peer)#port 0/0/0
Router(config-dial-peer)#end
Router#
In this example, both phones are analog and use FXS ports. In reality, for
an analog PLAR connection only the phone configured as the PLAR line
needs to be analog. The destination extension can be analog, IP based, or
somewhere off network if desired. PLAR can also be enabled on IP phones
running either SIP or SCCP, but their configuration setup is different and
outside the scope of this book.
Now when you take the PLAR phone off hook, it will immediately ring the security
phone at extension 2001.
Configuring FXO Ports
Configuring FXO ports is similar to configuring FXS ports at the interface level. Just like
the FXS interfaces, FXO interfaces can use the same station-id, cptone, and ring frequency
config-voiceport commands. The main difference in configuration setup between the two is
that we’ll want to use either the signal groundStart or signal cama config-voiceport command
for signaling to the PSTN or to the PSAP (Public Switch Answering Point). Remember that
in some circumstances in North America, you need to route your emergency service calls
out using a CAMA- configured port. I’ll show you how to configure your port for CAMA
operation later in this section. For now, we’ll configure our FXO port for ground-start
signaling. We’ll use Figure 7.3 as our FXO interface and dial-peer configuration example.
Configuring Analog FXS and FXO Ports with Basic Dial Peers
F I G U R E 7. 3
311
Configuring an FXO port with POTS dial peer
FXO
Voice-Port 0/1/0
Ext:
2211
Ext: 555-555-7777
CUCM
Express
Legacy
PBX
Ext:
2576
In our example, we’re going to keep most of our defaults and simply configure groundstart signaling and provide caller-ID information on the port:
Router#configure terminal
Router(config)#voice-port 0/1/0
Router(config-voiceport)#station-id name Acme Inc
Router(config-voiceport)#station-id number 5555557777
Router(config-voiceport)#signal groundStart
Router(config-voiceport)#end
Router#
Two additional commands are used only with FXO hardware configurations. The first
configuration option is the dial-type config-voiceport command, which allows you to set
the address signaling used on this FXO port for outgoing calls. We know that we have two
options for address signaling. DTMF is the default and is used virtually everywhere these
days. You can also set the address signaling to pulse dialing if you find yourself in a part
of the world that still uses it. Here are the command options as they appear on the CUCM
Express:
Router(config-voiceport)#dial-type ?
dtmf
touch-tone dialer
pulse pulse dialer
Router(config-voiceport)#
A second FXO-only configuration option when configuring voice ports is the ring
number command. The ring number command signifies the maximum number of rings
detected by the router before answering the call. Remember that on FXO interfaces, the line
actually terminates at the router and not at a phone endpoint, as was discussed in Chapter 3.
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In our example, the number 555-555-7777 belongs to the FXO port itself. The ring number
command specifies how many rings it should wait before answering. The default number
is 1, which means it will answer the call immediately. There are some situations where you
might want to set this higher, such as if your PSTN line is split from the wall. One line goes
to your CUCM Express and another to a standard analog phone. Let’s say that you would
like to be able to answer all incoming calls on your analog phone on the first one to three
rings. If nobody is there to answer the calls, then the FXO should answer and handle the call
accordingly. To accomplish this, configure the following on your FXO port:
Router#configure terminal
Router(config)#voice-port 0/1/0
Router(config-voiceport)#ring number ?
<1-10> The number of rings detected before closing loop
Router(config-voiceport)#ring number 3
Router(config-voiceport)#end
Router#
Now you have the ability to catch the call on your other analog phone before it is
handled automatically by the CUCM Express FXO port.
Reviewing FXO Port Configuration and Status
Just as with the FXS ports, we can view configuration and status information of our FXO
ports by using the show voice port privileged exec command. Here’s what our FXO port
0/1/0 looks like:
Router#sh voice port 0/1/0
Foreign Exchange Office 0/1/0 Slot is 0, Sub-unit is 1, Port is 0
Type of VoicePort is FXO
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
Configuring Analog FXS and FXO Ports with Basic Dial Peers
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 1000 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
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 Acme Inc, Station number 5555557777
Translation profile (Incoming):
Translation profile (Outgoing):
Voice card specific Info Follows:
Signal Type is groundStart
Battery-Reversal is enabled
Number Of Rings is set to 3
Supervisory Disconnect is signal
Answer Supervision is inactive
Hook Status is On Hook
Ring Detect Status is inactive
Ring Ground Status is inactive
Tip Ground Status is inactive
Dial Out Type is dtmf
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Digit Duration Timing is set to 100 ms
InterDigit Duration Timing is set to 100 ms
Pulse Rate Timing is set to 10 pulses/second
InterDigit Pulse Duration Timing is set to 750 ms
Percent Break of Pulse is 60 percent
GuardOut timer is 2000 ms
Minimum ring duration timer is 125 ms
Hookflash-in Timing is set to 600 ms
Hookflash-out Timing is set to 400 ms
Supervisory Disconnect Timing (loopStart only) is set to 350 ms
OPX Ring Wait Timing is set to 6000 ms
Router#
We can see that our FXO port is UP and in a DORMANT state. We can also see our
configuration settings, including station name, station number, and number of rings. Next,
you’ll learn how to configure a dial peer for our FXO interface.
Configuring POTS Dial Peers for the FXO Ports
Now that we have our physical FXO analog port set up, we need to configure our POTS
dial peer to defi ne whom we can reach on this interface. This task gets a bit more complex
because FXO ports typically connect to the PSTN or a legacy PBX. That being said, we can
use a POTS dial peer to control the calls that are allowed on this interface. We can use dialpeer wildcards to help accomplish our goal. Table 7.2 lists the available dial-peer wildcards
on a CUCM Express system.
TA B L E 7. 2
Dial-peer wildcards
Wildcard
Description
Common Example
.
A single digit 0–9 or *.
5... matches 5 plus three additional
numbers or *.
[ ]
A range of single-numbered
digits. Incorporates - for
consecutive range or , for
nonconsecutive ranges.
[4-7] matches any number 4, 5, 6, or 7.
[5,8] Matches either 5 or 8.
( )
Indicates a pattern. It is used in
conjunction with the ?, %, and/or
⫹ symbols.
N/A
?
The last digit or ( ) pattern occurs
zero or one time.
543 matches 54 or 543. 6(54)? matches
6 or 654.
Configuring Analog FXS and FXO Ports with Basic Dial Peers
TA B L E 7. 2
315
Dial-peer wildcards (continued)
Wildcard
Description
Common Example
%
The last digit or ( ) pattern occurs
0 or more times.
765% matches 76 or 765 or up to any
number of 5s to a total of 32 digits.
3(21)% matches 3 or 321 or up to any
number of 21s to a total of 32 digits.
⫹
The last digit or ( ) occurs 1 or
more times.
987⫹ matches 987 or 9877 or up to an
infinite number of 7s. 6(54)⫹ matches
654 or 65454 or up to any of 54s to a
total of 32 digits.
T
The CUCM pauses to collect any
number of digits entered 0–9 or *.
9T matches 9 plus up to 31 additional
digits 0–9 and *.
You’ll notice that the dial-peer wildcards are virtually identical to other telephone-digit
wildcards used on the system. Once you learn them, you can use them in multiple scenarios
and not just with dial peers.
In our example, we connect to a legacy PBX system with an FXO port. The PBX has
two phones attached to it. Using a simple destination-pattern dial peer, we can use a
wildcard to cover all of the phones that the legacy PBX has with a single command:
Router#configure terminal
Router(config)#dial-peer voice 2000 pots
Router(config-dial-peer)#destination-pattern 2...
Router(config-dial-peer)#no digit-strip
Router(config-dial-peer)#port 0/1/0
Router(config-dial-peer)#end
Router
Now the router is set up so that when any CUCM Express – configured phones dial
2XXX, those calls are routed over to the legacy PBX on FXO port 0/1/0.
We had to include the command no digit-strip in this FXO POTS dialpeer configuration. This command is explained in the “Dial- Peer Digit
Manipulation” section of this chapter. For now, all you need to know is that
the command is needed to forward all four digits to the PBX, which is the
next hop along the call leg.
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FXO PLAR Configuration
FXO interfaces cannot utilize direct inward dials (DIDs). Instead, they have only a
single PSTN number assigned to the interface. They also cannot use Dialed Number
Identification Service (DNIS). DNIS is a service the public phone company offers on digital
circuits that lets the CUCM Express determine which telephone number was dialed by a
customer. It is useful for inbound calls because it allows you to switch the calls internally
based on the DNIS number.
Because our FXO interfaces cannot use DIDs or DNIS, and because the PSTN numbers
terminate the phone number directly on the voice-port interface, it is very common to use
PLAR to forward the call transparently to either a receptionist or an automated attendant
system for all inbound calls from the PSTN or a legacy PBX. If PLAR is not set up on the
FXO port, when a call from a user on the PSTN hits the FXO interface, that caller receives
a second dial tone. If the user knows the internal extension they want to call, they can enter
it then. That might work for people who know the exact internal extension they want to
call, but it can be confusing to others. A better way is to use PLAR to forward the calls to
a receptionist or automated attendant. That way, the receptionist can receive the inbound
calls and forward them to the proper internal extension. The same situation holds true for
an auto attendant. The inbound calls can be forwarded to the AA, and then the caller can
use automated prompts to be connected to the internal extension they choose. Figure 7.4
illustrates our configuration example.
F I G U R E 7. 4
An example of PLAR set up on an FXO port
FXO
Voice-Port 0/1/0
DN: 555-555-1212
Receptionist
Ext: 1000
PSTN
CUCM
Express
Here we have a single FXO port connected to the PSTN. The port is configured with a
555-555-1212 number. We want all calls going to this number to be automatically switched
to our receptionist IP phone at extension 1000, which is already configured on the CUCM
Express. So all we need to do is configure our FXO port for proper signaling and set up the
PLAR to point to extension 1000:
Router#configure-terminal
Router(config)#voice-port 0/1/0
Router(config)#signal groundStart
Router(config-voiceport)#connection plar 1000
Router(config-voiceport)#end
Router#
Configuring Analog FXS and FXO Ports with Basic Dial Peers
317
In our previous example, the PLAR number we used is an IP-based phone.
If the phone were an analog phone attached to an FXS port, we would
want to use the following config-voiceport command when setting up our
PLAR:
Router(config-voiceport)#connection plar opx 1000
The opx option stands for off-premise exchange . When an outside call
comes into voice -port 0/1/0, typically the voice gateway completes the call
immediately and then forwards it to the PLAR number. This works for both
IPT and analog (FXS) numbers. The problem arises when nobody is able to
answer the phone. If this occurs, the voice gateway cannot roll the call over
to voice mail, because the call has already been terminated. If we include
the opx option, the FXO port waits until the FXS port goes off-hook before
connecting the call. We can then set up an option to roll the incoming call
to voice mail if the analog phone is not answered after a specified number
of seconds.
FXO CAMA Configuration
In Chapter 2 you learned that CAMA interfaces connect to the PSAP for E911 calling.
They connect to the PSAP so the call can be directed to the proper E911 dispatch center
based on the calling party’s phone number. This helps to ensure that emergency
service calls are routed to the proper dispatch station. In this section you will learn how to
configure FXS ports for CAMA connections to the PSAP. Figure 7.5 shows our previously
configured FXO port 0/1/0 going to the PSTN.
F I G U R E 7. 5
An example of FXO with CAMA
Other off-network calls
FXO
Voice-Port 0/1/0
DN: 555-555-1212
Receptionist
Ext: 1000
CUCM
Express
PSTN
CA
MA
e-P
or
91 t 0/1
1C
/
all 1
s
Voi
c
911 Operators
As you can see from the diagram, we want to configure 0/1/1 to go to the E911 PSAP
as a CAMA port. The diagram depicts 911 calls being routed out the CAMA interface on
voice-port 0/1/1 and all other calls going out to the PSTN on voice-port 0/1/0.
Configuration elements that are unique to the CAMA FXO interface involve signaling.
Instead of using loopStart or groundStart, we use the signal cama config-voiceport
318
Chapter 7
Configuring Voice Gateways for POTS and VoIP
command. In addition, we can choose from several options for the CAMA signaling. Here
are the options listed on the router:
Router#configure terminal
Router(config)#voice-port 0/1/0
Router(config-voiceport)#signal cama ?
KP-0-NPA-NXX-XXXX-ST
KP-0-NXX-XXXX-ST
KP-2-ST
KP-II-NPA-NXX-XXXX-ST-KP-NPA-NXX-XXXX-ST
KP-NPD-NXX-XXXX-ST
<cr>
Type
Type
Type
Type
Type
2
1
3
5
4
CAMA
CAMA
CAMA
CAMA
CAMA
Signaling
Signaling
Signaling
Signaling
Signaling
Router(config-voiceport)#signal cama
The type of signaling required will be specified by the emergency services technicians in
your area. Let’s assume that we want to configure KP- 0 -NXX-XXXX-ST as our CAMA
signaling type. Here’s how the configuration of the voice port looks:
Router(config-voiceport)#signal cama KP-0-NPA-NXX-XXXX-ST
Note: need to shut/no shut to complete the CAMA signal type configuration.
Router(config-voiceport)#
Notice that after we changed signaling to CAMA, the router gave us a console
message stating that we must perform a shut and no shut on the FXO port to put the
interface into CAMA mode. Once the FXO port is configured, we can create the dial
peer to send 911 and 9911 calls out the CAMA interface. Here’s how to accomplish
this task:
Router#configure terminal
Router(config)#dial-peer voice 911 pots
Router(config-dial-peer)#destination-pattern 911
Router(config-dial-peer)#no digit-strip
Router(config-dial-peer)#port 0/1/0
Router(config-dial-peer)#exit
Router(config)#dial-peer voice 911 pots
Router(config-dial-peer)#destination-pattern 9911
Router(config-dial-peer)#forward-digits 3
Router(config-dial-peer)#port 0/1/0
Router(config-dial-peer)#end
Router#
Configuring Digital T1 Ports
319
We introduce a new command called forward-digits here. This command
is explained in more detail in the “Dial- Peer Digit Manipulation” section of
this chapter. For now, you just need to understand that the command tells
the voice gateway to send the last three digits of the matched destination
pattern. Therefore, when a user dials 9911, only 911 is sent out the CAMA
port 0/1/0.
Now anytime a user dials either 911 or 9911, the call is routed out the CAMA port.
Notice that we have both 911 and 9911 set to route calls out the CAMA port. This is
common if users are accustomed to dialing 9 for an outside line. Because the dial peer for
our CAMA port matches a specific number, it takes precedence over any other 9XXX
numbers. You’ll learn more about how the CUCM Express selects the best destination
routes later in this chapter. For now, just know that the more specific dial peer is best.
Configuring Digital T1 Ports
T1 ports are more commonly found in larger environments because they can carry multiple
phone lines over a single trunk link. This section will show you how to configure a T1
CAS, which uses all 24 channels for voice. We’ll then move on to configure the T1 PRI
circuit, which uses 23 channels for voice and the 24th channel for out- of-band signaling.
Along with the physical configuration of the digital circuits, you’ll learn how to set up basic
dial peers to be able to call inbound and outbound on the T1s.
Configuring T1 CAS Ports
A T1 CAS circuit comes into your voice gateway on a single copper connection. It has the
ability to handle up to 24 concurrent calls. This fi rst section will show you the various
configuration steps and options available. Depending on your service provider, the options
you choose may be different. You’ll have to work closely with your PSTN provider to make
sure you have the correct settings to have the T1 function properly. This book will use the
most common configuration used in North America for framing and linecoding, which is
ESF and B8ZS.
Framing and linecoding are outside the scope of this book. Just keep in
mind that Cisco supports a few types, but you will most likely see B8ZS/
ESF circuits.
Once we have the physical interface set up, we’ll configure a basic POTS dial peer so
calls can be properly routed out our interface.
A T1 that carries multiple calls on a single connection is referred to as a trunk line.
The T1 hardware that is installed on your router will be seen as “controller” interfaces
320
Chapter 7
Configuring Voice Gateways for POTS and VoIP
in the IOS configuration. To configure the T1 CAS card, you need to enter into configcontroller mode and choose the slot/port where your T1 card is located on the router. Once
in config- controller mode, you must configure the following four settings:
Framing type
Linecode type
Clock source
Ds0 -group options
Let’s briefly discuss each of these options to understand what they are used for.
Framing Type
The framing type sets the framing that your PSTN provider has configured on their end.
You can see the options listed here while in config- controller mode:
Router#config t
Router(config)#controller t1 0/1/0
Router(config-controller)#framing ?
esf Extended Superframe
sf
Superframe
Router(config-controller)#framing esf
Router(config-controller)#
Super Frame (SF) is the older of the two framing types available, and most telco
providers now use Extended Super Frame (ESF).
Linecode Type
The linecode type you choose again depends on your PSTN provider. You have to set your
linecode to match whatever coding they provide to you on the circuit. Here are the options
available to you on the T1 card:
Router(config-controller)#linecode ?
ami
AMI encoding
b8zs B8ZS encoding
Router(config-controller)#linecode b8zs
Router(config-controller)#
The most common option in much of the world is B8ZS.
Clock Source
The clock source option allows you to determine where the T1 circuit synchronizes
its timing clock. We know that the digital T1 circuits use Time-Division Multiplexing
Configuring Digital T1 Ports
321
(TDM) to send multiple voice channels over a single circuit. A clock mechanism ensures
that both sides of the T1 remain in sync so TDM can function properly. Here are the
options available for setting the clock source:
Router(config-controller)#clock source ?
internal
Internal Clock
line
Recovered Clock
Router(config-controller)#clock source line
Router(config-controller)#
The line option specifies that the T1 uses the clock from the T1 line itself. This means
that it synchronizes using the clock configured from the PSTN. The internal option tells
us that the T1 uses its own internal interface clock. You should use this option if you are
connecting to an internal PBX and want the router to handle the clock.
Ds0 -group Options
The ds0-group options are where you can configure multiple or individual channels of the
24 -line T1 CAS. Each T1 CAS channel is called a timeslot. Listed here are the different
timeslot signaling types you can choose from:
Router(config-controller)#ds0-group 0 timeslots 1-24 type ?
e&m-delay-dial
E & M Delay Dial
e&m-fgd
E & M Type II FGD
e&m-immediate-start E & M Immediate Start
e&m-wink-start
E & M Wink Start
ext-sig
External Signaling
fgd-eana
FGD-EANA BOC side
fgd-os
FGD-OS BOC side
fxo-ground-start
FXO Ground Start
fxo-loop-start
FXO Loop Start
fxs-ground-start
FXS Ground Start
fxs-loop-start
FXS Loop Start
none
Null Signalling for External Call Control
<cr>
Router(config-controller)#ds0-group 0 timeslots 1-24
You can configure signaling for each timeslot to be identical to or different from one
another, depending on what you want to accomplish, by using the timeslots X-X option
where X-X lists the timeslot range. When you want to use a group of timeslots for the
same task, you should put them into the same ds0 -group. If another set of timeslots is for
322
Chapter 7
Configuring Voice Gateways for POTS and VoIP
a different task, you should put them into a second ds0 -group. To show what I’m talking
about, let’s configure a ds0 -group using the fi rst 12 channels on ds0 -group 0 and the
second 12 channels for ds0 -group 1. The fi rst group will use fxo-loop-start signaling,
and the second group will be set up using e&m-immediate-start:
Router#configure terminal
Router(config)#controller t1 0/1/0
Router(config-controller)#ds0-group 0 timeslots 1-12 type fxo-loop-start
Router(config-controller)#ds0-group 1 timeslots 13-24 type e&m-immediate-start
Router(config-controller)#
This example shows how you can split up the 24 channels into different ds0 -groups for
different purposes. In the next section, you’ll see how to configure POTS dial peers to route
off-network calls to the PSTN.
So far, we’ve been discussing T1 circuits that have a full 24 channels for
use. If your environment does not need this many channels, most PSTN
providers offer fractional T1 circuits, in which they provide a T1 circuit
but limit the number of usable channels. For example, if you need 12
POTS lines, the PSTN will give you a full T1 but only timeslots 1–12 will
be usable. If you were to define all 24 timeslots, but only 12 were usable,
you may reach a point where calls cannot go through. In this situation you
would need to configure your ds0 - group timeslots for only 1–12, as shown
in this example:
Router(config-controller)#ds0-group 0 timeslots 1-12 type
fxo-loop-start
Configuring POTS Dial Peers for T1 CAS Ports
One thing to keep in mind about digital circuits such as T1s is that they’re still considered
POTS lines because they do not use IP for transport. In that regard, configuring POTS
dial peers for T1 CAS interfaces is very similar to configuring POTS dial peers for FXO
interfaces. The difference is that the T1 CAS circuits have multiple timeslots. In the next
configuration example, we will assume that the CAS T1 is set up to use all 24 channels for
the same signaling type. The following command output shows how to route off-network
calls out the T1 CAS located at port 0/1/0:
Router(config-dial-peer)#dial-peer voice 91 pots
Router(config-dial-peer)#destination-pattern 91..........
Router(config-dial-peer)#port 0/1/0:D
Router(config-dial-peer)#end
Router#
Configuring Digital T1 Ports
323
This configuration looks very similar to a standard FXO line except for the fact that this
dial peer allows 24 calls to be placed at a single time.
Configuring T1 PRI Ports
Configuration of a T1 PRI circuit is very similar to that of a T1 CAS circuit. I must
mention a few differences, however. Because T1 PRIs use Q.931 ISDN signaling, we must
configure the ISDN switch type that our PSTN provider uses. Also keep in mind that T1
PRIs use common channel signaling (CCS), which is out of band. That means that channel
24 (timeslot 23) is set aside for signaling, so only timeslots 0 –22 are available for voice
calls. Nothing from a configuration standpoint needs to be addressed. As soon as you set
the T1 to use Q.931 signaling, timeslot 23 is automatically reserved for signaling. Here
are the different ISDN switch type options. Note that this is a global router configuration
option. It can be overridden on a per-interface basis if needed:
Router#configure terminal
Router(config)#isdn switch-type ?
primary-4ess
Lucent 4ESS switch type for the U.S.
primary-5ess
Lucent 5ESS switch type for the U.S.
primary-dms100 Northern Telecom DMS-100 switch type for the U.S.
primary-dpnss
DPNSS switch type for Europe
primary-net5
NET5 switch type for UK, Europe, Asia and Australia
primary-ni
National ISDN Switch type for the U.S.
primary-ntt
NTT switch type for Japan
primary-qsig
QSIG switch type
primary-ts014
TS014 switch type for Australia (obsolete)
Router(config)#isdn switch-type primary-ni
Router(config)#
In the United States, you will typically choose primary-5ess or primary-ni for your
switch type.
Just like the T1 CAS circuit, the T1 PRI is configured within config- controller mode.
Here are the four options you need to be aware of for configuring the T1 PRI:
Framing type
Linecode type
Clock source
Pri-group options
The framing type, linecode type, and clock source options are identical to those for the
T1 CAS configuration detailed earlier in this chapter. Notice, however, that instead of
ds0 -group options, the T1 PRI uses pri-group options. Let’s take a closer look at this
command.
324
Chapter 7
Configuring Voice Gateways for POTS and VoIP
Pri-group Options
The pri-group option simply sets the timeslots you wish to use. Here is how to configure a
pri-group for a full PRI:
Router#configure terminal
Router(config)#controller t1 0/1/0
Router(config-controller)#pri-group timeslots 1-24
Did you notice that the pri-group is just a single global group that cannot be broken up
into subgroups? This is one of the benefits that the T1 CAS has over the T1 PRI. The more
granular control over timeslot signaling provided by using multiple ds0 -groups is not an
option with the pri-group commands.
T1 PRIs that utilize the Q.931 ISDN signaling can allow for the phone endpoint’s
full DID number to be used as the identity for off-network calls. This functionality is
not available on the T1 CAS. This feature is nice when you want to use the PSTN call
accounting records to keep track of your outbound calls.
Just like T1 CAS circuits, T1 PRIs can be ordered as fractional T1s. Keep in mind,
though, that the signaling is always performed on channel 24, which is timeslot 23. When
you configure a T1 pri-group command, you’ll fi nd that the D channel has automatically
been created for you in the form of an interface serial and voice-port format. Here’s the
output showing what these two configurations look like:
Router# show run | section 0/1/0:23
Building configuration...
interface Serial0/1/0:23
no ip address
encapsulation hdlc
isdn switch-type primary-ni
isdn incoming-voice voice
isdn bind-l3 ccm-manager
no cdp enable
!
voice-port 0/1/0:23
We can also verify the operational status of each of the 23 voice channels on our T1 by
issuing the show voice port summary privileged exec command, as shown here:
Router1#show voice port summary
PORT
===============
0/1/0:23
0/1/0:23
CH
SIG-TYPE
ADMIN
== ============ =====
01 xcc-voice
up
02 xcc-voice
up
OPER
====
dorm
dorm
IN
STATUS
========
none
none
OUT
STATUS
========
none
none
EC
==
y
y
Configuring Digital T1 Ports
0/1/0:23
0/1/0:23
0/1/0:23
0/1/0:23
0/1/0:23
0/1/0:23
0/1/0:23
0/1/0:23
0/1/0:23
0/1/0:23
0/1/0:23
0/1/0:23
0/1/0:23
0/1/0:23
0/1/0:23
0/1/0:23
0/1/0:23
0/1/0:23
0/1/0:23
0/1/0:23
0/1/0:23
03
04
05
06
07
08
09
10
11
12
13
14
15
16
17
18
19
20
21
22
23
PWR FAILOVER PORT
=================
xcc-voice
xcc-voice
xcc-voice
xcc-voice
xcc-voice
xcc-voice
xcc-voice
xcc-voice
xcc-voice
xcc-voice
xcc-voice
xcc-voice
xcc-voice
xcc-voice
xcc-voice
xcc-voice
xcc-voice
xcc-voice
xcc-voice
xcc-voice
xcc-voice
up
up
up
up
up
up
up
up
up
up
up
up
up
up
up
up
up
up
up
up
up
dorm
dorm
dorm
dorm
dorm
dorm
dorm
dorm
dorm
dorm
dorm
dorm
dorm
dorm
dorm
dorm
dorm
dorm
dorm
dorm
dorm
none
none
none
none
none
none
none
none
none
none
none
none
none
none
none
none
none
none
none
none
none
none
none
none
none
none
none
none
none
none
none
none
none
none
none
none
none
none
none
none
none
none
325
y
y
y
y
y
y
y
y
y
y
y
y
y
y
y
y
y
y
y
y
y
PSTN FAILOVER PORT
==================
Router#
Notice that all the ports are labeled as port 0/1/0:23, indicating that timeslot 23 is the
out- of-band signaling channel. Keep this in mind as we configure the dial peers in the next
section.
Configuring POTS Dial Peers for T1 PRI Ports
In this section, we’re going to configure the pri-group for a full T1 and then configure a
POTS dial peer for off-network calling using a very basic destination pattern. There is very
little difference between configuring a T1 PRI POTS dial peer and any other POTS dial
peer, as you will see here:
Router(config-dial-peer)#dial-peer voice 91 pots
Router(config-dial-peer)#destination-pattern 91..........
Router(config-dial-peer)#port 0/1/0:23
Router(config-dial-peer)#end
Router#
326
Chapter 7
Configuring Voice Gateways for POTS and VoIP
The one difference in the PRI setup is the :23 that specifies the D timeslot of the PRI
circuit. This is because all of the channels within the PRI are labeled as port 0/1/0:23, as
we saw previously in the show voice port summary command.
Now that we have the POTS dial-peer configurations taken care of, let’s move on to
configuring VoIP dial peers.
Configuring VoIP Dial Peers
over WAN Connections
Now that we’ve covered how to configure physical analog and digital ports and pair
them with POTS dial peers, we’re going to move on to how to utilize an IP WAN circuit
and VoIP dial peers for transporting voice from one voice gateway to another.
Because this book focuses mainly on voice- configuration topics, I’m going to assume you
know how to configure IP WAN interfaces. From a voice standpoint, it really doesn’t
matter what physical medium your WAN link uses. It could be point-to -point data
T1s, SONET, Opt-E -MAN, MPLS, or any other IP technology, as long as it runs IP;
the only thing you have to consider is bandwidth and delay requirements for voice.
That being said, we’re going to mostly focus on configuring VoIP dial peers across the
WAN links using various options available to us.
We’re going to create a point-to -point scenario in which we have a two -site WAN
environment. The WAN is set up with a P2P link between two CUCM Express routers. See
Figure 7.6 for a visual representation of our scenario:
F I G U R E 7. 6
An example of a VoIP WAN dial-peer gateway
IP WAN
172.16.30.1/30
Site_A: CUCM
Express
Extensions
5XXX
172.16.30.2/30
Site_B: CUCM
Express
Extensions
6XXX
We can send calls across the IP WAN connection by configuring VoIP dial peers.
Figure 7.6 shows that all extensions at Site_A are within the 5XXX range and extensions
configured at Site_B are in the 6XXX range. We fi rst must configure a VoIP dial peer and
use a unique tag to identify it. Let’s assume we are on the CUCM Express router at Site_A.
We will configure a VoIP dial peer with the 6000 tag, as follows:
Configuring VoIP Dial Peers over WAN Connections
327
Site_A#configure terminal
Site_A(config)#dial-peer voice 6000 voip
Site_A(config-dial-peer)#
At this point, we need to configure our VoIP dial-peer options. You should be familiar
with three important configuration steps:
Destination pattern
Session target
Codec type
The destination pattern of a VoIP dial peer is identical to the POTS dial peer. This
is the command used to identify a number or range of numbers that are to be directed over
the WAN link. In our example, we will use the wildcard of 6..., which means that all
four-digit extensions beginning with 6 are sent over the IP WAN to Site_B. Also remember
that you must add the no digit-strip command so all digits are sent over to the remote
CUCM Express:
Site_A(config-dial-peer)#destination-pattern 6...
Site_B(config-dial-peer)#no digit-strip
The session target command is similar to the POTS port dial peer command. It tells
the dial peer that calls matching the defi ned destination pattern should be sent to the IP
address listed in the session target. Using our example, we’ll configure Site_A to forward
calls to the IP address of the WAN interface configured on the CUCM Express of Site_B,
which is 172.16.30.2:
Site_A(config-dial-peer)#session target ipv4:172.16.30.2
The session target command lets you configure session targets using
the IP address or DNS name. To configure using a DNS name, the syntax is
as follows:
Router(config-dial-peer)#session target dns:<dns-name>
If you want to use DNS names for targets, you must ensure that your
CUCM Express or voice gateway has a name server configured on it so
it can perform DNS lookups. To configure a DNS server, you use the ip
name-server <ip address> command.
The fi nal VoIP configuration option that needs mentioning is the codec type. This is
where you can set the codec you wish to use over the IP WAN link. The following output
shows all of the different codec options available:
Router(config-dial-peer)#codec ?
clear-channel Clear Channel 64000 bps (No voice capabilities: data transport
only)
g711alaw
G.711 A Law 64000 bps
328
Chapter 7
g711ulaw
g722-48
g722-56
g722-64
g723ar53
g723ar63
g723r53
g723r63
g726r16
g726r24
g726r32
g728
g729br8
g729r8
ilbc
Configuring Voice Gateways for POTS and VoIP
G.711 u Law 64000 bps
G722-48K 64000 bps - Only supported for H.320<->H.323 calls
G722-56K 64000 bps - Only supported for H.320<->H.323 calls
G722-64K 64000 bps
G.723.1 ANNEX-A 5300 bps (contains built-in vad that cannot be
disabled)
G.723.1 ANNEX-A 6300 bps (contains built-in vad that cannot be
disabled)
G.723.1 5300 bps
G.723.1 6300 bps
G.726 16000 bps
G.726 24000 bps
G.726 32000 bps
G.728 16000 bps
G.729 ANNEX-B 8000 bps (contains built-in vad that cannot be
disabled)
G.729 8000 bps
iLBC 13330 or 15200 bps
Router(config-dial-peer)#codec
Depending on the amount of bandwidth on your WAN, you may want to use a codec
that is optimal for low-bandwidth links (codec choices were discussed in Chapter 3). For
our example, we’ll set the codec to g729br8:
Site_A(config-dial-peer)#codec g729br8
That’s all we need to do to configure the VoIP dial peer on Site_A. Let’s move over to
the CUCM Express on Site_B and configure a dial peer so users at Site_B will send calls
destined to extensions 5XXX over the WAN to Site_A:
Site_B#configure terminal
Site_B(config)#dial-peer voice 6000 voip
Site_B(config-dial-peer)#destination-pattern 5...
Site_B(config-dial-peer)#no digit-strip
Site_B(config-dial-peer)#session target ipv4:172.16.30.1
Site_B(config-dial-peer)#codec g729br8
Site_B(config-dial-peer)#end
Site_B#
Now we have a fully operational VoIP dial peer that routes internal extension calls
between the two sites over an IP WAN. Next we’re going to look at how we should plan
our dialing strategy to meet current and future needs in a way to ensure that our dial-peer
destination patterns remain fairly simple.
Dial-Plan Strategy
329
Dial-Plan Strategy
You’ve already learned that phone routing decisions are made using either POTS or VoIP
dial peers. More specifically, the routing decision is made in the destination patterns
within the dial peers. Once the telephone number decision is made at the destinationpattern level, the dial peer is assigned a port (for POTS dial peers) or a session target
address (for VoIP dial peers) where the calls are directed to.
When you begin to roll out a new voice system, you should take care to ensure
that you have a solid plan in place for the assignment of internal and external DID
numbers. If you begin to randomly assign numbers to users without a plan, you
may fi nd that you need to create very elaborate dial-plan destination patterns to route
calls properly. Obviously you should avoid doing this. If you plan your dial-peer
strategy for not only today’s needs but future needs, you should be able to have
a fairly simple dial-peer structure that doesn’t require fancy wildcard setups or
intensive digit manipulation. Let’s look at a fictitious company voice network as depicted
in Figure 7.7.
F I G U R E 7. 7
The voice network where we’ll implement a dial-plan strategy
Remote:
Chicago
18 Phones
Remote:
SanFran
Remote:
Miami
78 Phones
80 Phones
IP WAN
150 Phones
Central Office:
Seattle
Here we have a central office with three remote sites. All of the sites have WAN
connections back to the central office located in Seattle. They’re using a distributed
call-processing design, so each office has its own call manager. The Seattle central
office currently has 150 phones and isn’t expected to grow much over the next five
years. The San Francisco office has 80 phones, and expected growth over five years is
5 percent. The Miami office has 12 phones and expects a 10 percent growth in
330
Chapter 7
Configuring Voice Gateways for POTS and VoIP
five years. The Chicago office currently has 18 phones and is not expected to grow
any further. Finally, an additional four remote offices are expected to pop up
throughout the United States, bringing the total number of offices to nine.
These new offices will have approximately 5 phones each. Our job is to take the
current and expected growth states into account and plan our internal dialing
structure. One additional requirement given is that the company desires to use three-digit
extensions.
The idea behind a dial-plan strategy is to be able to efficiently route calls
between the remote sites with the fewest number of destination-pattern commands.
Given our restriction of three-digit extensions, we have approximately 800 numbers
to play with and break up as we choose. Table 7.3 shows how the 800 number limit
was derived.
TA B L E 7. 3
Three-digit dialing scope
Extension Range
Availability
0XX
Not available: used for operator
1XX
Available
2XX
Available
3XX
Available
4XX
Available
5XX
Available
6XX
Available
7XX
Available
8XX
Available
9XX
Not available: used for off-network and 911
Generally, it’s best practice to exclude all internal extension numbers beginning with 0
and 9 so they can be used for operator and off-network/911 service dialing.
Because we want our dial-plan strategy to be usable in the future, we need to do some
simple math to determine the approximate state of our voice endpoints across the multiple
sites five years down the road, as shown in Table 7.4.
Dial-Plan Strategy
TA B L E 7. 4
331
Extension requirements (five-year plan)
Site Name
IP Endpoints
Seattle
150
SanFran
84
Miami
13
Chicago
18
New Site 1
5
New Site 2
5
New Site 3
5
New Site 4
5
This gives us a total of 285 endpoints. With 800 extension numbers available, this
should be no problem, right? Your fi rst thought might be to assign the one of the fi rst digits
to each location and call it a day. The Seattle office would receive 1XX numbers, SanFran
would receive 2XX, and so on. The problem with this is that there are only 100 usable
extensions per block. Because the Seattle office requires 150 extensions, they’ll need to have
two of these blocks. A second problem is that we need to plan for growth for up to nine
remote sites. Unfortunately, we have only eight usable blocks of 100 numbers. One obvious
option would be to dump the three-digit extension requirement and go with four- or even
five-digit extensions. But to prove that we can fairly easily create destination patterns in
this scenario using three-digit extensions, we’re going to stick with this rule. What we can
do is plan for blocks of 100 extensions for some locations and blocks of 10 numbers for
others. This way, we can properly segment the dial plan to meet our needs. Table 7.5 lists a
suggested breakdown of a dial-plan strategy that meets the needs of our example company.
TA B L E 7. 5
A dial-plan strategy
Site Name
IP Endpoints
Extension Blocks
Seattle
150
1XX, 2XX
SanFran
84
3XX
Miami
13
40X, 41X
Chicago
18
42X, 43X
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TA B L E 7. 5
Configuring Voice Gateways for POTS and VoIP
A dial-plan strategy (continued)
Site Name
IP Endpoints
Extension Blocks
New Site 1
5
44X
New Site 2
5
45X
New Site 3
5
46X
New Site 4
5
47X
Using this dial - plan strategy, we’ve successfully fit our 285 endpoints into a three-digit
extension plan with room for growth. Now that you understand the importance of a dialplan strategy, we can explore the details of how voice routers make call-routing decisions.
If you purchase blocks of DIDs from the PSTN, you may or may not be
able to get contiguous blocks, as shown in our example. This example is
a simplified scenario intended to give you an understanding of why it is
important to have a dial-plan strategy.
Understanding the Dial-Peer
Decision-Making Process
The process of routing voice calls boils down to understanding the decisions made
for handling both outbound and inbound dial peers. At each call leg, an inbound and
outbound match must be made prior to forwarding the call to the next call leg. Inbound
dial peers come into the voice gateway, and outbound dial peers leave the voice gateway.
This section will discuss the dial-peer attributes and call-setup elements that voice gateways
use to make matches.
The Selection Process for Outbound Dial Peers
The outbound dial-peer decision-making process is the easiest to understand. To match
outbound dial peers, the router uses the destination pattern to match the phone number.
Once a match is made, it then uses either the port number or the IP session target to
forward the call to the next destination. You should understand the two rules the router
follows in determining the best destination pattern to use:
Understanding the Dial-Peer Decision-Making Process
1.
The router will always choose the most specific destination pattern.
2.
Once a match is found, the router will immediately route the call.
333
To better understand these two rules, let’s look at an example. On our CUCM Express,
we have four configured dial peers with the following destination patterns:
Dial-peer 1: 555[4 -7] . . .
Dial-peer 2: 5554 . . .
Dial-peer 3: 5555 . . .
Dial-peer 4: 5555
Now let’s say that a call is made on the CUCM Express to 555- 6712. Looking at our
dial-peer options, the number matches only dial-peer 1, and the call is forwarded to the
port or session target configured for that dial peer.
A second call is made to 555- 4213. Now dial-peer 1 and dial-peer 2 are in the decisionmaking process because the number fits into both wildcard scenarios. Because of rule 1
stated previously, the router will choose dial-peer 2 as its best option. This is because dialpeer
2 matches 1000 different numbers within its wildcard setup, whereas dial-peer 1 matches
4000 different numbers. The most specific pattern wins, so in this case it’s dial-peer 2.
Finally, a third call is made to 555-5111. This time the number matches dial-peers 1,
3, and 4. Even though seven digits were dialed, the fi rst four digits were an exact match
on dial-peer 4. According to rule 2, once a match is found, the router immediately routes
the call. Therefore, the fi nal three digits play no role in determining the best destination
pattern to use and may not be sent.
Selection Process for Inbound Dial Peers
A voice gateway can utilize information found in call setup messages when a call arrives
at a voice gateway and matches this information against one of four dial-peer attributes
you can configure. First we’ll look at the types of call setup information we can gain from
inbound setup messages; then you’ll see how to match this information against the dialpeer attributes.
Inbound Dial-Peer Call-Setup Information
Depending on the type of connection your inbound dial peers are being received on, the
following call setup information can be used to route calls correctly based on inbound dial
peers:
Dialed Number Identification Service The DNIS is a number or extension that represents
the destination number the calling party wants to reach. DNIS is found only within Q.931
(ISDN BRI and PRI) and CAS signaling. Analog ports do not carry DNIS information.
Automatic Number Identification The Automatic Number Identification (ANI) is a
number or extension that represents the originating phone number. It is also referred to
as caller ID.
Inbound Port
This is the port number that a POTS call comes in on.
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Inbound Dial-Peer Configuration Attributes
We can gather the call setup information to help match our inbound call leg. Once we have
this information, we can match it against one of the four following dial-peer configuration
attributes. They are listed in the order in which the voice gateway checks them. As soon as
a match is made, the call is checked against the next outbound dial peer:
1. incoming called-number The number configured in this command is matched against
the DNIS if one is provided in the setup message.
2. answer-address The number configured in this command is matched against the ANI
if one is provided in the setup message.
3. destination-pattern The number configured in this command is matched against the
ANI if one is provided in the setup message.
4. port The port configured in this command is matched against the port that the
inbound call is made on if the call comes from a POTS line.
An inbound dial peer must have at least one match before it can move on
to the next call leg. If none of the four dial-peer configuration attributes match,
there is a fi fth “default” dial peer, known as dial-peer 0. It is explained in detail in the
next section.
When All Else Fails: Dial-Peer 0
Dial-peer 0 (or pid 0) is the last-resort method that POTS and/or VoIP inbound dial peers
are matched with if the fi rst four methods do not provide a match. It is needed because
the inbound dial peer must match something if the call is to move on to the next call leg.
Unfortunately, if no match is found using the fi rst four attributes, you have to play by the
dial-peer 0 default rules, which cannot be modified. Following are the dial-peer 0 rules,
which, as you will see, likely won’t be the optimal choice for your voice calls:
Uses any voice codec that the router can understand for VoIP dial peers.
No Resource Reservation Protocol (RSVP) support for VoIP dial peers.
This feature can reserve bandwidth along the call path to ensure that there is sufficient
bandwidth.
Uses fax-rate voice settings. This limits the amount of bandwidth available to fax calls
to the absolute minimum.
Does not use DTMF relay or any other nondefault voice-network capabilities that can
offer a more stable network infrastructure and thus ensure higher-quality calls. This is
true for both POTS and VoIP dial peers. The only difference is that VAD on VoIP dial
peers is enabled by default using dial-peer 0.
No DID support. You cannot use DIDs to forward off-network calls to on-network
phones.
No Interactive Voice Response (IVR) support for POTS dial peers.
Dial-Peer Digit Manipulation
335
Because most of these default settings are probably not ideal for your voice calls, you
should make sure that you attempt to match one of the four dial-peer configurations listed
previously.
Dial-Peer Digit Manipulation
The process of digit manipulation converts a dialed number into a different number to
reach the intended destination. There are many reasons to manipulate digits on your voice
system. You can manipulate dialed numbers by addition, subtraction, or substitution.
Listed here are some more common reasons:
To translate a full PSTN number (such as a 10 -digit E.164 number) to
a shorter extension so both internal and external calls can be made to a single
extension.
To have users dial an access code for PSTN calls. This access code must then be
stripped prior to actually placing the call on the PSTN.
To block calls to specific numbers.
To redirect calls to specific numbers.
The CUCM Express has several methods for manipulating numbers the way the
administrator wants them. This next section will cover digit-manipulation commands
including the following:
digit-strip
prefix
forward-digits
num-exp
translation-profile
Let’s look at each of these and use them in some situations that you will encounter in the
real world.
POTS Digit Manipulation Using Stripped Digits
When you configure a dial peer using wildcards, the leftmost digits in the destination
pattern that are explicitly defi ned are stripped off. This is because dial-peer statements have
digit stripping enabled by default. Here is an example POTS dial-peer configuration:
Router_A#configure terminal
Router_A(config)#dial-peer voice 10 pots
Router_A(config-dial-peer)#destination-pattern 555....
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Router_A(config-dial-peer)#port 1/0:1
Router_A(config-dial-peer)#end
Router_A#
This is a simple POTS dial peer that matches any dialed number beginning with 555.
When a call comes in that is destined for 555-1212, it matches the destination pattern
and is forwarded out port 1/0:1. Before the call is sent out the port along the call leg,
the leftmost explicitly defi ned digits are stripped, because digit stripping is enabled. This
means that only 1212 will be sent out port 1/0:1. For some situations, this may be desired.
But in other circumstances, you may want to send the 555 on the outbound call leg. To
accomplish this, we can add a no digit-strip command within the dial-peer statement.
Here is what the full configuration should look like if you want to send all seven digits out
the call leg:
Router_A#configure terminal
Router_A(config)#dial-peer voice 10 pots
Router_A(config-dial-peer)#destination-pattern 555...
Router_A(config-dial-peer)#no digit-strip
Router_A(config-dial-peer)#port 1/0:1
Router_A(config-dial-peer)#end
Router_A#
POTS Digit Manipulation Using Prefixes
Digit prefi x is the process of adding additional digits and/or pauses to the beginning of
the dialed number prior to passing it on. This can be useful in situations where you need
to add in specific digits that may have been stripped off when digit stripping is active in a
dial peer. Let’s look at an example where a DID number comes into the CUCM Express
as 1-312-555- 4773. The internal extension structure is configured for five-digit dialing.
Therefore we need to end up with 54773 as the number we wish to pass on. Here’s the
POTS dial peer that we have configured:
Router_B#configure terminal
Router_B(config)#dial-peer voice 50 pots
Router_B(config-dial-peer)#destination-pattern 131255.....
Router_B(config-dial-peer)#port 1/0:1
Router_B(config-dial-peer)#end
Router_B#
Because digit stripping is enabled, the CUCM Express will pass on only the last four
digits instead of the required five digits. To remedy this, we can use the prefix <numberstring> command to add our needed digit back in. Following is the full configuration with
the included prefix command:
Dial-Peer Digit Manipulation
337
Router_B#configure terminal
Router_B(config)#dial-peer voice 50 pots
Router_B(config-dial-peer)#destination-pattern 131255.....
Router_B(config-dial-peer)#prefix 5
Router_B(config-dial-peer)#port 1/0:1
Router_B(config-dial-peer)#end
Router_B#
Now our router can successfully pass on the required five digits for proper internal
extension dialing.
A second popular use for the prefix command is to add the number 9 to the beginning
of a dialed number so the router can use an outside line for off-network dialing. Let’s look
at this example:
Router_B#configure terminal
Router_B(config)#dial-peer voice 1 pots
Router_B(config-dial-peer)#destination-pattern 131255.....
Router_B(config-dial-peer)#no digit-strip
Router_B(config-dial-peer)#prefix 9,
Router_B(config-dial-peer)#port 1/1:1
Router_B(config-dial-peer)#end
Router_B#
Here you can see that we’re going to match 1312555 and then the 4 wildcard digits after
that. As you can see, we do not have digit stripping configured, so all the digits will be
sent to the PSTN. In addition, we have a prefi x configured to add 9, to the beginning of
the string. The 9 allows for off-network calling for PSTN calls. The comma (,) is called a
pause and informs the router to wait for one second before continuing to dial digits. In this
situation, the voice gateway will dial 9 and pause for one second. This allows time for the
PSTN to interpret the 9 as a signal to provide a second dial tone for off-network dialing.
After the voice gateway pauses, it continues to dial the remaining 11 digits.
POTS Digit Manipulation Using Forward-Digits
You have already learned that you can disable the default digit stripping of leftmost explicit
digits in a POTS dial-peer destination pattern. But what if you want only a certain number
of leftmost digits to remain? A handy command for this is the forward-digits number
command. With the forward digits command, you can inform the voice gateway to forward
either all digits received or a specific number of rightmost digits. Let’s say that you have the
following POTS dial peer configured:
Router_C#configure terminal
Router_C(config)#dial-peer voice 5 pots
Router_C(config-dial-peer)#destination-pattern 91312.......
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Router_C(config-dial-peer)#port 1/0:1
Router_C(config-dial-peer)#end
Router_C#
By default, the voice gateway will strip off 91312 and forward only the 7 remaining
digits. We want to be able to forward the 312 to the PSTN. Therefore, we can use the
forward-digits POTS dial-peer command to let the voice gateway know that we want
the last 10 dialed digits to be sent out the port. Here is the full POTS dial-peer
configuration to accomplish this goal:
Router_C#configure terminal
Router_C(config)#dial-peer voice 5 pots
Router_C(config-dial-peer)#destination-pattern 91312.......
Router_C(config-dial-peer)#forward-digits 10
Router_C(config-dial-peer)#port 1/0:1
Router_C(config-dial-peer)#end
Router_C#
POTS and VoIP Digit Manipulation
Using Number Expansion
The num-exp command is a global configuration command that you can use to match either
POTS or VoIP strings and change the number to anything you want. One good example of
number expansion is translating from extension numbers to full E.164 DIDs. In most PBX
environments configured today, internal dialing between phones that are both on network
can be accomplished by dialing a shortened extension as opposed to the full DID number.
Typically the internal extension is 3 to 5 digits in length. Let’s say that our ephone-DNs are
configured with the full E.164 10 -digit number, as shown with these two example ephoneDN configurations:
Router_D#configure terminal
Router_D(config)#ephone-dn 1
Router_D(config-ephone-dn)#number 7735553784
Router_D(config-ephone-dn)#ephone-dn 2
Router_D(config-ephone-dn)#number 7735553991
Router_D(config-ephone-dn)#exi
Router_D(config)#ephone 1
Router_D(config-ephone)#button 1:1
Router_D(config-ephone)#ephone 2
Router_D(config-ephone)#button 1:2
Router_D(config-ephone)#end
Router_D#
Dial-Peer Digit Manipulation
339
Now when ephone 1 wants to call ephone 2, the user must dial the full 10 -digit E.164
number to reach the destination. One way to get around this problem and allow users to
dial the preferred 4 -digit extensions is to use the num-exp command. Because the fi rst 7
digits (7735553) match both of our ephone-DNs, we can use the 7th digit for matching and
prepend the additional 6 -dial string to match the full 10 -digit ephone-DN. Here is how to
configure num-exp on a voice gateway to solve our example problem:
Router_D#configure terminal
Router_D(config)#num-exp 3... 7735553...
Router_D(config)#exit
Now the user at ephone 1 can dial 3991 and have the 6 additional digits added, which
will match ephone-DN 2 and ultimately ring ephone 2.
You can also completely change a number with the num-exp command. Let’s say we have
the following POTS dial peer configured:
Router_E#configure terminal
Router_E(config)#dial-peer voice 5 pots
Router_E(config-dial-peer)#destination-pattern 5040
Router_E(config-dial-peer)#port 1/0:1
Router_E(config-dial-peer)#end
Router_E#
The user at this extension wants to forward to 7447, which a POTS dial peer
configured on the voice gateway. We can do the following to forward all calls that match
5040 to 7447:
Router_E#configure terminal
Router_E(config)#num-exp 5040 7447
Router_E(config)#exit
To verify that your num-exp configurations are correct, you can use the show dialplan
number <number-string> command to verify that your phone number properly maps to a
dial peer, as follows:
Router_E#show dialplan number 5040
Macro Exp.: 7447
VoiceOverIpPeer7447
peer type = voice, system default peer = FALSE, information type = voice,
description = `’,
tag = 7447, destination-pattern = `7447’,
[output cut]
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The show dialplan number <number-string> command can be used
in verifying more than just show num-exp manipulations. This command
is useful anytime you want to verify which outgoing dial peer is reached
when a particular number is dialed.
You can also run a show num-exp to see all of the extension numbers you have mapped
to expanded numbers, as shown here:
Router_E#show num-exp
Dest Digit Pattern = ‘5040’
Translation =
‘7447’
Router_E#
POTS and VoIP Digit Manipulation
Using Translation Profiles
Translation profiles work similarly to access control lists (ACL) on a router. With access
lists, you create a unique ACL and provide permit and deny statements. You then apply the
ACL to a router interface using the access-group command.
Translation profi les use the same approach as ACLs. Your fi rst task is to create voicetranslation rules with a unique tag for identification. Each voice-translation rule can
contain up to 15 rules. The next step is to create a voice-translation profi le and set the
rules to be used for called or calling numbers. Lastly, the voice-translation profi le is applied
to the POTS or VoIP dial peer, using the translation-profile dial-peer command. It
can also be set globally on the voice gateway if you desire. This profi le can be applied to
incoming or outgoing calls on the dial peer.
Translation profi les are extremely flexible and can be used to provide more granular
control over modification scenarios than the other digit-manipulation tools previously
described. The creation of a translation profi le is a simple matter of entering into
configuration mode and using the translation-profile number command, where you can
tag the rule set by number. This example creates translation rule 1:
Router_F#configure terminal
Router_F(config)#voice translation-rule 1
Router_F(cfg-translation-rule)#
At this point, we are in cfg-translation-rule mode. Here we can create the individual
rules that reside in translation rule 1. The proper syntax for this is
rule <1-15> /match-number-string/ /replacement-number-string/
Let’s configure a few rules to match extensions 3111, 4111, and 5111 and set them
all to 6000:
Dial-Peer Digit Manipulation
341
Router_F(cfg-translation-rule)#rule 1 /3111/ /6000/
Router_F(cfg-translation-rule)#rule 2 /4111/ /6000/
Router_F(cfg-translation-rule)#rule 3 /5111/ /6000/
Router_F(cfg-translation-rule)#exit
Router_F(config)#
Keep in mind that it doesn’t matter what digits precede the match
numbers. The router is looking for this sequence anywhere it can.
Therefore, if you were to dial 5553111 on an interface where this translation
rule is applied, it would change the entire number to 6000.
Now that we have created our translation rules, we need to apply them to a voicetranslation profi le. We can select whether we want the profile to translate for called, calling,
redirect- called, or redirect-target numbers. Here’s an example of how to configure a voicetranslation profi le labeled to_6000 that includes all the rules from voice-translation rule 1:
Router_F(config)#voice translation-profile to_6000
Router_F(cfg-translation-profile)#translate ?
called
Translation rule for the called-number
calling
Translation rule for the calling-number
redirect-called Translation rule for the redirect-number
redirect-target Translation rule for the redirect-target
Router_F(cfg-translation-profile)#translate called 1
Router_F(cfg-translation-profile)#exit
Router_F(config)#
We have defi ned our voice-translation rules and have inserted those rules into a voicetranslation profi le, which specifies called or calling translations. We have yet to apply this
voice-translation profi le to anything, however. This is similar to creating a router ACL but
not applying it to an interface. The fi nal step is to use the translation-profile command
to apply the voice-translation profi le either to a POTS/VoIP dial peer or to an individual
ephone-DN. The profi le can be configured to translate for either incoming or outgoing
call legs. In our example, we will configure a POTS dial peer to use the to_6000 voicetranslation pattern on incoming calls. Here is the syntax:
Router_F(config)#dial-peer voice 100 pots
Router_F(config-dial-peer)#translation-profile ?
incoming Translation Profile for incoming call leg
outgoing Translation Profile for outgoing call leg
Router_F(config-dial-peer)#translation-profile incoming to_6000
Router_F(config-dial-peer)#end
Router_F#
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The rules can be applied to both incoming and outgoing call legs if you
wish. You need to have both configurations set to do so.
We can verify that our voice-translation rules actually work as described by issuing the
test voice translation-rule privileged exec command. Here you can see that all three
of our numbers configured in rule 1 are converted to 6000:
Router_F#test voice translation-rule 1 3111
Matched with rule 1
Original number: 3111
Translated number: 6000
Original number type: none
Translated number type: none
Original number plan: none
Translated number plan: none
Router_F#test voice translation-rule 1 4111
Matched with rule 2
Original number: 4111
Translated number: 6000
Original number type: none
Translated number type: none
Original number plan: none
Translated number plan: none
Router_F#test voice translation-rule 1 5111
Matched with rule 3
Original number: 5111
Translated number: 6000
Original number type: none
Translated number type: none
Original number plan: none
Translated number plan: none
VoIP to PSTN Failover Using Digit Manipulation and Preference Commands
Alexander is a lead IPT design consultant at a Cisco VAR. A customer based in Chicago
recently approached him to discuss a problem that arose after an outage took down all
PSTN lines at the company’s Denver location. Both sites use the PSTN for all off-network
calls. Because of the PSTN outage, employees in the Denver and Chicago locations could
not communicate with each other over the phone. The company is looking for a way to
provide redundancy using the established IP WAN that connects the two sites for data.
The solution that Alexander came up with was to utilize the WAN link for all calls
between the two locations and to fall back to the PSTN lines only if there is an outage.
Not only will this create a second path, which provides for high availability, it will also
lower the PSTN costs because calls will be routed over the IP WAN and no long-distance
Dial-Peer Digit Manipulation
charges will be incurred. The following diagram shows the new CUCM environment; you
can see the redundancy that the new VoIP to PSTN failover design offers.
er
f
re
c
en
e
IP WAN
0
pr
ef
er
en
ce
p
pre
fere
Chicago:
CUCME
nce
1
PSTN
0
nce
1
fere
pre
Denver:
CUCME
Extensions:
312-555-4XXX
Extensions:
303-555-5XXX
Using the preference command, Alexander can set the primary dial-peer path to be
the IP WAN connection. If the WAN were to go down for some reason, the dial-peer
would select the next-highest-preferred dial peer, which is the PSTN. Following are the
configuration options that Alexander needed to add to each CUCM Express in Denver to
configure redundant dial peers:
Denver-CUCME
Denver-CUCME#configure terminal
Denver-CUCME(config)#dial-peer voice 14000 voip
Denver-CUCME(config-dial-peer)#destination-pattern 4...
Denver-CUCME(config-dial-peer)#session target 172.16.1.1
Denver-CUCME(config-dial-peer)#preference 0
Denver-CUCME(config-dial-peer)#codec g729r8
Denver-CUCME(config-dial-peer)#exit
Denver-CUCME(config)#dial-peer voice 14001 pots
Denver-CUCME(config-dial-peer)#destination-pattern 4...
Denver-CUCME(config-dial-peer)#port 1/1:1
Denver-CUCME(config-dial-peer)#preference 1
Denver-CUCME(config-dial-peer)#prefix 1312555
Now on to the Chicago CUCM Express:
Chicago-CCME#configure terminal
Chicago-CCME(config)#dial-peer voice 15000 voip
Chicago-CCME(config-dial-peer)#destination-pattern 5...
Chicago-CCME(config-dial-peer)#session target 172.16.1.2
Chicago-CCME(config-dial-peer)#preference 0
Chicago-CCME(config-dial-peer)#codec g729r8
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Chicago-CCME(config-dial-peer)#exit
Chicago-CCME(config)#dial-peer voice 15001 pots
Chicago-CCME(config-dial-peer)#destination-pattern 5...
Chicago-CCME(config-dial-peer)#port 1/1:1
Chicago-CCME(config-dial-peer)#preference 1
Chicago-CCME(config-dial-peer)#prefix 1303555
Chicago-CCME(config-dial-peer)#end
Chicago-CCME#
Now both CUCM Express routers will always prefer to send data over the IP WAN. Notice
that Alexander is also requiring that any call destined for the IP WAN use the G.729 codec.
This codec requires less bandwidth and allows for more simultaneous calls over a fixedbandwidth link. The problem with H.323 and the G.729 protocol is that by default, H.323
sends the DTMF tones in band. This means that the tones will be compressed along with
the voice. Because of the additional compression that G.729 performs, the DTMF tones are
often unrecognizable when decompressed on the other side. To get around this problem,
Alexander configured both voice gateways to send DTMF out of band using the H.245
standard format. He used the dtmf-relay h245-alphanumeric command to accomplish
this. He added the commands to both voice gateways to complete the Denver configuration:
Denver-CUCME#configure terminal
Denver-CUCME(config)#dial-peer voice 14000 voip
Denver-CUCME(config-dial-peer)#dtmf-relay h245-alphanumeric
And on the Chicago CUCM Express:
Chicago-CCME#configure terminal
Chicago-CCME(config)#dial-peer voice 15000 voip
Chicago-CCME(config-dial-peer)#dtmf-relay h245-alphanumeric
If the WAN link were to have a failure, the session target IP address listed in the VoIP dial
peer would no longer be in the routing table. When this happens, the dial peer is taken
out of consideration from the call-routing process. Therefore, the next-highest preference
that matches the destination pattern takes precedence. This happens to be the T1 POTS
link that he configured at both sites. You’ll notice that he used the prefix command to
add a 1 and then the area code and office code numbers. These need to be provided
to the PSTN so it knows the location of the offices.
Understanding the Digit-Manipulation Hierarchy
In a complex voice environment where dialed strings of numbers may go through multiple
types of digit manipulation, you likely begin to question when one digit-manipulation
method is run before another method is performed. With all the different digit
Configuring a Trunk between Voice Gateways using H.323 and SIP Trunks
345
manipulations occurring, it can get confusing to figure out what the fi nal number will be.
Figure 7.8 shows the digit- manipulation hierarchy.
F I G U R E 7. 8
The digit-manipulation hierarchy
Forward
-digits
Processed Last
prefix
voice-translation profile
digit-strip
num-exp
Processed First
The method at the bottom (num-exp) is applied fi rst. Then the next layer is applied, and
so on until you reach the last manipulation process to be applied, which is the forwarddigits option. This should help you figure out what happens when a dialed number is
subject to two or more digit-manipulation processes.
Configuring a Trunk between Voice
Gateways using H.323 and SIP Trunks
You may be surprised to hear this, but you’ve already seen an H.323 trunk configured
between two voice gateways in the “Configuring VoIP Dial Peers over WAN Connections”
section of this chapter. In case you missed this little fact, we’ll go over how to configure
a simple H.323 trunk between two voice gateways. Then you’ll see how to configure those
same two voice gateways for simple SIP trunking.
H.323 Trunking
Anytime you configure a VoIP dial peer between two voice gateways or CUCM Express
systems, by default you are creating an H.323 trunk between the peer systems. Calls
are then routed over the IP network when the destination pattern is matched. This type
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of H.323 trunk is the simplest to implement and it’s all you will need to know how to
configure within the limits of the CCNA Voice certification. Here is another quick example
of how to configure H.323 trunking between voice gateways using the g.711 codec. We’ll
use Figure 7.9 as our example.
F I G U R E 7. 9
An example of H.323 gateway communication
Extensions:
4XX
H.323 Signaling
V
Site_A
10.0.0.1
10.0.0.2
V
Site_B
Extensions:
5XX
Here is the configuration syntax for site A:
Site_A#configure terminal
Site_A(config)#dial-peer voice 500 voip
Site_A(config-dial-peer)#destination-pattern 5..
Site_A(config-dial-peer)#codec g711ulaw
Site_A(config-dial-peer)#session target ipv4:10.0.0.2
Site_A(config-dial-peer)#end
Site_A#
And here’s the configuration for Site_B:
Site_B#configure terminal
Site_B(config)#dial-peer voice 400 voip
Site_B(config-dial-peer)#destination-pattern 4..
Site_B(config-dial-peer)#codec g711ulaw
Site_B(config-dial-peer)#session target ipv4:10.0.0.1
Site_B(config-dial-peer)#end
Site_B#
SIP Trunking
Configuration of a simple SIP trunk between two voice gateways involves using a command
to change the signaling protocol from the default H.323 to SIP version 2. The session
protocol sipv2 dial-peer command changes the trunk signaling to the proper format. As
soon as the signaling is changed on both sides, your VoIP dial peer will use SIP signaling
Summary
347
instead of H.323. Let’s configure the same two -site setup as described in Figure 7.9, but
this time we’ll configure SIP trunking on our dial peers. Here is Site_A:
Site_A#configure terminal
Site_A(config)#dial-peer voice 400 voip
Site_A(config-dial-peer)#destination-pattern 4..
Site_A(config-dial-peer)#session protocol sipv2
Site_A(config-dial-peer)#codec g711ulaw
Site_A(config-dial-peer)#session target ipv4:10.0.0.1
Site_A(config-dial-peer)#end
Site_A#
And here’s Site_B:
Site_B#configure terminal
Site_B(config)#dial-peer voice 500 voip
Site_B(config-dial-peer)#destination-pattern 5..
Site_B(config-dial-peer)#session protocol sipv2
Site_B(config-dial-peer)#codec g711ulaw
Site_B(config-dial-peer)#session target ipv4:10.0.0.2
Site_B(config-dial-peer)#end
Site_B#
That’s really all there is to connecting two Cisco SIP voice gateway peers. Later on
in this book, you’ll learn how to connect to an ITSP with a SIP trunk and SIP user
authentication using the UC500 series CUCM Express.
Summary
Chapter 7 began with a look at how to configure analog interfaces and then how to use
POTS dial peers to route calls over the voice ports. We then moved on to discuss how
to configure digital T1 CAS and PRI interfaces as well as how to use POTS dial peers to
direct calls over the digital connections. Then you saw how to use VoIP dial peers to route
calls to remote voice gateways across an IP WAN and learned some dial-plan strategies to
help simplify the dial-peer setup and manipulation. We then discussed the voice-gateway
decision-making process and various techniques to manipulate that process. Finally, we
took a closer look at connecting voice gateways over an IP network using both H.323 and
SIP signaling protocols.
While teleporting voice traffic across a voice gateway is not as spectacular as using the
starship Enterprise’s teleporter, both function on the same principle of deconstructing
something into a medium for transport so it can be reconstructed on the other end. Beam
me up, Scotty!
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Exam Essentials
Know how to configure FXS and FXO analog ports. Understand how to apply caller-ID
names/numbers, call progress tones, ring frequencies, and signaling information to analog
interfaces.
Understand the purpose of PLAR. interfaces and how to configure them on FXS and
FXO ports for both outbound and inbound calling. PLAR interfaces create an automatic
connection either when a phone goes off-hook (FXS) or when a call comes into the
interface (FXO).
Know how to configure POTS dial peers for FXS and FXO analog ports. With FXS
ports, you use the extension number as the destination pattern and apply it to an
FXS analog port. With FXO ports, you typically want to match many numbers for routing
outbound calls either to the PSTN or another PBX. To match multiple numbers in the
fewest destination patterns possible, you utilize dial-peer wildcards.
Know how to configure CAMA-signaled FXO ports for E.911 calling. CAMA ports
connect to a PSAP network rather than the PSTN. These connections use their own
signaling protocol.
Know how to configure T1 CAS ports. Understand the configuration options of framing
type, linecode type, clock source, and ds0 -group options of T1 CAS interfaces.
Know how to configure T1 PRI ports. Understand the ISDN switch-type options available
to you. Also understand the pri-group options, which set each voice timeslot for TDM. If
you configure a fractional T1 PRI, remember that the D channel is always timeslot 23.
Know how to configure POTS dial peers for T1 CAS and PRI circuits. Dial peers for T1
CAS and PRI circuits are very similar. The main difference is that the T1 PRI circuit always
specifies the D channel in the port configuration.
Understand and know how to configure VoIP dial peers. VoIP dial peers connect voice
gateways over an IP network. The configuration process is similar to creating POTS dial
peers except that instead of specifying a physical port to send data out of, you specify the
IP address of the remote voice gateway.
Understand the necessity of creating a dial-plan strategy. When designing a voice
network, you want to limit the complexity of your dial-peer setups. Therefore, it is
important to have a plan in place for phone number assignment in the environment that
meets the needs for a business today and into the foreseeable future.
Understand the dial-peer decision-making process. Know how voice gateways choose
to route calls based on inbound and outbound dial peers. Understand how dial-peer 0
functions as the last resort.
Know how to manipulate numbers using digit-manipulation techniques. Techniques such
as digit-strip, prefi x, destination-pattern, forward-digits, and translation-profi les can be
Hands - on Labs
349
used to help you forward the exact numbering scheme to the next voice gateway
or endpoint.
Know how to configure H.323 and SIP trunks between voice gateways. By default,
creating a VoIP dial peer between two voice gateways results in the setup of an H.323
trunk. You can also easily change that trunk to use SIP signaling using the session
protocol sipv2 config- dial-peer command.
Written Lab 7.1
Write the answers to the following questions:
1. What is the config-voiceport command to set signaling for ground start?
2. What is the config-voiceport command to configure a PLAR to automatically ring
extension 2111?
3. What config- dial-peer command tells the voice gateway to send all digits to the next
call leg?
4. What is the config- controller command to set the framing type to Extended Super
Frame?
5. When you want to set your full T1 CAS circuit timeslots to FXO Loop Start, what
config- controller command do you enter?
6. You are configuring a T1 PRI circuit. What is the config command to set the ISDN
switch to which your circuit connects to primary-ni?
7. How do you view a brief summary of the operational status of all of your POTS
circuits on a voice gateway?
8. What VoIP config- dial-peer command sets the next call leg to 10.1.1.100?
9. You’ve just created a translation rule on your voice gateway to translate 401 to 501.
How do you verify that the rule is working properly using the command line?
10. What config- dial-peer command sets the trunking signaling protocol to SIP version 2?
(The answers to Written Lab 7.1 can be found following the answers to the review
questions for this chapter.)
Hands-on Labs
To complete the labs in this section, you need a router with a voice- capable IOS, T1 PRI
interface, Fast Ethernet interface, and two FXS ports for analog phones. The CUCM
Express should be properly set up and ready for configuring IP phones. Each lab in this
section builds on the last and follows the logical CUCM Express PBX model design as
shown in Figure 7.10.
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F I G U R E 7.1 0
Configuring Voice Gateways for POTS and VoIP
Diagram for the voice gateway labs
FXS: 0/0/0
Analog Ext.
301
T1 PRI: 1/0/0
FXS: 0/0/1
V
VG_1
Analog Ext.
302
PSTN
Fa2/0:
172.16.1.1/30
IP WAN
H.323 Trunk
G.729
V
Remote_VG
Dial Plan 4XX
Here is a list of the labs in this chapter:
Lab 7.1: Configuring FXS Interfaces for Two Analog Phones
Lab 7.2: Configuring a T1 PRI Interface
Lab 7.3: Configuring an H.323 Trunk
Lab 7.4: Configuring Translation Profi les
Hands-on Lab 7.1: Configuring FXS Interfaces
for Two Analog Phones
In this lab, we’re going to configure a voice gateway that has two analog phones connected
via FXS ports. The phones are located in the United States, so we can use the default
cptone and ring frequencies. Table 7.6 lists the information needed to configure the ports:
TA B L E 7. 6
Information for port configuration
Extension
Caller-ID Number
Caller-ID Name
301
5558301
Marty Jones
302
5558302
Samantha Wilson
1. Log in to your CUCM Express router, and go into configuration mode by typing
enable and then configuration terminal.
2. Enter config-voiceport mode by typing voice-port 0/0/0.
Hands-on Labs
351
3. Configure FXS port 0/0/0 to have the seven- digit caller-id number by typing stationid number 5558301.
4. Enable caller ID by typing caller-id enable.
5. Configure FXS port 0/0/0 to have the correct caller-id name by typing station-id
name Marty Jones.
6. Configure the next FXS port by typing voice-port 0/0/1.
7. Configure FXS port 0/0/1 to have the seven- digit caller-id number by typing stationid number 5558302.
8. Enter config-voiceport mode for the next FXS interface by typing voice-port 0/0/0.
9. Enable caller ID by typing caller-id enable.
10. Configure FXS port 0/0/1 to have the correct caller-id name by typing station-id
name Samantha Wilson.
11. Exit config-voiceport mode by typing Exit.
12. Configure a POTS dial peer for FXS port 1 by typing dial-peer voice 301 pots. You
will now be in config-dial-peer mode.
13. Configure the destination pattern to match the three- digit extension and have the
dial peer point to FXS port 0/0/0 by typing destination-pattern 301 and then port
0/0/0.
14. Configure a POTS dial peer for FXS port 2 by typing dial-peer voice 302 pots. You
will now be in config-dial-peer mode.
15. Configure the destination pattern to match the three-digit extension and have the dial
peer point to FXS port 0/0/1 by typing destination-pattern 302 and then port 0/0/1.
16. Exit config- dial-peer mode by typing end.
Hands-on Lab 7.2: Configuring a T1 PRI Interface
1. Log in to your CUCM Express router and go into configuration mode by typing
enable and then configuration terminal.
2. Configure the ISDN switch type to be primary-ni by typing isdn switch-type
primary-ni.
3. Enter into config- controller mode by typing controller t1 1/0/0.
4. Set your pri-group to utilize all 23 POTs lines by typing pri-group timeslots 1-24.
5. Configure framing and linecoding types by typing framing esf and then linecode
b8zs.
6. Set the T1 to receive clocking from the PSTN equipment by typing clock source
line.
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7. Exit config- controller mode by typing Exit.
8. Configure a POTS outbound dial peer for the T1 PRI by typing dial-peer voice 10
pots. You will now be in config- dial-peer mode.
9. Configure the destination pattern to match any 10 - digit E.164 number and have
the dial peer point to the T1 interface port 1/0/0 by typing destination-pattern
91.......... and then port 1/0/0:23.
10. Exit config- dial-peer mode by typing end.
Hands-on Lab 7.3: Configuring an H.323 Trunk
This trunk connects to an already configured remote site that uses extensions in the 4XX
range. We need to set up the physical Fast Ethernet. Assume that the remote site only has
analog phones attached to FXS ports. Therefore, we don’t have to worry about routing IP
traffic to the remote site.
1.
Log in to your CUCM Express router and go into configuration mode by typing
enable and then configuration terminal.
2.
Enter config-if mode by typing interface fa2/0.
3.
Assign an IP address to match the diagram by typing ip address 172.168.1.1
255.255.255.0.
4.
Exit config-if mode by typing Exit.
5.
Configure a VoIP outbound dial peer for the WAN link by typing dial-peer voice 10
pots. You will now be in config- dial-peer mode.
6.
Configure the destination pattern to match any 10 -digit E.164 number and have
the dial peer point to the T1 interface port 1/0/0 by typing destination-pattern
91.......... and then port 1/0/0:23.
7.
Configure a VoIP dial peer for the WAN interface by typing dial-peer voice 400
voip. You will now be in config- dial-peer mode.
8.
Configure the destination pattern to match the four plus two additional
wildcard digits and have the dial peer point to the remote-side WAN IP
address of 172.16.1.2 by typing destination-pattern 4.. and then session-target
ipv4:172.168.1.2.
9.
Make sure that the 4 is not stripped off when the number is sent to the remote voice
gateway by typing no digit-strip.
10. Force the WAN to use a lower-bandwidth codec by typing codec g729r8.
11. Exit config- dial-peer mode by typing end.
Hands-on Labs
353
Hands-on Lab 7.4: Configuring Translation Profiles
In this lab, we’re going to configure translation profiles for our two analog
phones attached to FXS voice-ports 0/0/0 and 0/0/1. Right now, the phones will ring
when the three-digit extension is called. But callers from the PSTN will be dialing DIDs
that don’t correspond with our extensions. That means when PSTN calls come in, they
will not go through, because we don’t have a dial peer configured for the DID. To remedy
this, we’re going to create translation profiles for two PSTN DID numbers (555-4777
and 555-4952), mapping them to the three-digit FXS extensions, which will then
be matched against the POTS dial peers we created in Lab 7.1.
1.
Log in to your CUCM Express router and go into configuration mode by typing
enable and then configuration terminal.
2.
Enter cfg-translation-rule mode by typing voice translation-rule 1.
3.
Add two rules to translate the seven-digit PSTN DIDs to the three-digit extensions
assigned to the FXS interfaces by typing rule 1 /5554777/ /301/ and then rule 2
/5554952/ /302/.
4.
Exit cfg-translation-rule mode by typing Exit.
5.
Enter cfg-translation-profile mode by typing voice translation-profile from_pstn.
6.
Add our newly configured translation rule (rule 1) for called numbers to the translation
profile by typing translate called 1.
7.
Exit cfg-translation-profile mode by typing Exit.
8.
Enter config-dial-peer mode for our already created T1 PRI dial peer by typing dialpeer voice 10 pots.
9.
Add the translation profile (from_pstn) to search for and translate incoming DIDs
from the PSTN by typing translation-profile incoming from_pstn.
10. Exit config- dial-peer mode by typing end.
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Configuring Voice Gateways for POTS and VoIP
Review Questions
1.
What are the two types of dial peers?
A. PSTN dial peer
2.
B.
POTS dial peer
C.
VoIP dial peer
D.
IPT dial peer
E.
CUE dial peer
Which of the following outbound dial peers will finally be matched when a user at
extension 4555 calls 5888?
A. Destination pattern 4…
3.
B.
Destination pattern 5…
C.
Destination pattern 5T
D.
Destination pattern 4T
You want to configure a VoIP target to point to 192.168.18.10. What is the proper syntax?
A. voip target ipv4:192.168.18.10
4.
B.
session-target 192.168.18.10
C.
session target ipv4:192.168.18.10
D.
voip-target 192.168.18.10
What type of signaling is commonly configured on FXS ports?
A. SIPv2
B.
Loop start
C.
Line start
D.
B8zf
E.
Ground start
Review Questions
5.
355
Which two dial-peer configurations will correctly route emergency service calls out
CAMA port 1/0/1?
A. Router(config)#dial-peer voice 911 pots
Router(config-dial-peer)#destination-pattern .911
Router(config-dial-peer)#port 1/0/1
B.
Router(config)#dial-peer voice 911pots
Router(config-dial-peer)#destination-pattern 911
Router(config-dial-peer)#port 1/0/1
C.
Router(config)#dial-peer voice 911pots
Router(config-dial-peer)#destination-pattern 9911
Router(config-dial-peer)#forward-digits all
Router(config-dial-peer)#port 1/0/1
D.
Router(config)#dial-peer voice 911pots
Router(config-dial-peer)#destination-pattern 911
Router(config-dial-peer)#forward-digits all
Router(config-dial-peer)#port 1/0/1
E.
Router(config)#dial-peer voice 911pots
Router(config-dial-peer)#destination-pattern 9911
Router(config-dial-peer)#prefix 911
Router(config-dial-peer)#port 1/0/1
6.
What term defines the need to modify the voltage of analog FXS endpoints to match
the phones that require a different setting to properly ring?
A. Ring frequency
7.
B.
Cptone
C.
Dial plan
D.
DTMF
What describes the function when a user picks up a phone handset and the phone automatically dials a preconfigured extension?
A. PLAR
8.
B.
CAMA
C.
SRST
D.
Call leg
When configuring FXO dial-type, what two options do you have?
A. POTS
B.
VoIP
C.
Pulse
D.
DTMF
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9.
Configuring Voice Gateways for POTS and VoIP
What does the FXO interface command ring number do?
A. Defines the outbound PSTN number to call
B.
Sets the ring frequency
C.
Sets the number of rings before the voice gateway answers the incoming call
D.
Sets the number of rings before the voice gateway forwards the call
10. What command can be used within a dial-peer statement to ensure that all digits are
forwarded to the destination?
A. forward-digits all
B.
no digit-strip
C.
no prefix
D.
destination-pattern T
11. What configuration setup is commonly configured on FXO ports to forward incoming
calls to an operator or auto attendant?
A. digit-strip
B.
PLAR
C.
no digit-strip
D.
CAMA
12. When you physically install T1 hardware on your voice gateway, what type of inter-
faces will your T1 interfaces be described as?
A. Digital-port
B.
Session
C.
Controller
D.
Service -module
13. What term describes how an administrator determines where a T1 circuit will receive
clocking information?
A. NTP
B.
Time/date stamp
C.
Clock source
D.
Clock linecode
14. What global configuration option must be performed for T1 PRI circuits to communi-
cate properly with the PSTN or connected PBX?
A. Enable SIP signaling
B.
Enable SCCP signaling
C.
Set the E&M controller
D.
Set the ISDN switch type
Review Questions
357
15. What T1 PRI timeslot is always used as the destination port when configuring dial
peers?
A. Timeslot 1
B.
Timeslot 24
C.
Timeslot 30
D.
Timeslot 23
16. What is the VoIP equivalent to the POTS dial-peer port command?
A. destination-pattern
B.
session target
C.
ip route
D.
codec
17. Why is it important to design a flexible dial-plan strategy for your network?
A. To prevent routing loops.
B.
To comply with strictly enforced ITU -T E.164 guidelines.
C.
To efficiently route calls between sites with the fewest number of destinationpattern commands.
D.
Your PSTN or ITSP provider will give you your dial plan strategy. You simply need to
follow their design.
18. Which POTS circuits carry DNIS information? Choose all that apply.
A. FXO
B.
ISDN BRI
C.
ISDN PRI
D.
CAS T1
19. What is the default VoIP trunk-signaling protocol used on Cisco voice gateways?
A. SIP
B.
MGCP
C.
H.323
D.
SCCP
20. Within the voice-gateway digit-manipulation hierarchy, which method is always
applied last?
A. Translation profile
B.
Digit-strip
C.
Destination-pattern
D.
Forward- digits
E.
Number expansion
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Answers to Review Questions
1.
B, C. The two types of dial peers are POTS and VoIP.
2.
B. You can narrow the answer down to B and C based on the fi rst digit matching the
destination pattern. B is the more exact choice because it specifies that the extension is four
digits in length.
3.
C. The proper command syntax is session target ipv4:192.168.18.10.
4.
B. FXS ports are most commonly configured with loop start signaling.
5.
D, E. Don’t forget that explicit matches are not forwarded to the destination.
That means you must use the forward-digits command to send the three digits of 911 or
use the prefix command to explicitly send 911 to the destination.
6.
A. Changing the ring frequency informs the voice gateway what type of analog device
is being used. Depending on where you are located, different phones use different ring
frequencies, which are measured in Hertz.
7.
A. A private line automatic ringdown (PLAR) acts as a hotline phone. When the phone
goes off-hook, it will ring an extension without any user interaction.
8.
C, D. The dial-type command deals with the type of digit signaling the port expects to
hear from the PSTN. The two options are DTMF and pulse dialing.
9.
C. The ring number signifies the maximum number of rings the voice gateway waits before
answering the call.
10. B. The no digit-strip command ensures that all digits including those explicitly defi ned
are forwarded to the destination.
11. B. PLAR is often used so the FXO interface answers the call directly on the port and
immediately forwards it to the operator extension or AA pilot number.
12. C. The T1 hardware installed on a voice gateway will be seen as controller interfaces in
the IOS configuration.
13. C. Clock source is the term to defi ne where a T1 receives clocking information. The
choices are free-running, internal, and line.
14. D. The ISDN switch type must be configured globally so Q.931 signaling can be sent to
and received by our PRI peers.
15. D. PRI timeslot 23 (channel 24) is always used for out-of-band signaling. When
configuring the destination port, you point it toward the D channel.
16. B. Both the port and session target commands tell the voice gateway where the next
hop for calls should be routed.
Answers to Review Questions
359
17. C. Proper planning should go into any new voice network to limit the number of
destination-pattern commands. The fewer destination-pattern rules, the easier your
voice gateway will be to troubleshoot and manage.
18. B, C, D. DNIS information is found only on digital circuits. FXO ports are analog and
therefore cannot carry DNIS information.
19. C. By default, H.323 is used for signaling when you configure a VoIP trunk.
20. D. The forward-digits digit-manipulation method is performed after all other methods on
Cisco voice gateways.
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Chapter 7
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Answers to Written Lab 7.1
1.
signal loopStart
2.
connection plar 2111
3.
forward-digits all or no digit-strip
4.
framing esf
5.
ds0-group 0 timeslots 1-24 type fxo-loop-start
6.
isdn switch-type primar-ni
7.
show voice port summary
8.
session target ipv4:10.1.1.100
9.
test voice translation-rule 401
10. session protocol sipv2
Chapter
8
Unity Express
Overview and
Installation
THE FOLLOWING CCNA VOICE
EXAM OBJECTIVES ARE COVERED
IN THIS CHAPTER:
Implement voice mail features using Cisco Unity Express.
Configure the foundational elements required for Cisco
Unified Communications Manager Express to support
Cisco Unity Express.
Describe the features available in Cisco Unity Express.
Implement voice mail features using Cisco Unity Express.
Describe the Cisco Unity Express hardware platforms.
I have heard arguments that email has rendered voice mail
obsolete. Despite what critics say, voice mail is far from dead.
It’s a vital part of business communication that is still widely
used today. Think about it. Can you imagine calling a business and not being able to leave
a voice mail? Sure, you can send an email, but voice mail messages add a personal touch
that email cannot get across. It is also important to note that voice mail technology is
continuing to evolve to provide users with advanced features and functionality that make
it even more useful today than it was 5 to 10 years ago. In fact, I see email and voice mail
continuing to merge to the point where it is difficult to separate the two.
In this chapter we will explore the various features available to users and groups
in Cisco’s voice mail tool, the Unity Express system. We’ll then check out some of the
more advanced functionality such as the ability to provide integrated messaging with
IMAP email clients such as Microsoft Outlook. Then we’ll move beyond the voice mail
capabilities of Unity Express and explore other functions such as the auto attendant and
optional automated voice response system. Cisco Unity Express (CUE) has a fairly complex
licensing structure, and this chapter covers all the licensing for the various functions of
Unity Express. I will show you how to install, initialize, and upgrade the CUE software.
Finally, you’ll learn a quick way to restore your Unity Express to factory default settings.
Understanding Unity Express
Voice Mail Features
This section will describe voice mailbox users and groups that can be set up to use the
system. Users are assigned individual subscriber mailboxes, while groups use a shared
general-delivery mailbox. Once you’ve seen how users and groups are defined, I’ll detail
many of the features that mailbox owners can utilize. I’ll then describe the different
voice mail caller options available when people call in to Unity Express either to leave
messages for other users/groups or to log in to their own mailbox remotely to check messages.
Finally, I’ll touch on some of the more advanced features that Unity Express offers.
Users/Subscribers
Unity Express users, known as subscribers, are individual accounts that are created to
provide personal mailbox accounts for voice mail storage. The terms user and subscriber
can be used interchangeably. These individual mailbox accounts are known as subscriber
Understanding Unity Express Voice Mail Features
363
accounts. Owners of these accounts can customize their mailboxes to suit their needs.
Each subscriber can be assigned a username, PIN, and password. These credentials allow
a user to manage their account. The personal identification number (PIN) is used when the
subscriber manages their mailbox using the telephone user interface (TUI). The username
and password are used when the subscriber manages their account using the web GUI
interface or other email access protocols.
Later in this chapter you’ll see that users who have already been created as ephone-DN
owners can be imported automatically into Unity Express. When this import occurs, a
password and PIN can be any of the following:
Randomly assigned
Left blank
Manually entered by the administrator
If you change a user ’s password from within the CUE operating system,
the password for that user is updated on the CUCM Express automatically.
However, if the change is made on the CUCM Express OS side, the
password will not automatically be updated on Unity Express. You should
use the Administration Synchronize Unity Express GUI option to make
sure users and passwords are the same on both systems.
Voice mail subscribers configured on Unity Express use the following information to
create individual mailboxes:
User Name Full name: fi rst, last.
Group
Name of the group of which the user is a member.
Password Used for logging in to the Unity Express GUI.
PIN Used to authenticate a user who is using the TUI. When a subscriber logs in to the
TUI for the fi rst time, they are required to change the default PIN.
Groups
Groups are collections of subscribers who have some sort of commonality. Typically,
users within the same business function are bundled together to form a Unity
Express group. For example, there may be a regional sales group and a customer service
group. Members of a particular group can be either users or other groups. That is,
entire groups can be contained within another group. For example, the regional sales
group might be a member of the national sales group. A group is assigned a single
telephone extension with a shared mailbox. A mailbox that is assigned to a group is
called a general delivery mailbox (GDM). At any one time, only one member can access
the GDM. There is no PIN assigned. The user fi rst logs into their personal mail account
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and then can choose to access the GDM from the TUI interface. There are two types of
group users:
Group member Can access the general delivery mailbox to check messages and perform
other mailbox functionality.
Group owner Can add or delete group members and group owners. An owner is not
considered a group member and therefore cannot access the GDM.
A group’s owner is not automatically a member of the group. The owner account must
also be set up as a member in order to access the GDM.
Mailbox Owner Features
There are literally dozens of mailbox features for both subscriber and group voice
mailboxes. In this section, we’re going to discuss the features that are most commonly
deployed.
Spoken Name
When a user logs into their mailbox for the fi rst time, they can record their spoken name
to identify themselves as the owner of the mailbox. The spoken name is used when another
user forwards a voice mail message, when the user does not record a personal greeting, or
in the auto attendant when a caller chooses the user extension.
Personal Greeting
A user can set up a personal greeting for their voice mailbox. When a caller is transferred
to a user’s voice mailbox, the personal greeting is played, which typically asks the caller to
leave a name, number, and reason for calling. If the mailbox user does not create a personal
greeting, a CUE standard greeting is played in its place, which tells the caller to leave a
message, either for extension XXXX or using the spoken name if one was recorded.
Alternate Greeting
An alternate greeting can be created for occasions when the user wants to switch quickly
between a personal greeting and a different greeting. This is useful when the user needs to
have two different greetings but does not wish to rerecord the personal greeting each time.
When the alternate greeting is active, the personal greeting is in an inactive state but is still
stored on the system.
Operator Assistance
The operator assistance feature is also called a “zero out” feature. It allows callers to
press 0 to reach an operator. The actual extension that is called can be the default number
defi ned by the Unity Express administrator, or it can be locally defi ned by the mailbox
subscriber or group member. This feature lets callers opt out of leaving a message and
attempt to speak with a live operator or assistant instead.
Understanding Unity Express Voice Mail Features
365
Tutorial
When a new subscriber or group mailbox is created, a user or group member can log in
to their mailbox and use a TUI tutorial, which walks the user through basic setup of their
mailbox. The voice prompts will step the user through setting up the following:
Spoken name
Greeting
Change PIN
Operator assistance
Message Waiting Indicator
Two different indicators inform users of new voice mails. The first, the message waiting
indicator (MWI), is a red light found on all Cisco IP phones. This light is a prominent visual
notification to the phone owner that they have a message waiting on the extension assigned
to button 1 of a multiline phone. Only the extension assigned to button 1 uses the red light
MWI. A second notification found on Cisco IP phones with LCD displays is the flashing
envelope icon. This icon will flash next to the extension number on the button where the new
voice mail was left. Figure 8.1 shows the flashing envelope icon next to extension 5004.
F I G U R E 8 .1
Flashing envelope MWI
Flashing
Envelope
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This envelope icon indicates that a caller dialed extension 5004 and left a voice mail in
the mailbox assigned to that extension. Also note that the red light on the phone is not lit.
This is because extension 5004 is configured on button 2 of the phone, and the red light
is used only when messages are waiting on button 1. A phone with multiple extensions
may simultaneously have one extension tied to the MWI but have multiple lines with the
envelope icons showing.
Message Notifications
The MWI described previously is a great way to notify users of a new voice mail when they
are physically in front of their phone and can see the red light and flashing envelope icon.
But as we know, many of our users are mobile and constantly away from their desks. They
would have no idea that a new voice mail was waiting for them unless they periodically
dialed in to check, a method that is not cost or time efficient. Fortunately, with the Unity
Express version 3.1 and later versions, we can configure message notifications to alert users
to a voice mail on any of the following:
Another telephone (home, cellular, and so on)
Numeric or text pager
Email account
Message notification is globally enabled at the system level by the Unity Express
administrator. The administrator then has the ability to control how message notifications
are handled on a per-user or group basis. The administrator can also specify either that all
new voice mail messages are sent to notification destinations or that only messages flagged
as urgent are sent out.
Once the Unity Express administrator configures and enables the message-notification
parameters, subscribers and group members can log in to their mailbox using either
the TUI or GUI interface to specify the phone numbers and/or email addresses where the
notifications are to be sent.
For email notifications to be sent, it is important that you remember to
configure your Unity Express to point to an SMTP server. If you don’t,
emails will not be sent to users.
Users and/or group members can also configure cascading message notifications. This
technique sets up a list of multiple notification destinations that have priorities assigned to them.
The Unity Express will send an alert notification to the first destination and wait a defined
period of time. If the timer expires and the new message has not been checked, the Unity
Express system will move to the next destination and send a second notification, and so on.
Understanding Distribution Lists
Many times, you want to be able to send the same voice mail message to multiple people.
You could call every extension associated with the person you wish the message to
reach, but that would not be efficient. Instead, Unity Express offers the ability to create
Understanding Unity Express Voice Mail Features
367
distribution lists, which allow users to create lists of subscribers so they can send a
single voice mail message to multiple subscriber and/or GDM mailboxes. Think of it as
forwarding an email to multiple email recipients. Unity Express supports two types of
distribution lists:
Public distribution lists
Private distribution lists
Let’s look at each of these so you can better understand the differences between the two.
Public Distribution Lists
A public distribution list is a list that is typically defi ned by the Unity Express system
administrator. These lists are for any subscriber or group to use. It is common for
public distribution lists to be created for the various departments that would need
similar notifications. It is also common to see a public distribution list created grouping
department managers together.
Private Distribution Lists
Private distribution lists are created by an individual subscriber or group. These lists
can be used only by the owner of the private list. Users can customize a list that fits their
specific job functions. These lists are commonly much more granular in nature than the
public distribution lists available.
Mailbox Caller Features
When a caller dials an extension and is redirected because the user either is not available
to answer the phone or is busy on another call, the call is most commonly redirected to a
subscriber/group mailbox. At this point the calling party has several mailbox caller features
available. This section will cover message recording options, operator assistance, and the
ability to log in to the mailbox to check messages remotely.
Record Message Options
When a caller reaches a subscriber or group mailbox, they will be presented with a greeting
of some sort informing the caller to leave a message. The caller then hears an audible beep
on the phone handset as a signal that the Unity Express system has begun recording.
When the caller has fi nished leaving a message, they can simply hang up the phone. As long
as the message lasts two seconds or longer, the message is saved and is stored in the user’s
mailbox until they retrieve it.
Alternatively, after the caller leaves a message, they can press the # key on their
handset to utilize the Record Message options available on the Unity Express system.
These options are:
Review recorded message Used if the caller would like to hear the message they just left.
Set the message priority as urgent By default, mailbox messages are set with a normal
priority. A caller can change this setting to urgent here. Depending on the mailbox
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configuration, flagging a message as urgent can trigger the CUE to kick off message
notifications to quickly inform the mailbox owner of the urgent message.
Rerecord the message Used to delete the previously recorded message and rerecord it.
Cancel the message
Used to delete the previously recorded message without rerecording it.
Operator Assistance
The operator assistance feature gives the caller a chance to talk to another person rather
than leave a voice mail. For example, suppose a caller tries to reach extension 4172. The user
for that extension is already on a call, so the incoming call is redirected to voice mail. The
caller begins listening to the greeting, which tells them to leave their name and number. If
the caller presses the 0 key on their phone before the message recording beep goes off, they
are redirected to the extension configured as the operator for extension 4172. Individual
extensions can have separate and different operators configured.
Mailbox Login
The mailbox login feature is for mailbox subscribers who are away from their desk and
do not have access to their primary IP phone. The user can dial the phone number that
is tied to the mailbox they wish to check. When the user begins hearing the voice mail
greeting, they can press the * button. This action triggers the Unity Express system to ask
for the extension and PIN associated with the mailbox they wish to access. Once the user is
authenticated, they can access voice mail messages using the TUI as if they were using their
primary phone at their desk.
Unity Express Advanced
User Functionality
Besides using the standard TUI interface for checking voice mail messages on Unity Express,
two additional methods offer alternatives to the TUI. The fi rst method is called VoiceView
Express. This allows the user to log in to their voice mail from any Cisco IP phone that
supports XML. Using the LCD screen, the user can check their subscriber mailbox as well as
any GDMs they are members of. The other advanced user voice mail functionality is called
integrated messaging. This setup allows a user to check their voice mail and email from a
single IMAP client. Let’s take a closer look at each of these features.
VoiceView Express
The most common way to access and maintain voice mail features is through the TUI. An
alternative method that Unity Express offers is called VoiceView Express. The CUCM
Express and Unity Express software work together to let voice mail users listen to, send,
Unity Express Advanced User Functionality
369
and manage voice messages on a Cisco phone using the LCD display and softkeys.
Figure 8.2 shows an IP Communicator logged into VoiceView Express.
FIGURE 8.2
VoiceView Express
The service uses XML to deliver information to the LCD display of the phone. To access
VoiceView Express on a compatible Cisco phone, the user would press the Services button
on the phone.
Integrated Messaging
For many, email has taken over the title for most-used communications method. If you are
like me, you check your email far more frequently than your voice mail. To help streamline
functions, Unity Express offers integrated messaging so your voice mail messages can be
pulled off the CUE and placed in your Microsoft Outlook inbox or any compatible IMAP
email client.
Integrated messaging merges your email and voice mail systems into a single point
of communications reference. Unity Express uses IMAP to deliver messages to an email
client capable of running IMAP version 4 rev 1. Note that integrated messaging is not
unified messaging. With integrated messaging, you simply set up your email client to have
two IMAP profiles, one for email and a second for voice mail. Figure 8.3 shows how your
IMAP client software is then set up to pull from two different sources.
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FIGURE 8.3
Unity Express Overview and Installation
Integrated messaging
IMAP Service
Enabled
Me
ss
M
ail
es
s
Em
ag
es
VM
MS Exchange
ag
es
CUCM/Unity
Express
User PC running
MS Outlook
Profile 1: Email
Profile 2: CUE Voice mail
Once this connection is set up, the user will receive voice mail messages as attached
.wav fi les.
Understanding Unity Express Auto
Attendant Scripting Methods
The auto attendant (AA) feature of modern PBX systems has replaced a live operator as the
fi rst point of contact for many businesses. The AA is a script that greets callers and either
asks them to enter a known extension for the person they are trying to reach or guides
the caller through various prompts on the script by asking them to press buttons on the
keypad. This script helps the caller navigate to the right person or department that will best
handle their needs. All Unity Express installations come standard with two default scripts.
The audio of the default fi les can be rerecorded to customize them to fit most businesses or
organizations. Some prompts have prerecorded audio on them, but you’ll want to customize
them to personalize the AA experience. Let’s take a closer look at the structure of the two
Unity Express preinstalled scripts.
Preinstalled Scripts
Two scripts have a basic mapping structure laid out to meet many AA needs for small to
medium-size businesses:
Understanding Unity Express Auto Attendant Scripting Methods
Default Auto Attendant Script: aa.aef
Auto Attendant Simple Script: aasimple.aef
371
These preinstalled custom scripts are similar in nature. Figure 8.4 displays the process
flow for both of them.
The primary difference between the two preinstalled scripts is that the simple script
utilizes a PlayExtensionsPrompt, which can be set up to say, for example, “To reach John,
press 1. To reach Tammy, press 2.” This type of script is generally useful only in smaller
FIGURE 8.4
The process flow for preinstalled scripts
Start
welcomePrompt
Holiday?
Yes
holidayPrompt
Options
businessClosedPrompt
Options
No
Open for
Business?
No
Yes
businessOpenPrompt
Options
End
environments. The default script, on the other hand, allows callers to dial the extension if
they know it already or to go through the dial-by-name feature.
There are two ways to change the audio contained within the various prompts. First,
you can record your own audio using PC recording software. The fi les must be in the
following format to function on the Unity Express system:
G.711 u-law
8 kHz
8 bit
Mono
A second and simpler method is to use the administration via telephone (AvT) functions
by dialing into the TUI and recording prompts over the phone that are saved on the CUE in
the proper format. This feature will be discussed in more depth later in this chapter.
Editor Express
The Cisco Unity Editor Express is a slimmed-down version of the Custom Script Editor
that runs on Microsoft Windows. The Editor Express has the advantage of running directly
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on the Unity Express hardware. You simply log in to the web GUI, navigate to System Scripts, and click the New button. Figure 8.5 shows a screenshot of the Editor Express tool.
FIGURE 8.5
Editor Express
As you can see, the tool allows you to choose various options using drop -down menus.
This editor does not have nearly as many configuration options as the full-blown Cisco
script editor, but it may be enough to create the scripts that you need.
Unity Express Editor Application
If you require a fully customized AA script that the Editor Express tool cannot handle, you’re
in luck! Cisco also offers the Unity Express Editor, which is a far more robust application.
This tool allows you to build and customize an AA script that is as complex as you desire.
The Cisco Unity Express Editor is an external Microsoft Windows application that is
available for download on the cisco.com website. Figure 8.6 shows a screenshot of the CUE
Editor application running on Windows XP.
Understanding Unity Express Interactive Voice Response
FIGURE 8.6
373
The AA CUE Editor application
Once you create the script on your PC, you can save it as an .aef fi le and then upload
the fi le to Unity Express. You can then set the script to be used as the primary AA script
on the CUE.
Understanding Unity Express
Interactive Voice Response
The Interactive Voice Response (IVR) functionality allows callers to interact with a
voice menu much like an auto attendant, where the caller is prompted to press numbers
on the telephone handset to work their way through the script. IVR provides additional
capabilities, however, in the type of information a caller can receive. IVR has the ability to
do the following:
Query databases and present this information to the caller over the phone
Send emails/faxes to customers based on caller responses on the IVR application
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Unity Express Overview and Installation
To use the capabilities of IVR on your Unity Express system, you must
purchase the proper IVR licensing from Cisco. Licensing of the Unity
Express system is discussed in the next section.
Understanding CUCM
Express Licensing
As with all Cisco hardware, you need proper licenses to run the CUE software. There are
two separate Cisco license packages for legitimately running your Unity Express System
with full functionality:
1.
2.
Unity Express Mailbox License package (required)
Mailbox license to run with the CUCM Express
Mailbox license to run with the CUCM or CUCMBE
Unity Express IVR License package (optional)
Depending on which Cisco call-processing device you’ll be connecting to, you will
install either the cme license for interoperation with the CUCM Express or the ccm license
for interoperation with the CUCM and CUCMBE. There are also differences in the
number of mailboxes and sessions you can license depending on the type of hardware you
plan to run Unity Express on. Table 8.1 lists all of the currently available Unity Express
hardware devices and the maximum number of mailboxes, sessions, and storage hours
for each.
TA B L E 8 .1
CUE hardware comparison
Hardware
AIM-CUE
NM-CUE
NM-CUE-EC
NME-CUE
Max mailboxes
50
100
250
250
Max sessions
6
8
16
24
Total VM storage
14
100
300
300
Let’s now look at each of these in depth to see which software and licensing options are
available from Cisco.
Understanding CUCM Express Licensing
375
AIM-CUE
AIM- CUE offers the following licensing options:
XX is the number of mailboxes the license
supports. For the AIM module, the licenses can be for 12, 25, or 50 mailboxes. The cme
within the fi lename tells us that this Unity Express software is designed to work with the
CUCM Express system, and the notation Y.Y.Y represents the version of the Unity Express
software.
cue-vm-license_XXmbx_cme_Y.Y.Y.pkg
XX is the number of mailboxes the license
supports. For the AIM module, the licenses can be for 12, 25, or 50 mailboxes. The ccm
within the fi lename tells us that this Unity Express software is designed to work with either
the CUCMBE or CUCM call-processing system, and Y.Y.Y represents the version of the
Unity Express software.
cue-vm-license_XXmbx_ccm_Y.Y.Y.pkg
cue-vm-license_Xport_ivr_Y.Y.Y.pkg X is the number of simultaneous IVR sessions the
license supports. For the AIM module, the licenses can be for 2, 4, 8, 16, or 20 sessions.
NM- CUE
NM- CUE offers the following licensing options:
XX is the number of mailboxes the license
supports. For the AIM module, the licenses can be for 12, 25, 50, or 100 mailboxes. The
cme within the fi lename tells us that this Unity Express software is designed to work with
the CUCM Express system, and Y.Y.Y represents the version of the Unity Express software.
cue-vm-license_XXmbx_cme_Y.Y.Y.pkg
XX is the number of mailboxes the license
supports. For the AIM module, the licenses can be for 12, 25, 50, or 100 mailboxes. The
ccm within the fi lename tells us that this Unity Express software is designed to work with
either the CUCMBE or CUCM call processing system. Y.Y.Y represents the version of the
Unity Express software.
cue-vm-license_XXmbx_ccm_Y.Y.Y.pkg
cue-vm-license_Xport_ivr_Y.Y.Y.pkg X is the number of simultaneous IVR sessions the
license supports. For the AIM module, the licenses can be for 2, 4, 8, 16, or 20 sessions.
NM- CUE-EC
NM- CUE -EC offers the following licensing options:
XX is the number of mailboxes the license
supports. For the AIM module, the licenses can be for 12, 25, 50, 100, or 250 mailboxes.
The cme within the fi lename tells us that this Unity Express software is designed to
work with the CUCM Express system. Y.Y.Y represents the version of the Unity Express
software.
cue-vm-license_XXmbx_cme_Y.Y.Y.pkg
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XX is the number of mailboxes the license
supports. For the AIM module, the licenses can be for 12, 25, 50, 100, or 250 mailboxes.
The ccm within the fi lename tells us that this Unity Express software is designed to work
with either the CUCMBE or CUCM call-processing system. Y.Y.Y represents the version of
the Unity Express software.
cue-vm-license_XXmbx_ccm_Y.Y.Y.pkg
cue-vm-license_Xport_ivr_Y.Y.Y.pkg X is the number of simultaneous IVR sessions the
license supports. For the AIM module, the licenses can be for 2, 4, 8, 16, and 20 sessions.
NME-CUE
NME - CUE offers the following licensing options:
XX is the number of mailboxes the license
supports. For the AIM module, the licenses can be for 12, 25, 50, 100, or 250 mailboxes.
The cme within the fi lename tells us that this Unity Express software is designed to work
with the CUCM Express system, and Y.Y.Y represents the version of the Unity Express
software.
cue-vm-license_XXmbx_cme_Y.Y.Y.pkg
XX is the number of mailboxes the license
supports. For the AIM module, the licenses can be for 12, 25, 50, 100, or 250 mailboxes.
The ccm within the fi lename tells us that this Unity Express software is designed to work
with either the CUCMBE or CUCM call-processing system, and Y.Y.Y represents the
version of the Unity Express software.
cue-vm-license_XXmbx_ccm_Y.Y.Y.pkg
cue-vm-license_Xport_ivr_Y.Y.Y.pkg X is the number of simultaneous IVR
sessions the license supports. For the AIM module, the licenses can be for 2, 4, 8, 16,
or 20 sessions.
Installation and Initial Configuration
of Unity Express on CUCM
Express Routers
When either the AIM or NM Unity hardware module is properly inserted into a supported
Cisco router, a new physical interface appears within the Cisco IOS. The interface will
be labeled either Service-Engine or Integrated-Service-Engine, depending on the Unity
Express hardware you are using. Running show version confi rms that the service engine is
installed:
Router#sh version
Cisco IOS Software, UC500 Software (UC500-ADVIPSERVICESK9-M), Version
12.4(11)XW7, RELEASE SOFTWARE (fc2)
Installation and Initial Configuration of Unity Express on CUCM
377
Technical Support: http://www.cisco.com/techsupport
Copyright (c) 1986-2008 by Cisco Systems, Inc.
Compiled Wed 09-Apr-08 03:07 by prod_rel_team
ROM: System Bootstrap, Version 12.4(11r)XW3, RELEASE SOFTWARE (fc1)
Router uptime is 6 hours, 2 minutes
System returned to ROM by power-on
System image file is “flash:uc500-advipservicesk9-mz.124-11.XW7”
This product contains cryptographic features and is subject to United
States and local country laws governing import, export, transfer and
use. Delivery of Cisco cryptographic products does not imply
third-party authority to import, export, distribute or use encryption.
Importers, exporters, distributors and users are responsible for
compliance with U.S. and local country laws. By using this product you
agree to comply with applicable laws and regulations. If you are unable
to comply with U.S. and local laws, return this product immediately.
A summary of U.S. laws governing Cisco cryptographic products may be found at:
http://www.cisco.com/wwl/export/crypto/tool/stqrg.html
If you require further assistance please contact us by sending email to
export@cisco.com.
Cisco UC520-8U-4FXO-K9 (MPC8358) processor (revision 0x202) with 249856K/12288K
bytes of memory.
Processor board ID FTX130886A2
MPC8358 CPU Rev: Part Number 0x804A, Revision ID 0x20
14 User Licenses
10 FastEthernet interfaces
2 terminal lines
4 Voice FXO interfaces
4 Voice FXS interfaces
1 Voice MoH interface
1 cisco service engine(s)
128K bytes of non-volatile configuration memory.
125440K bytes of ATA CompactFlash (Read/Write)
Configuration register is 0x2102
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Remember that the Unity Express should be thought of as a separate network
device that just happens to be housed within the CUCM Express router. From a logical
standpoint, the Service-Module is just like an Ethernet port that connects the router to the
Unity Express hardware. The router side of the logical interface is the physical IntegratedService-Engine interface. On the other side of this network connection is the Service Module interface, which represents the IP endpoint of the Unity Express hardware.
Figure 8.7 diagrams the logical network between the CUCM Express router and Unity
Express hardware.
F I GU R E 8 .7
Logical network between CUCM and CUE
Logical network between CUCM
Express and Unity Express
Integrated-Service-Engine
1/0
CUCM
Express
ServiceModule
Unity
Express
There are two methods for setting up this IP network so we can communicate with the
Unity Express for operational and administration purposes. The fi rst method is to
use the ip unnumbered command on the Integrated-Service-Engine so we can use an
already configured network on the CUCM Express rather than create a new one just for the
purpose of connecting the Unity Express system. The second method is to create a separate
point-to -point IP network between the Integrated-Service-Engine and the Service-Module.
Let’s look at how to configure each of these methods.
Configuring IP Unnumbered to Use Existing
IP Network for Unity Express Connectivity
The IP unnumbered configuration method sets up the CUE so you don’t have to waste a
separate IP network for the sole purpose of Unity Express connectivity. By using the ip
unnumbered <interface> config-interface command on the Integrated- Service-Engine,
you can configure the Service-Module on an already configured IP subnet on the CUCM
Express router. Figure 8.8 shows what this logical setup looks like:
Installation and Initial Configuration of Unity Express on CUCM
FIGURE 8.8
379
Using ip unnumbered for CUE connectivity
Logical network between CUCM
Express and Unity Express
TelephonyService IP
Source:
192.168.10.1
Integrated-Service-Engine
1/0
IP Unnumbered
Service-Module
192.168.10.2
CUCM
Express
Unity
Express
Notice that the Telephony-Service source IP address and the Service-Module IP address
are on the same subnet. Both of these IP addresses use the /24 network that has been
configured on interface loopback 0. A static route pointing to the Integrated-Service-Engine
is then needed to tell the router how to reach the Unity Express system on the shared
network. Here are the configuration commands to set up the logical network properly using
IP unnumbered:
Router#configure terminal
Router(config)#interface loopback 0
Router(config-if)#ip address 192.168.10.1 255.255.255.0
Router(config)#interface integrated-Service-Engine 0/0
Router(config-if)#ip unnumbered loopback 0
Router(config-if)#service-module ip address 192.168.10.2 255.255.255.0
Router(config-if)#service-module ip default-gateway 192.168.10.1
Router(config-if)#exit
Router(config)#ip route 192.168.10.2 255.255.255.255 integrated-Service-Engine
0/0
Router(config)#
Now our Unity Express system is identified with the IP address 192.168.10.2/24, which
resides on the same network as the source IP address for the CUCM Express system.
This is the preferred method because we now have the CUCM Express and CUE sitting
on the same network. This method is simpler and more easily understood by system
administrators.
We can issue a show interfaces Integrated-Service-Engine 0/0 command to verify
that our interface is up and sharing the loopback 0 IP address:
Router#show interfaces Integrated-Service-Engine 0/0
Integrated-Service-Engine0/0 is up, line protocol is up
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Hardware is PQII_PRO_UEC, address is 0021.a02b.6081 (bia 0021.a02b.6081)
Interface is unnumbered. Using address of loopback 0 (192.168.10.1)
MTU 1500 bytes, BW 100000 Kbit, DLY 100 usec,
reliability 255/255, txload 1/255, rxload 1/255
Encapsulation ARPA, loopback not set
ARP type: ARPA, ARP Timeout 04:00:00
Last input 00:00:03, output 00:00:03, output hang never
Last clearing of “show interface” counters never
Input queue: 0/75/0/0 (size/max/drops/flushes); Total output drops: 0
Queueing strategy: fifo
Output queue: 0/40 (size/max)
5 minute input rate 0 bits/sec, 0 packets/sec
5 minute output rate 0 bits/sec, 0 packets/sec
21773 packets input, 9552774 bytes, 0 no buffer
Received 826 broadcasts, 0 runts, 0 giants, 0 throttles
0 input errors, 0 CRC, 0 frame, 0 overrun, 0 ignored
0 input packets with dribble condition detected
19172 packets output, 4603856 bytes, 0 underruns
0 output errors, 0 collisions, 3 interface resets
0 babbles, 0 late collision, 0 deferred
0 lost carrier, 0 no carrier
0 output buffer failures, 0 output buffers swapped out
Router#
We should also be able to ping our CUE Service-Module IP address of 192.168.10.2:
Router#ping 192.168.10.2
Type escape sequence to abort.
Sending 5, 100-byte ICMP Echos to 192.168.10.2, timeout is 2 seconds:
!!!!!
Success rate is 100 percent (5/5), round-trip min/avg/max = 1/3/4 ms
Router#
Configuring a Separate IP Network
for Unity Express Connectivity
The second method for connecting the CUE to the CUCM is to configure a separate IP
network between the Integrated-Service-Engine and the Service-Module. It is very similar
to creating a router Ethernet interface with an IP address on the Integrated-ServiceEngine physical interface. Then, within config-if mode, you configure an IP address and
Installation and Initial Configuration of Unity Express on CUCM
381
default-gateway within the same IP subnet space for the Unity Express hardware using
the service-module command. Finally, for proper routing, a static route is needed so the
router knows how to reach the Unity Express device on the other end of the network.
Figure 8.9 shows our logical network with a 172.16.1.X/24 subnet created on the logical
network segment.
FIGURE 8.9
Separate IP network for CUE connectivity
Logical network between CUCM
Express and Unity Express
TelephonyService IP
Source:
192.168.10.1
Integrated-Service-Engine
1/0
172.16.1.1
CUCM
Express
Service-Module
172.16.1.2
Unity
Express
Here are the commands for configuring a separate IP network for Unity Express
communication:
Router#configure terminal
Router(config)#interface integrated-Service-Engine 0/0
Router(config-if)#ip address 172.16.1.1 255.255.255.0
Router(config-if)#service-module ip address 172.16.1.2 255.255.255.0
Router(config-if)#service-module ip default-gateway 172.16.1.1
Router(config-if)#exit
Router(config)#ip route 172.16.1.2 255.255.255.255 integrated-Service-Engine 0/0
Router(config)#
We can once again issue a show interfaces Integrated-Service-Engine 0/0
command to verify that our interface is up using its own IP address:
Router#show interfaces Integrated-Service-Engine 0/0
Integrated-Service-Engine0/0 is up, line protocol is up
Hardware is PQII_PRO_UEC, address is 0021.a02b.6081 (bia 0021.a02b.6081)
Internet address is 172.16.1.1/24
MTU 1500 bytes, BW 100000 Kbit, DLY 100 usec,
reliability 255/255, txload 1/255, rxload 1/255
Encapsulation ARPA, loopback not set
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ARP type: ARPA, ARP Timeout 04:00:00
Last input 00:00:03, output 00:00:03, output hang never
Last clearing of “show interface” counters never
Input queue: 0/75/0/0 (size/max/drops/flushes); Total output drops: 0
Queueing strategy: fifo
Output queue: 0/40 (size/max)
5 minute input rate 0 bits/sec, 0 packets/sec
5 minute output rate 0 bits/sec, 0 packets/sec
21773 packets input, 9552774 bytes, 0 no buffer
Received 826 broadcasts, 0 runts, 0 giants, 0 throttles
0 input errors, 0 CRC, 0 frame, 0 overrun, 0 ignored
0 input packets with dribble condition detected
19172 packets output, 4603856 bytes, 0 underruns
0 output errors, 0 collisions, 3 interface resets
0 babbles, 0 late collision, 0 deferred
0 lost carrier, 0 no carrier
0 output buffer failures, 0 output buffers swapped out
Router#
We should also be able to ping our CUE Service-Module IP address of 172.16.1.2:
Router#ping 172.16.1.2
Type escape sequence to abort.
Sending 5, 100-byte ICMP Echos to 172.16.1.2, timeout is 2 seconds:
!!!!!
Success rate is 100 percent (5/5), round-trip min/avg/max = 1/3/4 ms
Router#
We must configure two additional options on the CUCM Express prior to configuring
the Unity Express system. First, we must create dial peers for our voice mail, auto attendant
(AA), and administration via telephone (AvT) pilot numbers. Second, we need to create
extensions for the message waiting indicator (MWI), which is the light on the phone that
turns on when a phone receives a new voice mail message.
Configuring Dial Peers for Unity Express Functions
We need to configure three distinct dial peers on the CUCM Express to take advantage of
voice mail, auto attendant (AA), and administration via telephone (AvT). We already know
about the voice mail and auto attendant features, but administration via telephone is a new
term for us. Administration via telephone is a way for telephone administrators to add/
delete and modify audio prompts used by the Unity Express system. AvT offers a way for
administrators to create these custom prompts without the need for a PC or sound- editing
Installation and Initial Configuration of Unity Express on CUCM
383
software. AvT also allows administrators to broadcast messages to all mailboxes on the
Unity Express system.
When an ephone-DN is configured for voice mail, you need to specify an extension for
the call-forward busy and call-forward noan commands to tell the CUCM Express to
send the calls to the Unity Express system. This unique extension must be tied to a VoIP
dial peer and forwarded over an SIP trunk to ultimately get to the Unity Express hardware
located logically on the other side of the logical network we just configured. The auto
attendant, AVT, and VoIP dial peers require their own unique extensions, but the setup for
each is identical.
Remember that the Unity Express system can use only SIP version 2
signaling between itself and the CUCM Express. No other signaling
methods can be used.
Here is an example of how to configure dial peers for all three Unity Express functions:
Router#configure terminal
Router(config)#dial-peer voice 188 voip
Router(config-dial-peer)#description voice mail
Router(config-dial-peer)#destination-pattern 188
Router(config-dial-peer)#session protocol sipv2
Router(config-dial-peer)#session target ipv4:172.16.1.2
Router(config-dial-peer)#dtmf-relay sip-notify
Router(config-dial-peer)#codec g711ulaw
Router(config-dial-peer)#no vad
Router(config-dial-peer)#exit
Router(config)#dial-peer voice 189 voip
Router(config-dial-peer)#description aa
Router(config-dial-peer)#destination-pattern 189
Router(config-dial-peer)#session protocol sipv2
Router(config-dial-peer)#session target ipv4:172.16.1.2
Router(config-dial-peer)#dtmf-relay sip-notify
Router(config-dial-peer)#codec g711ulaw
Router(config-dial-peer)#no vad
Router(config-dial-peer)#exit
Router(config)#dial-peer voice 190 voip
Router(config-dial-peer)#description avt
Router(config-dial-peer)#destination-pattern 190
Router(config-dial-peer)#session protocol sipv2
Router(config-dial-peer)#session target ipv4:172.16.1.2
Router(config-dial-peer)#dtmf-relay sip-notify
Router(config-dial-peer)#codec g711ulaw
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Router(config-dial-peer)#no vad
Router(config-dial-peer)#end
Router#
We’ve configured extensions 188, 189, and 190 as our destination patterns for each of
these Unity Express features. We also specified that SIPv2 is the signaling protocol that is
to be used, by including the session protocol sipv2 config-dial-peer command.
The dtmf-relay sip-notify command is required because Unity Express uses the
phone handset’s tones to gather information needed to navigate through the voice mail,
AA, and AvT menus. The command tells the CUCM Express to relay those tones over the
SIP trunk to Unity Express.
We’ve statically assigned our codec to G.711. Remember that by default, dial peers are
configured to use the G.729 codec. Unity Express can understand only the G.711 codec, so
we must configure this or the dial peer will not work.
Finally, the no vad command tells the CUCM Express not to use Voice Activity
Detection (VAD) across the SIP trunk. The Unity Express does not work properly with
VAD enabled and must be turned off. VAD is enabled on all dial peers by default.
Configuring MWI Ephone-DNs
The message waiting indicator (MWI) feature informs phone users when one or more
new voice mails are waiting in their voice mailbox. On almost every Cisco IP phone, a red
light on the handset of the phone will light up in the event of a new message waiting in the
phone owner’s mailbox. When a new message arrives, the Unity Express system will call a
configured ephone-DN that is specifically used for MWI. After all the new messages have
been read, the Unity Express system will dial a second number to turn the light off. That
means you set up MWI ephone-DNs similarly to any other standard ephone-DN; you must
include an extension for MWI to function.
You must configure two MWI ephone-DNs: one to turn the MWI lamp on and the other
to turn the lamp off. The numbers for these extensions have a static prefi x and a wildcard
ending where Unity Express appends the particular phone extension it wishes to reach. The
following example shows two ephone-DNs being configured for MWI duty. Ephone-DN
11 turns the MWI light on and ephone-DN 12 turns it off. The prefi x for the MWI DNs
includes the letter a in this example. This is so a phone user cannot accidentally dial the
MWI light extension to turn on/off a phone’s light when there are no new messages in
the user’s mailbox. An administrator can use the letters a, b, c, and d for DTMF priority
and override digits for the extension number. These are used in this situation to prevent
users from inadvertently dialing the MWI extension to turn on and off the message lamp.
Router#configure terminal
Router(config)#ephone-dn 11
Router(config-ephone-dn)#number a11....
Router(config-ephone-dn)#mwi on
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385
Router(config-ephone-dn)#exit
Router(config)#ephone-dn 12
Router(config-ephone-dn)#number a12....
Router(config-ephone-dn)#mwi off
Router(config-ephone-dn)#end
Router#
Now that we can reach our Unity Express system over IP and have our dial peers and
MWI extensions set up, we will go through the process of upgrading the CUE software and
then setting up the system using the GUI Initialization Wizard.
AU: Verifying MWI Functionality without Leaving Your Chair
Marty has just completed configuring MWI on ephone-DN 20 for his boss. To test the
MWI functionality, Marty calls his boss’s phone and leaves a brief test message that
should kick-start the MWI to turn on. When he walks over to his boss’s office, Marty
realizes that his boss is currently in a meeting. Not wanting to disturb his boss,
Marty goes back to his desk to figure out a way to see if the MWI indicators were properly
lit up. After trying out a few show commands on the CLI, Marty received the following
output when issuing a show ephone 20 command:
Router#show ephone 20
ephone-20[1] Mac:0021.A02E.7D9A TCP socket:[5] activeLine:0 REGISTERED in SCCP
ver 12/9
mediaActive:0 offhook:0 ringing:0 ringRate: 0 reset:0 reset_sent:0 paging 0
debug:0 caps:9
IP:192.168.10.11 50271 7965
button 1: dn 1
keepalive 6 max_line 6
number 5000 CH1
IDLE
CH2
IDLE
mwi
Preferred Codec: g711ulaw
From this output, we can see that Marty’s boss’s phone is a 7965, which has one
configured button with the extension 5000 assigned to it using ephone-DN 1. Also note
that this line is a dual-line phone, thus the CH1 and CH2 IDLE status notifications. Finally,
on the far right of the button 1 line, we see the mwi notification. This tells Marty that
his test voice mail indeed triggered the MWI light to turn on. Now Marty can continue
configuring other phones for MWI and be able to test the configurations without ever
having to leave his desk.
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Upgrading Unity Express Software
We’ve officially completed all the steps needed to get Unity Express networked with the
CUCM Express. We can fi nally focus on getting Unity Express software up and running.
To reach our Unity Express system via the command line, we log in to the CUCM Express
and get into privileged- exec mode. We can then issue the following command:
Router#service-module integrated-Service-Engine 0/0 session
Trying 192.168.10.1, 2002 ... Open
UC500-CUE en
Password:
UC500-CUE#
By default, no enable password is set, so you can just hit Enter to get into the CUE
privileged- exec mode.
One of the fi rst steps you will likely want to take is to install the latest version of the
Unity Express software, which can be downloaded from the cisco.com website. All of
the necessary Unity Express fi les can be downloaded in a single zip fi le. You then extract
that zip fi le and put it on an FTP server on your network in order to upload the new
software to the Unity Express system. Figure 8.10 shows a screenshot of all the CUE
software fi les needed to install the latest version of Unity Express on a UC500
series router.
F I G U R E 8 .1 0
CUE software files
Table 8.2 lists the Unity Express software fi les and their functions. The x.x.x in the
fi lename indicates the different version types available.
Installation and Initial Configuration of Unity Express on CUCM
TA B L E 8 . 2
387
Unity Express software files
Software Filename
Function
cue-bootloader.ise.x.x.x
Unity Express boot loader
cue-installer.ise.x.x.x
Unity Express install helper image
cue-vm.ise.x.x.x.pkg
Unity Express software package
cue-vm-en_US-langpack.ise.x.x.x.prt1
English (US) language package
cue-vm-full.ise.x.x.x.prt1
Voice mail application
cue-vm-installer.ise.x.x.x.prt1
Installer software
cue-vm-langpack.ise.x.x.x.pkg
Language package
cue-vm-license_50mbx_cme_ise.x.x.x.pkg
UC500 license package
To download and install a brand-new copy of the Unity Express software, we use the
software install command. This offers several different options, as shown here:
UC500-CUE# software install ?
abort
abort
clean
clean download
status
status
upgrade
upgrade download
UC500-CUE#
The clean option specifies that this will be a brand-new install and not simply an
upgrade. The difference between clean and upgrade is that an upgrade will keep any
previously set configuration options and reapply them to the upgraded software. The
status command lets us view the progress of a clean install or upgrade, and the abort
command allows us to stop a currently running clean install or upgrade.
Once we choose a download option, we must specify the location of our FTP server and
any username/password credentials needed.
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There is also a software download command, which will download the
necessary install files without running through the installation process.
You can then go back at another time to run the install process. In our
example the proper syntax to install software that has been locally
downloaded onto Unity Express storage is software install clean cuevm.ise.3.0.3.pkg. This command is very useful if you want to prepare for
an install during the day and bring down the voice mail system at night to
perform the upgrade.
Because this is a brand-new voice mail system, we’ll use the clean option to ensure that
our setup is configured with only the factory default settings. Here is the output of our
software download and installation process:
UC500-CUE# software install clean url ftp://192.168.1.100/cue-vm.ise.3.0.3.pkg
username cisco
password for cisco :
WARNING:: This command will download the necessary software to
WARNING:: complete a clean install. It is recommended that a backup be done
WARNING:: before installing software.
WARNING:: The system will briefly be brought to an offline state
WARNING:: This will terminate any active call and prevent new calls
WARNING:: from being processed.
Would you like to continue? [n] y
Downloading ftp cue-vm.ise.3.0.3.pkg
Bytes downloaded : 176977
Validating package signature ... done
- Parsing package manifest files... complete.
Validating installed manifests ............complete.
- Checking Package dependencies... complete.
Downloading ftp cue-vm-langpack.ise.3.0.3.pkg
Bytes downloaded : 575607
Validating package signature ... done
Found Add-On Subsystem SID: e2e81cc6-39b5-47e1-9f83-b83c897fc50c Name: CUE
Voice Mail Language
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389
Support Version: 3.0.0.0
Found Add-On Subsystem SID: c28339fa-f7ae-4732-85ab-fa6c68b5de0c Name: CUE
Voice Mail Italian Version: 3.0.0.0
Found Add-On Subsystem SID: 49f09114-e0b0-4721-8b85-04be2064920c
Name: CUE Voice Mail European Spanish Version: 3.0.0.0
Found Add-On Subsystem SID: 27e5e2ab-1622-4c02-8a0a-cfad0d932148
Name: CUE Voice Mail US English Version: 3.0.0.0
Found Add-On Subsystem SID: cf860289-67ac-4886-9295-a41e4c7a8487
Name: CUE Voice Mail European French Version: 3.0.0.0
Found Add-On Subsystem SID: f0a41398-3917-4d49-b5ab-c2b39a80c121
Name: CUE Voice Mail Latin American Spanish Version: 3.0.0.0
Found Add-On Subsystem SID: c4ca62e2-daff-40dc-b94e-bf20094bd700
Name: CUE Voice Mail Mexican Spanish Version: 3.0.0.0
Found Add-On Subsystem SID: 683674a5-e6ef-4c97-8e05-efbba1e6fe47
Name: CUE Voice Mail Canadian French Version: 3.0.0.0
Found Add-On Subsystem SID: fa803d25-9c89-4171-a14c-ec12d6ed6b8c
Name: CUE Voice Mail UK English Version: 3.0.0.0
Found Add-On Subsystem SID: 3f968fd0-6598-48e2-be1c-4af6c2e02e02
Name: CUE Voice Mail German Version: 3.0.0.0
Found Add-On Subsystem SID: 88f73a6c-884d-4838-b162-1b544dd6583f
Name: CUE Voice Mail Danish Version: 3.0.0.0
Found Add-On Subsystem SID: a2ba4f96-3452-40c3-83ad-c442cb6bf42f
Name: CUE Voice Mail Brazilian Portuguese Version: 3.0.0.0
- Parsing package manifest files... complete.
- Checking Package dependencies... complete.
- Checking Manifest dependencies for subsystems in the install candidate list...
complete
Starting payload download
File : cue-vm-full.ise.3.0.3.prt1 Bytes : 95600813
Validating payloads match registered checksums...
- cue-vm-full.ise.3.0.3.prt1
.............................................................verified
Extracting install scripts ...
starting_phase:
install-files.sh /dwnld/.script_work_order
add_file /dwnld/pkgdata/cue-vm-full.ise.3.0.3.prt1 13 /dwnld/scripts/e2e81cc639b5-47e1-9f83-b83c897fc50c usr/bin/products/cue/lang_ui_script.py tgz
Scripts extraction complete.
Remove scripts work order /dwnld/.script_work_order
Running Script Processor for ui_install
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At this point, the new software has been downloaded from the FTP server to storage on
the Unity express system and the installation process begins. The install script that is run
will ask you what language or languages you wish the Unity Express to run in. Depending
on the hardware platform used, a different maximum number of language add- ons is
allowed. This particular hardware allows for up to two languages to be installed. High- end
ISR routers with Unity Express network modules can support up to five languages. The
following shows that we ask that only US English be installed in the clean installation:
Maximum 2 language add-ons allowed for this platform.
Please select language(s) to install from the following list:
Language Installation Menu:
# Selected
SKU
Language Name (version)
---------------------------------------------------------------------1
ITA
CUE Voice Mail Italian (3.0.0.0)
2
ESP
CUE Voice Mail European Spanish (3.0.0.0)
3
ENU
CUE Voice Mail US English (3.0.0.0)
4
FRA
CUE Voice Mail European French (3.0.0.0)
5
ESO
CUE Voice Mail Latin American Spanish (3.0.0.0)
6
ESM
CUE Voice Mail Mexican Spanish (3.0.0.0)
7
FRC
CUE Voice Mail Canadian French (3.0.0.0)
8
ENG
CUE Voice Mail UK English (3.0.0.0)
9
DEU
CUE Voice Mail German (3.0.0.0)
10
DAN
CUE Voice Mail Danish (3.0.0.0)
11
PTB
CUE Voice Mail Brazilian Portuguese (3.0.0.0)
---------------------------------------------------------------------Available commands are:
# - enter the number for the language to select one
r # - remove the language for given #
i # - more information about the language for given #
x - Done with language selection
Enter Command:Enter Command:3
Language Installation Menu:
# Selected
SKU
Language Name (version)
---------------------------------------------------------------------1
ITA
CUE Voice Mail Italian (3.0.0.0)
2
ESP
CUE Voice Mail European Spanish (3.0.0.0)
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391
3
*
ENU
CUE Voice Mail US English (3.0.0.0)
4
FRA
CUE Voice Mail European French (3.0.0.0)
5
ESO
CUE Voice Mail Latin American Spanish (3.0.0.0)
6
ESM
CUE Voice Mail Mexican Spanish (3.0.0.0)
7
FRC
CUE Voice Mail Canadian French (3.0.0.0)
8
ENG
CUE Voice Mail UK English (3.0.0.0)
9
DEU
CUE Voice Mail German (3.0.0.0)
10
DAN
CUE Voice Mail Danish (3.0.0.0)
11
PTB
CUE Voice Mail Brazilian Portuguese (3.0.0.0)
---------------------------------------------------------------------Available commands are:
# - enter the number for the language to select one
r # - remove the language for given #
i # - more information about the language for given #
x - Done with language selection
Enter Command:x
ui_install scripts executed successfully.
UC500-CUE#
Once you see the message that the ui_install scripts executed successfully, the software
install process is complete. Unity Express will then reboot and come back online using the
newly installed software.
We’re not quite fi nished yet, however. Accompanying any new software install, we must
also download and install the software licensing information. When we downloaded and
extracted the zip fi le from cisco.com that contained the Unity Express software, a license
fi le for the version of CUE for the UC500 hardware came with it. Different licenses are tied
to the specific hardware platform Unity Express resides on. You must install the correct
license that does not exceed hardware specifications. To accomplish this task, we again
use the software install clean privileged exec command and specify the FTP location,
fi lename, and any required authentication parameters. Here is an example of this process
in action:
UC500-CUE# software install clean url ftp://192.168.1.100/cue-vm-license_50mbx_
cme_ise.3.0.3.pkg username cisco
password for cisco :
WARNING:: This command will download the necessary software to
WARNING:: complete a clean install. It is recommended that a backup be done
WARNING:: before installing software.
WARNING:: The system will briefly be brought to an offline state
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WARNING:: This will terminate any active call and prevent new calls
WARNING:: from being processed.
Would you like to continue? [n] y
Downloading ftp cue-vm-license_50mbx_cme_ise.3.0.3.pkg
Bytes downloaded : 6373
Validating package signature ... done
compatibility mode
- Parsing package manifest files... complete.
Validating installed manifests ............complete.
- Checking Package dependencies... complete.
- Checking Manifest dependencies for subsystems in the install candidate
list...complete
We can now run a show software license privileged exec command to verify that our
license is properly installed:
UC500-CUE# show software license
Installed license files:
- voice mail_lic.sig : 50 MAILBOX LICENSE
Core:
- Application mode: CCME
- Total usable system ports: 6
Voice Mail/Auto Attendant:
- Max system mailbox capacity time: 840
- Default # of general delivery mailboxes: 15
- Default # of personal mailboxes: 50
- Max # of configurable mailboxes: 65
Interactive Voice Response:
- Max # of IVR ports: Not Available
Languages:
- Max installed languages: 2
- Max enabled languages: 1
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393
This command shows us our license maximums for the Unity Express system. We fi nd
the following licensing information within the output of this command:
Maximum number of configurable mailboxes (65)
50 personal mailboxes
15 general delivery mailboxes
Total number of simultaneous ports (lines) into Unity Express (6)
Maximum number of mailbox minutes (840)
Maximum number of IVR ports (“Not Available” because we need to purchase and
install a separate IVR license to use this functionality)
Maximum number of enabled (1) and installed (2) languages
Our Unity Express system is now installed and licensed. We can issue a show run
privileged exec command to view the default configuration on the newly installed software.
Remember that because this is a clean install, these settings are prebuilt configurations by
the CUE software. We will run through the Installation Wizard to modify many of these
configuration parameters:
UC500-CUE# show run
Generating configuration:
clock timezone America/Los_Angeles
hostname UC500-CUE
ip domain-name localdomain
system language preferred “en_US”
ntp server 10.1.10.2 prefer
software download server url “ftp://127.0.0.1/ftp” credentials hidden
“6u/dKTN/hsEuSoEfw40XlF2eFHnZfyUTSd8ZZNgd+Y9J3x
lk2B35j0nfGWTYHfmPSd8ZZNgd+Y9J3x
lk2B35j0nfGWTYHfmPSd8ZZNgd+Y9J3xlk2B35j0nfGWTYHfgT”
groupname Administrators create
groupname Broadcasters create
username cisco create
groupname Administrators member cisco
groupname Administrators privilege ManagePrompts
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groupname
groupname
groupname
groupname
groupname
groupname
groupname
groupname
groupname
Administrators privilege broadcast
Administrators privilege local-broadcast
Administrators privilege ManagePublicList
Administrators privilege ViewPrivateList
Administrators privilege vm-imap
Administrators privilege ViewHistoricalReports
Administrators privilege ViewRealTimeReports
Administrators privilege superuser
Broadcasters privilege broadcast
restriction
restriction
restriction
restriction
Unity Express Overview and Installation
msg-notification
msg-notification
msg-notification
msg-notification
create
min-digits 1
max-digits 30
dial-string preference 1 pattern * allowed
backup server url “ftp://127.0.0.1/ftp” credentials hidden
“EWlTygcMhYmj5tXhE/VNXHCkplVV4KjescbDaLa4fl4WLSPFvv1rWUnfGWTYHfmPSd8ZZNgdY9J3xlk2B35j0nfGWTYHfmPSd8ZZNgd+Y9J3xlk2B35j0nfGWTYHfmP”
calendar biz-schedule systemschedule
open day 1 from 00:00 to 24:00
open day 2 from 00:00 to 24:00
open day 3 from 00:00 to 24:00
open day 4 from 00:00 to 24:00
open day 5 from 00:00 to 24:00
open day 6 from 00:00 to 24:00
open day 7 from 00:00 to 24:00
end schedule
ccn application autoattendant aa
description “autoattendant”
enabled
maxsessions 6
script “aa.aef”
parameter “busClosedPrompt” “AABusinessClosed.wav”
parameter “holidayPrompt” “AAHolidayPrompt.wav”
parameter “welcomePrompt” “AAWelcome.wav”
parameter “disconnectAfterMenu” “false”
parameter “dialByFirstName” “false”
parameter “allowExternalTransfers” “false”
parameter “MaxRetry” “3”
parameter “dialByExtnAnytime” “false”
Installation and Initial Configuration of Unity Express on CUCM
parameter “busOpenPrompt” “AABusinessOpen.wav”
parameter “businessSchedule” “systemschedule”
parameter “dialByExtnAnytimeInputLength” “4”
parameter “operExtn” “0”
end application
ccn application ciscomwiapplication aa
description “ciscomwiapplication”
enabled
maxsessions 6
script “setmwi.aef”
parameter “CallControlGroupID” “0”
parameter “strMWI_OFF_DN” “8001”
parameter “strMWI_ON_DN” “8000”
end application
ccn application msgnotification aa
description “msgnotification”
enabled
maxsessions 6
script “msgnotify.aef”
parameter “logoutUri” “http://localhost/voice mail/vxmlscripts/mbxLogout.jsp”
parameter “DelayBeforeSendDTMF” “1”
end application
ccn application promptmgmt aa
description “promptmgmt”
enabled
maxsessions 1
script “promptmgmt.aef”
end application
ccn application voice mail aa
description “voice mail”
enabled
maxsessions 6
script “voicebrowser.aef”
parameter “logoutUri” “http://localhost/voice mail/vxmlscripts/mbxLogout.jsp”
parameter “uri” “http://localhost/voice mail/vxmlscripts/login.vxml”
end application
ccn engine
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end engine
ccn reporting historical
database local
description “UC500-CUE”
end reporting
ccn subsystem sip
end subsystem
ccn trigger http urlname msgnotifytrg
application “msgnotification”
enabled
maxsessions 2
end trigger
ccn trigger http urlname mwiapp
application “ciscomwiapplication”
enabled
maxsessions 1
end trigger
security password lockout policy temp-lock
security pin lockout policy temp-lock
service phone-authentication
end phone-authentication
service voiceview
enable
end voiceview
voice mail default mailboxsize 775
voice mail broadcast recording time 300
voice mail notification restriction msg-notification
end
Some of these defaults will remain the same, and others will change to fit our
environment. We could go ahead and configure the Unity Express system entirely through
the command line. But the CCNA Voice exam tends to focus on the GUI for software
initialization and configuration, so that’s what this book will follow. Next, you’ll learn to
use the web GUI to run the Unity Express Initialization Wizard.
Installation and Initial Configuration of Unity Express on CUCM
397
Unity Express Setup Using the Initialization Wizard
Using our example setup of configuring a separate IP network for CUE connectivity, we
can open a web browser to http://172.16.1.2 and go through the GUI Unity Express
Initialization Wizard.
In order to use either the CUCM Express or the Unity Express web GUI
features, you must enable HTTP server functionality on the CUCM Express
router. This configuration information can be found in Chapter 5 under the
heading “Enabling the GUI Interface.”
The fi rst screen is the Authentication page, which asks you to log in with a User Name
and Password, as shown in Figure 8.11.
F I G U R E 8 .11
The Unity Express authentication page
Notice that the page says “System is not initialized. Only Administrator logins are
allowed” in bold red text. When we performed our upgrade, a general post-installation
script was automatically run, which created a generic Administrator User Name and
Password. The User Name is cisco and the Password is cisco. We enter the authentication
information into the fields and click the Login button to proceed. The next screen presents
four different options, as shown in Figure 8.12.
F I G U R E 8 .1 2
Initialization Wizard options
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The fi rst option is to view the current settings configured on the Unity Express system.
These will be the factory default settings. The second option is to run the Initialization
Wizard. The other two options let us skip the wizard or log off the system. We want to
run through the Initialization Wizard, so we will click that link and begin the
Initialization Wizard.
The Unity Express system wants to check the CUCM Express configuration for any
information already configured on it that should be pulled into Unity Express. Items such
as phone usernames/passwords and MWI configurations will automatically be read off the
CUCM Express configuration and pulled into Unity Express. Figure 8.13 shows the CUCM
Express login credentials page.
F I G U R E 8 .1 3
CUCM Express login credentials
The Unity Express wants us to enter the Hostname of the CUCM Express system to
which we will be connecting Unity Express. The IP address of the telephony-service source
address should be already populated here. The User Name required is one of the local
usernames created on the CUCM Express for managing the router. Enter the information
and the password and click the Next button.
Using the supplied credentials, Unity Express automatically reads the CUCM Express
configuration to see if any phone users have been previously configured on the system.
If there are, the usernames, extensions, and privileges are imported into Unity Express.
Figure 8.14 shows that our CUCM Express did not have any users configured at the time
the Initialization Wizard was run.
Installation and Initial Configuration of Unity Express on CUCM
F I G U R E 8 .1 4
399
Importing CUCM Express phone users
Click the Next button to continue the Initialization Wizard.
The next screen displays different default parameters for the Unity Express mailboxes.
Figure 8.15 shows the page with the default settings.
F I G U R E 8 .1 5
Unity Express mailbox default settings
In this page, we can change the default mailbox language spoken in the automated
voice mail prompts. We can also change the behavior for new mailbox password/PIN
settings as well as mailbox size, message size, and message expiration time limits. You can
either leave the default settings as is or change them to suit your preferences. Click the Next
button to continue.
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The Call Handling page lists the extension numbers for voice mail, auto attendant, and
MWI functions within the Unity Express system. These extensions should match the dialpeer numbers we created on the CUCM Express earlier in the configuration process. Note
that the MWI extensions were automatically pulled in by the CUCM Express. Figure 8.16
shows the screen with the proper extension numbers entered to match our CUCM Express
dial-peer configurations.
F I G U R E 8 .1 6
CUCM Express call handling
On this page, the only mandatory number to fi ll in is the voice mail pilot extension. All
other fields can be left blank if you choose not to use the features. Click the Next button to
continue the Initialization Wizard.
Finally, we are presented with a Commit page, which lists the settings that we just
configured. If you want to make any changes to the configuration at this point, you can
click the Back button to move backward through the wizard. If you are satisfied with
the configuration settings, you can click the Finally, Save To Startup Configuration check
box and click the Finish button shown in Figure 8.17.
F I G U R E 8 .17
Unity Express Initialization Wizard commit page
Restoring Unity Express to Factory Default Settings
401
The wizard will then save the configurations on both the Unity Express and CUCM
Express systems. Why does the process include saving the CUCM Express configuration?
The Unity Express system added a voice mail 188 command into the telephony-service
section of its configuration using the supplied username and password you gave it during
the wizard process. This command tells CUCM Express that to reach voice mail features, it
should dial extension 188.
Restoring Unity Express to Factory
Default Settings
Whether you’re in a lab environment or in production, there will come a time when you
will need to completely blow away your voice mail system and start from scratch. Cisco
provides an easy way to use the command line to fi rst suspend voice mail services and then
set the configuration back to factory default settings. Keep in mind that once the process
starts, there is no way to retrieve any voice mail, AA, or other configuration settings, so be
very careful that you have backups of anything you might need! The following steps show
how to restore Unity Express to factory default settings.
Step 1: Suspend Unity Express Services
The fi rst step is to suspend voice mail service processes. To accomplish this, you must
session in to the CUE device and get into privileged exec mode. You then can run the
offline command. Unity Express will ask if you are sure you want to take voice mail
offl ine. To proceed, type y and then press the Enter key, as shown in the following example:
Router#service-module integrated-Service-Engine 0/0 session
Trying 10.1.10.2, 2002 ... Open
UC500-CUE>
UC500-CUE> en
Password:
UC500-CUE# offline
!!!WARNING!!!: If you are going offline to do a backup, it is recommended
that you save the current running configuration using the ‘write’ command,
prior to going to the offline state.
Putting the system offline will terminate all end user sessions.
Are you sure you want to go offline[n]? : y
UC500-CUE(offline)#
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Step 2: Restore Factory Defaults on Unity Express
Now that Unity Express is in an offl ine state, it will no longer process voice mail or auto
attendant functions. You are now safe to reset the software to bring it back to a clean state,
that is, before you made any configuration changes. To do that, run the restore factory
default command, as shown here. Unity Express will again ask you if you are sure you
want to perform this task; press y and the Enter key to continue.
UC500-CUE(offline)# restore factory default
!!!WARNING!!!: This operation will cause all configuration and data
on the system to be erased. This operation is not reversible.
Do you wish to continue[n]? : y
Restoring the system. Please wait .....done
System will be restored to factory default when it reloads.
Press any key to reload:
System reloading ....
<output removed>
The Cisco Unity Express system will erase everything saved on the system and reboot
into factory default mode. It will then come up in an online state that is brand new and
ready to be configured from scratch.
Summary
This chapter covered the typical and advanced voice mail features of Unity Express. We
also looked at the additional CUE functionality of the auto attendant and Interactive Voice
Response tools and then covered the proper licensing structure for your Unity Express.
Finally, we got our hands on a Unity Express, and you learned how to upgrade, install, and
initialize Unity Express to the point where you will be ready to begin configuring features
on the system.
Now you have seen the types of voice mail features that Unity Express can provide.
Voice mail technology has evolved to be much more than a method for people to leave
messages while you are away from your desk. The capability to be notified of messages
virtually anywhere in the world and the continuing integration with email make this
technology a vital business communication tool that will continue to thrive for years
to come.
Exam Essentials
403
Exam Essentials
Know the difference between subscriber and group mailboxes. Subscriber mailboxes are
for individual users. Group mailboxes are shared by a group of users.
Know the different types of message notifications. Users can be informed of
messages waiting within mailboxes through the flashing envelope icon on the LCD
display or by the message waiting indicator light on the phone handset when using
Cisco IP phones.
Understand the purpose and different types of distribution lists. Distribution lists are
used to send a single voice mail message to multiple recipients at the same time. Public
distribution lists are globally set up by the voice administrator and can be used by anyone.
Private distribution lists are created by individual users and can be used only by that
individual.
Know the different types of mailbox caller features. When a person leaves a
message on a Unity Express voice mail system, they not only can leave a message
but can also perform other functions such as reviewing, rerecording, canceling, and
setting the message priority. They can also choose to use the operator assistance
feature or even log in to their own mailbox if they have the proper security credentials
in the form of a PIN.
Know the different types of Unity Express advanced user functionality. Besides using
the TUI for checking messages, administrators can configure VoiceView Express and/or
integrated messaging to increase the flexibility for message retrieval.
Understand the components that make up the CUE auto attendant. The auto attendant
provides a flexible, automated, voice answering service for businesses. There are multiple
ways of creating AA scripts, including using preinstalled scripts, the Editor Express, and
the Custom Script Editor application.
Understand Interactive Voice Response. IVR is an advanced automated voice
system that lets users navigate through a menu system to receive up-to -date information
that is pulled off a database and transformed into voice scripts. The Unity Express IVR
also can be configured to send emails or faxes automatically based on caller responses to
the IVR.
Understand CUE licensing. There are two types of Unity Express mailbox
licensing packages. Only one license is required based on the type of CUCM you
are running. You purchase these licenses based on the number of mailboxes,
ranging from 12 to 250. A separate IVR license is required if you plan to run the
IVR application.
Know how to configure the CUE to communicate with CUCM Express. The CUE
is a completely separate device from a network and software point of view. You must
be able to configure the CUE to properly communicate with CUCM Express over a
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logical IP connection. You then need to set up dial peers to forward voice calls to
the CUE.
Know how to configure MWI ephone -DNs. The message waiting indicator light requires
specially configured ephone-DNs to turn on and off the MWI light and LCD indicator on
Cisco IP phones.
Know how to upgrade Unity Express software. Understand the different Unity Express
software fi les and how to perform a software upgrade from scratch.
Know how to set up Unity Express using the Configuration Wizard. Unity Express
provides a web GUI wizard for the initial setup of the software. Understand what each step
in the wizard is for.
Know how to restore Unity Express to factory defaults. Understand the process required
to suspend and reload the software to bring it back to its factory default settings.
Written Lab 8.1
Write the answers to the following questions:
1.
What is the name of the mailbox that is shared by multiple users?
2.
What term refers to a message notification that will continue to trigger alerts to
different destinations until the notification is acknowledged?
3.
Name the type of distribution list that is significant only to the user who created it.
4.
What is a Unity Express voice mail retrieval method where users can use the Cisco IP
phone LCD display and softkeys to check messages?
5.
What is the name for the convergence of email and voice mail that is offered by Unity
Express?
6.
Name the AA script- editing software that runs on Microsoft Windows platforms.
7.
What is the command used to set the Unity Express integrated service engine to share
or borrow an IP address from the already configured loopback 0 interface?
8.
Name the three dial peers commonly configured to point to a Unity Express system.
9.
What command is used when setting up the SIP trunk between CUCM Express and
Unity Express to tell the CUCM Express to not use voice audio detection?
10. What Unity Express software install command is used to upgrade the CUE software
while maintaining and reusing the previous configuration on the system?
(The answers to Written Lab 8.1 can be found following the answers to the review
questions for this chapter.)
Hands - on Labs
405
Hands-on Labs
To complete the labs in this section, you need a CUCM Express router with a Cisco
Unity Express module. Each lab in this section builds on the last and will follow the
CUCM Express to Unity Express integration according to the information found in
Table 8.3.
TA B L E 8 . 3
Lab parameters
Description
Parameter
Loopback 0 IP address
172.16.10.1/24
Service-Engine 0/1 IP address
IP unnumbered
Service-Module IP address
172.16.10.2/24
Voice mail pilot
297
AA extension
298
AvT extension
299
MWI light on
A11
MWI light off
A12
Here is a list of the labs in this chapter:
Lab 8.1: Configuring IP Network Connectivity Using the ip unnumbered Command
Lab 8.2: Configuring Unity Express Dial Peers
Lab 8.3: Configuring MWI Ephone-DNs
Hands-on Lab 8.1: Configuring IP Network Connectivity
Using the ip unnumbered Command
In this lab, we’re going to configure IP connectivity between the CUCM Express and Unity
Express. We will be using the ip unnumbered technique for the virtual network connecting
the two logically separated devices. Use Table 8.3 for the proper IP address and subnet
mask information.
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1.
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Log in to your CUCM Express router and go into configuration mode by typing
enable and then configuration terminal.
2.
Configure an IP address on the loopback interface by typing interface loopback 0
and then ip address 172.16.10.1 255.255.255.0.
3.
Configure the Service-Engine to share an IP address with loopback 0 by typing
interface Service-Engine 0/1 and then ip unnumbered loopback 0.
4.
Configure networking on the service module by first adding the IP address and
then adding a default gateway that points to the CUCM Express Service-Engine. To
accomplish this, type service-module ip address 172.16.10.2 255.255.255.0 and then
service-module ip default-gateway 172.16.10.1.
5.
Exit config-interface mode by typing exit.
6.
Add a static route on the CUCM Express router that directs IP traffic destined to Unity
Express. Type ip route 172.16.10.2 255.255.255.255 Service-Engine 0/1.
7.
Exit configuration mode by typing end.
Hands-on Lab 8.2: Configuring Unity
Express Dial Peers
In the second lab, we will configure dial peers for the voice mail, AA, and AvT extensions
that point to the IP address we assigned as the Unity Express. These SIP trunk dial peers
will use the specific parameters required by Unity Express to function properly.
1.
Log in to your CUCM Express router and go into configuration mode by typing
enable and then configuration terminal.
2.
We want to create a dial peer for our voice mail pilot extension according to the
information in Table 8.3. To add a new dial peer, type dial-peer voice 297 voip.
3.
Add a description for our dial peer by typing description VM Pilot.
4.
Add our extension as the destination trigger by typing destination-pattern 297.
5.
Set the trunk protocol to SIP version 2 by typing session protocol sipv2.
6.
Point the dial-peer destination to the IP address of our Unity Express system by typing
session target ipv4:172.16.10.2.
7.
Set the proper DTMF signaling by typing dtmf-relay sip-notify.
8.
Set the codec to G.711 by typing codec g711ulaw.
9.
Ensure that VAD is disabled by typing no vad.
10. Repeat steps 2 through 9 for the AA and AvT extensions by replacing the descriptions
and extensions to match those found in Table 8.3.
11. Exit config- dial-peer mode by typing end.
Hands - on Labs
407
Hands-on Lab 8.3: Configuring MWI Ephone-DNs
In the third and fi nal lab, we will configure the message waiting indicator on and off
ephone-DNs. These will then be triggered by Unity Express to give a visual alert to a user
that they have a new voice message in their mailbox. Use the extensions given in Table 8.3.
Also assume that Cisco IP phones configured on this system are three digits in length.
1.
Log in to your CUCM Express router and go into configuration mode by typing
enable and then configuration terminal.
2.
Create a new ephone-DN for our MWI on signal. Use 111 as the label by typing
ephone-dn 111.
3.
Add the extension trigger and three-digit wildcard by typing number a11....
4.
Set this ephone-DN to be designated as our MWI on signal by typing mwi on.
5.
Create a second ephone-DN for our MWI off signal. Use 112 as the label by typing
ephone-dn 112.
6.
Add the extension trigger and three-digit wildcard by typing number a12....
7.
Set this ephone-DN to be designated as our MWI off signal by typing mwi off.
8.
Exit config- ephone-DN mode by typing end.
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Unity Express Overview and Installation
Review Questions
1.
What name is used to describe a personal mailbox on Unity Express?
A. General delivery
2.
B.
VoiceView Express
C.
Caller ID
D.
Subscriber
E.
Integrated messaging
Where on the CUE GUI can you go to synchronize configurations that are shared between
the CUCM Express and CUE Express?
A. Administration Synchronize
3.
B.
Administration Write Memory
C.
Synchronize CUCM Express
D.
Synchronize Write Memory
When a CUCM Express user is imported into Unity Express, which default PIN is not an option?
A. Randomly generated
4.
B.
Left blank
C.
Automatically set the same as the user extension
D.
Manually entered by the administrator
The red MWI light can be used on which Cisco phone button(s)?
A. Button 1
5.
B.
Button 1 or 2
C.
Any button with the feature ring configured
D.
Any phone button
When running the Unity Express Initialization Wizard, what information will not be
imported from the CUCM Express configuration?
A. MWI extensions
6.
B.
MoH audio files
C.
Usernames
D.
User passwords
E.
Ephone-DN
Which of the following devices cannot be configured on Unity Express for message notification?
A. PSTN telephone
B.
Cellular telephone
C.
Pager
D.
RFID
Review Questions
7.
409
Which of the following is not an audio file requirement necessary to make your own audio
prompts for the auto attendant?
A. 8-bit
8.
B.
8 kHz
C.
G.711 a-law codec
D.
Mono
Auto attendant scripts are saved as what type of format?
A. .wav
9.
B.
.au
C.
G.711 u-law
D.
.aef
What Cisco feature allows callers to interact with a voice menu system to receive customized information or to trigger tasks such as sending faxes or email messages?
A. AA
B.
IVR
C.
AvT
D.
TUI
10. When the Unity Express Initialization Wizard is complete and changes are being saved,
why does the wizard also perform a configuration save on the CUCM Express?
A. To ensure that the clocks are synchronized
B.
To update IP network changes that were made within the wizard
C.
To automatically update dial-peer rules on the CUCM Express
D.
To modify the MAC addresses of the ephones to match those found on Unity Express
11. Which of the following files is a CUE mailbox license for running up to 25 mailboxes on a
CUCM Express system?
A. cue-vm-license_25mbx_ccm_7.0.1.pkg
B.
cue-vm-license_25mbx_cme_7.0.1.pkg
C.
cue-vm-license_25mbx_ivr_7.0.1.pkg
D.
cue-vm-license_12mbx_ccm_7.0.1.pkg
12. What is the largest number of IVR licenses that can be purchased?
A. 10
B.
50
C.
20
D.
250
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Unity Express Overview and Installation
13. What is considered to be the logical interface of the Cisco Unity Express from an IP networking standpoint?
A. Loopback 0
B.
Service -Module
C.
Integrated- Service -Engine 0/0
D.
FastEthernet 1/0/1
14. Which two statements about the Unity Express Initialization Wizard are not correct?
A. Only administrators are able to log in and run the wizard for the first time.
B.
The wizard saves the configuration on Unity Express after each configuration section.
C.
Configuring an auto attendant extension is mandatory.
D.
Configuring a voice mail extension is mandatory.
15. A subscriber mailbox password serves what purpose?
A. To allow a user to log in to their mailbox using the TUI
B.
To allow a user to log in to their mailbox using the web GUI
C.
To allow a user to log in to the AvT
D.
A one-time password used to set the PIN
16. Which of the following configurations is not required to configure IP networking properly
for the Cisco Unity Express?
A. IP address on the Integrated- Service-Engine
B.
IP address on the Service-Module
C.
Integrated- Service -Engine
D.
Dial peer for the voice mail pilot number
E.
Static route pointing the Service-Module IP address to the Integrated- Service -Engine
interface
17. When configuring the SIP trunk between the CUCM Express and Unity Express systems,
what does the dtmf-relay sip-notify config- dial-peer command do?
A. Gathers and sends DTMF tones from Unity Express to the CUCM Express
B.
Gathers and sends DTMF tones from the CUCM Express to Unity Express
C.
Gathers and sends pulse tones from Unity Express to the CUCM Express
D.
Gathers and sends pulse tones from the CUCM Express to Unity Express
18. What type of dial peers are used to configure Unity Express pilot numbers for voice mail,
AA, and AvT?
A. PSTN dial peers
B.
POTS dial peers
C.
IPv4 dial peers
D.
VoIP dial peers
Review Questions
411
19. When performing a Unity Express software upgrade, what type of server can you use to
transfer the software files to Unity Express storage?
A. TFTP server
B.
CIFS server
C.
FTP server
D.
SMB server
20. The Call Handling page of the Unity Express Initialization Wizard lists all of the following
extensions except:
A. Voice mail number
B.
AvT number
C.
Voice mail operation extension
D.
MWI extensions
E.
Directory extension
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Answers to Review Questions
1.
D. A subscriber mailbox is a personal mailbox assigned to a single user.
2.
A. To synchronize shared configurations between the CUCM Express and CUE, you
navigate to Administration Synchronize within the Unity Express web GUI.
3.
C. The CUCM Express cannot automatically set the PIN to be the same as the user extension.
4.
A. The red MWI light can be used only for the primary extension, which is configured on
button 1.
5.
B. Music on Hold (MoH) is solely handled by the CUCM Express, so there is no need for
Unity Express to pull audio fi les over.
6.
D. All of the devices except RFID can be configured for message notification.
7.
C. The codec must be G.711 u-law.
8.
D. All scripts used by Unity Express are saved in the .aef format.
9.
B. The Interactive Voice Response (IVR) feature allows callers to interact by sending and
receiving tailored information.
10. C. Changes such as those to voice mail dial peers need to be made on the CUCM Express
for Unity Express to function properly. These changes are made to CUCM Express within
the Initialization Wizard and then are automatically saved.
11. B. Option B is the correct CUE license for the CME for up to 25 mailbox licenses.
12. C. IVR licenses can be purchased in bundles of 2, 4, 8, 16, and 20.
13. B. The Service-Module is the logical representation of the Unity Express interface.
14. B, C. The wizard saves the configuration only at the end when the administrator commits
to the changes. Also, the AA extension is not a mandatory field.
15. B. The subscriber password allows the user to log in to the web GUI to check messages or
make changes to mailbox settings. The TUI uses a PIN for login purposes.
16. D. The dial-peer statements are required for proper functionality of the Unity Express system, but they are not necessary for IP networking.
17. B. The command is used to notify the CUCM Express that DTMF tones should be sent to
Unity Express over the SIP trunk.
18. D. VoIP dial peers are configured for Unity Express pilot numbers and are directed over
the SIP trunk.
19. C. An FTP server can be used to transfer the new software over to Unity Express storage
for installation.
20. E. There is never a defi ned extension for either the local or corporate directory.
Answers to Written Lab 8.1
Answers to Written Lab 8.1
1.
General delivery mailbox (GDM)
2.
Cascading notification
3.
Private distribution list
4.
VoiceView Express
5.
Integrated messaging
6.
Custom Script Editor
7.
ip unnumbered loopback 0
8.
Voice mail, AvT, and AA pilot numbers
9.
no vad
10. software install upgrade
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Chapter
9
Unity Express
Configuration
THE CCNA VOICE EXAM TOPICS COVERED
IN THIS CHAPTER INCLUDE THE
FOLLOWING:
Implement voice mail features using Cisco
Unity Express.
Describe the features available in Cisco Unity Express.
Configure basic voice mail features using Cisco Unity
Express.
Configure Auto Attendant services using Cisco Unity
Express.
Communications Manager Express.
Explain basic troubleshooting methods for Cisco
Unity Express.
A large part of any Unified Communications (UC)
administrator’s role is the implementation phase. This phase
requires that the administrator have a solid understanding of
how to configure a device to do what they intend it to do. Essentially the design creates a
set of instructions, and the UC engineer must then carry out those instructions in the form
of a configuration.
Chapter 9 is basically a set of instructions you can use to build various features that
your business requires. Along the way you will learn what Unity Express is capable of by
configuring it with the web GUI. To administrate CUE, you will also need to learn how to
run fi le backups and restores as well as run and view trace fi les for troubleshooting.
Configuring Unity Express System
Settings and Voice Mail Defaults
This section will guide you through the basics of setting up the Cisco Unity Express system
settings and default voice mail capabilities. The system administrator will determine
the settings required based on the geographic location of the system, Unity Express
capabilities, and the needs of users and groups who utilize the mailboxes contained within
the system. I’ll fi rst cover the general system settings for the CUE. After that, you’ll learn
how to set the systemwide default settings that will be used when creating new mailboxes
for both subscribers and groups.
Configuring System Settings
System settings on the Unity Express are parameters that will likely be set only once
throughout the life of the system. Functions such as time zones, DNS server settings,
language settings, and call-in numbers fall within this category. Let’s step through some of
the most commonly used system settings. Keep in mind that you must be logged in with an
administrator account to perform any of the systemwide configurations.
NTP and Time Zone
Just as in the CUCM Express, configuring a Network Time Protocol (NTP) server is
vital for proper operation of Unity Express. Your mailbox subscribers need to have an
Configuring Unity Express System Settings and Voice Mail Defaults
417
accurate clock so they can determine the time a message was left for them. To point
Unity Express at an NTP server, you fi rst must log in with a user account that has
administrative rights. Once you’re logged in, go to System Network Time & Time Zone
Settings. Figure 9.1 shows the configuration options available when configuring NTP and
time zones.
F I G U R E 9 .1
Network time and time zone settings
You can configure multiple NTP servers on the system by clicking the Add button and
entering the IP address of the NTP server you wish to synchronize time with.
After making any change on the Unity Express GUI, you must click the
Apply button to activate the change and save it on the system.
Domain Name Settings
If you need your Unity Express to resolve hostnames to IP addresses within the system,
then you will need to configure Domain Name System (DNS) settings. Features such as
Integrated Messaging will often require the use of a DNS server in order to forward mail
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to an SMTP gateway that is configured with a domain name instead of an IP address. To
configure these settings, navigate to System Domain Name Settings. Figure 9.2 shows the
configuration options for setting the Unity Express local hostname and domain as well as
the option to add a DNS server for domain name lookups.
FIGURE 9.2
Domain name settings
You will notice that the Hostname and Domain fields have already been populated. This
is because the two settings are required and were given default names when the CUE was
initialized.
If you wish to configure a DNS server, click the Add button and enter the IP address of
the DNS server you want to resolve hostnames against.
Default Language
An administrator will want to set the System Default Language setting on the CUE to the
primary language their voice mail users speak. This setting can also be changed at the user
and group levels, but this is where the default is configured for all new users and groups set
up on the system. To set the default language, go to System Language Settings. As shown
in Figure 9.3, you select the default language from the drop -down list.
Configuring Unity Express System Settings and Voice Mail Defaults
FIGURE 9.3
419
Setting the default language
Depending on the capabilities of your Unity Express system and the language packages
installed, you may have anywhere from one to five languages to choose from. The
maximum number of language packages on a single CUE is five.
Call-in Numbers
Callers use three primary call-in numbers for Unity Express to reach CUE applications:
Voice mail
The access number used to access the Voice Mail application
Auto attendant
Promptmgmt
application
The access number used to access the Auto Attendant application
The access number used to access the Administration via Telephone (AvT)
Figure 9.4 shows the currently configured call-in numbers for each of
these applications.
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FIGURE 9.4
Unity Express Configuration
Call-in numbers
These numbers were previously configured in the Unity Express Initialization Wizard
(Chapter 8). You can add additional numbers by clicking the Add button. You can also
modify settings within the already configured numbers by clicking the number to pop up a
screen where you can enter and save the changes. Figure 9.5 displays the settings that can
be modified on call-in number profi le 188.
FIGURE 9.5
Call-in number profile
Configuring Unity Express System Settings and Voice Mail Defaults
421
The Application drop -down menu lists the three applications for which the number can
be configured. Maximum Sessions is the maximum number of users who can dial in to the
call-in number simultaneously. The voice prompt languages can also be modified here, and
you can enable/disable a particular dial-in number if you need to.
Restriction Tables
You’ll want to be able to control the telephone numbers that your users can modify
when using Unity Express features such as message notification. Doing so protects the
organization from users entering long-distance or international numbers. As you have
learned, message notification allows users to set CUE to call remote telephone numbers
for cellular phones or numeric pagers to notify them of new voice mails. Restriction
tables can limit the types of numbers that can be entered. To add or modify a
restriction table, navigate to System Restriction Tables. Figure 9.6 shows the
configuration options available.
FIGURE 9.6
Restriction tables
By default, a restriction table called msg-notification is already created for you. You can
choose to add a new table by clicking the Add button, or you can modify the one already
created. You have the ability to set the minimum and maximum number of digits allowed
as well as to create allow/deny call patterns. These patterns use the * and dot (.) wildcards
that we are already familiar with. Also keep in mind that the restriction table is read in
top-down order; the entry at the top of the restriction table is read fi rst, then the second,
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and so on. As soon as a match is made, the call is either allowed or denied based on the
rule that is matched. Therefore, it is important to set your more- exact patterns at the top
to match fi rst, followed by more-general patterns that take advantage of wildcards. Next,
we’re going to look at configuring voice mail default settings.
Configuring Voice Mail Default Settings
When voice mailboxes are created, you can configure various parameters to customize
them for each user or group who uses them. It is very likely, however, that most of your
mailbox parameters will be identical. Because of this, you’ll want to set the default settings
within Unity Express to fit what the majority of your users will require. In this section
you’ll see how to configure default call handling, voice mail configuration defaults for the
system, and voice mail defaults for individual mailboxes.
Voice Mail Call Handling
The default call-handling settings for voice mailbox users and groups are configured in the
Voice Mail > Call Handling window. You can modify the voice mail pilot, operator, and
AvT numbers. You can also specify the maximum number of sessions and languages used
for voice mail and AvT prompts. To make changes to voice mail call handling, navigate to
Voice Mail Call Handling. Figure 9.7 shows the available options.
F I GU R E 9.7
Call-handling settings
Configuring Unity Express System Settings and Voice Mail Defaults
423
The fields have already been populated based on the responses we gave when running
the Unity Express Initialization Wizard. If you need to modify these settings, you simply
change and apply them here.
Voice Mail Configuration
In the voice mail configuration section, you can set all of the CUE systemwide settings for
voice mail. To configure voice mail configuration default settings, go to Voice Mail VM
Configuration. Figure 9.8 shows what the configuration screen options look like.
FIGURE 9.8
VM Configuration settings
Looking at the figure, you can see that we can set the maximum time settings for overall
voice mail, individual mailbox, broadcast message, and expiration limits here. There are
also yes/no options for the following commonly used features:
Use MWI For Broadcast Messages You can determine whether the message
waiting indicator (MWI) light will be used when broadcast messages are sent to a mailbox.
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Play Caller ID For External Callers If off-network calls provide caller-ID information and
the caller leaves a message, you can choose to be able to play that caller-ID information to
the voice mail subscriber.
Mandatory Message Expiry You can determine whether messages will be deleted after
the absolute expiration time is up.
Another commonly used voice mail default setting we need to discuss is the Mailbox
Selection drop-down menu. This option determines which voice mail a caller will be sent to
when they are forwarded from one extension to another within Unity Express. By default,
the setting is for the last redirecting party. This means the mailbox assigned to the last
extension that redirected the call will get the voice mail. The other option is to use the
original called party. That means that no matter how many times a user is passed around,
if they end up going to voice mail, they will leave a message on the mailbox assigned to the
original extension that was called. Let’s look at an example.
Suppose a business has a single E.164 number that is used for customers to call.
This number is answered by a receptionist. The receptionist’s job is to figure out what
the customer wants and to forward that call to the employee who can best assist.
So a call comes into the receptionist at extension 3333. The receptionist answers the call
and determines that the customer needs to talk to the billing department at extension
3444. The call is forwarded, but nobody picks up the call. Depending on the Mailbox
Selection setting, the caller will then be directed to either the mailbox assigned to the
receptionist or the mailbox assigned to the billing department. Figure 9.9 is a visual
representation of what would happen in both the Last Redirecting Party and the Original
Called Party scenarios.
FIGURE 9.9
Mailbox selection process flows
Last Redirecting Party Process Flow
Customer
dials 444555-3333
Receptionist
answers and
forwards call to
3444
Call goes
unanswered and
is forward to Unity
Express
application
Unity Express runs
“Last Redirecting
Party” greeting
Customer leaves a
message regarding
a billing question
MWI Sent to Billing department phone
Original Called Party Process Flow
Customer
dials 444555-3333
Receptionist
answers and
forwards call to
3444
Call goes
unanswered and
is forward to Unity
Express
application
Unity Express runs
“Original Called
Party” greeting
MWI Sent to receptionist phone
Customer leaves a
message regarding
a billing question
Configuring Unity Express System Settings and Voice Mail Defaults
425
In our example, it is clear that we should configure our Mailbox Selection default
setting to Last Redirecting Party, because we want the billing department rather than the
receptionist to get the customer’s voice message.
Voice Mail Defaults
Voice mail defaults set parameters for individual mailboxes. To make changes to these
settings, navigate to Voice Mail VM Defaults. Figure 9.10 shows the settings that can be
modified here.
F I G U R E 9 .1 0
Voice mail default settings
You can set the default individual mailbox size and message size in seconds. You can
also set the message expiration time limit. If the system administrator has the Mandatory
Message Expiry set to Yes in the VM Configuration settings, all saved messages will be
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deleted after the message is kept past the expiry time. If this setting is set to No, Unity
Express will alert the user that the message will be deleted, but the user can choose to
resave the message, which essentially resets the expiry timer back to 0.
Creating Users, Groups,
and Mailboxes
We’re now ready to begin adding new users and groups to the Unity Express system. You
may already have users on the CUE from when you ran the Unity Express Initialization
Wizard. If any users had been created on the CUCM Express, they would have been
automatically imported into the CUE. At minimum, you should have the default cisco
administrator account configured as a user.
To learn how to create users and groups, we’ll use three different examples. First,
we’ll create a new user and assign a subscriber mailbox to them. Second, we’ll create a
group with multiple members who share a general delivery mailbox. Our last example
will be to create a group within Unity Express that is strictly for CUE administration
purposes using the AvT software feature. This administrator group will not have a
mailbox assigned.
User Creation with Mailbox
Our fi rst example will go through the process of creating a single Unity Express user with
a subscriber mailbox. We’ll configure user defaults that will be used as a template for all
subsequent users. We’ll then add a new user account, assign the user to an extension, and
create a subscriber mailbox that links a CUCM phone extension with a CUE subscriber
mailbox. Let’s get started.
Configuring User Defaults
When creating a new user, Unity Express uses a default template to fi ll in much of the
information required. As an administrator, you will want to modify this default user
template to match the settings you wish to use for the majority of your CUE users. To
modify user account default settings, navigate to Configure User Defaults. Figure 9.11
shows all of the default settings that can be modified.
Creating Users, Groups, and Mailboxes
F I G U R E 9 .11
427
User Defaults settings
In this section, you can change the default auto -generation policy for assigning user
PINs and passwords. You can also modify the minimum number of characters required
for each. If security is of high importance to your organization, you have the option to set
passwords to expire and force the user to use a password that has not been used previously.
The History Depth setting dictates how many past passwords are kept, which means that
the user cannot reuse them. For example, if History Depth is set to 2, when the user
needs to change their password, they cannot set it to either of the previous two passwords
that the account used.
You may also change the password/PIN lockout policy for all of your users. You can set
the number of attempts, lockout time period, and whether the lockout policy is temporary,
permanent, or disabled. If you choose to set the lockout policy to temporary or permanent,
you can also set the number of attempts tried before an account is locked.
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Creating a New User and Mailbox
We’re now ready to create a new user account. To do so, go to Configure Users. The
page will now display all of the currently configured users on Unity Express, as shown in
Figure 9.12. At this point, there is only the default cisco user.
F I G U R E 9 .1 2
The Configure > Users window
To create a new user, click the Add button. A new window will open that displays all of
the configuration parameters we can set for our new Unity Express user. Figure 9.13 shows
this window.
We can begin creating our new user (UserOne) by fi lling in the User ID, First Name,
Last Name, Nick Name, and Display Name fields. Keep in mind that there cannot be any
numbers, special characters, or spaces in the First or Last Name fields.
Creating Users, Groups, and Mailboxes
F I G U R E 9 .1 3
429
Adding a new user
Next, we want to assign an extension (CUCM Express ephone-DN) to this user so that
we can map the user to the extension and ultimately the extension to a voice mailbox. To
get a list of all the already defi ned ephone-DNs, click the Other radio button and click
the magnifying glass icon to the right of the blank field. A new window will open, titled
Extension Option. Click the radio button of the ephone-DN you want to assign to the user.
In this example, we chose extension 5001, as shown in Figure 9.14.
To save this extension assignment, click the Select Extension button in the upper left
of the window. This will close the window and bring you back to the Add A New User
window. We will fi nish the user configuration by choosing to specify a PIN and password
as opposed to assigning them randomly or leaving them blank. Lastly, we’ll check the
Create Mailbox check box. We’ll also leave the Forward CFNA & CFB Of Extension
(If Configured) To Voice Mail Number 188 check box checked. This option tells us that
our ephone-DN will be automatically set to forward calls that are not answered after
20 seconds or calls that cannot be answered because the user is already on the phone. The
calls will be forwarded to extension 188, which is the voice mail call-in number.
Figure 9.15 shows a completed user profi le.
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F I G U R E 9 .1 4
Selecting an extension for the new user
F I G U R E 9 .1 5
A completed new user form
Creating Users, Groups, and Mailboxes
431
Because we checked the Create Mailbox check box when we created our new UserOne
profi le, Unity Express will automatically bring up the mailbox configuration options as
soon as we click the Add button to actually create our new user.
A description of the mailbox will have been automatically created, containing the user
ID of the user the mailbox belongs to.
You can also customize the operator-assistance number to whatever the mailbox owner
wants. Maximum mailbox and message sizes can also be changed, although they must be
less than or equal to the maximum sizes you configured globally in the voice mail default
settings we configured previously in this chapter. Users can also modify the expiry time
for messages and indicate whether they want to use the tutorial when fi rst setting up the
mailbox.
If you attempted to check the Enable Notification For This User/Group
check box, you noticed that it didn’t work. When you attempt to save
the configuration by clicking the Add button, you will receive a warning
message that states “Notification disabled system wide, cannot enable
notification for owner.” You’ll learn how to enable Message Notification
systemwide later in this chapter, so we can later check this box and
successfully enable message -notification functionality.
Figure 9.16 shows the new mailbox configuration settings.
F I G U R E 9 .1 6
Adding a new mailbox
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When you have fi nished setting the mailbox parameters, click the Add button so that
Unity Express will build the mailbox for you.
Unity Express then takes you back to the original page of currently configured users.
Figure 9.17 shows UserOne as a user on the Unity Express system.
F I G U R E 9 .17
A newly configured user
That is how you can configure a user and subscriber mailbox on the CUE. Next, you’ll
learn how to create a CUE group to share a general delivery mailbox (GDM) as well as a
group used for administrative roles that do not require mailboxes.
Group Creation with Mailbox
In our fi rst group - creation example, let’s say that within the organization we have an
IT support team of three members (UserOne, UserTwo, and UserThree). Employees who
need technical support over the phone are told to call extension 5555. This extension is
Creating Users, Groups, and Mailboxes
433
configured on button 2 of all three phones that make up the IT support group. If for some
reason an employee dials 5555 and none of the three employees is able to answer the phone,
after five rings the call will be forwarded to voice mail. The IT support team wants to use a
GDM so that all three of them can be notified of a new voice mail as well as log in to check
the GDM box. With these requirements in mind, let’s create our IT support group with
a GDM.
The fi rst step in a new group creation is to navigate to Configure Groups. The next
page displays all of the groups currently configured on Unity Express, as shown in
Figure 9.18.
F I G U R E 9 .1 8
Configured groups
We want to build a new group from scratch, so click the Add button. A new window
will pop up, asking you to enter details that identify the group and allow privileges as
needed. Figure 9.19 shows the completed form for providing a new group for IT support.
Notice that Create Mailbox is checked; this will create a single GDM that all members
within the group will be able to access.
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Unity Express Configuration
Adding a new group
Don’t forget that GDMs are accessed differently from standard subscriber
voice mailboxes. With a GDM, a group member will log in to their personal
subscriber account first and then access the GDM through the subscriber
voice mail account. Only subscribers who are members of the GDM will
have access to the group mailbox.
After you click the Add button to create the new group, a window will open that allows
you to modify settings for our new general-deployment mailbox. Figure 9.20 shows this
mailbox configuration screen.
FIGURE 9.20
The Add A New Mailbox window for groups
Creating Users, Groups, and Mailboxes
435
When you have fi nished configuring the GDM, click the Add button to have CUE
build the new mailbox. You are then taken back to the main group page that lists all of
the currently configured groups. Figure 9.21 shows this screen with our newly configured
ITsupport group listed.
F I G U R E 9. 21
The newly configured group in the Configure > Groups window
Now that our group is created, we need to add our three members to it so they can
receive MWI and access the GDM to check messages. To accomplish this, click the
ITsupport link. The ITsupport group profi le will pop up as a new window. At the top of
this window are tabs that contain various configuration settings within the new group.
We want to add members to the group, so click the Owners/Members tab. The next screen
shows us that we do not have any configured members; the table is blank. To add members,
click the Subscribe Member button at the top. Now we are able to search for users we
wish to include in the ITsupport group. If you click the magnifying glass without entering
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any additional information, the window will display all users configured within the CUE.
Select the three users we want in the group, as shown in Figure 9.22.
FIGURE 9.22
Adding group subscribers
Once all three users are selected, click the Select Row(s) button. That’s all there
is to it. Now the three members have a shared mailbox for support calls that any of them
can access.
Group Creation for Administrative Roles
Groups are used not only for shared GDM setups, as shown in the last example; we can
also create administrator groups to give certain users access to various administrative roles
on Unity Express. Listed here are the administrative privileges that can be assigned. A
group can have one or many of the privileges assigned to a group:
Administration via Telephone (AvT)
AvT TUI access to change AA prompts.
Voice Mail Broadcaster— Local/Local and Network
messages.
AvT TUI access to send broadcast
Private/Public List Manager Web GUI access to view the existence and membership of
configured private or public distribution lists. No changes can be made.
Creating Users, Groups, and Mailboxes
Integrated Messaging
437
Access to Integrated Messaging configurations using the web GUI.
Historical Reports Viewer Access to view historical voice message using the web GUI.
Real Time Reports Viewer
web GUI.
Super Users
Access to view real-time voice message reports using the
Full access to the CUE web GUI; can perform any configuration task.
By default, two administrator groups are preconfigured, Administrators and
Broadcasters. Administrators have all the listed privileges assigned, while Broadcasters
have only the Voice Mail Broadcaster— Local privileges. In this example, we are going to
create an AvT group and assign a single member. This user will then be given the rights
to utilize the Administration via Telephone (AvT) application within the CUE.
The fi rst few steps of the group creation are identical to what we performed with the
ITsupport group. Navigate to the Configure Groups section and click Add to bring up
the group configuration settings. Fill out the Group ID, Full Name, and Description fields
for our AvT group. Do not click the box to make a GDM, because this group does not
need one. We will, however, give this group the right to access the AvT software contained
within Unity Express for modifying auto attendant prompts and other voice-related
recordings. Figure 9.23 shows the proper configuration settings for the AvT group.
FIGURE 9.23
Adding a new administrative group
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Click the Add button to create the new group. This action takes you back to the main
screen, which shows all the groups in the CUE. This screen is shown in Figure 9.24.
FIGURE 9.24
Groups window
The newly configured administrative group in the Configure >
As we did with the ITsupport group, we need to add members to the AvT group. Select
this group to modify it and then click the Owners/Members tab. The next screen shows us
that we do not have any configured members, because the table is blank. To add members,
click the Subscribe Member button at the top. Now we are able to search for users we
wish to include in the AvT group. If you click the magnifying glass icon without entering
any additional information, the window will display all users configured within the CUE.
Figure 9.25 shows the search results listing all users configured on Unity Express.
Configuring Auto Attendant
FIGURE 9.25
439
Adding group subscribers
Check the box next to UserOne to make that user a member of this group, and click
the Select Row(s) button. UserOne is now a member of the AvT group and can now
access the AvT software features by dialing into the AvT pilot number configured on CUE.
In the next section, we will go about configuring Auto Attendant. We will use UserOne in
this section to record new AA prompts.
Configuring Auto Attendant
As you’ll recall, when we ran through the Unity Express Initialization Wizard in the last
chapter we were asked to assign an extension for the Auto Attendant application. Once an
extension number is assigned, Unity Express will make the AA application available
to anyone who dials this extension. This section will cover how to administrate the
AA to change the AA scripts that users hear when they log in. Then we’ll go over how
to make changes to the system schedule to set your AA to function properly during both
working and nonworking hours as well as holidays. Finally, we will use the AvT application
to create the prompts to personalize the AA to our business.
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Administrating the Auto Attendant Application
By default, AA will use the prebuilt aa.aef script. To make general changes to the AA
setup, navigate to Voice Mail Auto Attendant Edit. Figure 9.26 shows the options
available.
FIGURE 9.26
Modifying Auto Attendant settings
In this location you can make the following changes:
Modify the AA call-in number.
Select the script you wish to use.
Upload custom-made scripts that were built with the Unity Express Editor PC
software.
Set the language used in the voice prompts.
Set the Maximum Sessions
Enable or disable AA.
Modify or upload parameters within the AA script you have chosen to use.
Configuring Auto Attendant
441
Change the prompts used by your AA script.
Choose the business schedule that informs the AA software of office hours and
holiday schedules. The AA uses this schedule to determine when to use various
prompts such as BusOpenPrompt or BusClosedPrompt.
As soon as you have made all the necessary changes, click the Apply button to activate
the modifications on the CUE.
Modifying the Business Hours Schedule
The business hours schedule is how the AA determines when to use the business open/
closed prompts that your AA script can utilize. Unity Express has a default schedule
that you can modify, or you can choose to create your own schedule. To make changes
to the business hours schedule, go to System Business Hours Settings. The default
systemschedule business hours schedule comes up, as shown in Figure 9.27.
FIGURE 9.27
Business hours schedule
A table consisting of the days of the week and the time in 30 -minute blocks is shown.
You can click these blocks to set the open and closed hours for the business. As soon as
you have made the necessary changes and clicked the Apply button, the office hour changes
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will be in effect and the AA script will change the prompts to use busOpenPrompt or
busClosedPrompt based on these settings.
Configuring the Holiday Schedule
The AA also can use different prompts when the office is closed for a holiday. This is
referred to as the holiday schedule. By default, no holidays are configured on the CUE.
You must manually configure the holidays your office observes. To add holidays to the
CUE, navigate to System Holiday Settings. Click the Add button to add a holiday. A new
window will open that asks you to enter the year, date, and a description of the holiday.
Figure 9.28 shows the Add A New Holiday screen.
FIGURE 9.28
Adding a new holiday to the holiday schedule
When you have entered all the proper information, click the Add button and the new
holiday will be active on the Unity Express system.
Creating Custom Prompts Using the AvT
You can record your own custom prompts for the AA by utilizing the AvT feature. When
you fi rst initialized Unity Express, you created a call-in number that was specifically meant
for AvT users to dial into. This application is referred to as promptmgmt on the CME web
GUI. A user must have AvT privileges assigned in order to be properly authenticated and
change the prompt messages. You learned how to assign those prompts in the “Group
Creation for Administrative Roles” section of this chapter. When this user dials the
Configuring Message Notification
443
call-in number and authenticates, they can enable alternate greetings on the fly in case
of an emergency, such as bad weather that might keep employees from coming into the
office. Users can also record their own prompts to use in the AA script for customization
purposes. The default AA prompts that come on the CUE are very generic. Most businesses
want a custom AA that at minimum says “Welcome to company XYZ” to inform callers of
the name of the business they are calling. When the AvT user creates and saves a recorded
prompt, this recording will be accessible on the CUE web GUI by going to System Prompts. Figure 9.29 shows the current saved custom prompts.
FIGURE 9.29
Custom prompts using AvT
The prompt at the bottom that begins with “UserPrompt” is a custom voice fi le an AvT
administrator has created and saved on the system. This new prompt can then be used
within the active AA script to personalize the AA experience.
Configuring Message Notification
The Message Notification feature allows mailbox subscribers to be notified of a new
voice mail message by calling a remote telephone such as a home or cellular phone. You can
also configure Message Notification to send an email or alphanumeric page. The details
here are subject to the restrictions discussed earlier.
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SMTP Settings for Message Notification
Manhattan Optical is a distributor of glasses and contact lenses throughout the United States.
The sales team is equipped with smartphones with email capabilities. A request has been sent
to IT to see if it is possible to notify the sales team of new voice mail messages over email.
Ann, the support engineer, has discovered with a little research that Unity Express has a feature
called Message Notification that can remotely alert users of new voice messages via email.
Ann uses the web GUI and finds where Message Notification is configured for voice mail
users. She enables Message Notification and clicks the Apply button to make the changes
active on the configuration. But when she does, an error message appears, stating that
an SMTP server must be configured for the proper sending of email notifications. This
error makes complete sense to Ann because previously there was no need to configure
SNMP within Unity Express. Now that Ann needs to send emails from Unity Express, she
needs to identify a mail server from which to send messages. Logging in to the Unity
Express command line, Ann issues the following command:
UC500-CUE#configure terminal
UC500-CUE(config)# smtp server address 10.68.20.10 authentication username
uc500 password VmP4ss
This command configures outgoing emails to use the SMTP server located at 10.68.20.10.
This mail server requires authentication, so a username of uc500 and a password of
VmP4ss were included.
Now that an SMTP server has been properly configured on Unity Express, the sales staff
can be sent email notifications when new voice mails arrive in their mailboxes.
By default, Message Notification is disabled systemwide. Navigate to Voice Mail Message Notification Message Administration. Here you can check the box to
enable Message Notification systemwide. Figure 9.30 shows this enable screen.
FIGURE 9.30
Message Notification systemwide settings
Configuring Message Notification
445
In addition to enabling Message Notification, the administrator can specify the
following global settings, as listed in Table 9.1.
TA B L E 9 .1
Message Notification systemwide settings
Setting
Description
Options
Enable system-wide
notification
Types of messages that will
trigger message notification
All Messages will trigger
message notification for
any voice mail message.
Urgent Messages will trigger
message notification only
when the calling party flags
the message as urgent.
Allow user to login
to voice mail box to
retrieve voice mail when
phone notification device
is notified
With this box checked, when
Unity Express calls a configured
number on the message
notification list, that user is
allowed to automatically log
in and check their messages.
Otherwise, the user would
have to terminate the call and
dial back into Unity Express to
retrieve messages.
Enabled or disabled.
Attach message
to outgoing email
notification
Will attach voice mail message
to email notifications in a .wav
format
Enabled or disabled.
Enable Cascading
Notifications
Unity Express waits a period of
time for the user to log in and
check their messages. After
a set time has expired, Unity
Express will notify the next
phone, email address, or pager
on the list.
Enabled or disabled.
If phone is not answered,
hang up after XX seconds
Amount of time Unity Express
Number of seconds between
attempts to reach a notification
12 and 96.
number before the system
considers the notification to
have failed to reach the intended
party.
Restriction Table Name
A restriction table can be added
to limit the phone numbers
that can be used for message
notification.
Any configured restriction
table in the drop-down list.
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In our example, we’re going to enable Message Notification for urgent messages. The
user will have the ability to log in to Unity Express automatically to retrieve messages.
Also, we have enabled Cascading Notifications to cycle through any configured notification
destination endpoints configured.
Now that we have Message Notification enabled globally, we can go into individual user
accounts and enable Message Notification. To enable this feature on existing accounts,
navigate to Configure Users and click the User ID of the user for whom you wish to
set up message notification. Figure 9.31 shows that we have decided to enable Message
Notification on UserOne’s account.
F I GU R E 9. 31
Enabling Message Notification for a user or group
To enable notification, click the Enable Notification For This User/Group check box.
Next, click the Notification tab for UserOne. Here we can go ahead and set up
the notification destination endpoints. There are several notification device types, for
notification of telephones, pagers, and email addresses. To configure these options, choose
the notification endpoint by clicking the link. The screen will then display options for you
to enable notification for the device and to enter the information Unity Express needs to
send alerts. For example, Figure 9.32 shows us enabling and configuring notification for a
home telephone.
Configuring Message Notification
FIGURE 9.32
447
Configuring Message Notification: home phone
We’ve checked the Enable check box and added the telephone number we want Unity
Express to call. We’ve also set the notification schedule to send alerts only when messages
come in Monday through Friday from 8:00 a.m. to 5:00 p.m. Once you have fi nished
setting up the notification endpoint, click the Apply button.
Unity Express will then bring us back to the main Message Notification screen for
UserOne. Figure 9.33 now lists the Home Phone device type as enabled and the telephone
number associated with this option.
Now that we have set up message notification for UserOne, when an urgent message is
stored in the subscriber mailbox, Unity Express will call the home phone number of
333 -555-1234 and inform UserOne that an urgent message is waiting for them. The user
can then authenticate to their mailbox using their PIN and retrieve the message.
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Chapter 9
FIGURE 9.33
Unity Express Configuration
Configuring Message Notification: home phone enabled
Administrating and Troubleshooting
Unity Express
Unity Express administrators have multiple tools available to them to help keep the
system running smoothly. This section will discuss how to synchronize configuration
information contained within the CUCM Express and CUE databases. Then we’ll talk
about how to set up external backups of your CUE data in case of a failure. Finally, we’ll
talk about how to start and view trace files in real time to help troubleshoot any issues on
the CUE.
Synchronizing Information
It is always important to keep in mind that the call-processing configuration of the
CUCM Express is separate from the configuration of Unity Express. They are two
completely separate devices that share a single hardware chassis. When you run the
Administrating and Troubleshooting Unity Express
449
Unity Express Initialization Wizard for the fi rst time (as described in Chapter 8), this
process pulls in information from the CUCM Express. This is the fi rst synchronization
performed between the two systems to ensure that all the data to be shared between
the two systems is known by both devices. Any subsequent additions to either the
CUCM Express or Unity Express may need to be synchronized; you can do this
using the Synchronize feature in the CUE web GUI. For example, when the CUCM
Express administrator creates a new ephone-DN, this information is not known to
Unity Express. You must perform a synchronization between the two databases.
To do this, navigate to Administration Synchronization Information.
The synchronization will then be performed, and any changes will be displayed once the
synch is complete.
Backing Up and Restoring Configurations
It is vital that you, as an administrator of a voice mail system, make regular backups of the
configuration and user/group mailbox contents. Unity Express gives you the ability to back
up data to an external server such as an FTP server and restore it from there.
To set up a server for external backups, go to Administration Backup/Restore Configuration. Figure 9.34 shows the configuration options.
FIGURE 9.34
The Backup/Restore > Configuration screen
The figure shows that our FTP server is located at 192.168.10.50. The FTP server
username and password are set here for proper authentication. You can set the Maximum
Revisions option to tell the system to keep a specific number of backups on the FTP server.
Any backups above this number will be deleted, with the oldest backups deleted fi rst.
Once your Backup/Restore server is set up, you can run a backup or restore by
navigating to Administration Backup/Restore Start Backup or Administration Backup/Restore Start Restore.
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Running a Unity Express Trace
A trace on Unity Express is the equivalent of the debug command on a Cisco router or
switch. To run a trace, you must connect to Unity Express using the command line. Once
you are connected, you run a trace for a particular module within the CUE. Modules are
segments of Unity Express that handle different processes within the system. The following
output shows us logging into the CUCM and displaying some of the trace modules
available:
Router#service-module integrated-Service-Engine 0/0 session
Trying 192.168.10.1, 2002 ... Open
UC500-CUE> en
Password:
UC500-CUE# trace ?
BackupRestore Module
all
Every module, entity and activity
caff-sip
Module
capi
Module
ccn
Module
config-ccn
Module
configapi
Module
dbclient
Module
dns
Module
editorexpress Module
entityManager Module
imap
Module
limitsManager Module
[output cut]
UC500-CUE#
To view the trace information in real time, we fi rst must initiate a trace. For example,
we’ll initiate a full trace of the ccn module by running a trace ccn all privileged- exec
command. The ccn trace is one example of many traces that can be run, as shown here:
UC500-CUE#trace ccn all
Now we can view the trace output in real time by doing a show trace buffer tail
privileged EXEC command. The tail portion of the command tells Unity Express to print
the last log message to the screen. Following is an example of this command:
UC500-CUE# show trace buffer tail
Press <CTRL-C> to exit...
Administrating and Troubleshooting Unity Express
1829 04/20 22:01:30.034 WFSP MISC 0 WFSysdbNdJCallStats::get exit
3313 04/20 22:07:34.670 DSSP LWRE 0 Received UDP packet on 192.168.10.2:5060 ,
source 192.168.10.1:60349
INVITE sip:188@192.168.10.2:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.1:5060;branch=z9hG4bK272504
Remote-Party-ID: <sip:5002@192.168.10.1>;party=calling;screen=no;privacy=off
From: <sip:5002@192.168.10.1>;tag=54322C-998
To: <sip:188@192.168.10.2>
Date: Fri, 03 Jul 2009 14:25:53 GMT
Call-ID: 4004C842-671411DE-80269C78-EC077092@192.168.10.1
Supported: 100rel,timer,resource-priority,replaces
Min-SE: 1800
Cisco-Guid: 1073935306-1729368542-2149686392-3959910546
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE,
NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1246631153
Contact: <sip:5002@192.168.10.1:5060>
Call-Info: <sip:192.168.10.1:5060>;method=”NOTIFY;Event=telephone-event;
Duration=2000”
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 191
v=0
o=CiscoSystemsSIP-GW-UserAgent 5216 7538 IN IP4 192.168.10.1
s=SIP Call
c=IN IP4 192.168.10.1
t=0 0
m=audio 18456 RTP/AVP 0
c=IN IP4 192.168.10.1
a=rtpmap:0 PCMU/8000
a=ptime:20
--- end of packet ---
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To end the trace, we exit out of the real-time monitoring and issue no trace ccn all.
Trace fi les can come in handy when you are troubleshooting multiple CUE problems. Also,
if you ever create a Cisco support case for a problem with Unity Express, one of the fi rst
pieces of information technical support requests is the output of a trace on the system.
Summary
In this chapter, we went over many of the different configuration, backup, and
troubleshooting options for Unity Express. Using the web GUI, administrators can
quickly and easily implement voice mail and Auto Attendant features that were laid out
in the design phase of the project. You also learned how to back up and restore not only
configurations but also any saved voice mail fi les. Finally, this chapter introduced you
to the trace command, which is a powerful tool used for troubleshooting problems that
might arise when implementing or supporting Unity Express.
In Chapter 10, you’ll learn how to work with the SBCS platform and Cisco
Configuration Assistant.
Exam Essentials
Know How to configure an NTP server for proper unity express timekeeping. Mailbox
subscribers require an accurate clock to ensure that they know when callers left messages.
Using the CUE GUI, navigate to System Network Time & Time Zone Settings.
Understand when it is important to configure a DNS Server on unity express. Whenever
Unity Express needs to talk to an external server such as an SMTP server for sending
voice mail as email attachments, you might configure the server using a domain name as
opposed to an IP address. If you use a domain name, you need to configure Unity Express
with a DNS server to resolve the domain name to an IP address.
Know how to modify the default language used on unity express. To modify language
settings, navigate to System Language Settings.
Understand the three common types of call-in numbers. The three primary call-in
numbers configured are the voice mail pilot, auto attendant, and AvT extensions.
Understand the purpose of restriction tables. Restriction tables control the telephone
numbers that Unity Express users can enter for message notification. These tables help to
protect a company from expensive long-distance charges.
Know what default VM settings can be made on unity express. Options such as mailbox/
message size and expiration time limits can be set here; they are applied to all subscriber
mailboxes by default.
Written Lab 9.1
453
Know how to create users, groups, and mailboxes using the unity express web GUI. Users
are individuals who usually are associated with their own personal subscriber mailbox.
Groups consist of two or more people who share a GDM box as well as have individual
mailboxes. All of these are set up using the Configure tab.
Understand that groups are created for both GDM and administrative purposes. Besides
their use for sharing GDMs, groups can also be created to serve administrative roles such
as an Administration via Telephone (AvT) Administrators group.
Know how to configure auto attendant features using the Unity Express web GUI. You
can configure Auto Attendant by navigating to Voice Mail Auto Attendant. Here you can
set up your scripts, set business/holiday hours, and manage custom prompts.
Understand the purpose of message notification. Message Notification is a feature that
alerts users of new voice messages by triggering automated calls or emails to a destination
the user chooses.
Understand the process of synchronizing CUCM Express with Unity Express The CUCM
Express and Unity Express share information with each other, such as ephone-DNs and
users. When you make an addition to shared information, you need to synchronize the two
systems to ensure they both are aware of the change.
Know how to back up and restore the unity express configuration and Files. You can
back up and restore information to a remote location by navigating to Administration Backup/Restore and then choosing to configure the settings or run a backup or restore.
Know what a unity express trace command is used for. A Unity Express trace is much
like a debug on a Cisco router or switch. You can obtain information that is helpful for
troubleshooting problems with your system.
Written Lab 9.1
Write the answers to the following questions:
1.
Sam just checked his voice mail and found a new message waiting in his mailbox that
said it was left at 2:00 p.m. today. This is impossible because it is only 1:15 p.m. What
should the administrator check?
2.
The mail administrator gave you the following SMTP gateway to use for Unity Express
Integrated messaging: mail.fakecompany.com. What else do you need to do to ensure
that SMTP is properly configured on the CUE?
3.
What is the maximum number of language packages that can be used at once on a
Unity Express system?
4.
What can administrators use to limit the types of numbers they can enter for Message
Notification?
5.
What feature is used to control the reuse of passwords?
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Unity Express Configuration
6.
What two types of groups are used on Unity Express?
7.
Using the Unity Express web GUI, where can you configure Auto Attendant features?
8.
Where can you modify the operational hours of a business within the Unity Express
web GUI?
9.
Name the type of Message Notification in which Unity Express notifies one device,
waits for the user to acknowledge the notification, and, if the message is not
acknowledged after a period of time, alerts the next device (phone, pager, or email).
10. What tool does Unity Express use for troubleshooting purposes that is similar to the
debug command used on Cisco routers and switches?
(The answers to Written Lab 9.1 can be found following the answers to the review
questions for this chapter.)
Hands-on Labs
To complete the labs in this section, you need a UC500 with a Cisco Unity Express AIM
module. Note that if you are using a different CUCM Express router and Unity Express
network module, the commands used will be slightly different. Lab 9.2 also requires an
FTP server to be connected to the network.
Here is a list of the labs in this chapter:
Lab 9.1: Viewing Real-Time Trace Logs
Lab 9.2: Saving and Retrieving Trace Log Files
Hands-on Lab 9.1: Viewing Real-Time Trace Logs
In this lab, we’re going to run a trace on Unity Express and view the fi les in real time.
We’re going to focus on troubleshooting problems when leaving voice mail messages. The
problem can be re- created, so our solution will be to turn on the trace, view the logs in real
time, and then disable the trace.
1.
Log in to your CUCM Express router and go into configuration mode by typing
enable, entering your password, and then typing configuration terminal.
2.
Session into the Unity Express AIM module by typing service-module integratedService-Engine X/X session. The X/X is the AIM module/slot where the hardware
resides within the CUCM Express router.
3.
Enter into privileged- exec mode by typing enable and entering your CUE password.
4.
Turn on voice mail debugging by typing trace voice mail debug all.
5.
Begin real-time viewing by typing show trace buffer tail.
6.
Re- create the voice mail problem that is being experienced, and watch the trace debugs
on the terminal screen.
7.
Once the problem has been re- created and captured in real time, disable the trace by
issuing no trace voice mail debug all.
Hands-on Labs
455
Hands-on Lab 9.2: Saving and Retrieving
Trace Log Files
Now let’s say that by viewing the real-time trace debugs we ran in lab 9.1, we were unable
to fi nd the problem. We’ve decided to get some help from the Cisco Technical Assistance
Center (TAC), and they have requested that we run the same trace and send it to them
for viewing. This time, instead of viewing the real-time logs, we’re going to have the logs
written to the trace log fi le. To do this, fi rst we will enable saving to a log fi le; then we will
start our trace and re- create the problem. Finally, we’ll disable the trace and FTP the log
fi le to our desktop so we can forward it to TAC for review.
1.
Log in to your CUCM Express router and go into configuration mode by typing
enable, entering your password, and then typing configuration terminal.
2.
Session into the Unity Express AIM module by typing service-module integratedService-Engine X/X session. The X/X is the AIM module/slot where the hardware
resides within the CUCM Express router.
3.
Enter into privileged- exec mode by typing enable and entering your CUE password.
4.
Enable saving trace logs to a file stored on the compact flash by typing log trace
buffer save.
5.
Turn on voice mail debugging by typing trace voice mail debug all.
6.
Re- create the voice mail problem that is being experienced, and watch the trace debugs
on the terminal screen.
7.
Once the problem has been re- created and captured in real time, disable the trace by
issuing no trace voice mail debug all.
8.
View the saved log files on the compact flash by typing show logs. The log file that
Cisco TAC will want to see is called atrace_save.log.
9.
Copy the atrace_save.log file from the compact flash to an external FTP server by
typing copy log atrace_save.log url ftp://user1:cisco@XX.XX.XX.XX/cue/atrace_
save.log, where XX.XX.XX.XX is the IP address of your FTP server and there is a
configured user named user1 with a password of cisco.
10. Once you have the log file on the FTP server, you can forward it to Cisco TAC
for review.
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Unity Express Configuration
Review Questions
1.
For Integrated Messaging to work properly, what must be configured on Unity Express?
A. DNS server
2.
B.
DHCP server
C.
HTTP server
D.
SSH server
Languages within Unity Express can be set at all of the following levels except:
A. Default level
3.
B.
Admin level
C.
User level
D.
Group level
Where can you set the maximum number of voice mail sessions allowed within the Unity
Express web GUI?
A. Voice Mail VM Configuration
4.
B.
Voice Mail Call Handling
C.
Configure User Defaults
D.
System User Defaults
By default, how is Unity Express configured to route a caller to the correct mailbox when
calls are forwarded within the CUCM Express system and the call is not answered?
A. Original called party
5.
B.
General delivery mailbox
C.
Operator mailbox
D.
Last redirecting party
How can the administrator set a limit on how long any voice mail is saved on the Unity
Express System?
A. Ensure that Mandatory Message Expiry is set to No
6.
B.
Ensure that Mandatory Message Expiry is set to Yes
C.
Ensure that Mandatory Message Expiry is set to Delete After 30 Days
D.
Ensure that Mandatory Message Expiry is set to Yes or Delete After 30 Days
What are the three types of user password/PIN lockout policies?
A. Temporary
B.
Notify
C.
Permanent
D.
Disabled
E.
Static
Review Questions
7.
457
When adding a new user mailbox, what is the result of creating a user mailbox by checking
the Create Mailbox check box but unchecking the Forward CFNA & CFB Of Extension To
Voice Mail Number option? Choose all that apply.
A. The user will have a new mailbox created with the default mailbox parameters.
8.
B.
The user will have a new mailbox, but it will be disabled.
C.
Calls going into the user extension will never go to voice mail when the user is either
unable to answer or already on the phone.
D.
Calls going into the user extension will be redirected to the voice mailbox when the
user is either unable to answer or already on the phone.
What does the Zero Out feature do?
A. Resets the keypad to Unity Express default settings
9.
B.
Resets the message expiry time to 30 days
C.
Provides an extension to forward calls within Unity Express voice mail to when the
caller presses the 0 button
D.
Provides a way to reset the user PIN/password while using the Unity Express TUI
We want a group of users to be able to listen to group voice mail messages and be notified
of new messages via the MWI. What type of GDM users are these?
A. GDM operators
B.
GDM owners
C.
GDM members
D.
GDM super users
10. What are the two Unity Express groups available by default?
A. Administrators
B.
Broadcasters
C.
AvT Administrators
D.
Super Users
E.
Historical Reports Viewers
11. What is the name of the Auto Attendant script used by default on Unity Express?
A. aa.aef
B.
aaDefault.wav
C.
aa.wav
D.
aaDefault.aef
12. On the web GUI for Unity Express, which holidays are configured by default?
A. Christmas Day
B.
New Year’s Day
C.
Independence Day
D.
No holidays are configured
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13. When an AvT administrator creates a new recording for the Auto Attendant, where can the
newly recorded files be accessed using Unity Express GUI?
A. Voice Mail Auto Attendant
B.
Voice Mail AvT
C.
System Prompts
D.
The files can be accessed only through the command line.
14. What step must be taken to properly configure Message Notification to work over email
using Unity Express?
A. Configure an SNMP server
B.
Enable network address translation (NAT) on Unity Express
C.
Configure an SMTP server
D.
Configure MWI on all ephones
15. What two types of messages can you choose for triggering message notification?
A. Urgent messages
B.
Broadcast messages
C.
GDM messages
D.
All messages
16. When configuring Message Notification on the Unity Express web GUI, what does it mean
when you check the “Allow user to login to voice mail box to retrieve voice mail when
phone notification device is notified” check box?
A. The user can call back in and access messages remotely.
B.
The user can have the voice mail automatically sent to an alphanumeric pager.
C.
The user can listen to their messages without terminating the call.
D.
The user can dial into their personal extension to retrieve messages.
17. Voice mail messages sent as email attachments on Unity Express are sent in what format?
A. .jpg
B.
. au
C.
.wav
D.
.mp3
E.
.mp4
Review Questions
459
18. You’ve just finished configuring a backup FTP server to save configuration and mailbox
contents onto a separate server. When configuring the backup server, you set the maximum
number of revisions to five. What does this mean?
A. Unity Express will monitor the number of backups on the FTP server and keep the five
oldest revisions.
B.
Unity Express will monitor the number of backups on the FTP server and keep the five
newest revisions.
C.
The FTP server will monitor the number of Unity Express backups and keep the five
oldest revisions.
D.
The FTP server will monitor the number of Unity Express backups and keep the five
newest revisions.
19. What is a trace in regard to Cisco Unity Express?
A. A tool to trace analog phone calls on the PSTN
B.
A tool to trace voice messages back to the original calling party
C.
A tool similar to the debug command used for troubleshooting purposes
D.
A methodology used by voice engineers to route voice traffic to the proper subscriber
or GDM
20. What two methods below will let you run a trace on Unity Express hardware?
A. Web GUI
B.
CCA
C.
Console port
D.
SSH/Telnet
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Answers to Review Questions
1.
A. Your Unity Express system must be set up to properly resolve hostnames. Therefore,
you must configure at least one DNS server.
2.
B. Languages can be set at a global or default level as well as at user and
group levels.
3.
B. You can configure language settings and maximum sessions by using the Unity Express
GUI and navigating to Voice Mail Call Handling.
4.
D. The calls are sent to the extension mailbox that the user was last redirected to.
5.
B. Mandatory Message Expiry can be set to either Yes or No. If you want all messages to
be deleted after a specified period of time, you should set this to Yes.
6.
A, C, D. The three types of lockout policies are temporary, permanent, and disabled.
7.
A, C. Checking the Create Mailbox check box indeed makes a new mailbox for the user.
However, if Forward CFNA & CFB Of Extension To Voice Mail Number is unchecked,
calls to that extension that are either no-answer or busy will not be forwarded to the
mailbox.
8.
C. The Zero Out feature is a way the user/administrator can set an extension to forward
operator-assistance calls to.
9.
C. Standard GDM users can listen to messages and be notified of new messages using message waiting indicators.
10. A, B. By default, Administrators and Broadcasters groups are defi ned.
11. A. The script used by default on Unity Express AA setups is aa.aef.
12. D. By default, Unity Express does not have any holidays configured.
13. C. Prompts are found by navigating to System Prompts on the Unity
Express GUI.
14. C. An SMTP server must be defi ned so that mail can be properly sent.
15. A, D. You can set Message Notification to trigger either on all messages or on those
marked by the caller as urgent.
16. C. When a user receives a notification call on their home or mobile phone, they have the
ability to listen to the message without having to dial back into Unity Express.
17. C. Messages attached to emails are sent as .wav fi les.
Answers to Review Questions
461
18. B. Unity Express is responsible for keeping track of the number of revisions saved on the
FTP server. Once the number of revision fi les goes over five, Unity Express deletes the oldest revisions.
Backups are a manual process, and you can kick off the backup by using the web GUI to
navigate to Administration Backup/Restore Start Backup.
19. C. A trace is similar to the debug command used for troubleshooting CUE.
20. C, D. The only way to run the trace tool is through the command line.
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Unity Express Configuration
Answers to Written Lab 9.1
1.
NTP and time zone settings
2.
Configure a DNS server on Unity Express
3.
Five
4.
Restriction tables
5.
History depth
6.
GDM groups and Administrator groups
7.
Voice Mail Auto Attendant
8.
System Business Hours Settings
9.
Cascading Notifications
10. Trace
Chapter
10
Introducing the
SBCS Platform and
Cisco Configuration
Assistant
THE FOLLOWING CCNA VOICE EXAM
OBJECTIVES ARE COVERED IN THIS
CHAPTER:
Implement UC500 using Cisco Configuration Assistant.
Describe the function and operation of Cisco Configuration
Assistant.
Describe the components of the Cisco Unified
Communications Architecture.
Describe how the Unified Communications components
work together to create the Cisco Unified Communications
Architecture.
Perform basic maintenance and operations tasks to
support the VoIP solution.
Explain basic maintenance and troubleshooting methods
for UC500.
Auto -pilot…no assembly required…fly-by-wire…Plug and Play.
All of these terms conjure up images of instant operability and
simplicity that take the user out of the experience so that focus
can be placed elsewhere. These same terms can be used to describe how Cisco envisioned
the SBCS lineup of products.
In this chapter you’ll get an overview of the Smart Business Communication System
product lineup Cisco offers, targeted toward small businesses and branch offices. You’ll
learn how all these products are engineered to take as many of the difficult configuration
and provisioning steps as possible out of the process. Last, the chapter will show you the
Cisco Configuration Assistant application, an easy-to -use tool for the configuration of
more advanced voice and data features that are not enabled by default.
The Smart Business
Communications System
The Cisco Smart Business Communications System (SBCS) is a separate lineup that
focuses on the needs of small- to medium-size businesses. Cisco’s goal with the SBCS is not
necessarily to be the cheapest solution. Instead, they’re marketing the lineup to businesses
that require feature-rich communications environments that are more commonly found in
the larger enterprise- class hardware and software. In this sense, you really do get a bargain
in terms of capabilities for your dollar. This section will briefly introduce the products that
make up the SBCS lineup including the UC500 Series, the only platform that offers voice
capabilities.
The SBCS Components
Cisco has spent a great deal of engineering time and effort to come up with the SBCS lineup
for small businesses. There is a product within their offerings for virtually every need that
a small business would require. Not only that, but they have designed all SBCS components
to be extremely easy to deploy to remote offices. In fact, many basic capabilities are
ready to go out of the box; essentially, the equipment can be almost Plug and Play if
desired.
Let’s take a closer look at each product currently available within the SBCS lineup.
The Smart Business Communications System
465
Unified Communications 500 Series
There are two distinct chassis form-factor types for the UC500. The UC520 -8U and
UC520 -16U models come in a “desktop” model chassis. The 8U and 16U in the model
number stand for the number of IP phone users licensed on the system. The height of
the units is 1.5 rack units. A rack unit (abbreviated U; don’t confuse this with the model
number) is a unit of measurement in the data/telecom world to describe the height of
equipment. Each rack unit is 1.75~IN. It’s possible to mount the desktop form-factor
models in a 19~IN rack, but that requires the purchase of special mounting brackets.
The other form-factor model is for the UC520 -24U, UC520 -32U, and UC520 - 48U. This
larger chassis is a full 19~IN wide, so it fits in a standard telecom rack, and no optional
mounting extensions are needed. The height of the unit is 2U. It also has a built-in power
supply; by contrast, the desktop version requires an external power brick. Table 10.1 breaks
down the different hardware options available for the UC520 Series.
TA B L E 1 0 .1
UC500 Series Hardware Options
Hardware Options
Description
UC520-8U and 16U
UC520-24U, 32U,
and 48U
Chassis form factor
Dimensions of
the unit
Desktop
Standard 19~IN
Height
Height of the unit
1.5U
2U
Power supply
Type of power supply
External power
brick
Built-in power
supply
Console port
Used to connect
directly to the
UC500 via PC serial
connection
1 port
1 port
Power over Ethernet
(PoE)
10/100 BaseTX for
802.3af and up to
15.4W
8 ports
8 ports
Fixed FXS ports
Used to connect
analog devices to the
network
4 ports
4 ports
Fixed FXO ports
Used to connect to
the PSTN for analog
service
Optional, 4 ports
Optional, up to 8
ports
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TA B L E 1 0 .1
Introducing the SBCS Platform
UC500 Series Hardware Options (Continued)
UC520-24U, 32U,
and 48U
Hardware Options
Description
UC520-8U and 16U
ISDN BRI ports
Used to connect to
the PSTN for digital
service
Optional, 2 ports
Optional, up to 6
ports
T1/E1 PRI ports
Used to connect to
the PSTN for digital
service
Not available
Optional 1 port
LAN expansion port
10/100 BASE-TX to
uplink to SBCS switch
for expansion
1 port
1 port
Voice expansion slot
VWIC slot to add
additional analog/
digital ports
FXS, FXO, ISDN BRI,
T1/E1
FXS, FXO, ISDN
BRI, T1/E1
WAN Ethernet port
10/100 BASE-TX
1 port
commonly used to
connect to DSL or
cable Internet services
1 port
MoH audio jack
3.5 mm port to
connect to external
audio source for MoH
such as CD player or
iPod
1 jack
1 jack
Compact flash (CF)
slot
Used to store
software and
configuration files
1 slot
1 slot
Integrated wireless
Integrated WAP with a
single antenna
1 WAP 802.11b/g
Not available
The fi rst two illustrations present a visual representation of the majority of the hardware
options available on the UC500 Series platform. The desktop and 19~IN rack form factors
have the ports laid out in the same manner, so I’ll show only the desktop form factor.
Figure 10.1 illustrates the UC520 in the desktop form factor with the optional four FXO
fi xed ports included, and Figure 10.2 details the same desktop form factor UC520 with the
optional two -port ISDN BRI included.
The Smart Business Communications System
F I G U R E 1 0 .1
467
UC500 desktop chassis with fixed FXO ports
Cisco Unified 500 Series
Router
SYS
POE
VM
WLAN
3
POWER over ETHERNET
1/7
1/6
1/5
1/4
1/3
1/2
2
FXO
1/1
1/0
1
3
0
CF
2
FXS
1
0
0
COMPACT FLASH
1/8
CONSOLE
F I G U R E 10. 2
UC500 desktop chassis with fixed ISDN BRI ports
Cisco Unified 500 Series
Router
SYS
POE
VM
WLAN
3
POWER over ETHERNET
1/7
1/6
1/5
1/4
1/3
1/2
1/1
1/0
CF
2
COMPACT FLASH
FXS
1
0
0
1/8
CONSOLE
Note that you can have only the four FXO ports or the two ISDN BRI ports in a fi xed
setup. If you require additional FXO/BRI ports, you’ll need to use the voice expansion slot.
The UC500 Series hardware is truly the heart of the SBCS lineup. With a single device,
you get the following functionality:
Cisco Unified Communications Manager Express
Voice gateway functions using supported built-in FXS, FXO, ISDN-BRI, and other
voice interfaces using the voice expansion port
Unity Express for voice mail, auto attendant, and IVR capabilities
Built-in Ethernet switch for voice and data connections
Power over Ethernet (PoE) support
VLAN configuration
Quality of Service (QoS) capabilities
IPSec VPN capabilities
Firewall capabilities
Static routing (no dynamic routing protocols are supported)
Optional built-in wireless autonomous access point (desktop chassis only)
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Keep in mind that the UC500 chassis devices are essentially fi xed, with no expandability
to add users. If you have 16 users and opt to purchase a UC520 -16U desktop device, as
soon as you require a 17th phone, you’ll be required to upgrade to the larger chassis.
Because of this lack of flexibility in field upgradability, you’ll need to pay special attention
to the future user requirements of the site where you are deploying the UC500.
Because this book focuses on voice, the UC500 Series is of utmost importance to us.
We will cover the UC500 series out- of-the-box capabilities in more detail later in this
chapter. The other non-voice features are not covered in this book, but it is important to
know the other features that are available to you, so we will look briefly at the other SBCS
series fi rst.
UC500 Power Failover Allows Good News to Reach Pet Owners
Heather is an employee at a veterinarian hospital in the Midwest. Her job is to keep
owners up to date on the status of their pets after various surgical procedures the
hospital performs there.
In the spring, the weather patterns can often lead to violent rainstorms with high winds.
On this particular day, the storm managed to knock out the power to the hospital.
The network equipment had proper UPS power, but after around 30 minutes, the
batteries ran out and power was still not restored.
Heather still needed to contact a pet owner to update them on the successful surgery
that was performed on a Labrador earlier that morning. Fortunately for Heather, the
voice system that was installed in the hospital was a Cisco UC500. Cisco incorporated a
high availability feature in the UC500 that allows users to make phone calls using analog
phones connected to FXS ports for outbound calling on FXO ports even when there is no
power to the system. The power failover feature allows the analog phone connected to an
FXS port be switched directly to the FXO ports that are connected to the PSTN. This way,
emergency calls can be made during occurrences when the UC500 loses power during an
outage.
Secure Router 520 Series
At fi rst glance, you might question why Cisco has included the Secure Router 520 (SR520)
Series in the SBCS lineup. While it’s true that the UC500 can provide router capabilities in
the form of static routes, the Secure Router 520 Series supports many of the most popular
dynamic routing protocols, which can greatly simplify larger and therefore more complex
routing environments. The following features are supported on this platform:
The Smart Business Communications System
Dynamic routing protocol support
Advanced firewalling and intrusion prevention system (IPS) capabilities
Built-in Ethernet switch for voice and data connections
Power over Ethernet (PoE) support
VLAN configuration
Quality of Service (QoS) capabilities
IPSec and SSL VPN capabilities
Optional built-in autonomous access point for wireless connectivity
469
Notice that voice gateway functions are not included. The 500 Series is a good option
when you don’t require the voice capabilities of the CUCM Express and Unity Express that
are found in the UC500. It also includes more advanced security capabilities, including IPS
functions that the UC500 does not support.
ESW 500 Series Switch
The Cisco ESW 500 Series switch lineup offers several hardware options for Fast or Gigabit
Ethernet connectivity for your end devices and servers. It also offers the capability to
connect to other switches using SFP uplink modules. Here is a list of features that the ESW
500 Series switch platforms offer:
Fast Ethernet and Gigabit Ethernet switch ports in 24 - or 48-port configurations
Power over Ethernet (PoE) for up to 48 ports of Fast Ethernet and 24 ports of Gigabit
Ethernet
QoS capabilities for Layer 2 traffic identification and tagging
Layer 2 security features such as IEEE 802.1X port security and access control lists
(ACLs)
VLAN configuration
Optional redundant power supply
Small Form-Factor Pluggable (SFP) expansion slots for uplinks to other network
devices
Expansion is the name of the game for the ESW 500 Series switches. Typically, an
office environment will include either the UC500 Series or the Secure Router 520 Series
hardware. Both of these devices have an integrated switch. If an office requires additional
switch ports, they’ll typically buy one of the ESW 500 Series switches to expand the
number of physical Ethernet ports available for use.
Cisco 500 Series Wireless Express
The SBCS lineup offers two different wireless options. One option is to purchase the
wireless component that is built into either the UC500 or SR500 Series hardware.
This provides a single autonomous access point. An autonomous access point (AAP)
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means that the wireless intelligence resides on the access point itself. A second way to
implement wireless is to purchase and implement the Cisco 500 Series wireless express
hardware, which consists of two pieces, the Cisco 521 and the optional 526 Wireless
Express Mobility Controller. First, the Cisco 521 is a stand-alone wireless access point
with the following capabilities:
802.11b/g functionality
Integrated antennas
Standards-based security
A Cisco 521 can run on its own in autonomous mode, and the Cisco Configuration
Assistant can manage up to three 521s in a single location.
If your wireless implementation may expand beyond three wireless hotspots at
a single location, you might want to consider the second piece of Cisco 500 Series
wireless express hardware/software, the 526 Wireless Express Mobility Controller.
When implemented, this device becomes the brain of your wireless network. The Cisco
521s are no longer considered autonomous. Instead, different software is used on the
521 hardware to make them “dumb” devices called lightweight access points (LWAPs).
All the configuration and maintenance are then performed at the 526 Wireless Express
Mobility Controller, and only basic radio and Ethernet transport functionality is
performed at the LWAP level. With this setup, you have the ability to control up to 12
LWAPs at a single location. The SBCS wireless controller also provides these additional
benefits:
A single point of wireless hardware and software management
The ability to monitor wireless coverage automatically and make real-time changes
to signal strength, gain, and wireless channel selection to optimize the wireless
network
Support for wireless mobility services to better support voice over wireless IP phones
such as the Cisco 7921 and 7925
It is important to know the entire SBCS suite of products not only for the CCNA Voice
exam but also to get an idea of the components available for designing and implementing
networks for commercial environments. Now we’re going to revisit the UC500 Series
platform in more detail to show you all the various options available when ordering and
setting up your voice network for small- to medium-size businesses.
Using the UC500 Series Platform
out of the Box
As mentioned earlier, one clear benefit the SBCS platform has over its competition is the
fact that straight out of the box, the devices are functional. No configuration is necessary
to provide basic capabilities. This Plug and Play functionality is also true for the UC500
Using the UC500 Series Platform out of the Box
471
Series. As soon as you plug in the UC520, it powers up and loads a default configuration
for voice and data usage. Also note that all of the licensing is already taken care of. Only
when you need to upgrade software or add additional user licenses or other capabilities
will you ever have to relicense the UC500.
The developers at Cisco made some assumptions about the default capabilities that
users of the UC500 Series platform would want to have. Here is a list of the preconfigured
features that you will fi nd on bootup of the UC520:
Separate voice and data VLANs.
DHCP server for voice (10.1.1.0/24) and data (192.168.10.0/24) VLANs.
Ethernet WAN port configured to receive IP address via DHCP for connection
to standard DSL, cable modem, or any other consumer/small-business
Internet service.
Network Address Translation (NAT) on WAN port.
Basic firewall access control list (ACL) protecting the inside network from the WAN
port and between VLANs.
HTTP and HTTPS GUI service setup.
TFTP server configuration for IP phones using option 150.
Basic FXS configuration of PLAR analog phones.
Basic dial-peer setup for off-network calling using FXO interfaces.
Telephony service set up to utilize CUCM Express in a key-system setup.
Multicast MoH setup using the default music-on-hold.au file stored in flash.
Auto -registration of phones enabled for Plug and Play setup of Cisco IP phones
for a single extension. IP phones receive extensions beginning with 201, and analog
phones receive assigned numbers beginning with 301.
Basic voice mail configuration.
With all of these features preconfigured, some businesses may not need to configure
the UC500 manually at all. It is highly recommend that you change at least the default
administrator password, however.
If you’ve already made modifications to your UC500, you can restore the
CUCM Express default configuration very easily. On the flash storage,
you should find a file listed with a name similar to UC520-8U-4FXO-K9factory-4.2.7.cfg. Depending on your hardware and factory-installed
software, this filename will be slightly different. Once you find the factory
default configuration, you can issue a copy flash: UC520-8U-4FXO-K9factory-4.2.7.cfg startup-config privileged EXEC command. Then
proceed to reboot your UC500 by issuing the reload command. Once the
UC500 reboots, your CUCME comes up with the default configuration.
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While the Plug and Play functionality is a nice option for quick or simple deployments,
you’ve made a substantial investment in the UC500 Series hardware, so you may as well
squeeze every useful feature you can out of it. In the next section, I’m going to introduce
you to the Cisco Configuration Assistant, a tool that greatly simplifies the configuration
and management of the SBCS platform. You will learn all about the software package and
then learn how to install and discover your UC500 Series environment so you can begin
the process of configuring the added features that aren’t preconfigured out of the box.
Introducing the Cisco
Configuration Assistant
Because the Smart Business Communications System is geared toward small- to mediumsize commercial environments, Cisco has anticipated that there may not be a highly skilled
network engineer on staff to configure and maintain complex networking equipment
using the command line or even the web GUI configuration tools. The SBCS offers a third
configuration and maintenance application called the Cisco Configuration Assistant (CCA)
that further simplifies the process to the point where you can configure and manage the
entire SBCS lineup using this single tool.
The Cisco Configuration Assistant application is simple to install and configure. This
section discusses the CCA application in detail to provide you with the requirements and
limitations of the tool.
CCA Requirements
This section details the system requirements for the CCA software to run on PC hardware.
It also lists the requirements of the devices that are to be managed within CCA.
CCA Software Requirements
You can find the CCA software on the included CD when you purchase any SBCS product.
If you no longer have the CD, you can download it for free from the cisco.com website,
provided you have a valid CCO account. The application is a “fat” client, meaning it runs
as an executable program on your Microsoft Windows PC. Here are the minimum system
requirements needed to run CCA version 2.0, the latest version as of the writing of this book:
Operating system: Windows XP Professional or Windows Vista Ultimate
PC processor: 1- GHz Pentium IV
Memory: 512 MB
Screen resolution: 1024 × 768
Disk space: 150 MB
LAN connectivity: Fast Ethernet
Introducing the Cisco Configuration Assistant
473
The CCA software also relies heavily on both Adobe Flash and Java to execute the
underlying CCA code. When you install the CCA application, both Flash and Java will be
installed on the PC if they are not already set up.
CCA-Managed Device Requirements
In addition to the SBCS lineup detailed earlier, the CCA can manage a handful of other
small-business Cisco devices. The CCA version 2.0 can currently manage the hardware
shown in Table 10.2.
TA B L E 1 0 . 2
CCA Version 2.0 Manageable Devices
Cisco Device
Model(s)
Routers
SBCS UC500Series
SBCS SR520 Series
Cisco 800 Series
Switches
SBCS ESW 500 Series
Catalyst Express 500 Series
Wireless
SBCS 500 Series Access Point
SBCS 500 Series Express Mobility Controller
CCA Limitations Per Site
There are limits to the number of devices that the CCA can manage per site. You can
support a maximum of 25 devices on a network. The types of devices managed are limited
as well. CCA can manage a maximum of the following devices:
5 Cisco UC500 Series or SR520 Series routers
15 ESW 500 Series switches
3 500 Series autonomous wireless access points
2 500 Series Wireless Express Mobility Controllers
Fortunately, these restrictions are a per-site limitation on the CCA. Within the CCA,
you have the ability to manage an unlimited number of sites with the same tool. Each site
then must abide by the maximum device limits.
Previous to version 2.0 of the CCA application, a “site” was called a
“community.” These two terms can be used interchangeably.
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Now that you have a better understanding of the capabilities and limitations of the CCA
software, we can now cover how to install and set up the CCA to support a site using a
UC500 Series router.
Setting Up CCA for Supporting
the UC500 Series Platform
In this section, we’ll go through the CCA software installation process and cover the main
user interface buttons of the CCA GUI. Finally, we’ll create a brand-new CCA site and take
it to the point where the CCA automatically discovers our lab network. Let’s get started.
Installing the CCA Software
Setting up the CCA software is very similar to installing any other Windows application.
The installation fi le comes as a Windows executable. Following are the steps you’ll go
through to install the application on Windows XP:
1.
Download the CCA version 2.x software, or insert the software CD that came with
your SBCS hardware. The easiest way to find the CCA software on Cisco’s website
is to go to http://www.cisco.com/go/configassist. You then click the Download
button on the right side of the screen. You will be required to log in to CCO to download the software.
2.
Locate the file labeled Cisco-config-assistant-win-k9-2_0-en.exe. This is your
installation file. It may look slightly different depending on the version you are about
to install.
3.
Double- click the executable and the installation process begins. The first thing the
InstallShield Wizard checks is to make sure you are running a compatible version of
Java and Flash.
4.
You will be presented with an end-user license agreement (EULA). You must accept the
EULA to continue the install process.
5.
The installer will ask you what physical directory you would like the software to be
installed in. By default, the location is C:\Program Files\Cisco Systems\CiscoSMB.
Figure 10.3 shows this part of the install process.
Either change the directory by clicking Browse and choosing an alternate location in
which to install the CCA software fi les or accept the defaults. When fi nished, click
the Next button. At this point the software will be installed on your PC. It may take
several minutes for this process to complete.
6.
When the installation is finished, you will be presented with the notification shown in
Figure 10.4.
Setting Up CCA for Supporting the UC500 Series Platform
F I G U R E 10. 3
Specifying the install directory location
F I G U R E 10. 4
Successful CCA installation
Click the Finish button to complete the installation process.
The installation will have created a shortcut on your desktop labeled “Cisco
Configuration Assistant.” The remainder of Chapter 10 will cover how to navigate the
application as well as how to set the CCA to discover a UC500 and any connected Cisco
IP phones. Chapter 11 will then cover the necessary steps to configure the UC500 system
using the Configuration Assistant.
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Navigating with the CCA User Interface
Before we start using CCA, let’s cover the CCA navigation bars and the functions of each
of the buttons. The two main navigation menus are the feature bar and the toolbar.
Figure 10.5 highlights the two main menu interfaces.
F I G U R E 10. 5
CCA GUI navigation bars
Toolbar
Feature bar
Let’s go over what each of the icons mean on these two navigation bars.
Understanding the Feature Bar Interface
The feature bar is vertical and by default resides along the left side of the screen.
Figure 10.6 shows the icons listed on the feature bar once a site has been discovered.
F I G U R E 10.6
The feature bar
Each button of the feature bar serves a different purpose. The bar shows the features
that can be configured for all of the devices you are managing at the site. Here’s a
breakdown of what can be done within each feature bar section:
Setting Up CCA for Supporting the UC500 Series Platform
477
Home Here you can fi nd access to the Dashboard, Topology, and Front Panel views. You
can also run the various setup wizards.
Configure Here you can manually configure routing, switching, security, telephony, and
other features of the SBCS lineup that is on your network. This is where an administrator
would commonly go to configure various options on the UC500 router.
Applications Here you can modify general site settings and configure setup options
for Smart Applications, which are optional applications on SBCS hardware. A Smart
Application example on the UC500 is Unified Messaging.
Monitor Here you can fi nd various monitoring tools and voice status reports.
Troubleshoot This button provides tools for troubleshooting network and voice problems.
Maintenance Here you’ll fi nd tools for maintaining the software of your SBCS equipment,
including software updates and license management.
Partners Connection This provides access to Cisco’s Small Business Support Community,
where you can fi nd product documentation, configuration information, and software
downloads.
Understanding the Toolbar Interface
The toolbar has icons that deal with the configuration, management, and monitoring of
your SBCS devices. Many of these buttons deal with the CCA application itself, whereas
others are duplicates of what is included in the feature bar but with graphical icons for ease
of understanding. Figure 10.7 displays the toolbar menu system with descriptions of each
icon button.
F I G U R E 10 .7
THE TOOLBAR
Connect
Refresh
Print
Preferences
Save
Config
VPN
Server
Voice
Smart
Ports
Firewall
and DMZ
Inventory
Port
Settings
Event
Notification
Health
Topolgy
Dashboard
Help
Front
Panel
Legend
Feedback
Let’s briefly break down what each icon represents from a setup, configuration, and
maintenance point of view.
Connect
Uses the CCA to connect to a different site or to an individual SBCS device.
Refresh Updates the CCA views with the latest information.
Print
Prints the currently active CCA window or help window.
Preferences
Modifies the default CCA display preferences.
Save Configuration Performs a copy running-config startup-config on the managed
device.
Voice
Provides configuration options for voice communication.
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Introducing the SBCS Platform
Provides configuration options to set up a virtual private network (VPN).
Firewall and DMZ Provides configuration options to set up fi rewall rules and to create a
network demilitarized zone (DMZ).
Smart Ports Allows you to configure various port security and management functions
based on Cisco suggested roles.
Port Settings
Provides View and Modify Port settings.
Inventory Displays the device hardware/software versions along with the management IP
address and other information that identifies the device.
Health Provides system measurements used to gauge the operational health of managed
devices. These measurements include bandwidth utilization, CPU utilization, memory
allocation, device temperature, and interface error statistics.
Event Notification Displays any event-driven notifications for all discovered devices. This
information can be useful when troubleshooting various problems. Events are considered to
be a triggered condition that occurs on CCA monitored devices that Cisco has determined
an administrator should know about. These events include:
Temperature that exceeds the recommended threshold
Fan malfunction
Port that was placed into administratively shutdown mode
FastEthernet port with a duplex mismatch
A monitored device that went into an “unknown” state
VLAN conflict
There are four different levels of event notification that correspond to syslog level types.
The lower the level type is, the more severe the alert. Within the Event Notification CCA
tool, Event Notification Types are defi ned as follows:
TA B L E 1 0 . 3
Event Notification Types
Syslog Level
Event Notification Type
0-1
Critical Error
2-3
Error
4
Warning
5-7
Informational
Setting Up CCA for Supporting the UC500 Series Platform
479
Dashboard Pulls up the Dashboard view, which is a great way to quickly display
information about the health of your network and attached devices.
Topology
Pulls up the Network Topology view for all discovered devices at a site.
Front Panel Displays a graphical representation of the physical front of your SBCS device.
This is great for checking the status of various ports and LEDs.
Legend Pulls up the Legend, which describes all icons, labels, and links available on
the CCA.
Help Pulls up the help utility for the active window, where you can search for
CCA-related information for configuration, monitoring, and maintenance assistance.
Feedback Pulls up a feedback page where you can leave feedback and suggestions
regarding the CCA tool. This information is then reviewed by Cisco so they can make
improvements to new versions of the CCA application.
Monitoring Your SBCS Equipment Using
the CCA Dashboard
A monitoring tool within CCA that is new to version 2.0 and above is the Dashboard. This
tool is designed to give the administrator quick and simple-to-understand monitoring tools that
show SBCS device system health displays using various GUI graphs and charts. Figure 10.8
shows the Dashboard displaying health information for a UC500 Series device.
F I G U R E 10. 8
The CCA Dashboard
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Now that you know the user interface a little better, we can go ahead and set up our fi rst
CCA-managed site.
Adding a New CCA Site
Figure 10.9 shows the initial screen you are presented with when you launch the CCA
application.
F I G U R E 10. 9
The Customer Sites tab is initially empty
The fi rst thing that we need to do to support an SBCS network is to add it by clicking the
Add A New Site button. Remember that each site, or community, is managed separately.
The CCA can manage multiple sites on the same application.
You should set your UC500 back to its default settings prior to adding it as
a new customer site in the CCA. This way, you can start with a clean slate,
using the default username/password for the administrator account (cisco/
cisco). You also have the default VLANs and IP ranges for the various
components.
A new window pops open, as shown in Figure 10.10.
Setting Up CCA for Supporting the UC500 Series Platform
F I G U R E 1 0 .1 0
481
Creating a new customer site
The next screen asks the administrator to fi ll out the following fields:
Site Name
Site Description
Discover
Devices Using A Seed IP Address
Devices On A Subnet
Devices On An IP Address Range
A Single Device By IP
The default (and recommended) network-discovery method is to enter a seed
IP address. The seed address is typically the heart or root of the network. In our example,
the seed IP address of our network is the UC500. When we enter the IP address of the
UC500 (192.168.10.1 by default) and click the Start button, the CCA will attempt to
connect to the system using either HTTP or HTTPS. That means you must have one of
these two services up and running to connect properly; otherwise, the discovery process
will fail.
Once the CCA software connects using HTTP or HTTPS, you will be prompted to
enter the proper administrator credentials for the UC500, as shown in Figure 10.11.
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Introducing the SBCS Platform
UC500 authentication
If you’re connecting to a UC500 that has the default configuration, the default username
and password are cisco/cisco.
As soon as the CCA software has been authenticated, a discovery process occurs.
This site discovery process uses the UC500 seed device to look for other CCA- compatible
devices such as ESW switches, IP phones, and other SCBS hardware. The term seed refers
to using the central source of a single device to branch out and fi nd other devices. But how
does the CCA actually discover these other attached devices? It uses the Cisco Discovery
Protocol (CDP). So another absolute requirement for proper CCA network visibility is that
CDP is running on the devices that you wish to discover!
It can take several minutes for the discovery process to complete. When the CCA
software is fi nished, it presents the user with a topology map of the seed device and any
other devices it found using CDP during the discovery process. Figure 10.12 shows our
small topology of a UC500 and two Cisco 7965 IP phones.
F I G U R E 1 0 .1 2
CCA topology
Make note of how the IP phones are considered “neighbors” of the seed UC500 device.
Exam Essentials
483
Summary
In Chapter 10 you learned the various hardware components that the SBCS lineup
comprises. Each SBCS hardware device serves a different purpose within a smallbusiness voice and data network. We then looked at the UC500 Series SBCS platform to
examine both its hardware and software capabilities. From a confi guration standpoint,
the UC500 can be used literally right out of the box for many basic voice and data
features. Last, the chapter introduced you to the Cisco Confi guration Assistant and
how it can configure and manage the different SBCS components from a simple -to -use
GUI application that runs on Microsoft Windows computers. Now you have a better
understanding of why the SBCS components that Cisco offers are an incredibly fl exible
hardware lineup both in their capabilities and in the way they are confi gured and
maintained. In the next chapter you’ll use the CCA to configure the UC500 platform’s
telephony functions in detail.
Exam Essentials
Know the different SBCS UC500 hardware options. The UC500 system provides voice
and data functionality for small businesses. Several different chassis and PSTN port
configurations are available. In addition, a voice expansion slot is available for additional
PSTN expandability.
Know the major SBCS hardware components. In addition to the UC500, the SBCS lineup
includes the Secure Router 500 Series, the ESW 500 Series Layer 2 switches, and the 500
Series wireless components.
Understand the UC500 Series preconfigured capabilities. The UC500 is preconfigured to
deliver voice and data services out of the box. This allows for extremely fast deployments
for environments that require only a basic implementation.
Know what the CCA software is used for and how to install it. The Cisco Configuration
Assistant is a Windows-based application that is used to simplify the configuration and
management of the SBCS lineup.
Know how to create and discover a new SBCS site using CCA. A site is a network that
utilizes SBCS hardware. You can use a seed address on the CCA to discover the SBCS
network for configuration and monitoring purposes.
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Written Lab 10.1
Write the answers to the following questions:
1.
The UC520 -16U comes in what chassis form factor?
2.
Which SBCS hardware supports dynamic routing protocols?
3.
What is the name of the feature that allows users to use analog phones connected to
the PSTN over FXO interfaces when the UC500 has no power?
4.
Which SBCS wireless component controls LWAPs?
5.
Which operating systems are required in order to use the CCA?
6.
Which two supplemental applications are required to be installed on the PC running
CCA?
7.
What is the maximum number of UC500 or SR500 Series SBCS devices that can be
managed at a single site?
8.
How many sites can be managed by CCA?
9.
Within the feature bar of the CCA, what button would you click to update the
software on your managed SBCS hardware?
10. On the toolbar of the CCA, what button would you click to display a graphical
representation of the SBCS system you are managing?
(The answers to Written Lab 10.1 can be found following the answers to the review
questions for this chapter.)
Review Questions
485
Review Questions
1.
What two UC500 models use an external power supply?
A. UC520 -8U
2.
B.
UC520 -16U
C.
UC520 -24U
D.
UC520 -32U
E.
UC520 - 48U
How many rack units (U) does the UC500 desktop chassis consume?
A. 1U
3.
B.
1.5U
C.
2U
D.
2.5U
What is the maximum number of fixed PRI interfaces that can be ordered on the desktop
form factor UC500 system?
A. 1
4.
B.
0
C.
2
D.
4
E.
8
What is the name of the UC500 port that is used to uplink to an Ethernet switch such as
the ESW 500 Series?
A. WAN Ethernet port
5.
B.
Compact flash slot
C.
Voice expansion slot
D.
FXO port
E.
LAN expansion port
What type of phone can be used when there is no power to a UC500 that is connected to
the PSTN using FXO ports?
A. Cisco 7900 Series IP phones
B.
Cisco 500 Series IP phones
C.
Cisco 7921 or 7925 wireless phones
D.
Analog phones
Chapter 10
486
6.
Introducing the SBCS Platform
Which UC500 models support an integrated wireless access point? Choose all that apply.
A. UC520 - 48U
7.
B.
UC520 -8U
C.
UC520 -32U
D.
UC520 -16U
E.
UC520 -24U
What SBCS series hardware supports dynamic routing protocols?
A. UC500 Series
8.
B.
ESW 500 Series
C.
SR500 Series
D.
500 Series Wireless Express Mobility Controller
If your site needs to manage more than three wireless access points, what additional device
is recommended?
A. 500 Series Express Wireless Mobility Controller
9.
B.
SR500
C.
ESW 500
D.
LWAP
When the intelligence of a wireless access point is moved from the access point to a
Wireless Express Mobility Controller, what term describes the access point?
A. Hot spot
B.
Integrated antenna
C.
Autonomous
D.
Controlled
E.
Lightweight
10. Which of the following is not a benefit of using SBCS 500 Series Wireless Express Mobility
Controllers?
A. Single point of management.
B.
Ability to monitor wireless coverage of multiple access points at one time.
C.
Cisco IP 7921 and 7925 phones can be used only on wireless designs that use the Wireless Express Mobility Controller.
D.
Support for wireless mobility services.
11. What part of the out- of-the-box UC500 Series configuration is recommended to be
changed?
A. Auto -registration
B.
Default administrator password
C.
Network Address Translation
D.
Default access control list (ACL)
Review Questions
487
12. Which of the following is not a system requirement for installing the CCA on
a Windows PC?
A. Processor 1GHz Pentium IV
B.
512 MB RAM
C.
150 MB disk space
D.
LAN or WLAN connectivity
13. Which of the following hardware cannot be configured or supported using CCA?
A. 800 Series router
B.
ESW 500 Series switch
C.
SR500 Series
D.
500 Series Express Mobility Controller
E.
1800 Series router
14. CCA requires which two additional applications to be installed on the Windows PC?
A. Flash
B.
JavaScript
C.
Java
D.
Silverlight
E.
SBCS
15. What is the maximum number of ESW 500 Series devices that can be supported using CCA
in a single site?
A. 2
B.
5
C.
10
D.
15
16. What are the names of the two main CCA navigation bars?
A. Feature bar
B.
CCA bar
C.
Wizard bar
D.
Toolbar
17. Which of the following is not a feature bar button in CCA?
A. Home
B.
Monitor
C.
Partners Connection
D.
Applications
E.
Wizard
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Introducing the SBCS Platform
18. Which of the following tasks can be achieved by using the Front Panel button on the CCA
toolbar?
A. Verify that the MoH lamp is lit on an IP phone
B.
View the network topology of the site
C.
Verify link up/down status of a Fast Ethernet port
D.
Reload the UC500 remotely.
19. What is the default method for a site discovery using CCA?
A. Single device by IP
B.
IP address range
C.
Seed IP
D.
Subnet range
20. When using the seed IP address CCA site- discovery method, what does the seed device use
to discover additional devices on the network to manage?
A. ICMP
B.
HTTP or HTTPS
C.
SSH
D.
Telnet
E.
CDP
Answers to Review Questions
489
Answers to Review Questions
1.
A, B. The two desktop chassis models of the UC520-8U and UC520-16U use an external
power brick.
2.
B. The desktop chassis form factor UC500 uses up 1.5U. Each rack unit is 1.75 inches.
3.
B. The desktop model of the UC500 does not support any fi xed PRI interfaces.
4.
E. The LAN expansion port is used to uplink to a switch such as the ESW 500 Series for
Ethernet port expandability.
5.
D. Only analog phones connected to the FXS interfaces can utilize the power failover
feature.
6.
B, D. Only the desktop UC500 chassis supports the integrated wireless access point.
7.
C. The Secure Router 500 Series hardware supports dynamic routing protocols. The
UC500 supports only static routes.
8.
A. The Wireless Express Mobility Controller is recommended for sites that
have more than three Cisco wireless access points when using the CCA. The CCA can
support only 3 autonomous APs at a single site. If the wireless APs are controlled by a
Wireless Express Mobility Controller, then the CCA can support up to 12 LWAPs at
a single site.
9.
E. When the intelligence of an access point resides at the Wireless Express Mobility Controller, the access point is referred to as a lightweight access point (LWAP).
10. C. The Cisco 7921 and 7925 wireless IP phones can be used with wireless access points
in both AAP and LWAP architectures. The Wireless Express Mobility Controller is not
required.
11. B. The default administrator password should be changed at minimum.
12. D. A Fast Ethernet connection is required to connect the PC to the network. A WLAN connection is not supported when using the CCA.
13. E. All of the devices are supported except for the 1800 Series routers.
14. A, C. Java and Adobe Flash are required on the Windows desktop. If they
are not installed on the system when CCA is installed, the CCA application installs them
for you.
15. D. Up to 15 ESW switches are supported within a single site using CCA.
16. A, D. The feature bar is found vertically along the left side of the screen, and the toolbar is
horizontal across the top of the application screen.
17. E. The Wizard is not a feature bar button within the CCA application.
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18. C. The Front Panel button of the CCA toolbar displays a graphical representation of SBCS
hardware such as the UC500. The only task that can be accomplished by using the Front
Panel button is to verify the up/down link status on a Fast Ethernet port.
19. C. The default method for CCA site discovery is to enter a single seed IP address.
20. E. The seed device uses CDP to discover additional devices that the
CCA can manage.
Answers to Written Lab 10.1
Answers to Written Lab 10.1
1.
Desktop
2.
Secure Router 500 Series
3.
Power failover feature
4.
Wireless Express Mobility Controller
5.
Microsoft Windows XP or Vista Ultimate
6.
Flash and Java
7.
Five
8.
Unlimited
9.
Maintenance
10. Front Panel
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Chapter
11
Configuring
Telephony Functions
Using the Cisco
Configuration
Assistant
THE FOLLOWING CCNA VOICE EXAM
OBJECTIVES ARE COVERED IN THIS
CHAPTER:
Implement UC500 using Cisco Configuration Assistant.
Describe the function and operation of Cisco Configuration
Assistant.
Configure UC500 device parameters.
Configure UC500 network parameters.
Configure UC500 dial plan and voice mail parameters.
Configure UC500 SIP trunk parameters.
Configure UC500 voice system features.
Configure UC500 user parameters.
In Chapter 10 you learned that the Cisco Configuration
Assistant (CCA) is a PC -based application that allows
administrators of small networks to configure and administer
various products within the Cisco SBCS lineup. In this chapter we’re going to focus on
configuration of the SBCS UC500 Series platform for voice functionality. The chapter will
start by running through the Telephony Initialization tool settings to prepare the UC500
for proper configuration using the CCA. The remainder of the chapter details the different
CCA voice- configuration options available to you. By the end of the chapter, you will have
a thorough understanding of setting up a UC500 for various voice capabilities using the
Cisco Configuration Assistant application.
Telephony Initialization
When you click any configuration option under Configure Telephony, a window pops
open labeled Telephony Initialization, as shown in Figure 11.1.
F I G U R E 11 .1
The Telephony Initialization window
Because you’re using the CCA for the fi rst time, you are given the option to set up your
CUCM to function as either a PBX or a key system. You can also choose the number of
digits your phone extensions will have. This field is auto -fi lled with 3, indicating that your
phone extensions will be three digits in length. Finally, you can optionally choose to add a
voice mail access extension (pilot) number. In our example, we used 700 for our voice mail
access extension. Click the OK button to continue.
At this point the CCA communicates with the CUCM and CUE to configure various
default settings. This typically takes several minutes to complete. Once it is fi nished, you
will receive a pop -up message that states “Voice system initialized.” Click OK to continue.
Now that our CCA has initialized our UC500, we can configure telephony functions
starting within the Configure Telephony portion of the Feature bar.
Configuring Telephony Voice Features Using CCA
495
Configuring the Telephony Region
Using CCA
The fi rst thing we’re going to configure is the telephony region where our UC500 resides
geographically, to match the telephone signaling and notification standards that users are
accustomed to. To reach the Telephony Region configuration area, use the Feature toolbar
and navigate to Configure Telephony Region. Figure 11.2 shows the Telephony Region
options available to us.
F I G U R E 11 . 2
The Telephony Region options
All of the configuration parameters displayed in the Region area should be familiar to
you by now. You can modify the UC500 to fit the language, call-processing tones, and
time/date formats of the local area of the user base.
Configuring Telephony Voice Features
Using CCA
The next Telephony configuration area we’re going to investigate comprises the voice
features. To configure voice telephony options, navigate to Configure Telephony Voice
in the Feature toolbar. When the voice configuration options open, you’ll notice several
tabs that segment the various voice features.
The next sections will cover the following tabs to show what you can configure within
the CCA:
System
Network
SIP Trunk
Voice Features
User Extensions
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Configuring Voice System Options
The System tab is a great place to view the hardware setup of your UC500 system.
During the CCA discovery process, the UC500 is analyzed and all hardware components
are detected. Figure 11.3 shows the layout of the System tab. Within the Hardware
Configuration section, you can see all of the built-in components and whether any module
slots are fi lled or empty.
F I G U R E 11 . 3
The Voice System options
As you can see, from a configuration standpoint, the only portion that can be modified
within the Setup tab is the System Message, which is the message displayed at the bottom
of the Cisco IP phone just above the softkeys. In Figure 11.3, we’ve changed our System
Message field to display “Hello World!”
Last, the System tab has a section labeled System Type Settings. This section shows the
administrator whether the UC500 is set up to function in PBX or keysystem mode and how
many digits per extension are used by default for the various phone and pilot extensions on
the system. Recall that during the CCA Telephony Initialization stage, we were required
to choose both the UC500 mode and the number of extension digits the CUCM Express
should use for auto -assignment of analog and IP phones. These choices cannot be modified
unless the administrator wants to reset the UC500 to factory defaults. That is why these
choices are grayed out within the System tab. It is important to emphasize that, because of
this lack of simple renumbering capability, you need to properly plan before installation.
The next tab we’re going to look at is Network, which configures the IP networking
capabilities for our voice VLAN on the UC500.
Configuring Voice Network Options
The Network tab within Voice Telephony configuration allows you to configure
IP addressing for the call-processing unit and IP phones. Figure 11.4 displays the IP
networking configuration options available.
Configuring Telephony Voice Features Using CCA
F I G U R E 11 . 4
497
The Voice Network options
The Voice VLAN section lets the administrator choose which configured VLAN to use
for IP phones. Remember that by default on the UC500, two VLANs are defi ned. One
VLAN is for data traffic and the other is for voice. CDP is used to automatically detect
Cisco IP phones. If a Cisco phone is detected, that phone will be placed into the VLAN
that is set here. If CDP is not enabled or not supported by the phone, the administrator will
have to configure the phones manually later on.
The voice DHCP scope is also configured here. You defi ne the IP network and
subnet mask. You can also configure excluded addresses so the DHCP service does not
accidentally hand out an IP address that is hard- coded, which would cause an IP address
confl ict. Last, you can set the Communications Manager Express IP address and subnet
mask. If you choose to configure this IP address to be in the same subnet as your DHCP
pool, as shown in the previous figure, make sure this IP address is one of the DHCPexcluded IP addresses.
Configuring SIP Trunk Options
The SIP Trunk tab is where you configure a trunk to another voice gateway or Internet
Telephony Service Provider (ITSP) connection. For ITSP configurations, the CCA greatly
simplifies the configuration process by providing SIP configuration templates for certified
ITSP providers. As of the writing of this book, the list of certified ITSP providers includes
the following:
AT&T
British Telecom (BT)
Broadview
Cbeyond
Covad
Fibernet
Nuvox
One Communications
PAETEC (McLeod)
XO Communications
Depending on the ITSP used, you will be required to fi ll out different information. For
example, Figure 11.5 shows the options required if you set up an SIP trunk with AT&T’s
ITSP voice services.
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F I G U R E 11 . 5
Configuring Telephony Functions Using the CCA
AT&T ITSP SIP trunk configuration
The Advantages of Using a Certified ITSP
A small Chicago-based business was tired of the expensive long-distance charges they
were experiencing with their PSTN provider. Because the organization recently had a
Cisco UC520 SBCS device installed, they wanted to pursue the option of setting up an
SIP trunk with a Cisco-certified ITSP. After doing some research, the company’s network
administrator chose Covad as the ITSP for the business.
After calling Covad and signing an ITSP contract for an SIP trunk with eight DIDs, the
network administrator was given detailed instructions on how to configure the UC520 to
connect to the SIP gateway at the other end of the Covad SIP trunk using the Covad SIP
Trunk template, as shown here:
Configuring Telephony Voice Features Using CCA
499
Setting up SIP trunks from Cisco-certified ITSPs is a snap because when you sign up
for one of these services, the ITSP tells you the exact information to enter into the
various SIP Trunk fields. It’s just a matter of plugging the information into the fields
and clicking the Apply button.
One additional SIP Trunk screen you should be aware of is the one used for all
noncertified ITSPs. The Generic SIP Trunk Provider option provides many more fields that
may be required by the non- certified ITSP. Figure 11.6 shows the SIP configuration fields
available.
F I G U R E 11 . 6
Generic SIP trunk configuration
Even though the Generic SIP Trunk Provider option might take some trial and error
by both you and the ITSP, it is still an available option so you are not solely locked into
choosing a certified service provider.
Configuring Voice Features Options
The Voice Features tab is a bit of a mish-mash of configuration options. Cisco has
determined that this grouping of features is commonly found in small to medium-size
environments and therefore has made them easily configurable here using the CCA.
Figure 11.7 displays the Voice Features tab layout.
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F I G U R E 11 . 7
Configuring Telephony Functions Using the CCA
Voice Features options
The tab is divided into four separate voice options. Let’s briefly go over each feature and
its configuration with CCA.
Call Park
Call parking is a very popular feature in small offices because it allows the user to
quickly put a caller on hold and resume the call from another phone on the system. For
example, let ’s say a clothing store clerk answers a call from a customer asking whether
the store has a specific pair of shoes in a size 8. The clerk needs to check the inventory in
the back of the store. Instead of placing the caller on hold, the clerk parks the caller. The
clerk then goes to the back of the store to check the inventory. The clerk can then pick
up a different phone located at the back of the store and dial the parking slot number to
resume the call.
You can set call park slots for the temporary holding of calls onto special extensions. In
Figure 11.7, we’ve enabled four call park slots with the extensions of 701 to 704.
Music on Hold
The Music On Hold section allows for a few modifications within the CCA application.
The Audio File drop -down menu allows you to select music fi les that are stored on the
UC500 flash in either a .wav or .au format.
Configuring Telephony Voice Features Using CCA
501
The Enable External Music On Hold Port check box allows you to enable or disable the
3.5mm jack that allows an external audio source such as a CD or MP3 player to be used
for MoH. When this box is checked, the music from the external jack takes precedence
over the audio fi le stored on the UC500 flash.
The Enable Music On Hold For Internal Calls check box allows you to enable or disable
MoH for calls that are on internal IP phones. When MoH is disabled for internal calls, any
call placed on hold will hear the hold beep signal every few seconds instead of music. This
saves UC500 resources such as memory and processor cycles that can be used for other
tasks and features.
Conferencing
The Conferencing section allows for two types of multi-party conference calls. Recall
that DSP resources are required for conferencing. Because of this, the CCA automatically
calculates the amount of available resources and sets limits for the number of conferences
that are allowed to be configured on the UC500. You can play around to balance the
number of sessions with the maximum number of participants and the codec mode to set
up conferencing as you see fit. Each of these three settings uses up a different amount of
DSP resources.
G.711-only mode is commonly used for deployments when conference
call participants are mostly on-network. Mixed mode (G.711/G.729) is
recommended for off-network participants that use an ITSP setup with
SIP trunking. If the ITSP supports G.729, then your calls will use less
bandwidth. Keep in mind that mixed mode uses more DSP resources than
G.711 mode alone.
Once you have these settings configured to allocate the various DSP resources, you can
choose between the two different types of conference- call sessions.
AdHoc conference calls let a user call one party and then call another so that all can
talk to one another. This is often referred to as three -way calling, but this term can be a
bit misleading, because often there are more than three parties in an AdHoc conference,
depending on the Maximum Participants Per Conference setting. In our example, we allow
up to eight parties per conference.
MeetMe conferences allow parties to dial an extension to “meet” others for a conference
call. This is also called a conference bridge. In our example, we’ve enabled a single MeetMe
conference bridge number of 771. Users who wish to establish a conference call on a Cisco
IP phone can press the MeetMe softkey, which becomes available as soon as this feature is
enabled. A confi rmation tone is then generated back through the phone handset to the user.
At this time, the user dials the 771 code to access the conference bridge.
Because we’ve set our codec to G711 and the maximum number of participants to 8, our
DSP resources allow us to have two simultaneous conference sessions. Using the slider bar,
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we’ve decided to allow for one AdHoc and one MeetMe conference session in the
example shown.
Night Service
The fi nal section of the Voice Features tab is Night Service. We’ve already gone over the
purpose of night service, when we discussed Call Forwarding options in Chapter 6. Just as
a reminder, night service allows for simple forwarding to a different extension (such as the
voice mail pilot number) based on the time of day. Because this feature is based on the time
of day, there is a notification statement in the configuration section stating that you must
tell the CCA what the office hours are. You will learn how to configure telephony schedules
later in this chapter.
After you’ve made the configuration changes for the various voice features you’ve
enabled, you can click the Apply button to save your changes. When you do this, you’ll
see an error message identifying errors that must be corrected before the CCA will make
any configuration changes to the UC500 system. Figure 11.8 shows the configuration error
message sent by the CCA.
F I G U R E 11 . 8
CCA configuration error notification
As you have seen throughout this book, many configuration features depend on the
proper configuration of other features. Some configurations serve as building blocks to
various features. The CCA has built-in intelligence to inform you of these dependencies.
The 10 errors that CCA is complaining about were flagged because we must fi rst fi ll out
user extension fields prior to enabling conferencing on the system. Let ’s look at the User
Extensions tab to correct the errors.
Configuring User Extensions Options
The User Extensions tab is where an administrator can add and delete analog and IP
phones on the UC500. There also is an import feature for bulk additions of phones using
standard .csv fi les.
During the initial seeding process, the CCA probed the UC500 and found that it had
four FXS analog ports and two Cisco 7965 IP phones attached to Ethernet interfaces.
Figure 11.9 shows that the CCA added the FXS ports and IP phones to the configuration.
In addition, it assigned the analog FXS port extensions between 301 and 304 and
extensions 201 and 202 for the Cisco phones. Generic fi rst/last names were also added
to the analog phone ports. As stated in the error notification, CCA is alerting us that we
must fi ll in the remaining fi rst/last names as well as assign a user ID to all phone devices
configured on the UC500.
Configuring Telephony Voice Features Using CCA
F I G U R E 11 . 9
User extension options
Let’s go ahead and assign user IDs to the four analog phones. Figure 11.10 shows the
CCA screen after adding these user IDs.
F I G U R E 11 .1 0
Adding analog phone user IDs
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In the Details section for the PhoneD analog phone port, you can see that we can
configure various call-blocking permissions as well as call-forward extensions. Also note
that as we begin to correct our “errors,” the error count in the alert drops. After we added
the four user IDs to the analog phones, the number of errors detected by CCA went from
10 to 6.
To appease the CCA application, let’s go ahead and configure usernames and IDs for the
two IP phone users to correct the fi nal six errors. Figure 11.11 shows the two users properly
configured.
F I G U R E 11 .11
Adding IP phone user IDs
In the Details section of the Cisco 7965 IP phone, we have the ability to set up the
phone button expansion module if we have one. We can also set up or modify any of the
six phone buttons that are standard on the 7965 phone. Also note that there is a button
to add any phones that the CCA did not detect initially when going through the discovery
process.
When all configurations are finished, click the Apply button to apply the changes to the
running configuration of the UC500. Next we’ll modify, configure, and view voice mail
features using the CCA.
Configuring Telephony Voice Mail Features Using CCA
505
Configuring Telephony Voice Mail
Features Using CCA
The third Telephony configuration area we’ll examine covers the voice mail options. To
configure voice mail options, navigate to Configure Telephony Voice Mail in the Feature
toolbar. You’ll notice that there are two tabs within the voice mail configuration options:
Setup and Mailboxes.
Obviously, since we’re working with voice mail, we’re dealing with Unity Express, which
is integrated into the UC500 hardware. We’re going to look at each of these tabs to see
what voic email configuration options are possible using the Cisco Configuration Assistant.
Voice Mail Setup Options
The Unity Express setup options available on this tab deal with systemwide settings.
Figure 11.12 displays the configuration options available to us.
F I G U R E 11 .1 2
Voice Mail Setup options
The Voice Mail Access Extension field is populated with extension 700. As you’ll recall,
when we initialized the telephony features within the UC500 using the CCA, we set the
access extension at that time. If you want to, you can modify this number here.
There is also a place to add a PSTN number so users could dial in and check messages
while they are off-network. This setting is optional. We added 5155557897 in our example.
Last, there are two check box options. The fi rst is to enable or disable VoiceView
Express so users can check their voice mail using softkeys on Cisco IP phones. The second
check box is labeled Live Reply. This feature allows the user to return calls to parties who
left voice mails directly from Unity Express. The alternative method of returning calls if
Live Reply is disabled is that the user disconnects (hangs up the phone) from Unity Express
and redials the number manually.
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Voice Mail Mailbox Options
The second Voice Mail tab allows us to modify the preconfigured mailboxes for each of the
extensions that were assigned in the User Extension tab of the Telephony Voice configuration
options. Figure 11.13 shows the Mailboxes tab with extension 201 highlighted to illustrate
the various configuration options available on the UC500 using CCA.
F I G U R E 11 .1 3
Voice Mail Mailboxes options
At the top of the screen is the Storage section. Here we can view the Unity Express
storage capabilities. We can see how much storage has been allocated compared to the
amount available. Capacity for Unity Express is expressed in minutes on a UC500.
Looking at user Jeff Thompson’s individual mailbox, we see that by default, the mailbox
is enabled. By unchecking the Select/De-select option, we can effectively disable the
mailbox if we choose. Other parameters that can be modified include extension number,
mailbox type, user ID assigned to the mailbox, and size of the mailbox itself. By default,
each mailbox will hold 12 minutes of recorded messages.
Configuring Telephony Phone
Groups Features Using CCA
The Phone Groups section is where we can configure various clusters of users to better handle
communication based on the grouping of users/extensions that have similar roles in the
organization. The following groups can be configured, modified, and deleted in this section:
Configuring Telephony Phone Groups Features Using CCA
Hunt groups
Paging groups
Pickup groups
507
To locate this configuration area, use the Feature bar to navigate to Configure Telephony
Phone Groups. Here’s a look at how each of these sections is laid out in the CCA.
Hunt Groups
You set up hunt groups by using the Feature bar and navigating to Configure Telephony
Phone Groups Hunt Groups. Up to 10 hunt groups can be defined using the CCA. As
a reminder, hunt groups use a pilot number that people dial to reach a group of people who
perform similar duties, such as a customer call center. Customer call centers can be ideal
situations for hunt groups because the caller is looking to speak not to a specific person but
rather to one of many people who are capable of servicing the caller’s needs. The CCA then
attempts to ring an extension that belongs to the hunt group based on various algorithms
that can be set, including sequential or longest-idle time. If the fi rst extension called does not
answer after a defi ned period, CCA attempts to ring the next extension in the hunt group.
In the next example, the three-digit pilot numbers are already defi ned, but you can
change them if you like. Figure 11.14 displays the Hunt Groups configuration screen.
F I G U R E 11 .1 4
Hunt Groups configuration
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By default, all the hunt groups are disabled. To enable a hunt group, check the Enable
check box located on the left. In this example, we’ve enabled hunt group pilot number 501.
We’ve also changed the Hunt Type setting to Longest-Idle Time. You can select hunt group
members from the Available section and click the right-arrow button to move them over
to the Selected section. Extensions 201, 202, and 301 have been added as members of this
group. Finally, we’ve set the No Answer Forward To drop -down list to ring extension 302
if none of the group members answers the call.
Paging Groups
To set up paging groups, use the Feature bar and navigate to Configure Telephony Phone Groups Paging Groups. The Paging Groups section is very similar in setup to
the options for configuring hunt groups. And again, the purpose of these paging groups
is to provide for a one-way broadcast of real-time voice to multiple group subscribers.
Figure 11.15 shows the CCA layout for this section.
F I G U R E 11 .1 5
Paging Groups configuration
The Enable check box allows you to enable up to four paging groups on the UC500 with
paging numbers that are predefi ned but can be modified. You can also describe each paging
group so that administrators can better keep track of the purpose of the group. Finally,
there is a listing of available phones and selected phones assigned to the paging group.
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Notice that only IP phones can be part of a paging group. Therefore, the analog ports on
our UC500 system are not included as Available choices.
Pickup Groups
Finally, you set up pickup groups in the Feature bar by navigating to Configure Telephony Phone Groups Pickup Groups. Users within a pickup group can use the
GPickUp softkey to answer any ringing phone that belongs to the same pickup group.
Again, this is a nice feature to have in call centers or places where multiple people perform
the same functions. Figure 11.16 displays the CCA layout for configuring pickup groups.
F I G U R E 11 .1 6
Pickup Groups configuration
Up to eight pickup groups can be configured. In the screenshot, three group members
have been added to pickup group 1. There is no enabling/disabling of the groups. To
activate a pickup group, you simply move IP or analog phones from the Available list to the
Selected list and click the Apply button.
That covers the different groups that can be set up within CCA version 2.0. Next, we’ll
explore how to set up business schedules including business hours, holidays, and nightservice hours.
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Configuring Telephony Schedules
Using CCA
The SBCS UC500 system needs to be instructed about business and holiday working hours
for two purposes. The fi rst is to play the correct Auto Attendant prompts. One standard
set of audio prompts can be played when the office is open, and another can be played to
inform callers that the office is closed. The same goes for holiday hours when the business
is closed.
The second reason for fi lling out a work schedule within the UC500 is to take advantage
of the Night Service tool, which is a convenient feature to automatically forward incoming
calls on an extension directly to voice mail when someone calls during non-working hours.
You can set these schedules using the CCA by using the Feature toolbar and navigating to
Configure Telephony Schedules. Three tabs set up scheduling, as shown here: Business
Hours, Night Service, and Holiday.
Let’s look at each of these tabs to see how to set up our office schedule for proper
functionality of both the AA and Night Service features.
Business Hours
The Business Hours tab lets you set the hours when the office is open and people are able
to take calls. Both of the standard AA scripts that come with the UC500 incorporate
the business hours functionality. Times can be configured at half-hour intervals. A
checked box means the office is open, while an unchecked box indicates closed times.
So, for example, if your office has working hours from 7:30 a.m. to 5:30 p.m. Monday
through Friday, then you would want to have the boxes checked between those hours
on the schedule. By default, all boxes are checked, indicating your office is open seven
days a week for all 24 hours of the day. You could manually go through and uncheck the
hours that the office is closed, but you’ll quickly see that it becomes time consuming and,
honestly, quite boring. To make the setting of working hours a bit easier, the CCA has a
widget to check or uncheck boxes based on time and day of the week. In Figure 11.17 I am
using the widget to uncheck the boxes between the hours of 17:30 (5:30 p.m.) and 24:00
(12:00 a.m.).
When you click the Update Table button, all of the half-hour time boxes on Monday
between those times become unchecked. To complete Monday’s closed hours you should
set the time from 00:00 to 08:30 using the time widget and click the Update Table button.
Also note that you can have up to four different business-hour schedules. To enable a
Configuring Telephony Schedules Using CCA
F I G U R E 11 .17
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The Business Hours tab
particular schedule, highlight it on the left and click the Enable Business Schedule check
box. Only one schedule can be enabled at a time, however, so you must fi rst uncheck the
schedule that is currently enabled before enabling a new schedule.
Night Service
The night service schedule goes hand-in-hand with the Night Service option described in
the “Configuring Voice Features Options” section of this chapter. Figure 11.18 displays the
Night Service Schedule layout.
By default, all seven days of the week are set up for night service between the hours
of 5:00 p.m. and 9:00 a.m. Since our office is open Monday through Friday from 8:30
to 5:30, we’ll need to modify these settings. We can use the night service schedule to set
the nonworking hours per day as 17:30 to 08:30 for Monday. We can then use the Copy
Selected Row To drop -down list to select Tuesday and apply Monday’s settings to Tuesday.
We can continue to do this for Wednesday, Thursday, and Friday. Finally, our office is not
open at all on Saturday and Sunday, so we can set our night service hours from 00:00 to
24:00 for Saturday and then highlight Saturday on the schedule and copy the hours over
to Sunday. Once this setup is complete, any extensions that have the Night Service feature
enabled will have calls immediately routed to voice mail (or any other configured extension
chosen) when the office is closed.
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The Night Service Schedule tab
Holiday
The Holiday tab allows you to configure business holidays for use by the Auto Attendant.
When these dates are set, the AA uses an alternate “holiday” greeting, informing callers that
the office is closed for that holiday. Figure 11.19 displays the Holiday tab and its features.
F I G U R E 11 .1 9
The Holiday Schedule tab
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By default, no holidays are configured on the UC500 when using the CCA. If you want
to add holidays, click the Add button, and the window shown in Figure 11.20 opens.
F I G U R E 11 . 2 0
The Add A Holiday window
You can click on the calendar icon to select the holiday you wish to configure. You can
then add a description to the holiday. In our example, we’ve configured July 4, 2009, and
labeled it Independence Day. Keep in mind that because some holidays do not fall on the
same calendar date every year, you’ll have to manually configure holidays for each calendar
year. The American holiday of Thanksgiving, for example, falls on the last Thursday in
November of each year. Because the calendar date changes, you must configure holidays
every year. The CCA allows you to configure holidays for the current year and one
additional year out. The writing of this book occurred in 2009, so the CCA gave us the
ability to configure holidays for 2009 and 2010.
Configuring Telephony Auto Attendant
Features Using CCA
You can fi nd the Auto Attendant configuration features within CCA by using the Feature
toolbar to navigate to Configure Telephony Auto Attendant. When the window opens,
you see three tabs for configuration of different AA options within CCA: Auto Attendant,
Prompt Management, and Script Management.
Let’s go over these tabs to see what the configuration screens look like.
Auto Attendant
When you initially look at the Auto Attendant tab options, there are only three radio
buttons to choose from, defi ning the mode you want for your auto attendant. The fi rst
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mode is to turn Auto Attendant off, the second choice is to use the AA in standard mode,
and the third is to use the AA in the more advanced multi-level mode. The difference
between standard and multi-level mode is that standard has a single menu system whereas
multi-level mode uses a more complex menu system for navigation. By default, the Auto
Attendant mode is set to Off. Figure 11.21 shows the options available to you in standard
AA mode:
F I G U R E 11 . 2 1
Auto Attendant Standard mode settings
As you can see, the CCA allows you to configure all the standard AA options you’ve
learned how to set using the Unity Express web GUI. Among the options to point out,
the AA Extension number is the pilot extension to dial to use the AA. You also have a
section to add the PSTN number for off-network calls to come into the AA. There also is
an option to choose the business hour schedule you want to implement. Finally, you can
configure dial pad key mappings so that when users press the digit on their handset, they
move the AA script forward. For example, we can set the number 1 on the keypad to dial
extension 201. The audio menu prompt will then have to be modified to tell the user to
press 1 for Jeff Thompson, who resides at extension 201. So just how can we change the
Configuring Telephony Auto Attendant Features Using CCA
515
various prompts used by the AA? That question is answered in the next Auto Attendant
tab, Prompt Management.
Prompt Management
The Prompt Management tab is where you can view, modify, add, or delete AA menu
prompts. Figure 11.22 shows the Prompt Management tab configuration options.
F I G U R E 11 . 2 2
The Prompt Management tab
As you can see, five default AA prompts are available. You can choose to modify these
default prompts or add new prompts if you desire. Either way, there are two different
prompt-recording options using the CCA. The fi rst method is to use the integrated CCA
Sound Recorder option. This allows you to use your PC microphone to record the prompts
and have them saved to the local drive of the PC where you have CCA running. Then, when
you click the Apply button, CCA uses its built-in FTP service to transfer the new prompt
fi le to the compact flash of the UC500. Figure 11.23 shows the recording window when you
click the Open button next to Record Prompts Using Sound Recorder.
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The CCA Auto Attendant Prompt Sound Recorder
The recorder has all the standard Record, Pause, Play, and Stop buttons, as you would
expect. Also note the limit of 60 seconds per prompt. When you save the prompt fi les for
new recordings, make sure you are descriptive with the fi lenames so you can easily recall
what the prompts are used for.
The second prompt-recording option is to enter a prompt-recording extension for
users to dial in to record their own AA prompts using the same AvT functionality
within Unity Express. We need to fi rst assign at least one member to the AvT group.
To do this within the CCA, we select the users we want to add as AvT users within the
Prompt Administrators section. As soon as we’ve assigned our prompt administrator
and applied the changes, we can use the prompt administrator’s phone to dial the AA
prompt-recording extension and use AvT to record, play back, and save our new
AA prompts.
Script Management
Auto Attendant scripts were detailed in Chapter 8, so we won’t go over their purpose here
again in great detail. But as you will recall, we discussed three different methods to develop
scripts. First, the administrator could use one of the two predefi ned scripts within Unity
Express. The second and third methods are useful if an administrator needs to create their
own customized scripts. The administrator can create scripts using the Editor Express tool
that is built into Unity Express. Or, for even more scripting options, they can download
and install the custom Script Editor application for Microsoft Windows. The Script
Management tab within CCA allows users to upload, rename, or delete AA prompts. The
CCA does not provide any way to create or modify the prompts, however. Figure 11.24
displays the Script Management tab with the predefi ned scripts available for use.
Configuring Telephony Dial Plans Using CCA
F I G U R E 11 . 2 4
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The AA Script Management tab
Please note that a maximum of 12 scripts can be stored on the UC500 at one time.
You now have a good understanding of the AA configuration options available within
the Cisco Configuration Assistant. Next, you’ll learn how to configure dial plans for both
incoming and outgoing calls.
Configuring Telephony Dial Plans
Using CCA
Incoming and outgoing dial peers can easily be configured using the CCA. To do so, use
the Feature toolbar and go to Configure Telephony Dial Plans. You will see two
sections, conveniently labeled Incoming and Outgoing.
Let’s look at both of these configuration sections to see how we can create dial plans for
inbound and outbound calls on our UC500 system.
Creating an Incoming Dial Plan
When we navigate to Configure Telephony Dial Plans Incoming, we see two
configuration tabs: Incoming FXO Calls and Direct Dialing.
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The Incoming FXO Calls tab is used to configure inbound rules for use by any idle
FXO ports that are connected to the PSTN. The Direct Dialing tab is used for the creation
of translation rules to map incoming DID numbers from digital PSTN trunks to internal
extension numbers configured on the UC500 system. Let’s take a closer look at the
incoming dial-peer configuration options available to us using the CCA.
Incoming FXO Calls
If your UC500 comes equipped with FXO ports, the incoming dial peers are defi ned here.
By default, each FXO port is assigned to extensions beginning with 201, as shown in
Figure 11.25.
F I G U R E 11 . 2 5
The Incoming FXO Calls tab
Why are the ports configured to forward to extensions beginning with 201? As you’ll
recall, the UC500 default auto -assign configuration sets IP phones with three-digit
extensions beginning with extension 201. Because we have two 7965 phones configured,
those phones use extensions 201 and 202. That means that if we have all four FXO ports
configured, calls coming into 0/1/0 and 0/1/1 would be forwarded to the 7965 phones
Configuring Telephony Dial Plans Using CCA
519
at extension 201 and 202, respectively. Calls that come into FXO ports 0/1/2 and 0/1/3
would receive a busy signal, because these extensions do not have an ephone -DN associated
with them. To remedy this, we can change the ports to forward to 201 or 202. An even
better idea would be to set all four FXO ports to dial the Auto Attendant extension of
700. That way, all external calls will fi rst hit the AA to be directed to the intended person
without any human interaction. To do this, you can manually enter the extension 700 into
the Destination field or, better yet, use the Destination Type drop -down menu to choose
AUTO_ATTENDANT. Once you do this, the destination extension is automatically set to
the AA extension. Several Destination Type options are available for the administrator
to choose from. Table 11.1 describes the most commonly used Destination Type options.
TA B L E 11 .1
FXO Destination Type options
Destination Type
Description
CO_LINE
Used for key system–configured UC500 systems
OPERATOR
Manually defines an extension to forward to using the
Destination field
AUTO_ATTENDANT
Forwards to the configured AA extension
HUNT_GROUP
Forwards to a configured hunt group extension
Now that our FXO ports are configured, we can move on to discuss other inward
direct-dialing configurations, using the Direct Dialing tab.
Direct Dialing
Chapters 3 and 7 made us well aware that for the PSTN, more options are available than
just FXO ports. The Incoming FXO Calls tab helped us set up incoming dial peers for our
four FXO ports, but what if we have a T1/E1, SIP, or ISDN BRI connection to our PSTN?
The Direct Dialing tab is used to configure dial peers for these digital circuits. Figure 11.26
shows the CCA format for the Direct Dialing configuration tab.
Notice that the configuration options are grouped into two distinct sections. One
section is called Direct Dial To Internal User Extensions. This essentially lets you create
a one-to - one mapping of a digital line to an internal extension. For ease of configuration,
the CCA allows you to group DID and extension ranges together in a single dial peer. Of
course, this can work only if both your DIDs and internal extensions are contiguous.
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The Direct Dialing tab
The second Direct Dialing tab section is Direct Dial To Auto-Attendant, Groups,
Operator. This lets you direct digital circuits with DIDs to predefi ned AA, hunt group, or
operator extensions. Again, if your DID ranges are contiguous, you can add multiple DIDs
to a single dial-peer rule.
Creating an Outgoing Dial Plan
When we speak of off-network calling, we typically are referring to outbound calls to
the PSTN using one or more PSTN interfaces such as FXO, ISDN, T1/E1, or even an IP
WAN connection to an ITSP using SIP. In order to route calls out of these PSTN interfaces
properly, we must create rules that trigger an off-network call when specific digit- entry
conditions are matched. A very common method of implementing this trigger is to have
all calls intended to reach someone off-network begin with the number 9. A prefi x digit to
trigger outgoing calls is called an access code. For example, if a user on our UC500 system
wishes to make a local call to 555-8934, the person would dial 95558934. The prefi x 9
would strictly be to let the UC500 know that the user wishes to make an off-network call.
This section shows how to use the CCA to set up outgoing dial plans on the UC500. To
configure these dial plans, use the Feature toolbar on the left and navigate to Configure Telephony Dial Plans Outgoing. You’ll fi nd three outgoing dial plan configuration
tabs: Outgoing Call Handling, PSTN Trunk Groups, and Caller Id.
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521
Let’s review each of these outgoing dial plan tabs to understand what can be configured
using the CCA.
Outgoing Call Handling
The CCA software includes prebuilt outgoing dial plan templates to greatly simplify
standard dial plans for many countries or regions around the globe. The software
predefi nes a dial plan that commonly suits the needs of most small businesses. Figure 11.27
displays the Outgoing Call Handling tab with the NANP template selected.
F I G U R E 11 . 2 7
The Outgoing Call Handling tab showing the North American template
You can delete or modify any of the dial-plan rules in the template to meet your
specifi c dial-plan requirements. You can also add rules using the Add Number template.
You also have the ability to import .csv fi les or export the template for use on other
systems if you choose. Also note that the default access code of 9 is chosen for this
template by default, and the collection timeout for any dial patterns that use the
T wildcard is 5 seconds.
The Trunk Priority column in the Outgoing Call Handling tab is where you can set a
priority for the different PSTN trunk lines you have installed on the UC500. Here is a list
of the Trunk Priority settings:
PSTN Only
SIP Only
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PSTN Then SIP
SIP Then PSTN
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Configuring Telephony Functions Using the CCA
So, for example, if your company has both an SIP trunk and standard PSTN lines for
off-network calling, it is likely that the SIP trunk calls will cost less. Therefore, you should
choose the SIP Then PSTN trunk priority for all of your long-distance dial-plan rules. That
way, the UC500 will route all calls over the lower- cost SIP trunk unless it is down. If the
SIP trunk is unavailable, the UC500 will route calls over the PSTN instead. This also gives
you site redundancy for off-network dialing.
PSTN Trunk Groups
When an on-network caller dials an off-network number, that call is matched by an
outgoing dial plan. When a match is made, the voice gateway has the responsibility to
choose one of the open PSTN lines for the outbound call. If the office has a single FXO line
to the PSTN, the choice is obvious because there are no alternatives. But what if the site has
two or more FXO lines and they both happen to be idle? Or what happens if you have a
23 - channel T1 PRI and 19 of those channels are idle? How does the voice gateway choose
which line to use next? The PSTN Trunk Groups tab allows you to create various trunk
groups, and you can then choose from a list of hunt schemes how the UC500 will choose
a phone line or channel to use. Figure 11.28 shows an example of a UC500 with a single
group, labeled ALL_FXO, which includes four FXO lines.
F I G U R E 11 . 2 8
The PSTN Trunk Groups tab
Summary
523
The hunt scheme that the ALL_FXO group is using is longest-idle. Here are all of the
hunt schemes available for use:
Longest-idle
Round-robin
Sequential
Random
Least-idle
Any PSTN connections that the UC500 supports (FXO, ISDN BRI, T1/E1, SIP) can be
configured into trunk groups. Each group can utilize a different hunt scheme if you choose.
Caller Id
The Caller Id tab is used to assign a standard telephone number for all outbound calls to
the PSTN when using digital circuits such as ISDN or T1/E1 lines. That means that a single
main PSTN number will used for outgoing calls, which will show up on the called party’s
caller ID display.
Remember that the Caller Id section is used only for digital circuits. Analog
PSTN lines such as FXO ports can use only the assigned PSTN number
defined by your provider for external caller ID.
Your choices for the main PSTN number are as follows:
None
Automated Attendant
Other (Enter any PSTN number you choose)
There also is an option to set up a Caller ID blocking code. When it is activated, users
can punch in this four-digit passcode (which must begin with an *) and then the PSTN
number they wish to call. When the call is made, the caller ID information is still sent,
but the sent caller ID number will be changed to say “Restricted” instead of an actual
telephone number.
Last, the Caller Id tab has the ability to translate DIDs that were configured within the
Direct Dialing tab in the Incoming Dial Plan section of the Feature toolbar. For each DID,
you can choose whether to display the main PSTN number or the more unique DID of the
extension that is making the call.
Summary
The UC500 is the sole SBCS device that provides voice functionality in the form of CUCM
Express and Unity Express. It is very important for CCNA Voice students to familiarize
themselves with the various UC500 configuration options available when using the CCA.
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In this fi nal chapter of the book, you have learned how to initialize the UC500 to prepare
the device for setup using the CCA. Many of the most critical voice and voice mail features
can be configured within the CCA Voice and Voice Mail sections. A section called Groups
is where you can configure various hunt, paging, and pickup groups by placing voice users
into different group categories. You then learned the two purposes for defi ning officehour and holiday schedules and how these can be used by both the Auto Attendant and
Night Service features. Last, the chapter showed how to defi ne the different incoming and
outgoing dial plans for off-network call routing on the UC500.
Exam Essentials
Know how to use the Telephony Initialization tool within CCA. The CCA has a tool
that is specifically designed to prepare the UC500 hardware to be configured using the
CCA. Within this tool, you choose to set your UC500 up as either a PBX or a key system.
You also determine the extension digit length and voice mail pilot extension if you want to
enable Unity Express functions.
Know how to configure Telephony Region settings within CCA. The CCA has a
Telephony Region section, where you defi ne the region the UC500 will be located in for
signaling and telephone-notification options found in different parts of the world.
Know how to configure Voice features within CCA. The Telephony Voice section is
where you can configure system, IP network, SIP trunking, popular voice features, and
user options.
Know how to configure Voice Mail features within CCA. The CCA contains a
Voice Mail section where you can configure global Unity Express settings such as pilot
numbers and whether to enable features such as VoiceView Express and Live Reply. In this
section, you can also modify mailboxes that were preconfigured based on the extensions
that were configured in the Telephony Voice configuration section.
Know how to configure Phone Group features within CCA. Hunt groups, paging groups,
and pickup groups can be configured using CCA. In its Phone Groups section you can
globally enable, add, delete, and modify groups as you see fit.
Understand why telephony schedules are important within the UC500 and how to
configure them within CCA. The UC500 needs to be aware of office hours to properly
utilize the Auto Attendant office open/closed and holiday prompts. Another reason is
to utilize the Night Service functionality. Three separate tabs within the CCA allow you to
configure hours and dates for the office being open and night service hours as well as
holidays when the office is closed.
Know the AA configuration capabilities that CCA offers. CCA has several configurable
options for Auto Attendant functionality. Within the Auto Attendant CCA section, you can
enable AA for either standard script functionality or using a multi-level mode. In addition,
Written Lab 11.1
525
you have several methods to create and manage AA voice prompts. Finally, you can manage
custom scripts within CCA. Keep in mind, however, that the CCA does not offer any way
to create custom scripts from scratch.
Know how to configure dial plans within CCA. CCA provides separate Telephony
configuration sections to configure incoming and outgoing dial plans. The Incoming dial
plans section allows you to set up dial plans for analog and digital PSTN. Here you can
determine how PSTN calls are handled when they arrive at the UC500. Outgoing dial
plans deal with outgoing call handling, PSTN trunk groups, and caller ID information that
can be controlled locally on the UC500 system.
Written Lab 11.1
Write the answers to the following questions:
1.
When running through the CCA Telephony Initialization tool, what are the two
CUCM model options available?
2.
If you are going to configure a UC500 outside the United States, to what CCA
Telephony section would you navigate to change the signaling and notification settings
on the system?
3.
What Telephony Voice configuration tab would you use to configure VLAN
information on the UC500 within the CCA?
4.
What Telephony Voice configuration tab would you navigate to if you wanted to set up
a PSTN connection to an ITSP?
5.
What SIP template drop -down choice would you select if you wanted to use a
noncertified ITSP?
6.
When configuring conferencing features on the CCA, which available codec mode uses
the least amount of DSP resources?
7.
When configuring the Night Service feature within CCA, what additional step are you
notified to perform in order for Night Service to function properly?
8.
What Unity Express feature that can be enabled using the CCA allows users who dial
into their mailbox to automatically return calls from parties through the UC500?
9.
What three groups can be configured on a UC500 using CCA?
10. When you use the CCA version 2.0 and navigate to Configure Telephony Schedules, what three tabs do you see?
(The answers to Written Lab 11.1 can be found following the answers to the review
questions for this chapter.)
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Review Questions
1.
When configuring global Telephony Initialization settings using CCA, what two system
mode options are available for you to choose between?
A. Key system
2.
B.
Hybrid
C.
Express
D.
Advanced
E.
PBX
Within the CCA, where can you modify region settings on the UC500?
A. Configure Voice Region
3.
B.
Setup Voice Region
C.
Configure Telephony Region
D.
Setup Telephony Region
When configuring Voice features by using CCA, what tab would you use to configure a
connection to an ITSP on a UC500?
A. User Extensions
4.
B.
SIP Trunk
C.
Voice Features
D.
Network
E.
System
You wish to modify the voice VLAN ID used on your UC500. What Voice configuration
tab would you use to modify this setting?
A. Analog Extensions
5.
B.
Setup
C.
Voice Features
D.
Network
E.
System
Using the CCA, you want to specify a DHCP pool of IP phones between 192.168.1.100
and 192.168.1.254. How would you accomplish this? Choose all that apply.
A. Set the DHCP pool to 192.168.1.0 and the subnet mask to 255.255.255.0.
B.
Set the DHCP pool to 192.168.1.0 and the subnet mask to 255.255.255.128.
C.
Set the Excluded parameters to be from 192.168.1.100 to 192.168.1.254.
D.
Set the Excluded parameters to be from 192.168.1.1 to 192.168.1.99.
Review Questions
6.
527
What determines the maximum number of simultaneous AdHoc and MeetMe sessions
available on a UC500?
A. Available CPU resources
7.
B.
Available system memory
C.
Number of DSP resources
D.
Number of codec resources
What conference type requires users to dial a unique extension to join the call with other
users?
A. MeetMe
8.
B.
Mixed Mode
C.
AdHoc
D.
DSP
What file type can be used to import multiple voice users at once using the Import button
within CCA?
A. .doc
9.
B.
.pdf
C.
.csv
D.
.exe
E.
.dat
During the Telephony Initialization process, the CCA detects what type of ports and
assigns generic first and last names to them?
A. FXS
B.
FXO
C.
PRI
D.
BRI
E.
VoIP
F.
SIP
10. If you have made configuration errors, how does the CCA notify you?
A. Any fields that have errors are bordered in green.
B.
Any fields that have errors are bordered in red.
C.
A pop -up window identifies any errors that need correcting.
D.
A new error tab is created that identifies any errors that need correcting.
Chapter 11
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11. What is the purpose of configuring a voice mail access PSTN number when configuring
Unity Express Voice Mail features within the CCA?
A. It enables on-network users to dial in to check their messages.
B.
It enables off-network users to dial in to check their messages.
C.
It enables on-network users to use Live Reply.
D.
It enables off-network users to use Live Reply.
12. What are the three CUCM Express phone groups that can be configured using CCA?
A. Call group
B.
Conference group
C.
Paging group
D.
Hunt group
E.
Pickup group
13. How many years out can an administrator configure holidays for using the CCA?
A. One
B.
Unlimited
C.
Zero
D.
Two
E.
Four
14. Which Telephony Schedule tabs within the CCA have configurations that are often used by
Auto Attendant scripts? Choose all that apply.
A. Business Hours
B.
Night Service
C.
Holiday
D.
Call Forward
15. When configuring Auto Attendant features using CCA, what tab would you navigate to in
order to select the AA script you want to use?
A. Holiday Schedule
B.
Auto Attendant
C.
Script Management
D.
Prompt Management
16. What are the three Auto Attendant modes within CCA that an administrator can choose
from?
A. Off
B.
On
C.
Standard
D.
Enable
E.
Multi-level
Review Questions
529
17. An administrator needs to record a new AA menu. What CCA Auto Attendant tab would
they navigate to in order to accomplish this task?
A. Auto Attendant
B.
Prompt Management
C.
Recording Management
D.
Script Management
18. Which of the following is not an FXO Destination Type option when configuring incoming
FXO calls using the CCA?
A. AUTO_ATTENDANT
B.
OPERATOR
C.
EMERGENCY_SERVICES
D.
HUNT_GROUP
E.
CO_LINE
19. What three options can be set up or modified on a UC500 using the CCA under the
Telephony configuration section of the Feature bar?
A. Voice VLAN
B.
Voice and Data VLANs
C.
Access Control Lists
D.
ITSP SIP Trunk
E.
Paging Groups
20. When configuring outgoing call handling on your UC500 using the CCA, what is the
purpose of the default access code?
A. An administrator password used to access the TUI
B.
Individual user codes for remote voice mail access
C.
A code to inform the UC500 that you wish to make an off-network call
D.
An administrator password used to modify the AA script
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Answers to Review Questions
1.
A, E. The two Telephony Initialization system mode options are PBX and key system.
2.
C. All CCA configuration parameters specific to the UC500 are found within Configure Telephony on the Feature toolbar.
3.
B. The UC500 has templates for certified Internet Telephony Service Providers (ITSP).
There also is a configuration template for generic ITSPs. These templates can be found
within the SIP Trunk tab when configuring Voice features using CCA.
4.
D. When using the CCA to configure Voice features of the UC500, you would navigate to
the Network tab to modify the voice VLAN.
5.
A, D. The best option is to use a /24 subnet mask. You will then need to exclude the
addresses from 1 to 99 so they are not handed out by the DHCP service.
6.
C. DSP resources are hardware resources that ultimately determine the number of simultaneous conference sessions possible.
7.
A. The MeetMe conference call type uses an extension that is dialed to meet other members who can join the call.
8.
C. The CCA supports .csv fi les for bulk adds of voice users.
9.
A. Generic fi rst and last names are assigned to all FXS ports that are found during the
Telephony Initialization process.
10. B. Errors are highlighted with a red border. These errors must be corrected before the CCA
applies the changes.
11. B. This configuration setting lets you defi ne a public PSTN number that users can dial
when off-network so they can remotely check voice mail messages.
12. C, D, E. The three phone groups that can be configured within CCA are hunt, paging, and
pickup.
13. D. You can configure holidays for the current year and one additional year out.
14. A, C. Both the Business Hours and Holiday schedule tabs contain configurations that can
be used by AA scripts to play different voice messages depending on whether the office is
open or closed. The Night Service tab configurations are not used by AA scripts because it
is designed to immediately forward calls of individual user extensions to
voice mail or an operator number without having that extension ring.
15. B. The Auto Attendant tab lets administrators choose the AA script they want to use by
selecting one from a drop-down menu that lists all scripts on the UC500.
16. B, C, E. The three choices for AA script modes are Off, which disables AA on the system;
Standard, which is a single-level AA script; and Multi-level, for a tiered AA script.
Answers to Review Questions
531
17. B. Administrators can view, modify, add, or delete AA menu prompts on the Prompt Management tab.
18. C. All of the Destination Type options are valid except for EMERGENCY_SERVICES.
19. A, D, E . Within the Telephony configuration section of the Feature bar, you can modify
the voice VLAN, set up an ITSP SIP trunk, and configure paging groups. The other two
options can also be configured using the CCA but not within the Telephony section.
20. C. The default access code is used to trigger an off-network call. By default, this number is
set to 9.
Chapter 11
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Configuring Telephony Functions Using the CCA
Answers to Written Lab 11.1
1.
PBX and key system
2.
Region
3.
Network
4.
SIP Trunk
5.
Generic SIP Trunk Provider
6.
G.711
7.
Configure the Night Service schedule
8.
Live Reply
9.
Hunt groups, paging groups, and pickup groups
10. Business Hours, Night Service, and Holiday
Appendix
A
Design and
Configuration Using
the CCA Telephony
Setup Wizard
According to the U.S. Small Business Administration, more
than 600,000 new small businesses are started each year.
Most of these businesses will require voice and data solutions,
for which the SBCS lineup is perfectly tailored. One of the beauties of working on the
design and build- out of a brand-new SBCS site is that you are free of having to deal with
currently implemented voice and data systems. In engineering circles, a project that is brand
new and lacks any constraints imposed by prior network equipment or designs is referred
to as a “greenfield” project. New businesses that are just starting up and need a voice
and data network give the design engineer the opportunity to design and configure a
voice/data network as they see fit.
The Telephony Setup Wizard is an alternative configuration tool within CCA that strives
to simplify basic voice and data provisioning of greenfield sites for small offices using the
UC500 system. Cisco clearly is of the opinion that there is more than one way to skin a cat.
An administrator can configure the UC500 using the command line, web GUI, CCA, or
the Telephony Setup Wizard within CCA. Think of the Telephony Setup Wizard as training
wheels for the CCA.
This appendix begins with an overview of the Telephony Setup Wizard and the
requirements for its use. We’ll then go over a mock greenfield case study, using a fictional
business to help understand the types of questions you will need to ask the business owners
when designing a voice and data network. Once we have compiled the information, you
will learn how to use the Telephony Setup Wizard to rapidly implement a basic UC500
system. The wizard does not offer anything new from a configuration standpoint; rather, it
gives you an additional method to configure your UC500 hardware.
CCA Telephony Setup Wizard
Overview and Requirements
The CCA Telephony Setup Wizard (TSW) is a relatively new tool that became available in
CCA version 2.0. This tool is intended for brand-new deployments with up to 24 users. It is
used only for brand-new (greenfield) configurations. When you connect to a UC500 using
the CCA for the fi rst time, the TSW automatically loads. At this time, you can choose to
configure your hardware using the wizard or the other CCA configuration methods you
learned in Chapter 11. The following requirements must be met in order to use the TSW for
your initial configuration:
CCA Telephony Setup Wizard Overview and Requirements
535
The UC500 must have a factory default configuration. If the UC500 was purchased
new from Cisco, then the device will be ready to go. If the UC500 has been used at a
different site or changes have been made to the factory default configuration, then the
administrator will need to reset both the Unity Express and CUCM Express to factory
default settings.
The PC running the TSW software within the CCA application must have only a
single network connection enabled. If the PC has both an Ethernet and a wireless
connection enabled, one of them will have to be disabled; otherwise, the TSW will
detect both network connections and you will be denied the opportunity to run the
wizard.
All of the IP phones that you wish to configure must be connected, fully booted,
and their firmware upgraded before running the wizard. You should make sure that
each phone is up and properly functioning with the factory default settings before
beginning.
When the CCA detects that the UC500 has the default configuration, it will
automatically run the wizard as soon as the CCA discovery process completes. Figure A.1
shows the TSW welcome screen.
F I G U R E A .1
The Telephony Setup Wizard welcome screen
Appendix A
536
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At the top of the TSW screen you can see the five steps of the configuration process:
1.
Networking
2.
Users/Extensions
3.
Auto Attendant
4.
Trunks
5.
Call Routing
We’ll cover these configuration steps in detail as we go through the informationgathering process from the owner of the fictional business we are setting up.
Running from top to bottom along the left side of the wizard screen within CCA is the
Help section. Here you can fi nd detailed descriptions of configuration parameters that
the wizard is asking for on the screen. You can collapse the Help section to give yourself
additional room on the screen.
Lastly, at the bottom of the screen are the wizard navigation buttons, used to progress
forward and backward through the wizard. It is important to keep in mind that no changes
to the UC500 are made until the administrator has run through the wizard in its entirety. If
for some reason you close the CCA prior to applying the configuration, any configuration
changes will not be saved and you can rerun the wizard without having to reset the UC500
to factory default settings.
Now that you understand the layout of the Telephony Setup Wizard, let’s begin our
fictional case study. In this case we will have a discussion with the owner of CC -NAV Inc.,
a company that is starting up a widget distribution center. Our goal with this discussion is
to gather the information required to configure the UC500 for both voice and data needs.
We will then walk through the configuration process using the CCA TSW.
The Information-Gathering
Meeting for CC-NAV Inc.
Abe Jefferson, owner of CC -NAV Inc., set up a meeting to discuss the voice and data
requirements for his new widget distribution center. The following is a review of how the
meeting went:
CC -NAV is a brand-new business that sells and distributes widgets. It is located in
Chicago, USA. The small company will have a single site with six employees. Each of these
employees will need their own telephone for making/receiving calls as well as basic Internet
access for email and web browsing. Abe has already contacted an ISP, and a standard DSL
connection will be installed at the site using DHCP.
Abe also contacted his local phone company and is having four analog telephone
lines run into the small business. Three of the lines will be used for customers calling for
sales or shipping questions. The number that CC -NAV will advertise to potential
customers is 312-555-8861. The fourth line will be dedicated for a fax machine. This
number is 312-555-8863.
Configuring Networking Parameters Using the TSW
537
Abe has hired five employees to handle the daily operations of the new start-up. He
provided us with an employee list that includes the name and job title of each employee.
There will be two salespersons to handle customers wanting to order widgets. Mr. Jefferson
also has two employees to handle warehouse and logistical duties to ensure that the
widgets are shipped properly to customers once the orders are finalized. Overseeing both
the sales and warehouse employees will be an operations manager, who will also be
required to fi ll in as a member of the sales or warehouse staff when needed.
Abe foresees customers calling into a main CC -NAV telephone number that will be
handled by an Auto Attendant system, where the customer can contact a salesperson for
new sales or the warehouse for any shipping questions. A hunt group will be set up for both
the sales and warehouse teams. If nobody in either hunt group is able to answer a call, the
fi nal extension tried will be that of the operations manager.
Unless there is a shipping problem, Abe believes that the shipping employees will not
receive many calls. On the other hand, the sales department will (hopefully) receive many
calls from customers wishing to order widgets. Because of this, Abe wants to provide high-end
Cisco 7965 IP phones to his sales team and the operations manager. Abe also will receive
a 7965 because, well, he’s the boss! The two warehouse employees will be supplied with
less-expensive analog phones that will be deployed in the warehouse section of the office.
All other parameters needed for the UC500 setup will be left up to the consultant (you)
to determine. Now that we have had a thorough conversation with the business owner,
let’s use this information we’ve gathered to set up our voice and data network for CC -NAV
using the Telephony Setup Wizard. In the next five sections of this case study, we will go
through each TSW configuration step, lay out the configuration plan using what we learned
during the information-gathering process, and configure our UC500 according to business
specifications.
Configuring Networking
Parameters Using the TSW
Now that we’ve gone through the information-gathering process to discover what we need
to configure the UC500 for CC -NAV Inc, we can turn our attention to using the Telephony
Setup Wizard to quickly configure the voice and data capabilities for the site. We’ll start
our configuration back at the TSW welcome screen you saw in Figure A.1. As stated
earlier, it is important that the IP phones we wish to configure using the TSW are up and
functioning. Once you have verified this, click the Next button to begin configuring the
system and network parameters.
Configuring System Access
This screen requires that you name the UC500 and set up an administrator account. Abe
never gave specific instructions on these steps, so we’ve taken the liberty of devising our
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own. As a consultant, it is very important that you document your configurations so you
can provide them to the customer. You never know when you might have to come back and
use this documentation for troubleshooting purposes! Table A.1 lists the required System
Access fields and the information we will configure within them.
TA B L E A .1
System Access parameters
Required Fields
Discovered Information
System Host Name
UC500_CC-NAV_Inc
Admin Username
Admin
Admin Password
cisco1
Figure A.2 shows the first configuration screen within the Networking parameters section.
FIGURE A.2
The System Access configuration screen
Enter the information into the fields as shown in the figure, and click the Next button to
continue.
Configuring Networking Parameters Using the TSW
539
Configuring the System Locale
The next Networking configuration screen is Choose Locale. Here we set regional
parameters that dictate the dial plan, time zone, and language used on the system. While
we did not specifically ask Abe all of these questions, we did learn from our conversation
that the business will be operated in Chicago, USA. Given this single piece of information,
we can infer the configuration parameters shown in Table A.2.
TA B L E A . 2
System location parameters
Required Fields
Discovered Information
UC500 Location
Chicago, USA
Language
English
Dial Plan Locale
NANP
Time Zone
GMT -06:00
Daylight Savings Mode
Enabled
Figure A.3 shows the Choose Locale screen with the proper settings for a business
located in Chicago.
FIGURE A.3
The Choose Locale configuration screen
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Enter the information into the fields as shown in the figure, and click Next
to continue.
Configuring WAN/LAN Settings
The following two configuration screens deal with WAN and LAN settings. In our
conversation with Abe, we learned that he has contacted an ISP and has ordered a DSL
connection that requires our end device to be set to receive an IP address using DHCP.
We also know that employees will use the network for both voice and data, so we should
separate this traffic into separate VLANs, following Cisco best-practice methodology.
There were no specific requirements for the addressing scheme used for the voice and
data IP subnets, so we can configure them as we see fit. Table A.3 lists the IP network
parameters we’ve decided to implement.
TA B L E A . 3
IP network parameters
Required Fields
Discovered Information
WAN/Internet
Standard DSL (using DHCP)
Data VLAN
1
Data IP Subnet
192.168.10.0/24
Data IP Gateway
192.168.10.1
Data Subnet DHCP Scope
192.168.10.10-240
Voice VLAN
100
Data IP Subnet
10.1.1.0/24
Data IP Gateway
10.1.1.1
Data Subnet DHCP Scope
10.1.1.10-240
The Configure WAN Connection screen is fi rst. Figure A.4 shows the different WAN
configuration options available to us, including these:
DHCP
Static IP
Configuring Networking Parameters Using the TSW
PPPoE IP Negotiate
PPPoE Static IP
FIGURE A.4
541
The WAN configuration screen
Because our site will be receiving an IP address from a DHCP server hosted by the ISP,
we can simply choose the DHCP option and click Next to continue. If we were to choose
one of the other options, we would have to add additional information including static IP
addresses, subnet masks, or authentication information for Point-to -Point Protocol over
Ethernet (PPPoE).
The next screen in the wizard allows us to configure the LAN portion of our IP network
including IP subnet settings. Figure A.5 shows how we’ve set up our network to use the
settings defi ned in Table A.3.
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Appendix A
FIGURE A.5
Design and Configuration Using the CCA
The LAN configuration screen
Notice how the VLAN information is listed but cannot be changed. This is one of the
limitations to the TSW. To make configurations as simple as possible, Cisco removed
the ability to change the VLAN ID because for most environments, the default VLANs of
1 and 100 will work just fi ne. In fact, we’ve decided not to modify any of the default LAN
settings given. In our conversation with the owner, there was nothing that he said that
would require us to stray away from the default settings provided by Cisco. Click the Next
button to continue.
This fi nal Networking configuration screen is a summary of all the decisions we’ve made
thus far. Figure A.6 shows the Networking Summary screen.
Configuring User and Extension Parameters Using the TSW
FIGURE A.6
543
The Networking Summary screen
You can review your choices and click the Back button to make any modifications. Once you
are satisfied, click the Next button to move on to the Users/Extensions configuration section.
Configuring User and Extension
Parameters Using the TSW
The Users/Extensions configuration screens are used to set up all of the internal (nonPSTN) settings, including extensions, FXS ports, and telephone users for assigning phones
and voice mail boxes to users. There are five configuration screens and one summary screen
in the wizard. Let’s plan out our configuration settings for each option and use the TSW to
configure the UC500. First up on the list is defi ning the internal dialing structure.
Configuring Internal Dialing
Internal dialing refers to the extensions that will be configured on analog and IP telephones
on the system. Internal extension lengths can be three, four, or five digits in length. By
default, three digits are used. Because there are only a handful of phones on the system, the
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Appendix A
Design and Configuration Using the CCA
choice of three digits is logical for this environment. In addition, Table A.4 lists the other
decisions we’ve made.
TA B L E A . 4
Internal dialing parameters
Required Fields
Discovered Information
Internal Extension Length
Three digits
Access Code
9
Prefix Digit for Voice Mail
6
Voice Mail Pilot Extension
299
AA Extension
298
The access code of 9 is for off-network calls, and the voice mail prefi x digit is used for
direct calls to the voice mail system for use by the AA (Automated Attendant). It will also
be used by analog phones that do not have a default voice mail button, as Cisco IP phones
do. Both the access code and prefi x digit are restricted to a single number from 1 to 9, and
the digit cannot be the fi rst number of any configured extension or access code. Figure A.7
shows the internal dialing screen with the proper configurations for our case.
F I GU R E A .7
The Define Internal Dialing screen
Configuring User and Extension Parameters Using the TSW
545
Enter the information into the fields as shown in the figure, and click the Next button to
continue to the next screen in the wizard.
Configuring Analog Station (FXS) Ports
During our conversation with Abe Jefferson, it came to light that the business requires
one shared fax machine and two analog phones for the warehouse workers. Given
this information, we’ve come up with the necessary information to configure the FXS
interfaces, as shown in Table A.5.
TA B L E A . 5
FXS port parameters
Required Fields
Discovered Information
Number of fax machines
1
Number of user analog phones
2
Our UC500 has four FXS ports, so we will disable 0/0/3 because we do not need it
at this time. Port 0/0/0 will be set up as a common area phone/fax for our analog fax
machine, and 0/0/1 and 0/0/2 will be configured as user ports. Figure A.8 shows the analog
station ports configuration screen configured to match our business requirements.
FIGURE A.8
The Configure Analog Station Ports screen
546
Appendix A
Design and Configuration Using the CCA
When we configure an FXS port as a common area phone or fax, you’ll notice that a
default extension of 301 appears. The Cisco default settings for three-digit extensions are
that user extensions begin at extension 201 and common-area phones begin at 301. You
can change these numbers to suit your needs as long as they don’t interfere with other
previously configured extensions or access codes such as voice mail pilot numbers or the
access code.
Enter the information into the fields as shown in the figure, and click the Next button to
continue through the wizard.
Configuring Phone Users and Extensions
Now it’s time to defi ne our users and user extensions on our PBX. The Defi ne Users
and Extensions screen allows us to enter the full fi rst and last names of our phone
users as well as assign an extension to them. Because we chose to use three-digit extensions
on this system, the CCA auto-filled the user extension field with numbers beginning
with 201. Using the information we gathered in our information-gathering conversation
with the owner, we’ve come up with the user parameters in Table A.6 that we can use to
configure the system.
TA B L E A . 6
User parameters
Name
User ID
Job Function
Phone Type
Extension
Abe Jefferson
ajefferson
Owner
Cisco 7965
201
David Miller
dmiller
Sales
Cisco 7965
202
Wendy Davis
wdavis
Sales
Cisco 7965
203
Sarah Foreman
sforeman
Operations manager
Cisco 7965
204
Chen Lee
clee
Warehouse/logistics
Analog
205
Michael Cross
mcross
Warehouse/logistics
Analog
206
Figure A.9 shows the Defi ne Users and Extensions screen properly fi lled out for our
six employees.
Configuring User and Extension Parameters Using the TSW
FIGURE A.9
547
The Define Users and Extensions screen
By default, each user password will initially be 12345. The user will be required
to change this password the fi rst time they log in. Also notice that we’ve decided
to create individual subscriber voice mail boxes for each employee. These will be
automatically created for us by the CCA. Click Next to move on to the next
configuration screen.
Now we have users and extensions configured, but none of these users are assigned a
phone. Figure A.10 shows us assigning a phone to each user. Each user configured is listed
on the left of the screen, and a drop -down list shows all of the discovered IP phones as
well as any enabled FXS ports available. Abe stated that he, the operations manager, and
the sales team would be using Cisco 7965 phones. The two warehouse employees would
use analog phones attached to the FXS ports on the UC500. Use Table A.6 to determine
whether a user is assigned a Cisco 7965 or an analog phone.
548
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Design and Configuration Using the CCA
The Assign Phones screen
If we did not care which employee was assigned to a particular phone, we could have
clicked the Speed Pick button, which randomly assigns a phone to a user. But because
specific people are to receive a specific type of phone, we manually assigned them to our
users. The system message that is displayed at the bottom of the LCD screen on the Cisco
IP phones is also configured on this screen. Click the Next button to move on to the next
configuration screen.
Configuring Hunt Groups
The last Users/Extensions configuration screen is used to create hunt and blast groups.
A blast group is simply a hunt group that will ring multiple phones simultaneously. The
fi rst extension to answer the call gets the connection. Now that we have users assigned to
phones, we can go ahead and create our hunt groups for the business. Abe would like
to see two hunt groups created for business operations. One group would be for sales and
the other for warehouse/logistics. We’ll use the longest-idle-time method to determine
which phone in the hunt group rings. Also keep in mind that Abe wants the “catch-all”
Configuring User and Extension Parameters Using the TSW
549
extension to be the operations manager if nobody is able to field the call. Figure A.11 shows
the hunt group setup screen with the proper configuration parameters entered in.
F I G U R E A .11
The Define Hunt Groups and Blast Groups screen
This hunt group setup may not be ideal for some businesses. If the first
two members fail to field the call, the forward will go directly to our
operations manager (Sara Foreman). If she is not able to answer the call,
it will be redirected to Sara’s personal mailbox. Keep situations like this in
mind when you are planning your hunt or blast groups.
The CCA configures hunt/blast groups to begin with extension 501 by default. You
should also make sure to name your group so you can clearly understand what its purpose
is. Once you have fi nished configuring your groups, click the Next button to continue.
The fi nal Users/Extensions screen is a summary of all the settings made in this section.
Figure A.12 shows the Users Summary screen.
550
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F I G U R E A .1 2
Design and Configuration Using the CCA
The Users Summary screen
You can review your choices and click the Back button to make any modifications. Once
you are satisfied, click the Next button to move on to the Auto Attendant configuration
section.
Configuring Auto Attendant
Parameters Using the TSW
Our client has clearly stated that an automated attendant should be used to direct calls to
either the sales or warehouse teams. Beyond that, it is up to us to determine the AA settings
that a typical business such as CC -NAV would require. We’ll fi rst set up basic AA settings,
such as the choice between a single and a dual schedule AA. We’ll then move on to defi ne
the AA prompts and actions to the prompts. Finally, we’ll defi ne how to customize the AA
prompts for CC -NAV Inc.
Defining the AA and Setting Working Hours
Figure A.13 shows the fi rst of a series of AA choices that need to be made.
Configuring Auto Attendant Parameters Using the TSW
F I G U R E A .1 3
551
The Define Auto Attendant screen
We must check the box to enable AA in the fi rst place. We’ve already determined the AA
pilot extension in previous TSW configuration steps, but we can modify it here if needed.
The TSW allows for two AA types, which differ as follows:
Single Schedule The single schedule presents the same AA script 24 hours a day regardless
of the business being open or not.
Dual Schedule The dual schedule provides two different AA scripts. One script is played
during office hours, and the other is for after-hours and holiday calls. If this choice is
selected, the administrator must defi ne when the business is open as well as the holiday
dates when the office is closed.
Since this is a small business that is unlikely to be run 24 hours a day, we’ve decided
to configure a dual-schedule setup. Once you’ve configured our choices, click Next to
continue.
Because of our choice to configure a dual schedule, we contacted Sarah Forman, the
operations manager, who provided us with office hours and days when the office will be
closed because of a holiday. This information will be used in the next two Auto Attendant
configuration screens. Figure A.14 shows us setting the office working hours to be Monday
through Friday from 8:00 a.m. to 5:00 p.m.
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F I G U R E A .1 4
Design and Configuration Using the CCA
The AA Business Hours definition screen
To modify the hours on the screen, you simply click and drag across the day you wish to
modify. The working hours portion is then highlighted in green. Set the hours according
to Sarah’s input, and click the Next button to continue.
This next screen is to configure holiday hours for the business. The CCA offers a
12-month calendar where you can fi nd the days you wish to defi ne as a holiday. Click
the holiday, name it, and that entire day will then use the after-hours AA script. By
default, holiday hours are set for New Years and Christmas Day only. We’ve added
additional holidays that are common in the United States. Figure A.15 shows our
setup.
Make the necessary holiday additions, and click Next to continue to the next screen.
Defining AA Prompts and Actions
The Defi ne Auto Attendant Prompts and Actions screen is where we defi ne how the AA
script progresses for each call. Because we decided to use a dual-schedule AA, both a
Company Greeting and an After Hours Greeting are defi ned in Figure A.16.
Configuring Auto Attendant Parameters Using the TSW
F I G U R E A .1 5
The AA Holiday Schedules definition screen
F I G U R E A .1 6
Defining AA prompts and actions
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The Auto Attendant Actions section is where we can configure the prompts that will
guide the user navigating through to reach the representative of CC -NAV they wish to
contact. The numbers 1 through 8 to the left of the actions represents the keypad button a
caller will press to initiate the action. In our example, we’ve mapped the Sales hunt group
to button 1 and the Warehouse hunt group to button 2. Callers can also press 3 to hear a
recording of office hours. Buttons 4 and 5 are used for the caller to either dial by number (if
known) or to dial by name by using the keypad to spell out the last name of the employee.
Also very important to notice is the key in the upper-right corner of the screen. This key
is used to defi ne whether a button option is available for callers based on when they call.
Defi ned office hours are considered to be “day” hours, and hours when the office is closed
are “night” hours. Following are the different options you can apply to a particular button
setup depending on when the user calls:
Day Same As Night
The AA button option is available regardless of when the user calls.
Day Different From Night The AA button option is used for different purposes depending
on the time of day the call is placed.
Day Only
Night Only
The AA button option is available only during office hours.
The AA button option is available only after office hours.
As you can see in Figure A.16, we’ve set our hunt groups to be available only when the office
is open. During working hours the phone buttons used by the AA are set up as the following:
Button 1: Contact the Sales hunt group
Button 2: Contact the Warehouse hunt group
Button 3: Company location and business hours recording
Button 4: Dial user by extension
Button 5: Dial user by name
When the office is closed, the following button mappings are used on the AA system:
Button 1: Dial user by extension
Button 2: Dial user by name
Button 3: Company location and business hours recording
As you can see, buttons 1 and 2 are using the Day Different From Night option to create
different actions for the button based on time of day. Button 3, which is a recording of the
business location and hours, remains the same 24 hours a day and therefore uses the Day
Same As Night option.
When you have made the proper configuration settings, click Next to continue.
Managing Auto Attendant Prompts
We have our AA scripts all defined, but we don’t have any voice scripts to help the user
navigate through them. Figure A.17 shows the different ways we can manage AA prompts using
the TSW. As you can see, we will need to record three different AA prompts according to the
Configuring Auto Attendant Parameters Using the TSW
555
settings we defined in the last step. The prompts we need to create are the Company Greeting,
the After Hours Greeting, and the Company Location and Business Hours Recording.
F I G U R E A .17
The Manage Auto Attendant Prompts screen
There are three ways we can add custom AA prompts to our UC500. One method is to
create and save audio fi les using a separate voice-recording application. We can then upload
the fi les as .wav or .au to the UC500 at this screen. To upload the files, click the folder icon
in the Action column and locate the stored fi le to upload it. The fi les will then be stored
onto the UC500 compact flash.
A second method is to use the TSW to record prompts directly to the UC500. This
requires that the PC you are running the CCA on have a voice card and microphone. If
it does, you can use this hardware to record the prompts and save them to the UC500
compact flash.
Finally, the last method is to use the UC500 Prompt Manager to record your custom
prompts. This is the method we’ve chosen to create our recordings for our three prompts.
Wendy Davis has agreed to record the AA prompts, so we have selected her phone
extension to be the Prompt Management User. What this means is that Wendy can use
her phone (and only her phone at extension 203) to dial into the prompt management
pilot number and walk through the recording of the three custom prompts our AA needs.
As you can see in the Help section, the TSW has assigned extension 297 for the prompt
556
Appendix A
Design and Configuration Using the CCA
management pilot number. Wendy will dial this extension and use the built-in TUI to
follow the directions to record and save our prompts to the UC500 compact flash.
When you have made the proper configuration settings, click Next to continue.
We are then presented with the Auto Attendant Summary screen, as shown in Figure A.18.
F I G U R E A .1 8
The Auto Attendant Summary screen
As in all other summary screens we’ve seen, we can review all our settings and go back
to modify anything that we wish. Once you are satisfied, click the Next button to move on
to configure the business PSTN trunk parameters.
Configuring Analog PSTN Trunk
Parameters Using the TSW
The PSTN trunk parameters screen for our mock setup is fairly straightforward. We
know that Abe has already contacted his local PSTN and ordered four PSTN lines. Three
of those lines are to be used for incoming and outgoing calls, and the fourth line will be
dedicated to a fax machine. All of the FXO interfaces will act as PBX- connected ports,
meaning that incoming calls will be directed to one specific extension. And as we
Configuring Analog PSTN Trunk Parameters Using the TSW
557
know, PBX extensions are unique to a single phone (unlike key system setups, where a
number is defi ned on multiple phones and the fi rst one to answer has control over that
particular line).
Figure A.19 shows the PSTN configuration that we have defi ned for CC -NAV Inc.
F I G U R E A .1 9
The analog trunk setup screen
There are very few options to configure here, which limits our flexibility but also
helps to simplify the TSW process, and that is what the tool is all about. We can enable
or disable any of the four interfaces as well as defi ne the action of the interface as being
either a PBX or a key system port. In our case, we want to use all four FXO ports and we
also want them to use the PBX model of functionality. Once you have made the necessary
configuration changes, click Next to continue.
As you can see, we’re already at the Trunking Summary page shown in Figure A.20.
This screen shows you how many ports will be disabled, set up as key system ports, or
set up as PBX model ports. When you are satisfied, click Next to move to the last TSW
section, Call Routing.
558
Appendix A
FIGURE A.20
Design and Configuration Using the CCA
The Trunking Summary screen
Configuring Call Routing Parameters
Using the TSW
In the Trunks section we defi ned all four FXO ports to be PBX-modeled interfaces. In
order to work when calls come inbound from the PSTN, the interfaces now must be
assigned to an extension. In our initial conversation, we learned that CC -NAV would have
two telephone numbers: the main number, which will be assigned to three PSTN lines, and
a fourth line for a fax machine. We also know that all customers calling to place orders or
inquire about shipping/logistics should go through an automated attendant to help them
reach the right employee. We’ve created Table A.7 to assist us with defi ning which FXO
port will be routed to the Auto Attendant or to the FXS port with the attached fax machine.
TA B L E A . 7
FXO port parameters
FXO Port
PSTN Number
Routed To
0/1/0
312-555-8861
Auto Attendant
0/1/1
312-555-8861
Auto Attendant
0/1/2
312-555-8861
Auto Attendant
0/1/3
312-555-8863
Fax machine
Configuring Call Routing Parameters Using the TSW
559
Now that we have the ports defi ned, we just need to configure them to be routed either
to the AA pilot number or to the FXS fax port. Figure A.21 shows how we configure call
handling for CC -NAV ’s FXO ports using the TSW.
F I G U R E A . 21
The incoming call-handling screen
The configuration process is quite simple; all of the enabled FXO ports are listed on
the left of the screen, and a drop -down list of all configured extensions is available to
choose from. When inbound calls come into one of these four ports, the calls are promptly
redirected to the extension without any manual intervention. The fi rst three ports (FXO
0/1/0 to 0/1/2) are redirected to the AA so customers can use it to contact either the Sales
or Warehouse group. Fax requests come in on the alternate PSTN number, which is then
routed to the FXS interface with the attached analog fax machine on it. Once you have set
the interfaces for proper routing, click Next to continue.
This again was another short configuration section. Figure A.22 shows the summary
screen, which is little more than a confi rmation of how the ports are defi ned as being
routed either to the AA or to the FXS interface.
Once you are sure about your settings, click the Next button to move on to the fi nal
TSW configuration screen.
560
Appendix A
FIGURE A.22
Design and Configuration Using the CCA
The Call Routing Summary screen
Final Review and Applying
the Configuration Using the TSW
The Apply Configuration screen is the fi nal opportunity to review all of the configuration
settings we wish to make using the TSW. Up to this point, we’ve only defi ned what we want
to configure, and no changes have actually been made on the UC500 system. Only when
we click the Apply Configuration button, as shown in Figure A.23, will the wizard apply
all the modifications onto the CUCM Express and Unity Express systems.
On this screen there are tabs for each of the five wizard configuration sections we
went through. At any time, you can go back and modify any of these settings before you
apply them to the system. Also note the red text stating that we have modified the VLAN
information in the wizard, so we will lose IP connectivity to the UC500 system and will
have to relaunch CCA using the new IP address on the VLAN.
Click the Apply Configuration button to put our new system configuration onto your
UC500. The wizard then converts our settings to command-line form and applies and saves
them to both the CUCM Express and Unity Express configurations. This process can take
up to 28 minutes to complete, so please be patient.
As soon as the wizard fi nishes applying and saving the configuration, the employees at
CC -NAV Inc. can test it out. Any modifications can then be made using the CCA telephony
Summary
FIGURE A.23
561
The Apply Configuration screen
tools, web GUI tools, or even the command-line tools that have been taught throughout the
course of this book. Any of these methods will get the job done and get our customers off
and running on their new voice system.
Summary
The mock consulting example in this case study was used to show you what the Telephony
Setup Wizard was designed to do. For brand-new and relatively straightforward setups for
small businesses, the wizard is a great way to get a site up and running very quickly.
Appendix
B
About the
Companion CD
IN THIS APPENDIX:
What you’ll find on the CD
System requirements
Using the CD
Troubleshooting
What You’ll Find on the CD
The following sections are arranged by category and
summarize the software and other goodies you’ll fi nd on the CD. If you need help with
installing the items provided on the CD, refer to the installation instructions in the “Using
the CD” section of this appendix.
Some programs on the CD might fall into one of these categories:
Shareware programs are fully functional, free, trial versions of copyrighted programs.
If you like particular programs, register with their authors for a nominal fee and
receive licenses, enhanced versions, and technical support.
Freeware programs are free, copyrighted games, applications, and utilities. You can
copy them to as many computers as you like — for free — but they offer no technical
support.
GNU software is governed by its own license, which is included inside the folder of
the GNU software. There are no restrictions on distribution of GNU software. See the
GNU license at the root of the CD for more details.
Trial, demo, or evaluation versions of software are usually limited either by time or by
functionality (such as not letting you save a project after you create it).
Sybex Test Engine
For Windows
The CD contains the Sybex test engine, which includes all of the assessment test and
chapter review questions in electronic format, as well as two bonus exams located only on
the CD.
PDF of the Book
For Windows
We have included an electronic version of the text in .pdf format. You can view the
electronic version of the book with Adobe Reader.
Using the CD
565
Adobe Reader
For Windows
We’ve also included a copy of Adobe Reader so you can view PDF fi les that accompany
the book’s content. For more information on Adobe Reader or to check for a newer version,
visit Adobe’s website at www.adobe.com/products/reader/.
Electronic Flashcards
For Windows
These handy electronic flashcards are just what they sound like. One side contains a
question or fi ll-in-the-blank question, and the other side shows the answer.
System Requirements
Make sure your computer meets the minimum system requirements shown in the following
list. If your computer doesn’t match up to most of these requirements, you may have
problems using the software and fi les on the companion CD. For the latest and greatest
information, please refer to the ReadMe fi le located at the root of the CD-ROM.
A PC running Microsoft Windows 98, Windows 2000, Windows NT4 (with SP4 or
later), Windows Me, Windows XP, or Windows Vista, and Windows 7.
An Internet connection
A CD-ROM drive
Using the CD
To install the items from the CD to your hard drive, follow these steps:
1.
Insert the CD into your computer’s CD-ROM drive. The license agreement appears.
Windows users: The interface won’t launch if you have autorun disabled.
In that case, click Start Run (for Windows Vista, Start All Programs
Accessories Run). In the dialog box that appears, type D:\Start.exe.
(Replace D with the proper letter if your CD drive uses a different letter.
If yo