Configuring Cisco Unified Communications Manager and Unity

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Configuring Cisco Unified
Communications Manager
and Unity Connection:
A Step-by-Step Guide
David Bateman
Cisco Press
800 East 96th Street
Indianapolis, IN 46240
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ii
Configuring Cisco Unified Communications Manager and Unity Connection: A Step-by-Step Guide
Configuring Cisco Unified Communications Manager
and Unity Connection: A Step-by-Step Guide
David Bateman
Copyright © 2011 Cisco Systems, Inc.
Cisco Press logo is a trademark of Cisco Systems, Inc.
Published by:
Cisco Press
800 East 96th Street
Indianapolis, IN 46240 USA
All rights reserved. No part of this book may be reproduced or transmitted in any form or by any means,
electronic or mechanical, including photocopying, recording, or by any information storage and retrieval system, without written permission from the publisher, except for the inclusion of brief quotations in a review.
Printed in the United States of America 1 2 3 4 5 6 7 8 9 0
First Printing May 2011
Library of Congress Cataloging-in-Publication Number is on file.
ISBN-10: 1-58714-226-0
ISBN-13: 978-1-58714-226-0
Warning and Disclaimer
This book is designed to provide information about configuration and administrative tasks related to
Communications Manager and Unity. Every effort has been made to make this book as complete and as
accurate as possible, but no warranty or fitness is implied.
The information is provided on an “as is” basis. The author, Cisco Press, and Cisco Systems, Inc. shall have
neither liability nor responsibility to any person or entity with respect to any loss or damages arising from
the information contained in this book or from the use of the discs or programs that may accompany it.
The opinions expressed in this book belong to the author and are not necessarily those of Cisco Systems, Inc.
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Trademark Acknowledgments
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Configuring Cisco Unified Communications Manager and Unity Connection: A Step-by-Step Guide
About the Author
David J. Bateman is a certified Cisco Systems instructor and the director of curriculum
development for Skyline-ATS. He has more than 20 years of internetworking experience.
For more than 10 years, David was a senior LAN/WAN engineer, working on small, medium, and large networks. Later in his career, he took on the responsibility of running the
business operations of a technical services company, while maintaining his existing client
base. David has always enjoyed sharing his knowledge, and in 1999, he added to his list
of accomplishments by becoming a technical seminar leader. After many successful seminars, he decided to become a full-time Cisco instructor for Skyline Advanced Technology
Services. He has been teaching and implementing Cisco voice technologies since 2000.
David’s years of real-world technical and business knowledge allow him to bring a unique
perspective to the classroom, where he not only delivers critical technical knowledge but
can also explain how technologies can be used to address various business issues.
About the Technical Reviewers
David L Mallory, CCIE No. 1933, is a technical leader for Learning@Cisco, where he is
responsible for content development strategy. For the last seven years, David has been
primarily focused on UC certifications and was the technical lead for the Cisco 360
Learning Program for CCIE Voice. Prior to joining Learning@Cisco, David was a systems
engineer supporting global accounts. David is a frequent presenter at Cisco Live and has
obtained four CCIEs—Routing & Switching, WAN Switching, Security, and Voice.
Toby Sauer is the lead voice instructor and voice curriculum manager for Skyline
Advanced Technology Services. He brings 30 years of experience in the traditional voice,
data, and VoIP arenas. Toby has been involved in Cisco VoIP since the beginning, working with traditional VoIP, and he was involved in the earliest installations of Cisco
Communications Manager. He has installed many different implementations of
Communications Manager and was responsible for converting most of the Midwest’s
Cisco offices from traditional PBX to Communications Manager.
Toby became a Cisco voice instructor in 2000. As the Communications Manager product
continued to grow and develop, he was a key instructor to many of the original deployment partners.
Toby currently holds CCNP-Voice, CCNA-Voice, CCNA-RS, CCSI, and various partnerlevel certifications. He teaches all the Cisco Standard Voice courses and many custom
variations of these courses.
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Dedications
I’d like to dedicate this book to my parents, who taught me unconditional love; to my
wife, Nikki, who is my life, my love, my all; and to Matthew, a young man that I am
proud to call my son.
Acknowledgments
There are a number of people that I would like to thank in helping me complete this
book. Often the greatest help that can be received is when someone is willing to sacrifice
so that you can succeed. With this in mind, I would like to thank my wife, Nikki. She has
sacrificed many beautiful summer days that we could have spent out on the motorcycle
so that I could work on this book. She sacrificed hours each week reading what I had
written in order that I might deliver a more readable copy to the editors. I know it was
not always fun for her, but it helped me complete this book. Without her sacrifice, this
book would not have been possible.
I would also like to thank the technical editors. Their keen insight and willingness to ask
me what the heck I was thinking on some subjects have helped make this a much better
book than it was when I first wrote it.
Of course I’d like to thank those at Skyline-ATS, where I work. I would especially like to
thank them for the skill they showed in increasing my workload as deadlines for the book
drew near. I guess they figured I would do better under pressure. But seriously, I would
like to thank Mike Maudlin and Mike Zanatto for their understanding and cooperation
during this project. I also need to thank all the others that I worked with at Skyline-ATS.
The awesome amount of knowledge that we hold as a team is incredible, and to have such
a resource at my disposal has been invaluable.
A big thank-you to the folks at Cisco Press: Brett Bartow, who assisted from the beginning of this project and was always there to remind me of upcoming deadlines long
enough in advance so that I had time to either meet the deadline or come up with a really
good excuse. Also Marianne Bartow, who acted as my development editor and was
always helpful and encouraging.
Thanks one and all for all you’ve done.
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Configuring Cisco Unified Communications Manager and Unity Connection: A Step-by-Step Guide
Contents at a Glance
Part I
Communications Manager Configuration
Chapter 1
CUCM and Unity Connection Overview
Chapter 2
Preparing CUCM for Deployment
Chapter 3
Deploying Devices
Chapter 4
Implementing a Route Plan
Chapter 5
Configuring Class of Service and Call Admission Control
Chapter 6
Configuring CUCM Features and Services
Part II
Messaging Configuration
Chapter 7
Unity Predeployment Tasks
295
Chapter 8
User/Subscriber Reference
377
Chapter 9
Call Management
Chapter 10
Implementing Unity Networking
Chapter 11
Exploring Unity/Connection Tools
Part III
Leveraging the Power of Communications Manager
and Unity
Chapter 12
Maximizing CUCM and Unity/Connection
Appendix
Additional Reference Resources
Index
1
41
77
151
193
231
495
567
587
651
657
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Table of Contents
Part I
Chapter 1
Communications Manager Configuration
CUCM and Unity Connection Overview
Ensuring a Reliable Foundation
Infrastructure Overview
Inline Power
1
2
3
4
Voice VLANs
4
CDP Support
4
Voice Gateways
4
Creating a Reliable VoIP Infrastructure
Communications Manager Overview
5
7
Defining Communications Manager Components
Communications Manager Business Edition
Communications Manager Devices
Phones
10
10
11
Gateways Overview
Gatekeepers
12
14
Media Resources
15
Conference Bridge (CFB)
Transcoders
MoH
8
15
16
16
Annunciator
16
Understanding Communications Manager Deployment Models
Single-Site
17
Multisite WAN with Centralized Call Processing
17
Multisite WAN with Distributed Call Processing
17
Route Plan Overview
Typical Call Flow
Wildcards
18
19
20
Calling Privileges
21
Unified Messaging Overview
Software Architecture
22
23
Unity Software Architecture
23
Unity Connection Architecture
Following the Call Flow
25
Exploring Call Handlers
26
25
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Configuring Cisco Unified Communications Manager and Unity Connection: A Step-by-Step Guide
Defining Various Types of Users
Unity Connection Users
Unity Subscribers
User Parameters
29
29
30
31
Networking Overview
Unity Networking
33
33
Unity Connection Networking
Securing the Environment
35
35
Securing the Operating System
35
Communications Manager Security Issues
Unity Security Issues
Summary
Chapter 2
36
38
39
Preparing CUCM for Deployment
41
Configuring Communications Manager for Maximum Performance
Activating Communications Manager Services
42
Configuring Communications Manager’s Enterprise Settings
Removing DNS Dependencies
50
50
Communications Manager Administrator Parameters
CCMUser Parameters
CDR Parameters
43
48
Defining Enterprise Parameters
General Parameters
41
52
53
55
Localization Parameters
55
Multi-Level Precedence and Pre-Emption (MLPP) Parameters
Security Parameters
56
Prepare Cluster for Roll Back
57
Phone URL Parameters and Secured Phone URL Parameters
User Search Parameters
58
CCM Web Services Parameters
Trace Parameters
59
59
User Management Parameters
60
Service Manager TCP Ports Parameters
CRS Application Parameters
Cluster Domain Configuration
Denial-of-Service Protection
TLS Handshake Timer
Cisco Support Use
55
60
60
60
60
60
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57
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IPv6 Configuration Modes
Cisco Syslog Agent
60
61
CUCReports Parameters
61
Logical Partitioning Configuration
61
Preparing Communications Manager for Device Registration
Device Pools
62
Common Device Configuration
66
Creating Communications Manager Groups
Defining Date/Time Groups
Configuring Regions
Chapter 3
69
70
Building Device Pools
Summary
66
72
75
Deploying Devices
Adding Clients
77
78
Defining Device Settings
78
Phone Button Templates
78
Softkey Template
Device Defaults
80
83
Adding Phones
84
Autoregistration
86
Manually Adding Phones
Add a Line to a Phone
89
99
Using BAT to Add Devices
106
Activating the BAT Service
107
BAT CSV and Template Overview
Creating a CSV File for BAT
110
Adding Phones Using BAT
113
Adding Phones Using TAPS
118
Adding Gateways
108
119
Adding H.323 Gateways
Device Information
119
121
Call Routing Information—Inbound Calls
123
Call Routing Information—Outbound Calls
Geolocation
125
Intercompany Media Engine
126
Incoming Calling/Called Party Settings
Adding MGCP Gateways
126
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124
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Configuring Cisco Unified Communications Manager and Unity Connection: A Step-by-Step Guide
Adding IOS MCGP Gateways
127
Adding Non-IOS MGCP Gateways
Adding Intercluster Trunks
Device Information
132
143
144
Call Routing Information—Inbound Calls
147
Call Routing Information—Outbound Calls
147
Remote Cisco Communications Manager Information
UUIE Configuration
149
Geolocation Configuration
Summary
Chapter 4
149
150
Implementing a Route Plan
Understanding Call Flow
151
152
Understanding Route Groups and Route Lists
Creating Route Groups
Creating a Route List
154
157
158
Understanding Route Patterns
163
Creating Basic Route Patterns
166
Using Pattern Wildcards to Create a Basic Dial Plan
Advanced Route Plan Components and Behavior
Creating Route Filters
Creating CTI Route Points
183
Directory Number Information
Directory Number Settings
185
185
186
187
Call Forward and Pickup Settings
Park Monitoring
187
189
MLPP Alternate Party Settings
Chapter 5
173
179
Adding a Line to a CTI Route Point
AAR Settings
171
174
Creating Translation Patterns
189
Line Settings for All Devices
190
Line Settings for This Device
190
Multiple Call / Call-Waiting Settings
190
Forwarded Call Information Display
191
Summary
149
191
Configuring Class of Service and Call Admission Control
Rights and Restrictions
193
Understanding Call Search Spaces and Partitions
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Creating Calling Search Spaces and Partitions
202
Applying Calling Search Spaces and Partitions
Assigning a CSS to a Phone
Assigning a CSS to a Line
205
206
206
Assigning a CSS to a Gateway or Trunk
207
Assigning a Partition to a Line (Directory Number)
Assigning a Partition to a Pattern
210
Implementing Call Admission Control
211
Configuring CAC for a Distributed Deployment
Configuring a Gatekeeper
211
213
Configuring a Gatekeeper-Controlled Trunk
215
Call Routing Information—Outbound Calls
219
Gatekeeper Information
220
Configuring CAC for a Centralized Deployment
Creating Locations
Special Services Configuration
Special Services Overview
223
224
224
Configuring Special Services Route Patterns
Chapter 6
221
221
Assigning a Location to Devices
Summary
209
225
229
Configuring CUCM Features and Services
Configuring Features
231
231
Creating Call Pickup Groups
231
Add a Call Pickup Number
232
Assign a Call Pickup Group to a Line
Creating Meet-Me Patterns
235
Creating Call Park Numbers
237
Creating Directed Call Park Numbers
Creating Intercoms
234
239
240
Creating Intercom Partitions
241
Intercom Calling Search Spaces
Creating Intercom Numbers
241
241
Assigning an Intercom DN to a Phone
242
Creating Forced Authorization Codes
244
Create a Forced Authorization Code
244
Assign a Forced Authorization Code to a Route Pattern
Configuring Client Matter Codes
246
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Configuring Cisco Unified Communications Manager and Unity Connection: A Step-by-Step Guide
Create a Client Matter Code
246
Assign a Client Matter Code to a Route Pattern
Configuring Voice Ports and Profiles
Creating Users
247
248
259
Configuring Advanced Services
262
Implementing Advanced Features
263
Configuring IP Phone Services
Extension Mobility
263
265
Creating and Managing Media Resources
Configuring an MOH Server
273
273
Assign an MOH Audio Source to a Phone
Creating Conference Bridges
Configuring MTPs
276
276
279
Creating Transcoders
279
Configuring Annunciators
281
Media Resource Management
282
Assign a Media Resource Group List to a Phone
285
Assign a Media Resource Group List to a Device Pool
Configuring Remote Site Failover
SRST Overview
Configuring SRST
287
287
Creating an SRST Reference to a Device Pool
Assign an SRST Reference to a Device Pool
Configuring AAR
288
290
290
Creating an AAR Group
291
Assign an AAR Group to a Line
Summary
286
286
292
294
Part II
Messaging Configuration
Chapter 7
Unity Predeployment Tasks
295
Accessing and Navigating Unity Administrator
296
Accessing and Navigating Unity Connection
Administrator 301
Unity Integration Verification
304
Communications Manager Integration
Voicemail Port Configuration
305
305
Unity Telephony Integration Manager (Communications
Manager) 307
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SIP Integration
311
SIP Configuration
311
Unity Telephony Integration Manager (SIP)
PIMG/TIMG Integration
312
315
PIMG/TIMG Configuration
315
Unity Telephony Integration Manager (PIMG/TIMG)
Defining Unity System Configuration
317
Creating Schedules and Holidays
View and Change a Schedule
Add a Schedule
319
320
Define a Default Schedule
Add a Holiday
320
321
Modify or Delete a Holiday
322
Defining Configuration Settings
Settings
322
322
Software Versions
Recordings
Contacts
318
326
326
328
Phone Languages
GUI Languages
328
330
Message Security
330
Message Subjects
330
Configuring Authentication Settings
Configuring Ports
331
332
Configuring Unity System Access and Policies
Defining Account Polices
334
Configuring Class of Service
Adding a CoS
334
337
337
Modifying a CoS
338
Creating and Managing Unity Public Distribution Lists
Creating Public Distribution Lists
Managing PDL Members
347
350
Unity Connection Integration Verification
Communications Manager Integration
351
351
Defining Unity Connection System Configuration
Defining General Configuration
Defining Mailbox Quotas
355
358
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347
315
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Configuring Cisco Unified Communications Manager and Unity Connection: A Step-by-Step Guide
Configuring Message Aging Policy
Creating Schedules and Holidays
View and Change a Schedule
359
361
361
Configuring Unity Connection System Access and Policies
Configuring Authentication Rules
Configuring Restriction Tables
Configuring CoS
366
371
Defining the Dial Plan
Chapter 8
363
368
Understanding Roles
Summary
363
372
375
User/Subscriber Reference
377
Defining Various Types of Subscribers
Exchange
378
Networked Subscribers
Unity Connection Users
Creating Users
377
378
378
378
Exploring Templates
379
Creating Unity Subscriber Templates
381
Configuring Subscriber Template Profile Settings
384
Configuring Subscriber Template Account Settings
386
Configuring Subscriber Template Passwords Settings
Configuring Subscriber Template Conversation
388
Configuring Subscriber Template Call Transfer
Configuring Subscriber Template Greetings
386
394
398
Configuring Subscriber Template Caller Input
402
Configuring Subscriber Template Messages Settings
405
Configuring Subscriber Template Distribution Lists Settings
407
Configuring Subscriber Template Message Notification Settings
Configuring Subscriber Feature Settings
Creating New Unity Subscribers
Importing Unity Subscribers
414
417
Creating Unity Connection User Templates
Configuring User Template Basics Settings
Configuring Password Settings
Configuring Template Passwords
Configuring Roles
412
420
424
426
427
427
Configuring User Template Transfer Rules
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427
408
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Configuring User Template Messages Settings
Configuring User Template Message Actions
Configuring User Template Caller Input
430
432
434
Configuring User Template Mailbox Settings
Configuring User Template Phone Menu
435
437
Configuring User Template Playback Message Settings
Configuring User Template Send Message Settings
Configuring User Template Greetings
439
442
444
Configuring User Template Post-Greeting Recording
446
Configuring User Template Message Notification Settings
Creating New Unity Connection Users
Importing Unity Connection Users
Unity Connection Contacts
454
456
Creating Unity Connection Contact Templates
Creating Unity Connection Contacts
Managing Users
447
451
456
457
460
Managing User Access
Unlocking an Account
Resetting Passwords
460
460
461
Changing a Subscriber’s Extension
Changing a Subscriber’s CoS
462
463
Granting Access to Licensed Features (FaxMail,
Text-to-Speech, CPCA) 463
Granting Additional System Access Rights
Managing Call Transfer and Greetings
466
Allowing Screening and Hold Options
Changing Maximum Greeting Length
Enabling and Disabling Greetings
Modifying Caller Input Options
465
466
467
468
469
Managing Message Access, Notification, and Indication
Allowing Subscribers to Send to Distribution Lists
472
473
Allowing Messages Deleted from the Phone to Be Saved in the Deleted
Items Folder 474
Enabling Live Reply for a Subscriber
Creating Private Lists
474
475
Configuring Message Notification
Adding Alternate Extensions
Adding Alternate Names
476
480
482
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Configuring Cisco Unified Communications Manager and Unity Connection: A Step-by-Step Guide
Assigning External Service Accounts (Unity Connection Only)
Add SMTP Proxy Addresses (Unity Connection Only)
Changing Maximum Outside Caller Message Length
Adjusting Urgent Message Marking
484
Enable MWI on Another Extension
485
483
484
Adding and Removing Users from a Distribution List
Conversation Management Settings
483
486
487
Changing Menus from Full to Brief
487
Changing How a User Searches for Other Users
488
Changing What Message Count Is Played to a User
489
Changing the Order in Which Messages Are Played
490
Changing What Header Information Is Heard While Listening to
Messages 492
Summary
Chapter 9
493
Call Management
495
Understanding Call Flow
495
Call Flow Architecture
496
Call Handler Overview
497
Creating Basic Call-Routing Systems
Call Handlers
499
500
Creating and Configuring Unity Call Handlers
Configuring Unity Call Handlers
Profile Settings
500
502
502
Call Transfer Settings
Greetings Settings
504
507
Configuring Call Handler Caller Input Settings
Configuring Call Handler Messages Settings
510
513
Creating and Configuring Unity Connection Call Handlers
Configuring Unity Connection Call Handlers
Configuring Call Handler Basics Settings
Configuring Call Handler Transfer Rules
Configuring Call Handler Caller Input
Configuring Call Handler Greetings
516
517
518
520
522
Configuring Call Handler Post-Greeting Recording
Configuring Call Handler Messages Settings
Configuring Call Handler Owners
526
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525
514
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Directory Handlers
527
Configuring Unity Directory Handlers
528
Directory Handler Search Options Settings
529
Directory Handler Match List Options Settings
Directory Handler Caller Input Settings
531
533
Configuring Unity Connection Directory Handlers
Unity Connection Directory Handler Greeting
Configuring Auto-Attendant
540
Creating Advanced Call-Routing Systems
Using Interview Handlers
534
539
542
543
Creating and Configuring Interview Handlers in Unity
543
Creating and Configuring Interview Handlers in Unity Connection
Creating an Audio Text Application
549
Remotely Managing Call Handlers
Configuring Call Routing
551
552
Creating and Configuring a Call Routing Rule in Unity
553
Creating and Configuring a Call Routing Rule in Unity Connection
Managing Restriction Tables
560
Configuring Unity Restriction Tables
561
Configuring Unity Connection Restriction Tables
Summary
Chapter 10
563
565
Implementing Unity Networking
Unity Networking Overview
567
Networking Components
568
Locations
567
568
Message Addressing
568
Network Subscribers
568
Voice Connector
568
Interoperability Gateway
Schema Extensions
546
569
569
Unity-to-Unity Networking Overview
569
Unity–to–Legacy Voicemail Networking Overview
Unity Networking Configuration
Defining Digital Networking
571
571
Unity to Non-Unity Networking Concepts
Defining AMIS Networking
577
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570
557
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Configuring Cisco Unified Communications Manager and Unity Connection: A Step-by-Step Guide
Defining VPIM Networking
578
Defining Bridge Networking
580
Unity Connection Networking Overview
581
Networking Unity Connection to Unity Connection
Networking Unity Connection to Unity
583
Networking Unity Connection to Other Systems
Summary
Chapter 11
586
Exploring Unity/Connection Tools
Using Unity Tools
Monitoring
Reports
587
587
Unity Web-Based Tools
587
588
592
Subscriber Reports
System Reports
593
595
Using Advanced Tools
599
Administration Tools
600
Audio Management Tools
Diagnostic Tools
608
609
Reporting Tools
612
Switch Integration Tools
613
Using Unity Connection Tools
614
Unity Connection Administration Tools
Task Management
616
Custom Keypad Mapping
617
Migration Utilities
618
Grammar Statistics
618
SMTP Address Search
Show Dependencies
619
619
Unity Connection Reports
619
Phone Interface Failed Logon Report
Users Report
622
Message Traffic Report
Port Activity Report
622
622
Mailbox Store Report
Dial Plan Report
614
615
Bulk Administration Tool
622
623
Dial Search Scope Report
582
623
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585
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User Phone Login and MWI Report
User Message Activity Report
Distribution Lists Report
User Lockout Report
623
623
623
623
Unused Voice Mail Accounts Report
Transfer Call Billing Report
624
624
Outcall Billing Detail Report
624
Outcall Billing Summary Report
Call Handler Traffic Report
624
624
System Configuration Report
625
SpeechView Activity Report By User
625
SpeechView Activity Summary Report
Summary
625
626
Part III
Leveraging the Power of Communications Manager and Unity
Chapter 12
Maximizing CUCM and Unity/Connection
627
Advanced Communications Manager Features
627
Configuring Administrative Rights
Time-of-Day Routing
627
631
Creating a Time Period
632
Creating a Time Schedule
633
Assigning a Time Schedule to a Partition
Hunt List
634
635
Creating a Line Group
635
Creating a Hunt List
637
Creating Hunt Pilots
639
Advanced Unity/Unity Connection Features
Enabling Call Queuing
642
Configuring Destination Call Screening
Unique Solutions
642
643
644
Enhanced Vacation Schedules
644
Configuring Unity/Connection as a Meet-Me
Conference Manager 647
Managing Multilocation Overlapping Extensions
Summary
Appendix
649
Additional Reference Resources
Index
651
657
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Configuring Cisco Unified Communications Manager and Unity Connection: A Step-by-Step Guide
Icons Used in This Book
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Ring
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Command Syntax Conventions
The conventions used to present command syntax in this book are the same conventions
used in the IOS Command Reference. The Command Reference describes these conventions as follows:
■
Boldface indicates commands and keywords that are entered literally as shown. In
actual configuration examples and output (not general command syntax), boldface
indicates commands that are manually input by the user (such as a show command).
■
Italics indicate arguments for which you supply actual values.
■
Vertical bars (|) indicate separate alternative, mutually exclusive elements.
■
Square brackets [ ] indicate optional elements.
■
Braces { } indicate a required choice.
■
Braces within brackets [{ }] indicate a required choice within an optional element.
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xxi
Introduction
On March 10, 1876, Alexander Graham Bell made the first successful telephone call. As
with many things, the test was purely accidental. Graham spilled acid on his leg, and
Watson, his assistant, heard his call for help through the telephone. So, what has changed
over the last 129 years? It would be easier to discuss what hasn’t changed. The world of
telephony has undergone some significant changes but none as exciting as Voice over IP
(VoIP) solutions from Cisco. There are still those who believe we were all a lot better off
in an analog world, but you can’t stop progress, and the Cisco Unified Communications
solutions are starting to grow faster than many had believed.
This new technology brings with it the need for individuals to learn how it works.
Although there are many fine Cisco Press books on this technology, I noticed many of
my students requesting a task-oriented book. They were looking for a book in which they
could look up a specific task and be walked through it. This was the initial goal of the
book. Through the writing process, the book evolved from offering only a step-by-step
guide into also offering easy-to-understand explanations for many of the Cisco Unified
Communications concepts and components.
Goals and Methods
New technologies bring new opportunities and challenges. One of the challenges that we
are faced with in the Cisco Unified Communications world is the capability to easily
understand the many facets of the configuration and integration process. Because this
platform can be deployed in so many different configurations and environments, system
administrators and system engineers need a resource that offers quick access to step-bystep solutions. In an environment such as this, it is nearly impossible to keep track of the
exact steps for each configuration task. Those tasks that you do on a daily basis are easy
to perform, but when you are called upon to perform unfamiliar tasks, you don’t always
have the time to learn the proper steps. Configuring Communication Manager and
Unity Connection shows readers how to complete many of the common tasks, and
some not-so-common tasks, performed within a Cisco Unified Communications solution.
Who Should Read This Book
The book is aimed at individuals who are required to configure Communications
Manager and Unity and Unity Connection solutions as a primary part of their jobs. The
book is unique because it covers Communications Manager, Unity, and Unity
Connection.
Although this book focuses on the tasks that must be performed, it also offers easy-tounderstand explanations for many of the technologies that are commonly found with
Cisco Unified Communications environments, which makes it an excellent resource for
individuals who are new to this technology.
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How This Book Is Organized
Within the book, tasks are organized in the same order in which they would naturally be
performed. Some tasks include cross-references to prerequisite tasks. Whenever possible,
however, all tasks are presented within the same section.
Different people, depending on their knowledge and background, will use this book in
different ways. Many will find it a useful reference tool when completing an unfamiliar
task, and those new to this technology will find that reading this book from cover to
cover will help them gain a solid understanding of this technology. Although the step-bystep guides were written with the assumption that you have access to a Communications
Manager while reading the steps, this is not required. This book includes numerous screen
shots, which enable you to see what is happening in the administration interface even if
you do not have access to a Communications Manager.
Chapter 1 offers you a high-level overview of most of the concepts and components that
are found within Communications Manager and Unity. Basically, the information found in
two weeks of classes has been compressed to quickly bring you up to speed. This by no
means is a replacement for these classes—just a quick overview.
Chapters 2 through 6 cover Communications Manager configuration, whereas Chapters 7
through 11 discuss Unity and Unity Connection configuration. The last chapter speaks to
more advanced features of both technologies and offers a few ways to leverage the
strengths of both to create a more feature-rich environment.
The following is a brief description of each chapter.
Chapter 1: CUCM and Unity Connection Overview
This chapter offers a broad overview of the Cisco Unified Communications solutions to
ensure that you are comfortable with what follows in the book. The intent of this chapter is
to offer you an overview of the various components of a Cisco Voice over IP solution. You
are strongly encouraged to refer to suggested reference material for additional information on
any topic with which you might be unfamiliar. You can find this material in the appendix.
Chapter 2: Preparing CUCM for Deployment
To ensure a smooth deployment, tasks must be performed in a certain order. In this chapter, you learn what tasks must be completed before adding devices. As with most things,
if you fail to create a solid foundation, you will encounter problems in the future. This
chapter ensures that the proper foundation is created and future problems are avoided.
Topics covered include services configuration, enterprise parameters, and device registration tasks. Additionally, this chapter includes step-by-step instructions for each task.
Chapter 3: Deploying Devices
After the predeployment tasks are completed, you are ready to add devices. This chapter
focuses on the tasks required to add various devices to your Communications Manager
environment. Devices have been divided into two major categories: clients (IP phones,
softphones, and so on) and gateways. The chapter includes step-by-step instructions for
adding each device.
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Chapter 4: Implementing a Route Plan
Before you can place calls to destinations that are not directly connected to your
Communications Manager environment, you must configure a route plan. This chapter discusses all the components of a route plan, such as route patterns, route lists, and route
groups and the tasks that are needed to implement an efficient dial plan. The step-by-step
tasks show how to create and configure route patterns, route lists, and route groups and
more advanced components, such as CTI route points, translation patterns, and route filters.
Chapter 5: Configuring Class of Service and Call Admission Control
After a dial plan is created, you might want to limit what destinations certain devices can
reach. This chapter discusses how to do this by configuring Calling Search Spaces and
partitions. It is also necessary that some types of Call Admission Control be deployed on
WAN links so that the quality of voice is maintained. To this end, Call Admission Control
features are covered. Finally, this chapter discusses the importance of special services,
such as 911, and describes how to properly configure the dial plan to handle these types
of calls.
Chapter 6: Configuring CUCM Features and Services
After basic call-processing functions are configured and working properly, you need to
add new features and monitor the health of the system. This chapter explores a number
of the features that can be implemented, including IP phone service, media resources, and
Extension Mobility. The need for, and the functions of, SRST is also covered in this chapter. Furthermore, this chapter examines some of the monitoring services that are included
in Communications Manager. Step-by-step instructions that explain how to add each feature and service are included.
Chapter 7: Unity Predeployment Tasks
The first step to proper configuration is verifying that the integration is correct and that
all predeployment tasks are complete. This chapter includes step-by-step instructions for
completing predeployment tasks, such as verifying integration, defining system parameters, and creating templates, distribution lists, and CoS.
Chapter 8: User Reference
After a proper integration between Unity/Connection and Communications Manager is
achieved and the predeployment tasks discussed in the previous chapter are completed,
the user can be added. In this chapter, the different types of users are examined. Then,
the process for adding, importing, and managing users is explored. Within the “Managing
Users” section, various administrative tasks are discussed, which range from “How to
reset a user’s password” to “How to properly remove users.” Each task includes step-bystep instructions.
Chapter 9: Call Management
One of the system’s most useful and often underutilized features is call management. This
chapter ensures that the reader understands the way that the system processes a call. The
most basic object of the call management system is a call handler. A brief review of how
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call handlers work is included in the beginning of this chapter. Additionally, a common
use of the system’s call management feature is to deploy a basic auto-attendant, which is
described within this text. The chapter also addresses some of the more advanced call
management features, such as call routing rules and audio-text applications. Complete
step-by-step instructions are included within this chapter.
Chapter 10: Implementing Unity Networking
Because many organizations are migrating to Unity/Connection from a voicemail system
or have other voicemail systems deployed at other locations, Unity/Connection must
communicate with them. Unity can be integrated with these systems through a number
of industry-standard protocols. This chapter discusses the different types of networking
that can be deployed and looks at how to determine the proper one to use.
Chapter 11: Exploring Additional Tools
Although most day-to-day tasks can be accomplished using the system administrative
interface, it is often more efficient to use one of the many tools that are included with
Unity/Connection. The tools help accomplish tasks that range from making bulk user
changes to migrating users to another server. This chapter introduces the reader to these
tools and includes step-by-step details on how to use each of them.
Chapter 12: Maximizing CUCM and Unity Connection Capabilities
As Communications Manager and Unity/Connection evolve, more and more advanced
features are added. This chapter looks at a few of these more advanced features, such
time-of-day routing and call queuing. In addition, the chapter offers a few examples of
features that can be created by taking existing features of each application and adding a
new twist to them, such as using Unity as a conference manager.
Target Version
This book was written for Communications Manager, Unity, and Unity Connection versions 8.0 and 8.5. This is not to say that you must run any of these versions for this book
to be of value to you. It does, however, mean that some of the step-by-step guides might
be slightly different. With each new version, the menus are sometimes moved or slightly
changed, or there might be an additional field in the new version. However, none of these
issues should cause you great concern. If the field isn’t there, don’t worry about it. If a
menu isn’t exactly where you expect it, just look above or below, and you are sure to find
it. Including the exact steps for every version of these applications would have made the
book larger than you would care to lift, let alone read. Remember that the value of this
book goes beyond the step-by-step guides, because it also provides easy-to-understand
explanations of many Cisco Unified Communications concepts.
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CUCM and Unity
Connection Overview
Before embarking on any worthwhile adventure, it is important that you have a good map
and a solid understanding of the purpose of your trip. This chapter provides just that—an
introduction to some of the many components that make up a Cisco Unified
Communications environment.
Technical books can be divided into one of two categories: “why” books and “how-to”
books. Why books provide you with a solid understanding of the technology and explain
why you would want to deploy it. How-to books tell you how to deploy a given technology. This is a how-to book. The main purpose of this book is that of a configuration reference. However, it is important that you have a solid understanding of the technology. This
chapter provides you with a broad overview of this technology and references to further
information. If you are new to this technology, you are strongly encouraged to pursue
more in-depth information than is presented in this chapter before deploying this technology. If you haven’t been involved in this technology for a while, you might be thinking of
skipping this chapter and moving on to the meat of the book. This, of course, is your
decision, but reading this chapter can give you a better understanding of the specific
technologies discussed later in this book.
After reading this chapter, you should have a high-level understanding of the
Communications Manager, Unity, and Unity Connection components and how they fit
into a Cisco Voice over IP (VoIP) solution. This chapter has been divided into the following sections:
■
Reliable foundation
■
Communications Manager overview
■
Unity overview
■
Unity Connection overview
■
Security concerns
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Because this technology is a mixture of two preexisting technologies, traditional telcom
and traditional data, it is likely that you started out solely in one of these disciplines.
Often when you start to learn a new technology, you try to compare it to technologies
you’ve learned. This sometimes causes learners to miss an important point because they
were preoccupied with trying to make this new information fit in with previous learning.
If you are new to this technology, you should take any current knowledge you have and
place it aside while reading. After you have read this chapter and feel that you understand
it, you should then integrate it with your current knowledge base.
At first, this can be difficult because we all seem to want to fall back on what we already
know. So each time you find yourself doing this, just stop reading for a moment and refocus on acquiring new information, knowing that later you can integrate it with what you
already know. Also, try not to make judgments while reading. Many times people have
made up their minds about a product or technology before they have even seen it. Even if
you are learning this technology because “you have to,” be as open to it as possible.
Regardless of any person’s resistance, technology will not stop or even slow down.
Ensuring a Reliable Foundation
Whether you are building a house or a network, a solid foundation is crucial. In a VoIP
network, the foundation is even more crucial because both data and voice will use the
same network. This means that you need to implement an even higher level of redundancy than you feel is necessary in a traditional data network. The term five 9s is used a lot
in the traditional telcom world; this stands for 99.999 percent uptime. The expectation is
that any network that carries voice should be up 99.999 percent of the time. This calculates to just a little more than 5 minutes a year of downtime, not including planned downtime for upgrades and maintenance. You might be saying, “That’s impossible,” but actually it is possible. With the proper planning and design, you can expect to see nearly no
downtime. Make note that I said, “with the proper planning and design.” There have been
a number of VoIP deployments that failed solely because a proper infrastructure was not
implemented. Typically, a VoIP environment is broken into four layers. Each layer plays a
vital role. An example of the devices that are in each layer follows:
■
Clients: IP phones
■
Applications: Unity
■
Call processing: Communications Manager
■
Infrastructure: Switches and gateways
Note Calculations of five 9s varied by telephony vendor and typically discounted issues
that affected a single user or a small group of users. For example, if you lost a desktop
switch and 24 users had no phone service for 3 hours, this wouldn’t count against your
five 9s.
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The foundation of the network is at the infrastructure level, where components such as
switches, routers, and gateways reside. A solid understanding of these components is
needed to design a solution that can withstand common day-to-day problems that arise
on most networks. The discussion begins with a look at these components.
Infrastructure Overview
A properly deployed infrastructure is the key to a reliable network. This section begins
by examining the foundation of the infrastructure. The cable is one of the most often
overlooked components of the network. This is often because it rarely causes problems
after it is installed. Cabling problems normally don’t appear until some new type of technology is added to the network. I remember one client that was running a 4-megabit network with no trouble. When he upgraded to 16 megabits, the network started failing and
he had to rewire the entire network.
Nowadays, twisted-pair Ethernet is installed in most environments. The Cisco VoIP solution is designed with the assumption that twisted-pair Ethernet is installed at each desktop.
One of the common issues that arises with cabling is when the installer takes a few shortcuts. A common shortcut is failing to terminate all the pairs of the cable. The installer
assumes that because Ethernet uses only pins 1, 2, 3, and 6, there is no need to terminate
the others. In most cases, the network can function when cabled this way. The problem is,
however, that such a network is not installed to industry standards, and all Cisco solutions
are based on the assumption that the existing infrastructure is installed according to
industry standards. In an environment such as this, you cannot use the Cisco power patch
panel because it relies on pins 4, 5, 7, and 8 to deliver power to the phone. Ensure that
you have all cabling tested and certified before the deployment begins. As the saying
goes, “An ounce of prevention is worth a pound of dropped calls” (or something like that).
After you have the cabling under control, you need to look at the equipment to which the
cabling connects. On the one end of the cable you have phones, which is quite straightforward. On the other end, you have the phone plugged into a switch.
Note This discussion assumes that the phone is plugged into a switch, not a hub.
Plugging phones into hubs is not advised because all devices on a hub share the same
bandwidth, and this can lead to poor voice quality. In addition, do not daisy-chain phones
(plug one phone into another).
When deciding which switch to use, a few things must be considered. First, it is recommended that all switches you plan to use within the Cisco VoIP solution are Cisco
switches. This is not simply because Cisco wants to sell more switches, but because certain Cisco switches include special features that allow greater functionality within your
network. These features include inline power, voice virtual LANs (VLAN), and Cisco
Discovery Protocol (CDP) support. This does not mean that switches from other manufacturers cannot be used. It simply means that some features discussed in the following
sections might not be supported.
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Inline Power
This is the ability to provide power to the phones through the Ethernet cable. There are
two inline power schemes:
■
The Cisco inline power convention, which uses pins 1, 2, 3, and 6 to provide power.
These same wires are used for the transfer of data.
■
Power over Ethernet (PoE), as defined in IEEE Standard 802.3af. This is an approved
industry standard, which uses pins 1, 2, 3, and 6 or 4, 5, 7, and 8 to provide power.
The standard differs from the Cisco inline power scheme in a number of ways, the
most significant being the way that power requirement is detected. Current Cisco
phones support the 802.3af standard, but keep in mind that older models might not.
The net effect of either standard is the same—power is supplied to the phone through the
Ethernet cable. The switches that support either of these conventions can detect whether
the attached device requires inline power, and if the device does require inline power, the
switch provides it. Having two methods of supplying inline power can be confusing, so it
is best that the phones and switches you purchase use the same method.
Voice VLANs
This allows the use of a single switch port to simultaneously support both a phone and a
PC by allowing a single port to recognize two VLANs. The PC is plugged into the back
of the phone, and the phone is plugged into the switch. The switch then advertises both
VLANs. The phone can recognize the voice VLAN and use it. PCs cannot recognize
voice VLANs and use the native VLAN.
CDP Support
CDP is a Cisco-proprietary protocol that allows Cisco equipment to share certain information with other Cisco equipment. The phones use CDP to determine whether a voice
VLAN is present on that port. It also shares other information such as port power information and quality of service (QoS) information with the Cisco Catalyst switch.
Make sure that the switch you choose supports these features. Most currently shipping
Catalyst switches are capable of supporting all these features.
Voice Gateways
After you ensure that the cabling and switches are adequate for a VoIP solution, you are
ready to deploy the endpoints. Endpoints can be any of the following: phones, soft
clients, or gateways. Of these devices, only gateways are considered to reside at the infrastructure level. Phones and Communications Managers are covered later in this chapter.
In its simplest form, a gateway is a device that allows connectivity of dissimilar networks.
In the VoIP world, a gateway connects the Communications Manager voice network to
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another network. The public switched telephone network (PSTN) is the most popular network with which IP phones must communicate. The job of the gateway is to convert the
data traveling through it to a format that the other side understands. Just as a translator is
needed when a person speaks German to someone who understands only Spanish, a gateway is needed to convert VoIP to a signal that the PSTN understands.
The hardware that acts as a gateway varies, depending on what type of network you connect to and what features you require. When choosing a gateway, ensure that it supports
the following four core gateway requirements:
■
DTMF relay: Dual-tone multifrequency (DTMF) are the tones that are played when
you press the dial pad on a phone. Many people refer to this as touch tones. Because
voice is often compressed, the DTMF can become distorted. The DTMF relay feature
allows the DTMF to be sent out of band, which resolves the distortion problem.
■
Supplementary services: Include hold, transfer, and conferencing.
■
Cisco Unified Communications Manager (CUCM) redundancy: Supports the capability to fail over to a secondary Communications Manager if the primary
Communications Manager fails.
■
Call survivability: Ensures that the call will not drop if the Communications
Manager, to which either endpoint is registered, fails.
Later, the various types of gateways are discussed. For now, understand the purpose of a
gateway and the required features.
Creating a Reliable VoIP Infrastructure
In the summer of 2003, the northeastern portion of the United States experienced a
widespread power outage. The power outage lasted from 6 hours to 3 days depending
upon the area. One of the most impressive and yet understated events that occurred during this time is what didn’t happen. For the most part, the PSTN didn’t fail, and no one
even noticed. Because no one actually noticed shows how much people expect the
phones to always work. The power was out, and yet most people didn’t think for a second
that the PSTN might fail. The system didn’t fail because of the highly reliable and redundant infrastructure that has been developed over the years. This is the type of reliability
that people have come to expect from the phone system. It has been stated that many
people view dial tone as a God-given right, or even one of the inalienable rights in the
constitution. (I doubt anyone thinks that, but you get the idea.) With this in mind, you
must make every effort to ensure that nothing short of a natural disaster prevents your
customer from having dial tone.
The most important thing to keep in mind is that individual components of the system
will fail. It is not a question of if something will fail, but when. Because components will
fail, it is up to you to determine how to prevent the failure from affecting dial tone. This
is done during the design phase of the project.
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Note The design phase is perhaps the single most important part of any deployment.
Countless times I’ve had panicked customers, whom I inherited from other integrators,
calling me with problems that could have been averted if dealt with during the design
phase. Often when I ask clients how this problem was dealt with during the design phase,
they answer, “What design phase?” Make certain that you cover as many foreseen and
unforeseen eventualities as possible during the design phase. Although your customers
might never see how good you are at fixing a system when it fails, they will know how
good you are because it doesn’t fail.
Redundancy is the core component in a reliable infrastructure. The system design should
include redundancy at every level. This starts in the wiring closet.
Reducing the cable infrastructure and allowing ease of cable management are two of the
motivating factors for migrating to a VoIP solution. Therefore, it does not make sense
that redundancy is extended to the cable level. Remember, your goal is to achieve the
same level of reliability that people expect from a phone system. People understand that
if there is a cabling problem, the phone won’t work. This is one of the few acceptable reasons for a phone system to fail. So, as far as the cabling goes, you just need to ensure that
the existing network cabling infrastructure is certified as previously mentioned.
Switches are the next piece of the infrastructure that needs to be considered.
Redundancy at the switch level is nothing new. Although redundancy has always been
encouraged in data networks, it is no longer just a suggestion; it is required to achieve the
expected level of reliability. In smaller environments that might have only a single switch,
redundancy at the switch level doesn’t apply; however, in large networks, make sure that
you design a highly available network by building redundancy in at the core and distribution switch level. This means that there will be multiple paths a packet can take to get to
its destination. Because of a protocol called Spanning Tree Protocol (STP), only one path
is available at any given time. STP ensures that if a link fails, an alternative path will be
opened. To find out more about STP, see the additional references listed in the appendix
“Additional Reference Resources.”
A redundant path can ensure that a packet reaches its destination, but it is also important that it gets there in a timely manner. Voice traffic does not handle delay very well.
If too much delay is introduced, the quality of the conversation tends to degrade rapidly. You have probably noticed the effect delay can have on a conversation when watching a TV news reporter through a satellite link. It seems to take the reporter a few seconds to respond to a news anchor’s question. This is because there is a several-second
delay between the time the question is asked and the time when it reaches the reporter’s
destination.
Many things can affect the delay that is introduced into a conversation. One of the most
common is the competition between voice and other traffic for bandwidth. To help alleviate this, QoS must be implemented within the network. QoS gives certain traffic priority
over other traffic. The proper configuration of QoS is essential for any network that has
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both voice and data on the same wire. A detailed discussion on QoS is beyond the scope
of this book. Refer to Appendix A, for suggested references on this subject.
Before leaving the wiring closet, one more thing requires attention: power. Remember
that a power failure is not an acceptable reason to lose dial tone. A power outage is not
necessarily an acceptable reason for data networks to fail. There was a time when people
expected and accepted the loss of data during a power outage. They were never happy
about it, but they weren’t surprised either. Nowadays with the reasonable price of uninterruptible power supplies (UPS), data networks are no longer as susceptible to power
outages as they were in the past. It is nearly unheard of not to have a UPS on file servers
and, in many cases, throughout the network. Switches are no exception. As with any
equipment, you need to do some research to determine the proper size of the UPS you
need. To do this, determine the amount of power that the switch draws and then determine the amount of time you want the switch to run without power. You don’t need to
worry about redundant power at the phone if you use inline power. Keep in mind that the
more phones that draw power from the switch, the larger the UPS you need.
As mentioned previously, gateways are also considered part of the infrastructure.
Therefore, whenever possible, redundancy should be included at the gateway level. In
some cases, such as an environment that has only a single trunk from the PSTN, redundancy is not feasible. If the environment has other Cisco routers, try to use the same
model router for your PSTN gateway. This way, if the PSTN gateway does fail, you might
swap equipment for a short-term solution or, at the very least, use the other router for
testing purposes after hours. If you do have multiple trunks, it is a good idea to have at
least two physical gateways connecting the network to the PSTN. A level of redundancy
can be added by using multiple service providers. For example, if you have two trunks,
use a different service provider for each. This way, if either of the service providers has a
widespread outage, the other trunk will still be functional.
This section dealt with the reliability of the infrastructure. This is only a portion of the
solution that must be considered when implementing a reliable system. A system is only
as good as its weakest link, so you need to ensure that the entire system is designed with
the same goal in mind—“Don’t affect dial tone.” In the next section, you look at the callprocessing layer, more specifically, the Communications Manager.
Communications Manager Overview
In the previous section, the infrastructure was discussed, and you learned what was necessary to create a solid foundation on which to build the rest of the system. As when
building a house, you can move to the heart of the project after the foundation is set.
The Communications Manager is considered the heart of the Cisco Unified
Communications solution. It is responsible for device registrations and call control.
Communications Manager is an application that runs on a media convergence server
(MCS). Often the term Communications Manager is used to refer to the physical device
that the application runs on, but the hardware should be referred to as the MCS.
Communications Manager is the software running on the hardware.
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Configuring Cisco Unified Communications Manager and Unity Connection: A Step-by-Step Guide
Note Cisco has certified certain servers for use as MCSs. Currently only certain HP and
IBM servers are certified. Servers can be purchased from Cisco or directly from IBM or
HP. The different platforms offer various features, such as redundant hard drives and power
supplies. Be sure to take this into consideration when choosing a server. Keep in mind that
not all IBM and HP servers are approved, so be certain to check with Cisco to ensure that
the server you choose is approved. Also be aware that the HP servers that are supported
are older models, and Cisco does not plan on approving newer HP models. Many integrators choose to purchase the MCS from Cisco to have a single-vendor solution.
Every system should have at least two Communications Managers, and the two are
referred to as a Communications Manager cluster. The exception to this rule is if you run
Communications Manager Business Edition. Later in this chapter, you learn why a minimum of two Communications Managers is strongly recommended. Based on the previous
section, you might guess for yourself. Does the word redundancy come to mind?
Defining Communications Manager Components
Communications Manager is responsible for all device registration and call control. Much
of the configuration is performed through the Communications Manager administrative
interface. This section introduces you to the various components of Communications
Manager and the devices that it controls.
Most configuration and administration is performed through Communications Manager’s
web browser interface. Using this interface, you can configure phones, add users to
Communications Manager’s directory, define the dial plan, and perform various other
tasks. The majority of the tasks that you learn how to perform later will be done using
this interface. The interface is fairly simple to navigate, and after a short time, most people are quite comfortable using it. It is important to remember that it is a web-based interface and hence might sometimes not be as responsive as you would expect. The delays
are more noticeable when you access a remote Communications Manager over a WAN
link. Each evolution of this interface improves the end-user experience, and nowadays it
is much more enjoyable to use than in years past.
All the information that you enter through the web interface must be stored.
Communications Manager uses IBM Informix to store this information. All configuration
information is stored in this database.
As mentioned earlier, each Communications Manager cluster should have at least two
Communications Managers. The reason for this is redundancy. Remember, the system
needs to deliver the same level of reliability that people are used to with a traditional
phone system. Having multiple Communications Managers also provides a more scalable
system, which will be explained shortly. For now, the focus is on the role that the various
Communications Managers play in regard to the database. A Communications Manager
is referred to as either a publisher or a subscriber. Each Communications Manager cluster
has only one publisher. All other Communications Managers within that cluster are
referred to as subscribers.
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The job of the publisher is to maintain the most current copy of the database. Whenever
anything is added to the database, the information is sent from the publisher to all the
subscribers. The data is never written to the subscribers first and then transferred to the
publisher.
So far, we have discussed only the roles that the Communications Managers play in the
database. The other job of the Communications Manager is device control. All devices
register to a Communications Manager. This Communications Manager is known as that
device’s primary Communications Manager. Each device also has a secondary
Communications Manager that it can register to if the primary fails. Devices use a subscriber as their primary Communications Manager. This leaves the publisher alone so that
it can take care of its main responsibility, which is to maintain the database. In some
cases, a device can have a tertiary server to which it can fail over if both the primary and
secondary are not available. In most cases, primary, secondary, and tertiary
Communications Managers should be subscribers.
If the primary Communications Manager fails, the device registers to the secondary. The
device registers with the secondary Communications Manager only if it is not on a call
when the Communications Manager fails. If the device is active when the Communications
Manager fails, it registers with the secondary Communications Manager when the call
ends. In most cases, a call stays up even if the participating Communications Manager
that is controlling devices in the call fails. The reason is that during a call, the communication is point to point, meaning that the Communications Manager is not involved with
the actual voice stream. The device has no idea that the Communications Manager has
failed because it does not communicate with the device again until either the call is over
or a feature that requires Communications Manager is invoked, such as hold or transfer. If
a device whose Communications Manager has failed tries to invoke such a feature, the
phone display indicates a Communications Manager failure. The feature either fails or is
unavailable (grayed out), depending on phone type. The call itself is not affected. A message also appears on the phone stating that the Communications Manager is down and
the feature is not available.
Note In certain cases, the failure of a Communications Manager could cause the call to
drop. One example is if the call were connected through a Media Gateway Control
Protocol (MGCP) PRI.
In small environments where there are only two Communications Managers, it is acceptable to use the publisher as a secondary Communications Manager. If you have more than
1250 users, it is not recommended to use the publisher as a secondary Communications
Manager. Figure 1-1 shows a typical Communications Manager environment that can support up to 5000 phones. This figure is an example of what is referred to as one-to-one
redundancy. In this configuration, 2500 phones register to Communications Managers B
and C. Communications Managers D and E are secondary servers for these phones.
Communications Manager A is the publisher, and no phones register to it. This example is
based on the assumption that all the servers on which the Communications Manager is
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loaded are MCS-7835s or equivalent. Other server models support a different number of
phones per server. For example, the MCS 7845 supports up to 7500 phones per server.
A
Publisher
B
2500
Phones
Primary/Subscriber
D
Secondary/Subscriber
C
2500
Phones
Primary/Subscriber
E
Secondary/Subscriber
Figure 1-1
One-to-One Redundancy Communications Manager Cluster
Communications Manager Business Edition
Cisco offers a version of Communications Manager that is called Communications
Manager Business Edition (CMBE). It is just like the standard Communications Manager
with a few limitations. CMBE can only support up to 575 phones and does not support
redundancy. In other words, there are no subscribers and only one publisher. It comes
packaged with Unity Connection, which runs coresident (on the same server). This is an
excellent “all in one” solution for small- to medium-size companies. All administration
tasks and features configurations are performed the same as they are on the standard
Communications Manager. They both use the same administration interface, and the software architecture is identical.
Communications Manager Devices
A large number of devices register with a Communications Manager, but they typically
fall into one of the following categories:
■
Phones
■
Gateways
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■
Gatekeeper
■
Media resources
Each of these devices has its own unique role within the Communications Manager environment, and this section briefly describes each one. For more information on these
devices, refer to Appendix A.
Phones
A number of different model phones can be used in a Communications Manager environment. Table 1-1 lists and describes some of the more popular models.
Table 1-1
Phone Models Used in a Communications Manager Environment
Phone
Description
Model 7911
The 7911 is a single-line, entry-level phone. It offers a switch port in the
back to which you can attach a PC. It only supports audio and text
eXtensible Markup Language (XML) support.
Model 7942/7945
These phones support two lines and XML applications. These are considered midrange phones and are typically used in environments where
two lines are adequate. They also have a switch port in the back for
attaching a PC. The main difference between these phones is that the
7945 offers a color screen and the 7942 is only grayscale.
Model 7962/7965
These phones support four lines and XML applications. They also have
a switch port in the back for attaching a PC. The main difference
between these phones is that the 7965 offers a color screen while the
7962 is only grayscale.
Model 7975
This is an eight-line phone that offers a touch screen; this allows you to
invoke certain features by touching the screen. It also supports XML
applications.
Model 7925
The 7925 is a color-screen wireless phone that connects to the network
through a wireless access point. The phone’s shape and size are similar
to a cell phone, but it only works in a Communications Manager environment.
IP Communicator
The Cisco IP Communicator is an application that runs on a PC and
allows the PC to be used as a phone. Typically, a headset is attached to
the PC, and the user can make and receive calls using the PC.
6900 Series
This is a series of cost-effective phones that work well for companies
that are switching from a traditional analog phone system. While not as
feature-rich as other phones, they offer a lower-cost entry yet still provide access to many Cisco Communication Manager’s features.
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Table 1-1
Phone Models Used in a Communications Manager Environment
Phone
Description
8900 Series
The 8900 series is a newer series of phone from Cisco that offers highdefinition voice in addition to a high-resolution adjustable display. This
series also includes USB ports, which support wired headsets.
9900 Series
This series is very similar to the 8900 but also supports a directly
attached USB camera for video calls and Bluetooth. The 9971 also
offers Wi-Fi.
Note One of the advanced features of many Cisco IP Phones is their capability to parse
XML. These phones have an LCD screen on which the user can look up others in the
directory, receive messages, log in and out of services, and perform many other functions.
Through the use of XML programming, many companies have developed a variety of
applications such as time clocks and inventory lookup.
Gateways Overview
As mentioned earlier, gateways are used to connect dissimilar systems together, such as
connecting Communications Manager to the public switched telephone network (PSTN).
The core requirements were discussed earlier, so this section examines the different types
of gateways and how they communicate, meaning the protocol they use. There are three
main protocols that are used today for communicating between Communications Manager
and gateways. They are Media Gateway Control Protocol (MGCP), H.323, and Session
Initiation Protocol (SIP). These are industry-standard protocols and offer similar features.
The type and number of trunks that a customer has also affects the type of gateway you
select. Communications Manager is connected to the PSTN using either an analog or a
digital trunk. The trunk used also affects the type of equipment you use for the gateway.
Gateways differ in interface types and capacities. If analog trunks are used, typically a
Foreign Exchange Office (FXO) port is used for each line. With analog lines, each call
takes up a port on a gateway. This is not always a practical solution in a large environment. Typically, a T1 or E1 line is used to connect to the PSTN if a company needs more
than a few lines. These types of trunks are normally more cost-effective if more than
eight simultaneous connections to the PSTN are required.
So far, only using gateways as a way to connect to the PSTN has been discussed.
Gateways are also needed to connect Communications Manager to traditional phones
systems. In many cases, customers choose to integrate the Communications Manager
into their existing voice solution and slowly replace the traditional PBX. This is done by
connecting the Communications Manager to the traditional PBX through either analog or
digital interfaces. The interface used depends on the volume of traffic expected to travel
between the two phone systems and the interface available. For environments that expect
large volumes of calls to travel between the phone systems, a T1 or E1 line is used. Figure
1-2 shows how this integration might look.
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Cisco
IP Phones
CallManager Cluster
Router w/T1
Interface
V
Switch
Ethernet Cable
T1
PBX Handset Lines
PBX
Figure 1-2
Communications Manager–to–PBX Integration
Note The physical connection used to connect a traditional PBX with Communications
Manager through T1 interfaces is a crossover T1 cable from the T1 interface of the traditional PBX to the T1 interface on the Communications Manager gateway.
Gateways are used not only to connect Communications Manager to traditional PBXs
but also to connect multiple Communications Manager environments together. As mentioned earlier, two or more Communications Managers are known as a Communications
Manager cluster. All the IP devices within a cluster can communicate with each other
without a gateway. However, when two Communications Manager clusters need to be
connected, a gateway must be configured. The connection between the two
Communications Manager clusters is called an ICT (Intercluster Trunk). In earlier versions
of Communications Manager, these were configured under gateways. Now they are
referred to as trunks in the configuration menu. Chapter 3, “Deploying Devices,” discusses these gateways more fully.
Gateways are also used to provide analog connectivity within a Communications
Manager environment. Although the goal of VoIP is to use IP to transport voice whenever possible, there are times when an analog connection is required; modems and fax
machines are examples. To connect an analog device such as a fax machine, a Foreign
Exchange Station (FXS) port is required. A gateway with FXS ports allows analog devices
to operate within a VoIP network.
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Gatekeepers
Now that you understand how to connect Communications Manager to the other systems, you need to make sure that the path used to connect to another system does not
become congested. It is not possible to allow more calls than a connection can handle
when connecting to the PSTN or a traditional PBX using analog lines or voice T1s.
However, when connecting devices using an IP connection, oversubscribing is possible.
Oversubscribing occurs when more calls connect across a link than the link can adequately handle. When connecting multiple Communications Manager clusters together,
you can use gatekeepers to prevent oversubscribing. This is referred to as Call Admission
Control (CAC).
A gatekeeper is an H.323 device and typically runs on a router such as a 2800. Hence,
Communications Manager communicates with it using H.323. Figure 1-3 shows a typical
deployment. This diagram shows two Communications Manager clusters connected
through an ICT. The gatekeeper manages the available bandwidth between the sites. The
total allowable bandwidth for voice calls is configured in the gatekeeper.
Detroit
New York
H.323
V
H.323
V
Gatekeeper (HSRP)
DTW Communications Manager Cluster A
NYC Communications Manager Cluster B
Inter Cluster Trunk
Figure 1-3
Gatekeeper
An example call flow would go something like this:
Step 1.
Joe, who resides in Detroit, attempts to place a call to Fred, who resides in
New York. The Communications Manager in Detroit sends a request to the
gatekeeper to see whether there is enough bandwidth to place the call.
Step 2.
The gatekeeper replies with either a confirmation (bandwidth is available) or a
rejection (bandwidth is not available).
Step 3.
If the bandwidth is available, the call setup proceeds and the Communications
Manager in New York is informed that a call is being placed to Fred.
Step 4.
The Communications Manager in New York then sends a request to the gatekeeper to see whether there is enough bandwidth for its side of the call.
Step 5.
If the gatekeeper sends a confirmation, the call setup is complete, and Joe and
Fred can talk about how the Lions actually won a game that weekend.
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Much more occurs during setup. The previous example was presented to help you understand how a gatekeeper enforces CAC.
Assume for a moment that the gatekeeper in the previous example failed. What happens
when the Detroit Communications Manager sends a request to the gatekeeper? Because
the gatekeeper isn’t working, the Communications Manager does not receive a confirmation and the call does not go through. How do you prevent a failed gatekeeper from negatively affecting call setup? The best solution is to have two gatekeepers. Once again, the
theme running throughout this chapter is redundancy. Redundant gatekeepers can be configured by using Hot Standby Router Protocol (HSRP) or gatekeeper clustering. It is recommended that one of these solutions be implemented.
Not only can the gatekeeper be used for CAC purposes, but it can also determine the
location of a requested device. This feature is discussed more fully in the CAC sections
of Chapter 5, “Configuring Class of Service and Call Admission Control.”
There are times when CAC is needed within a cluster. Because the gatekeeper is used
when communicating outside a cluster, some other type of CAC must be implemented.
For example, some type of CAC is needed within a cluster when a number of offices use
the same Communications Manager at a central site.
CAC is accomplished in an IP WAN with centralized call processing deployment by configuring locations in Communications Manager. When configuring locations, the amount
of bandwidth that is available for voice calls is entered in each location. When calls are
placed between locations, bandwidth is deducted from the available bandwidth and calls
are allowed or disallowed based on the available bandwidth. Locations are configured
within Communications Manager, and no additional hardware is required. Often people
ask whether they can use locations instead of a gatekeeper to save money. Remember that
locations are designed to be used when the call is placed between two devices within the
same cluster. It is best to use a gatekeeper for calls placed between separate networks.
Media Resources
To accomplish certain tasks such as conferencing and Music on Hold (MoH),
Communications Manager needs to call upon additional resources. The core
Communications Manager application does not have the capability to perform these
tasks, so it relies on other resources, which are either hardware or software. Some
resources reside on the same server as Communications Manager and others require additional hardware. The sections that follow describe the various resources.
Conference Bridge (CFB)
Conference bridges (CFB) are required for a caller to have a conference call with at least
two other callers. CFBs can be either software or hardware; however, hardware is recommended. Software CFBs run as a process in Communications Manager, whereas hardware
CFBs require additional equipment. Hardware CFBs require digital signal processors
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(DSP). Not all devices that have DSPs can be configured as a CFB. The following devices
can be configured as CFBs:
■
Catalyst 6000 T1/E1 ports
■
Catalyst 4000 Access Gateway Modules
■
Supported Cisco routers with DSP farms
Note The Catalyst 600 and 4000 can no longer be purchased from Cisco.
Transcoders
A transcoder allows devices that are using different codecs to communicate.
Transcoders can change an incoming codec to another codec that the destination device
can understand.
Note A codec (which stands for compression/decompression) is used to express the format used to compress voice. An algorithm is used to compress voice so that it requires less
bandwidth. The two most common codecs used in a Communications Manager environment are G.711 and G.729a.
MoH
Music on Hold (MoH) allows an audio source to be streamed to devices that are on hold.
It is a process that runs on a Communications Manager. The audio can be streamed either
multicast or unicast, and the codec can be configured. Up to 51 audio sources can be
configured, including live audio that is plugged into a sound card (USB device) installed
in the Communications Manager. It is also possible to configure the audio source to be
streamed from a router.
Annunciator
The Annunciator is a software service that runs on the Communications Manager and
provides audio announcements to callers. An example of this is if a caller dialed a number
that Communications Manager could not process, the caller would hear a message something like, “The number you have dialed cannot be reached; please hang up and try your
call again.”
Now that you have an overview of the components required in a Communications
Manager environment, you should review various deployment models. As this technology
matures, the way it is deployed continues to evolve. It is essential that it be deployed
only in a supported fashion. The next section discusses the various supported deployment models.
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Understanding Communications Manager Deployment Models
There are essentially three main Communications Manager deployment models currently
supported by Cisco. Although during the past few years these models have evolved to
create what appear to be new deployment models, all support models fit into one of the
following three categories of sites as described in the sections that follow:
■
Single-site
■
Multisite WAN with centralized call processing
■
Multisite WAN with distributed call processing
Single-Site
In this model, a Communications Manager is deployed within a single building or perhaps
in a campus area. However, some would argue a campus area would typically fall into the
centralized model. In the past, the core theme behind this model was exclusive connectivity. There was no VoIP connectivity to any system outside its own. At present, service
provides are offering SIP trunks that allow you to send calls outside of your system using
VoIP. Any call placed to a destination outside its own is sent to the PSTN or an Internet
Telephony Service Provider (ITSP).
Multisite WAN with Centralized Call Processing
A centralized deployment includes remote locations that have IP phones registering to
the Communications Manager at the main site. Normally, there is only one
Communications Manager cluster in a deployment such as this. Remote phones send all
requests across the WAN to the Communications Manager. If the WAN fails and the
phones have no local device to which to register, the phones are unusable.
Note Technology called Survivable Remote Site Telephony (SRST) has been developed
that allows enables remote offices to have dial tone and make calls across the PSTN if the
WAN or remote network fails. SRST runs on a router such as a 3800. The number of
phones that can register to an SRST system depends on the hardware on which it is running. SRST is discussed in Chapter 6, “Configuring CUCM Features and Services.”
Multisite WAN with Distributed Call Processing
There are multiple sites in this model, each having its own Communications Manager
cluster. There is IP connectivity between them, and ICTs are configured to send the voice
across. A gatekeeper is highly recommended in this deployment to prevent oversubscribing and assist in call routing.
Often companies that need distributed deployment also have remote sites with just a few
phones. A Communications Manager cannot be justified for the remote site, so the
phones register to a remote Communications Manager just as they would in a centralized
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deployment. When centralized and distributed deployments are merged together like
this, it is referred to as a hybrid deployment.
Another form of this deployment is referred to as a Clustering over the IP WAN. In this
scenario, a single Communications Manager cluster is split among multiple sites. For
example, a publisher and a subscriber are in the Detroit office and a subscriber is at the
New York office. The phones within the respective offices register with the local
Communications Manager.
One of the major requirements in a deployment such as this is that, as of
Communications Manager 6.1, the round-trip delay can be no greater than 80 ms (no
greater than 40 ms one delay). The standard one-way delay allowed enabled for voice is
150 ms, so the requirements in this type of deployment are much more stringent.
Figure 1-4 shows an example of a Clustering over the IP WAN deployment.
Publisher
Subscriber
Maximum RTD < 80 MS
Subscriber
Detroit
Figure 1-4
New York
Distributed Single Cluster
The environment in which you deploy Communications Manager typically dictates which
deployment model you choose. Regardless of which model is being deployed, it will most
likely connect to an outside system such as the PSTN. When a call is placed outside the
local system, the Communications Manager must know where to send the call based on
the numbers dialed. The next section discusses how these call-routing decisions are made.
Route Plan Overview
Communications Manager knows about all devices that are registered within the cluster
and knows how to route calls to any destination within the cluster. However, if a call is
placed to a destination outside the cluster, Communications Manager needs to know
where to send the call. This is the purpose of a route plan. A route plan is configured
with Communications Manager to determine where to send calls based on the number
that is dialed.
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A route plan consists of a number of components, which are discussed in this section.
Here Following is a brief description of each of the components that make up a dial plan.
The list is in the order in which these devices must be configured. Later, the actual flow
will be discussed.
Step 1.
Device (gateway): Gateways were explained earlier as devices that connect dissimilar systems. The gateway is the last component in a route plan because it
sends the call to an outside system. As mentioned before, the gateway connects
the Communications Manager system to the PSTN and to other destinations.
Step 2.
Route group: A route group is used to route the call to a gateway. Each route
group includes at least one gateway, but because there is often more than one
gateway, a call can be routed through a route group that can point to multiple
gateways. The order in which the gateways appear in the route group determines which gateway the call is routed to first. If the intended gateway is not
available, the route group sends the call to the next gateway in its list.
Step 3.
Route list: A route list is used to route the call to a route group. Each route
list must have at least one route group in it. Just as a route group can point to
multiple gateways, a route list can point to multiple route groups. The order in
which the route groups appear in the route list determines which route group
the call is routed to first. If all the devices in the first route group are unavailable, the route list routes the call to the next route group in the list. If all gateways in the route groups within the list are unavailable, the call fails.
Step 4.
Route pattern: When a user dials a number, Communications Manager compares the digits dialed against all the patterns it knows. As mentioned earlier,
Communications Manager knows about all devices within the cluster. If the
number dialed matches the number assigned to a device in the cluster,
Communications Manager sends the call to that device. If the call is placed to
a device outside the cluster, a matching route pattern must be configured in
the Communications Manager. Because it is impossible to enter every possible number that might be dialed, wildcards are used to allow a single route to
pattern-match multiple numbers. An example of one of the wildcards used in
Communications Manager is an X. In a route pattern, an X matches any single
digit 0–9. For example, a route pattern of 5XXX would match all numbers
from 5000 to 5999.
Typical Call Flow
Now that you understand the basic components of a dial plan, examining a typical call
flow is in order. Communications Manager in this example has the route pattern
9.1248547XXXX:
Step 1.
A user dials 912485479000.
Step 2.
Communications Manager analyzes the dialed digits and determines that the
closest match it knows of is 91248547XXXX.
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Step 3.
Communications Manager routes the call to the route list that has been configured for the route pattern 91248547XXXX.
Step 4.
The call is then routed to the first route group in the route list.
Step 5.
If there is more than one gateway in the route group, the call is sent to the first
gateway in the list. If the gateway is available, the call is sent out that gateway.
At times, a dialed number matches more than one pattern. When this occurs,
Communications Manager selects the closest match. For example, if a user dials 5010 and
Communications Manager has 5XXX and 50XX as route patterns, 50XX is selected
because there are only 100 possible matches, whereas there are 1000 matches with the
route pattern 5XXX.
Wildcards
Up to this point, the only route pattern wildcard that has been discussed is the X.
Chapter 4, “Implementing a Route Plan,” discusses wildcards further. For now, the following sections present wildcards that can be used in route patterns.
The Wildcard @
This wildcard matches any phone number that is part of a public number plan such as the
North American Numbering Plan (NANP). The easiest way to understand what numbers
are part of NANP is that any number you can dial from a home phone in North America
would be part of the NANP. This includes numbers such as 911 and all international numbers. The NANP is installed by default, but other numbering plans can be
downloaded/installed.
The Wildcard !
This wildcard represents any digit and any number of digits. At first, people think that “!”
is the same thing as the X wildcard, but remember that the X represents only a single
digit. The “!” can represent any number of digits.
The Wildcard [x-y]
This wildcard represents a single digit range. For example, 5[3-5] would match 53, 54, or
55. The numbers inside the brackets always represent a range and match only a single digit.
Note In a pattern such as 5[25-7], the only matches are 52, 55, 56, and 57. When people
first look at this pattern, they think it matches 525, 526, and 527, but remember that the
brackets can only represent a single digit.
The Wildcard [^x-y]
This wildcard represents an exclusion range, that is, any single digit that is not included in
the range matches. For example, the pattern 5[^3-8] matches 50, 51, 52, and 59.
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Calling Privileges
While creating a dial plan, you will find that some individuals within the company are
allowed to place calls that others are not. For example, some companies do not allow all
employees to make long-distance calls. Also, it is highly unlikely that you would want to
allow international calls to be placed from a lobby phone. These issues are addressed by
assigning calling privileges to devices. Calling privileges are configured by creating what
is referred to as a class of control. The term class of service (CoS) was used in previous
versions of Communications Manager, but it is now configured within the class of control area.
Note The terms class of service (CoS) and class of control are often used when talking
about calling privileges. These two terms are synonymous with calling privileges. Those of
you that come from a data background should not confuse this with the QoS component
known as CoS. Although it stands for the same thing, class of service, it does not mean the
same thing. Think of it this way: CoS in the data world is about prioritizing; CoS in the IP
telephony world involves rights and restrictions.
Class of control is essentially a way to determine what destinations a given device can
reach. For example, if you want to call someone in Germany, your CoS would have to
allow international calls. In the same regard, if you want to prevent a device from calling
Germany, you would assign a CoS that does not allow international dialing.
A class of control comprises two components, a Calling Search Space (CSS) and partitions. A solid understanding of these two components is essential to create an effective
class of control. These two components seem to cause some confusion for those who are
new to the concept, but at the core, they are simple concepts.
The simplest way to view this concept is to imagine a partition as a lock and a CSS as a key
chain. If a number has a lock on it, you need the key on your key chain to reach that number. Simple, right? Well, as with anything, as things grow they can become more complex.
Partitions are assigned to anything that can be called, such as a directory number or a
route pattern. Calling search spaces are assigned to anything that can place a call, such as
phones and gateways. The configuration of these components is discussed in Chapter 5.
Taking a closer look at how the components are configured can help clarify the concept.
Figure 1-5 shows how assigning partitions to a pattern and CSS to phones affect dialing
privileges.
The two phones in Figure 1-5, Phone A and Phone B, each have a different CSS that
determines where they can call. They both dial 912485479000. Based on the CSS of the
phone, a match is determined. The only pattern that the dialed digits match is the
9.1248547XXXX pattern, which has the LocalLD partition. Because Phone A’s CSS
allows access to the LocalLD partition, its call goes through. However, Phone B’s CSS
does not allow access to the LocalLD partition, so that call fails.
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CSS = Employees
Phone A
Employees CSS
LocalLD
Local
9.124854XXXX
Partition=LocalLD
CSS = Lobby
x
Lobby CSS
Local
Phone B
Figure 1-5
CoS Example
Note If you feel a little uncertain on this concept, fear not. I have had many students who
worked with CSS and partitions for years and still didn’t have a handle on it. The best advice
is to start small, such as the example in Figure 1-5, and build on that. Although the design
can become complex, the way that CSS and partitions work does not change and is simple.
As you can see, the components that are required to implement a Cisco Communications
Manager solution are quite extensive. This section has presented only a high-level
overview. By now, you should understand the importance of a solid infrastructure and be
able to describe the role that redundancy plays in providing reliable service. Furthermore,
you should be comfortable with the various devices that can register to Communications
Manager and the various deployment models of Communications Manager. Finally, you
should understand the basics behind dial plans and see how CoS can affect a device’s
capability to reach certain destinations.
Unified Messaging Overview
Typically, when you have a phone, you want to have a way for someone to leave a message when you cannot take the call. This means that most Communications Manager
environments will have some type of voicemail system. Cisco offers three voicemail solutions: Unity, Unity Connection, and Unity Express. That is right; they all contain the
word Unity in their name. This sometimes causes some confusion. This section looks at
the two systems that are most often deployed with CUCM. Unity Express is a solution
more often found in environments that have CUCM Express deployed and, for that reason, is outside of the scope of this course.
First, there is Unity. Many people view Unity as a voicemail solution, and although it
does handle all the functions of a voicemail system, it is much more than that. Unity is a
completely unified communication system. That is, it not only handles voicemail functions for a company, but it also can integrate into an existing email and faxing infrastructure. Because the voicemail, email, and faxing systems can be integrated with one another, they can use a single message store. This means that end users can now retrieve all
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their voicemail, email, and faxes from one location. In the past, users would have to use
the phone to get their voicemail, the PC to get their email, and a fax machine to retrieve
their faxes. With unified messaging, all these messages can be retrieved from a single
device, be it a PC or a phone. Using a PC, users can read their email and faxes or listen to
their voicemail over the PC speakers. The users can also use a phone to retrieve their
voicemail and email, because Unity’s text-to-speech engine converts the email to a synthesized voice and streams it over the phone.
The other voicemail system is Unity Connection. Recently it has risen in popularity. It
offers many of the same features as Unity and a few that Unity does not. The main difference, though, is not in the features it offers but rather in its infrastructure. Although
Unity requires an external mail store (Exchange or Dominos), Unity Connection does not.
It stores the messages on-box within its own store. However, even though Unity
Connection does not require an email server such as Exchange, it can still integrate with
an email server so that features such as listening to your email over the phone or checking
your voicemail from your email client are still available.
Both products are powerful applications that require a variety of components to accomplish its tasks. The topics covered in this section provide you with a good understanding
of the components within these systems and how these components fit into the solution.
The areas covered in this section are as follows:
■
Software architecture
■
Call flow
■
Call handlers
■
Subscribers
■
Networking
Software Architecture
Although Unity and Unity Connection offer similar services and features, their software
architecture is not similar at all, as you see in the following sections.
Unity Software Architecture
Unlike some applications, Unity is not a standalone application. That is, in and of itself, it
cannot provide voicemail services. Unity depends on a number of other applications to
provide its array of services. If Unity were merely a voicemail system, it could have been
designed not to require other applications. However, because it is a unified communications solution requires that it integrate with a company’s email server. Because both a
voicemail system and an email server have to send, receive, and store messages, Unity
simply uses the existing email server to handle these functions. This is not to say that
Unity does not handle the messages. This is discussed later. The various software components with which Unity interfaces include a mail store; a directory, such as Active
Directory (AD); Structured Query Language (SQL); and Internet Information Server (IIS).
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A closer look at the software architecture can help you understand the need for each
component.
The discussion begins at the lowest level of the software architecture. The first layer of
any software architecture is the operating system (OS). The current version of Unity
requires Windows 2003 Server OS. Unity integrates tightly with Microsoft’s AD when
using Exchange 2003. Previous versions of Unity supported Windows 2000, so don’t
worry if the version you run isn’t on Windows 2003.
Note Much like Communications Manager, Unity is supported only on Cisco-approved
servers. Currently there are a number of models approved by Cisco. These servers can be
purchased from Cisco or directly from HP or IBM. You can find the most current list of
approved servers by searching for “Cisco Unity supported platforms list” at Cisco.com.
After the OS is on the server, other supporting applications have to be available before
Unity can be configured. Unity depends on an outside message store to store messages.
Currently you can use Microsoft Exchange or Lotus Notes as the message store. If
Exchange is used, it can run on the same server as Unity, but this should be done only if
you use Unity as a voicemail-only solution. If Unity is used as a unified communications
solution, Exchange should be loaded on a separate server. If Notes is the message store, it
must run on a separate server.
Note As of Unity 8.0, Domino is no longer supported as a mail store.
Note If Notes is the message store, an additional piece of software is required. It is called
Domino Unified Communication Services (DUCS). This can be purchased from IBM and is
used to allow the transfer of information between Unity and the Domino environment.
All the information entered into Unity must be stored. Unity uses SQL as its database,
and most of the information you enter, such as usernames and phone passwords, are
stored in SQL. Hence, SQL must run on the Unity server.
As mentioned earlier, Unity integrates with AD when using Exchange 2003, so the server
must be part of an AD. If the Unity server is a standalone computer, that is, it is not part
of an existing AD, it must be configured with AD and act as its own domain controller.
Some of the information—for example, name, telephone number, and alias stored in
SQL—is replicated to the AD.
Unity is similar to Communications Manager in that most of the configuration of Unity
is done through a web browser interface. This requires that IIS run on the Unity server.
During the installation process, IIS is installed. Additional tools perform many of the
same tasks that are accomplished using the web browser interface. These tools are discussed in Chapter 11, “Exploring Unity/Connection Tools.” The main advantage to using
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the browser interface is that no additional software has to be loaded on a PC to administer Unity. This is a useful feature when you find yourself away from your desk needing to
change something in Unity.
Unity Connection Architecture
In contrast to Unity, Unity Connection is a standalone application. That is, it can be a
self-contained voicemail solution without any dependency on any separately installed
third-party software. Installing Unity Connection is as simple as inserting the disk into a
new server, powering it up, and answering a few questions. This install process installs the
OS and all required applications. The install for Unity Connection is much easier and
faster than that of Unity.
The OS that is used is often referred to as the VTG OS. VTG is an acronym for Voice
Technology Group. This is the same Linux-based OS that is used for Communications
Manager. Unity Connection stores all its information in an Informix database, just as
Communications Manager does. Unity Connection and Communications Manager share
a lot of the same software infrastructure. This is one reason why Unity Connection and
Communications Manager can run on the same server if you deploy Communications
Manager Business Edition.
Just as with Communications Manager and Unity, most of the configuration for Unity
Connection is done through a web browser.
Following the Call Flow
Before discussing the actual components that determine the path a call takes, now take a
high-level look at the various ways that a call can enter either of these systems and see
how each type is dealt with. Because Unity and Unity Connection handle calls in much
the same way, this section applies to both products. To keep things simple, both Unity
and Unity Connection are referred to as “the system” for the rest of this section.
Typically, calls are forwarded to voicemail systems because the phone called is either
busy or unanswered. When the system receives a call, it examines the reason that it is
receiving the call. The phone system that forwarded the call includes a call-forwarded reason to the system. In this case, it was forwarded because the called party was busy or did
not answer. If the call is forwarded because the caller did not answer, the system plays
the called party’s standard greeting. If the call were forwarded because the called party
was on the phone, the system plays the called party’s busy greeting.
In Unity, each user is known as a subscriber, but in Unity Connection, each user is known
as a user. To keep consistent in this section, they are referred to as users. Each user can
have five different greetings, which are as follows:
■
Standard: Played during open hours when the user does not answer the phone. This is
the default greeting and is always enabled.
■
Busy: Played when the user is on another call.
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■
Closed: Played after hours.
■
Internal: Played when another user reaches voicemail.
■
Alternate/holiday: If enabled, always plays regardless of the forwarded reason or
time of day. This greeting can be used as a vacation greeting. Note: Unity refers to
this greeting as the Alternate; UC refers to it as the Holiday.
The system is often used as an auto-attendant, which allows all incoming calls to be
answered by the system and then forwarded to the desired destination. In this instance,
the call is not forwarded and is considered a direct call. The system handles direct calls
differently than forwarded calls. The first thing the system does is attempt to determine
whether the caller is a user. It does this by identifying whether the caller ID matches any
phone number that is associated with a user. If it finds a match, it assumes that the user is
calling and asks for the user’s password.
Note All users have at least one phone number associated with them. This is their phone
extension. Additional numbers can be assigned to subscribers. They are referred to as alternate extensions. A common use for alternate extensions is to associate a subscriber’s cell
phone or home number so that when they call in to check messages they are taken directly
to the login prompt.
If the system determines that the caller ID is not associated with a user, the call is sent to
the system’s opening greeting. An opening greeting is the main greeting that outside
callers hear when they reach the auto-attendant. It might say something like “Thank you
for calling Bailey, Inc. If you know your party’s extension, you can dial it at any time during this greeting.” The options offered to the caller are determined by the system administrator based on the company’s needs.
Exploring Call Handlers
Often companies that use the system as an auto-attendant create a menu that can lead
outside callers to the proper department. You have probably experienced memorably
frustrating menus. To create menus in the system, objects titled call handlers are created.
One call handler was mentioned in the previous section. The opening greeting is a call
handler. The easiest way to think of a call handler is as an object that can be used to help
route calls. You can also think of call handlers as the building blocks that make up the
menu system.
There are primarily three types of call handlers that can be used within the system:
■
System call handlers
■
Interview handlers
■
Directory handlers
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Table 1-2
Call Handler Parameters
Configuration Page
Parameter
Profile (Handler Basics in Unity Connection)
Name, Creation date, Owner (Unity only),
Recorded voice, Schedule, Extension,
Language, Switch (PBX)
Transfer
Status, Transfer incoming call, Transfer type,
Rings to wait, If busy, Announce, Introduce,
Confirm, Ask caller’s name
Greeting
Greeting, Status, Source, Allow caller input,
After greeting action, Reprompt, Number of
reprompts
Caller Input
Allow extension dialing during greeting,
Milliseconds to wait, Lock key, Action
Message
Message recipient, Max message length, After
message action, Caller edit, Urgent marking
Call Handler Owners (Unity Connection only) Owner of the call handler
The system call handler is used to build the menu system and typically contains a recorded prompt and asks the caller for input, such as “Press 1 for sales, press 2 for technical
support, or hold on the line for further assistance.” Within Unity, these types of call handlers have no special classification and can be referred to as general call handlers. They
are simply known as call handlers. Within Unity Connection, they are called system call
handlers. Table 1-2 lists five configurable pages for a call handler and some of the parameters found in each. These parameters vary slightly between Unity and Unity Connection,
but this table gives you a good idea of the number of parameters that need to be configured for each call handler. More detail is given to these parameters in the “Configuring
Call Handlers” section found in Chapter 9, “Call Management.”
The next type of call handler is called an interview handler. It is used to extract information from the caller that requires asking more than one question. An interview handler
can contain up to 20 questions. It allows the user time to answer between the questions.
Unlike the call handler described previously, there are only two sets of parameters that
need to be configured for the interview handler. Table 1-3 shows these parameters in the
“Creating Advanced Call Routing Systems” section found in Chapter 9.
Note An example used for an interview handler would be a class survey hotline. After
students take a class, they can call a number and answer five questions. The answers are
later transcribed and entered into the school’s database.
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Table 1-3
Interview Handler Parameters
Parameter
Description
Profile (Handler Basics in Unity Connection)
Name, Creation date, Owner (Unity only),
Recorded voice, Extension, Language, Deliver
response to, Response urgency, After interview action
Question
Questions number, Question text, Maximum
length in seconds (for response), Recorded
question
The third type of call handler is known as a directory handler. The directory handler
allows callers to dial by name. As the caller spells the person’s name using the dial pad,
the system searches to find all matching names. The system then offers the caller name
choices and allows the caller to choose the party to which they would like to be forwarded. The system allows multiple directory handlers so that each department could have a
separate one if wanted. Directory handlers have four sets of configurable parameters, as
shown in Table 1-4. More detail is given to these parameters in the “Configuring
Directory Handlers” section found in Chapter 9.
Figure 1-6 shows a typical system menu system flow chart. The various types of call handlers that have been discussed are shown in this figure. Each of the boxes represents a
call handler.
Table 1-4
Directory Handler Parameters
Parameter
Description
Profile (Handler Basics in Unity Connection)
Name, Creation date, Owner, Recorded voice,
Extension, Language, Play all names, Search
scope (Unity Connection only), Search result
behavior (Unity Connection only)
Search Options (Unity only)
Search in, Search by
Match List Options (Unity only)
On unique, Announce matched using,
Announce extension
Caller Input
No input timeout, Last input timeout, Repeat
prompt, Send on to exit
Greeting (Unity Connection only)
Greeting
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Opening Greeting
Call Handler
Thank you for calling…
Press 1 for Sales.
Press 2 for Customer Service.
Press 3 to dial by name.
Press 0 for an operator.
Operator
Call Handler
Directory
Handler
Customer Service
Call Handler
Press 1 for returns.
Press 2 for repairs.
Repair
Call Handler
Figure 1-6
Returns
Call Handler
Sales
Call Handler
Press 1 for a catalog.
Press 2 to place an order.
Sales Person
Call Handler
Interview
Call Handler
Example Menu System Flow Chart
Tip It is a good idea to draw your menu system in a flow chart form before creating it
on the system. This helps ensure that all the call handlers are linked and that the flow
makes sense.
Typically call handlers are used to route a call through the system. However, they can be
configured to take messages. If a call handler is configured to take a message, the message must be sent to a user. Call handlers are simply database objects and do not have
mailboxes.
As you can see, call handlers are a fundamental part of the system. Even the most basic
system contains a number of call handlers. So a solid understanding of how they are configured and work is essential. This section has only touched the surface of the power and
flexibility of call handlers. Once again, you are encouraged to review the listed references
in Appendix A for further study of this technology.
Defining Various Types of Users
Now that you have a good understanding of how calls are routed through a system, we
take a look at the users. As mentioned earlier, in Unity, users are referred to as subscribers, while in Unity Connection, they are simply referred to as users. The following
sections look at each type separately to help you better understand the differences.
Unity Connection Users
Within Unity Connection there are three types of users: those with voice mailboxes,
those without voice mailboxes, and contacts. Users with voice mailboxes are the type of
users that you will be creating the majority of the time. Users without voice mailboxes
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are users for people that need administrative access to the system but do not need voicemail on the system. The main purpose of contacts is to add someone to the directory
who does not have an account on the system. It is common to list users who have a voicemail account on a different system. This allows voicemails to be forwarded to them.
Unity Subscribers
When it comes to Unity, there are a number of different types of subscribers. This is
because Unity relies on another application such as Exchange or Dominos to store the
messages. The following sections explore the various types of subscribers that can be
created within Unity.
A subscriber is anyone who has an account on the Unity system. Typically, Unity subscribers’ messages are stored on the message store that is associated with Unity.
However, subscribers are sometimes created where the mailbox actually resides on a different voicemail system. The following sections explore all the types of users who can be
configured in Unity and indicate the appropriate uses for each. Six different types of
subscribers can be configured as follows: Exchange, Domino, Internet, AMIS-a, VPIM,
and bridge.
Exchange Subscriber
An Exchange subscriber is the most common type of subscriber. This subscriber, as the
name implies, has its messages stored on an Exchange server. Exchange subscribers have
access to voicemail using the phone, also called the telephone user interface (TUI). If
licensed, they might also have access to voicemail using a PC. This is referred to as the
graphical user interface (GUI).
Note Exchange subscribers can be divided into two subtypes of subscribers. One type
is referred to as voicemail-only. These subscribers use the Exchange server for voicemail
messaging purposes only. Furthermore, they cannot retrieve their email through Unity or
their voicemail through a PC. The other type is known as unified messaging. These subscribers have both their email and voicemail stored on the Exchange server and can access
both from either a phone or a PC.
Domino Subscriber
A Domino subscriber is similar to an Exchange subscriber except that it uses Domino as
the message store, not Exchange. Unlike Exchange subscribers, a Domino subscriber
must be a unified messaging user. Domino is not supported as a voicemail-only solution.
Note As of Unity 8.0 Domino is no longer supported.
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Internet Subscriber
An Internet subscriber does not have a mailbox on the message store associated with
Unity. The mailbox is an Internet email account. This type of account allows outside
workers to be listed in the directory and receive voicemail without any type of access to
Unity. This type of subscriber is generally used to provide voicemail for contract employees who work from their homes. When messages are left for Internet subscribers, they are
delivered to the Internet email account as WAV file attachments, and the subscribers
retrieve them when they download their mail.
AMIS-a (Audio Messaging Interchange Specification–Analog) Subscriber
This type of subscriber is used when Unity is connected to a non-Unity voicemail system
using the AMIS protocol. The subscriber’s mailbox is located on the non-Unity voicemail
system. This type of subscriber has no direct access to Unity.
Voice Profile for Internet Mail (VPIM) Subscriber
This type of subscriber is used when Unity is connected to a non-Unity voicemail system
using the VPIM protocol. The subscriber’s mailbox is located on the non-Unity voicemail
system. This type of subscriber has no direct access to Unity.
Bridge Subscriber
This type of subscriber is used when Unity is connected to an Octel voicemail system
using bridge networking. The subscriber’s mailbox is located on the Octel voicemail system. This type of subscriber has no direct access to Unity.
User Parameters
When creating and configuring a user, you notice that a number of the parameters are
similar to those found in a call handler. This is because each user has an associated call
handler. When you create a user, a call handler is also created. So, when you are configuring certain user parameters, you are actually configuring the call handler associated with
that user. To help illustrate this, Table 1-5 shows all the Unity parameter sets. Those
marked with an asterisk (*) are the parameters used to create the call handler. While the
names of the parameters in Unity Connection vary slightly, the concept is the same.
Note Not all Unity subscribers have all the parameters listed in Table 1-5. The parameters available depend on the type of subscriber that you are creating. In some cases, there
are some parameters that are not listed in this table. Only the most common parameters
are listed.
A user is more than just a call handler, as you can see by the additional parameters that
need to be configured. A user can be seen as a collection of objects, one being the call
handler. It is the call handler’s job to offer the caller any available choices and play the
greeting. Another piece of a user is the mailbox where the voice messages are stored. For
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a Unity subscriber, the mailbox typically resides on an Exchange server. There are also
additional user settings. These settings are not part of the mailbox or the call handler.
These attributes deal with things such as phone passwords, message notification, and
CoS. These settings, much like the call handlers, are stored in the database.
Table 1-5
Subscriber Parameters
Parameter
Description
Profile*
Name, Display name, CoS, Extension, FAX ID, Recorded voice,
Schedule, Time zone, Self-enrollment setting, Directory, Language,
Switch (PBX)
Account
Account status, GUI access status, Creation date, Last phone contact, Billing ID, Call handlers owned, Windows NT account status
Phone Password
User cannot change password, Must change on next login, Never
expires, Password, Date of last change
Private Lists
Private list number, Name of list, Recorded name, Current members
Conversation
Menu style, Volume, Language, Time format, Send to, Identify by,
Play recorded name, Message counts, Message type menus, Sort by,
Message number, Time Message was sent
Call Transfer*
Transfer to, Transfer type, Rings to wait, If busy, Announce,
Introduce, Confirm, Ask caller’s name
Greetings*
Greeting, Status, Source, Allow caller input, After greeting action,
Reprompt, Number of reprompts
Caller Input*
Allow extension dialing during greeting, Milliseconds to wait, Lock
key, Action
Messages*
Max message length, After message action, Caller edit, Urgent
marking, Language, MWI
Message Notification
Device, Phone number, Extra digits, Dialing options, Status,
Schedule
Alternate Extensions
Alternate extensions
Alternate Names
Alternate name
Features
Broadcast messages, Message locator, Message security
*Parameters used to create the call handler
Note The term class of service (CoS) is sometimes confusing because it means different
things depending on the technology it is referring to. When the term is used with Unity/
Connection, it is referring to rights and access, such as what features a user has access to.
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As you can see, many parameters need to be configured for each user. If you had to configure each parameter for every user, it would take far too long. To simplify this task,
objects known as templates are used when creating users. A template defines how the
parameters should be set by default. Typically there are large groups of users that need
similar settings. A template is created for each group, and after creation, a user can be
edited as needed.
Note It is important to understand that templates are significant only at the point of user
creation. Changing a template after a user is created has no effect on that user.
As previously stated, not all subscribers have all the parameters listed in Table 1-5.
Parameters depend on the type of subscriber that is being created.
Networking Overview
Both Unity and Unity Connection have the capability to network with each other and
other voicemail systems. The following sections explore the types of networking that are
supported by each system.
Unity Networking
Unity has the capability to network with other types of voicemail systems. This feature
allows employees of large companies that have multiple voicemail systems to efficiently
exchange voicemails regardless of the system on which they are homed. Cisco understands that companies have a substantial investment in current solutions and might not be
ready to do a total cut-over. Often Unity is deployed for a portion of the users as a pilot.
It is easier to convince the client to start with a pilot because Unity can talk with other
systems and allow the end user to use the most common features.
Following are four types of Unity networking:
■
Digital
■
VPIM
■
AMIS
■
Bridge
Typically, people think of networking as plugging two PCs into an Ethernet jack that
allows enables them to communicate. Unity networking is not just plugging an Ethernet
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cable into the Unity server. Unity networking refers to connecting Unity to another voicemail system. This can be done in a variety of ways, which are described in the following
sections. The overall goal is to have Unity interface with another voicemail system as
seamlessly as possible.
Digital Networking
Digital networking allows multiple Unity systems within the same directory, such as AD,
to seamlessly interact with one another. When using digital networking, the Unity systems have the ability to share the directories of each other. This allows outside callers to
search for any subscriber, regardless of which Unity server they are associated with. This
type of networking offers many other features, which is discussed in Chapter 10,
“Implementing Unity Networking.”
VPIM Networking
VPIM networking allows Unity to interface with non-Unity voicemail servers through a
TCP/IP link. This link can be an Internet connection or a private network. The non-Unity
voicemail system must support VPIM, which is often an add-on that requires an additional investment.
AMIS Networking
AMIS networking allows Unity to interface with a non-Unity voicemail system across
analog lines. This is typically done across the PSTN, but it might also be used for inhouse migration. AMIS is not as efficient as VPIM, because it uses analog lines, so transmissions can be more time-consuming. For example, a 5-minute message sent to five people through AMIS takes 25 minutes, plus setup and teardown time. With VPIM, the
transmission time is much shorter. (The actual transmission time depends on the speed of
the connection.) The non-Unity voicemail system must support AMIS, which is often an
add-on that requires an additional investment.
Bridge Networking
Bridge networking is used to connect a Unity system with an Octel voicemail system.
Bridge networking is unique in that it requires an additional server. This is called the
Unity bridge server. The bridge server communicates with the Unity server through a
TCP/IP connection and with the Octel server across analog lines. The bridge server acts as
a translator between the Octel analog protocol and Unity’s digital networking protocol.
The overall message delivery can experience the same type of delay that is common with
AMIS because they both use analog connectivity. Octel networking adds features that
are not found in VPIM or AMIS and creates a more feature-rich environment. Bridge networking is recommended if a company is going to gradually migrate from Octel to Unity.
So which one should you use? The voicemail solution that you choose to integrate with
will often dictate the answer. If you have a choice between AMIS and VPIM, most often
VPIM is recommended. As mentioned previously, when integrating with Octel, bridge
networking should be used. When connecting multiple Unity systems, digital networking
is the solution if the Unity systems share the same Active Directory (AD). If they do not
share the same AD, the solution is VPIM networking.
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Unity Connection Networking
Unity Connection offers two types of networking. The first, digital networking, is used
when networking UC systems within a company. The other, VPIM networking, is used to
network UC to non-UC voicemail systems and UC systems not within the same company.
Unity Connection digital networking is similar to Unity’s digital networking; the big difference is that UC does not use AD. This means that each UC needs to synchronize its
directory with the other UC systems on the network. The net effect is the same as it is
with Unity, but a little more configuration might be required.
Unity Connection also supports VPIM networking. This is used when networking with a
Unity server or other voicemail system. The remote voicemail system must support
VPIM. Keep in mind that the remote system will most likely require a license for this
feature. VPIM is also used when networking UC systems when digital networking is not
an option.
Securing the Environment
With the proliferation of viruses and malicious attacks on computer systems today, it is
not just wise, but mandatory, to protect your systems. Many network administrators
remember Nimda and Blaster all too well. Even some admins that thought they were fairly
well protected got hit. Sometimes it seems that just when one type of attack can be
defended against, another begins. To make matters worse, in an IP telephony environment, you need to protect the system not only from computer-related attacks but also
from the types of attacks common to voice systems, such as toll fraud. You might as well
face it now: There is no silver bullet that completely and forever protects you. But there
are steps you can take to make sure that you are not an easy mark. The following sections
examine some of the current security concerns that you should be aware of. When possible, solutions are offered. Remember, this is an ongoing battle. The goal of these sections
is to make you aware of a few of the security issues and encourage you to be vigilant in
protecting your system.
Securing the Operating System
Because Unity uses Windows as its operating system, it is vulnerable to the security
issues that exist for Windows. Because Microsoft is undeniably the most commonly
installed OS, it is a large target. For various reasons, some people hold grudges against
Microsoft and find it fun to attack systems running on a Microsoft OS. Whether they are
valid grudges, it is important to protect your systems. What is important is that attacks
do occur, and you need to protect your system.
Typically an attack is virus driven. The damage viruses cause can range from something
benign to data corruption and destruction. All viruses should be considered dangerous.
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For this reason, virus protection software should be installed on all systems. Cisco has
qualified the following list of antivirus software for use with Unity 8.0:
■
CA Anti-Virus for the Enterprise version 8.0 and later (formerly called eTrust
Antivirus)
■
Computer Associates InoculateIT for Microsoft Windows
■
McAfee
■
ePolicy Orchestrator
■
GroupShield Domino (Unity 7.x and earlier only)
■
NetShield for Microsoft Windows
■
Symantec
■
AntiVirus Corporate Edition
■
Norton AntiVirus for IBM Lotus Notes/Domino (Unity 7.x and earlier only)
■
Norton AntiVirus for Microsoft Exchange
■
Norton AntiVirus for Microsoft Windows
■
Trend Micro
■
ScanMail for Lotus Notes (Unity 7.x and earlier only)
■
ScanMail for Microsoft Exchange
■
ServerProtect for Microsoft Windows
Because Unity uses SQL, keep in mind that in the past, SQL has been the target of
attacks. So, attention should be given to this. The best way to secure any SQL installation
is to ensure that the latest approved security patches are applied.
Note Remember to check that a patch has been tested and approved by Cisco before
you apply it. In some cases, there are special instructions on how to load certain patches.
It could even be required that you load only patches available directly from Cisco. Don’t
assume that all Microsoft patches can be loaded on Unity.
Communications Manager Security Issues
Protecting the system from outside threats is only half the battle. There are also internal
threats that exist. Ever since a telephone system has existed, people have tried to exploit
it. These exploitations range from illegal wiretapping to toll fraud.
Earlier in this chapter, it was mentioned that a PC can be connected to the back of the
phone, allowing both devices to share a single Ethernet port. Although this is an efficient
use of ports, if not properly configured, both devices could be using the same VLAN.
When both devices are on the same VLAN, it is possible for a PC to capture voice
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traffic. A tool called Voice Over Misconfigured Internet Telephones (VOMIT) can then
reassemble the captured data into a WAV file. The conversation could be played back on
a PC. By ensuring that the voice traffic is on a separate VLAN, you help prevent this from
occurring, but that alone is not enough. You also need to prevent data and voice networks from communicating with each other as much as possible. It is understood that
there are times when the networks might need to send packets to each other, but this
traffic should be managed by a firewall to ensure that only the desired traffic is allowed
through. Earlier, it was recommended that voice and data traffic be separated on different
VLANs so that the voice could be prioritized. Now you see that it can protect not only
the quality of the voice but also protect the voice from prying ears.
PC-based phones, such as Cisco IP Communicator, introduce the same problem that having voice and data on the same VLAN produces. Because a PC is being used as phone,
the voice traffic is sent on the data VLAN. This allows prying ears on the data network to
capture the conversation. For this reason, a properly designed network has voice and data
on separate VLANs.
Another often-overlooked area that can be exploited is allowing rogue phones to autoregister. This occurs when autoregistration is used during deployment and not turned off or
restricted later. Autoregistration is a useful tool during some deployments. It allows a
phone to be plugged into the system and register without having to configure it in
Communications Manager first. It is most often used in greenfield deployments. However
if autoregistration is not disabled after the initial deployment, rogue phones could register to Communications Manager. This seems like a minor issue until you factor in that if
the dial plan has not been secured, a user on a rogue phone could place a call anywhere
in the world without detection until the bill arrives.
The administration of Communications Manager should also be considered when securing the voice system. If people get access to Communications Manager administration,
they can do anything from changing users’ class of control to shutting down gateways, or
even the entire system. Security is often a double-edged sword: Too little and anyone can
do anything, and too much and no one can do anything. Often, you find that you want to
allow limited access to the administration interface. An example might be to allow someone to add users to the Communications Manager directory. You do not want this person
to change the dial plan. A Communications Manager add-on, called Multi-Level Access
(MLA), enables you to do this by granting limited administration access to individuals.
An even higher level of security can be implemented by restricting which physical system
can access the Communications Manager through the use of an access list.
The preceding few paragraphs discussed only a few security issues that you need to
address. The task of completely securing a Communications Manager environment can
seem daunting at times, but all possible efforts must be taken to ensure that the system is
protected from both external and internal threats.
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Configuring Cisco Unified Communications Manager and Unity Connection: A Step-by-Step Guide
Unity Security Issues
Unity offers a new way of looking at message management. It is now possible to store all
voicemail, email, and faxes in a single location. In addition, it is possible to retrieve them
from nearly anywhere in the world. Of course to do this, there must be a way to access
the system. Although this comes as no surprise, remember, at every entry point, there are
people trying to exploit that point of entry. Because of the many features that Unity
offers, there are many points of entry. By default, Unity enables access through the
phone, web browser, and email client. All of these entry points add to the task of securing the system. Securing Unity is not a trivial task, and sufficient attention must be given
to this task. The following are just a few examples of areas that must be secured.
Unity uses a third-party message store, and currently this is either Microsoft Exchange or
Lotus Notes. Both of these systems are subject to the same types of attacks as any email
systems. There are attacks aimed specifically at these systems. Securing these systems
requires that you stay current on all security-related patches.
Note Remember to check that a patch has been tested and approved by Cisco before
you apply it. In some cases, there are special instructions on how to load certain patches.
It could even be required that you load only patches available directly from Cisco. When a
patch is related to a high-level security issue, Cisco is quick to test and report these patches on Cisco.com. You can find available patches in the software download section of
Cisco.com. You can also set up to have alerts emailed to you automatically. Search
“Product Alert Tool” at Cisco.com to set this up.
Just as with Communications Manager, administration access has to be secured. By
default, the administrator selected when configuring Unity has full administrative access.
Often you need to allow others access for limited administrative tasks. You can do this by
creating a CoS that enables them only the access that you want them to have.
Unity has the capability to track all administrative access and create reports showing this
access. This can prove useful when troubleshooting to determine whether changes were
made and who made them. For this reason, it is necessary for every person accessing the
administration interface to use individual login credentials. Don’t just create one administrative login and allow everyone to use the same one because there is no way to determine who made what changes.
When you create subscribers, you determine which features they can access. A few of
these features can affect the security of your system. One of these features is the user’s
capability to define one-key transfer options that can be performed during his greeting.
This enables the outside caller to be transferred to another extension or number by pressing a single key. However, this also opens up the possibility of toll fraud. A subscriber
could set up a one-key transfer that transfers the call to a long-distance number, perhaps
a friend who lives out of state. This option, of course, would not be announced during
the greeting, but the subscriber could dial into his voicemail from home at night and be
transferred to his friend’s number all on the company’s dime. The configuration of such
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Chapter 1: CUCM and Unity Connection Overview
features should be restricted to administrative personnel only. Even then, you might want
to take further action to ensure that individuals with the ability to configure this feature
do not exploit the privilege. To prevent possible exploitation, you can restrict the numbers to which Unity can transfer these users. Chapter 7, “Unity Predeployment Tasks,”
discusses how to perform these tasks.
Each subscriber in Unity has two passwords. The first is the phone password. This is the
password that is used when accessing Unity through the phone. This password is only
numeric because it is entered using the digits on a phone. Unity should be configured to
require minimum-length passwords for each subscriber. The longer the minimum length,
the harder it is for someone to figure it out. It is important that you don’t take Unity
phone passwords lightly. Often people don’t think that their voicemail password is as
important as their email password, but with unified message, if you can access one, you
can access the other. The second password subscribers use to access Unity is their AD or
NT account password. This password is used when Unity is accessed using a PC, such as
when checking messages over the Internet. This password should contain both letters and
numbers, and a minimum length of five digits should be used.
Note All accounts have default passwords. Make sure that these are changed the first
time an account is accessed. There are a few default accounts that are created when Unity
is installed. Make certain that the default password on these accounts is changed immediately following the installation. On more than one occasion, I have logged in to systems at
clients’ sites and retrieved messages, unknowingly sent to these accounts, simply by using
the default extension and password. This, of course, was done with the full knowledge of
the customer and only to show him that his system was not secured by the installer.
This section has touched on only a few security concerns to bring attention to the fact
that these are valid concerns that must be addressed. Additional reference material is listed in Appendix A.
Summary
A Cisco IP telephony deployment comprises a number of technologies and platforms.
Before beginning to deploy such a solution, much thought must be given to the proper
design. It all begins with a reliable, solid infrastructure. A proper infrastructure begins
with a redundant physical topology and proper backup power. To allow access to outside
systems, gateways are used and methods of Call Admission Control must be implemented.
At the call-processing layer, redundant Communications Managers should be deployed.
These systems must run on Cisco-approved hardware called MCSs. Devices such as
phones, gateways, and conference bridges register to Communications Manager. When a
call is placed or a resource is requested, the request is sent to the Communications
Manager to which the device is registered. The Communications Manager uses the configured dial plan to determine how to route the call to the desired location. After the call
is complete, Communications Manager ensures that it is properly torn down.
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Configuring Cisco Unified Communications Manager and Unity Connection: A Step-by-Step Guide
Unity is a unified communications server that enables individuals to have all their voicemails, emails, and faxes stored in a single location. This enables more efficient access to
all messages. Unity has the capability to function as an auto-attendant and have outside
callers routed through the system when they make selections from the available menu. In
the future, unified communications will enable calls to be routed based on predefined
rules that individual users can configure.
Unity Connection is a voicemail system that offers many of the same features as Unity
but does not use a third-party message store. This prevents it from offering true unified
messaging, but it can still offer comparable feature and, in some cases, features that are
not offered by Unity.
The advances being made in IP telephony are exciting. As with most new technologies,
there are those who find ways to exploit them. It is critical that sufficient attention be
given to securing all IP telephony deployments.
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Chapter 2
Preparing CUCM for
Deployment
To ensure a smooth deployment, certain tasks must be performed in a certain order. In
this chapter, you learn what tasks need to be completed before adding devices. As with
most things, if you fail to create a solid foundation, you will encounter problems. The
topics covered in this chapter give you that firm foundation.
Before adding any devices to the system, ensure that a number of predeployment tasks
are accomplished. Because this book assumes that Communications Manager is already
installed, most settings discussed in this chapter should already be properly configured.
The goal of this chapter is to help you understand these settings and see how changing
them can affect your system. This chapter covers services configuration, enterprise
parameters, and device registration tasks. In addition, the chapter includes step-by-step
instructions for many of these tasks. It is worth the time to review these settings and
make sure that they are configured properly for your system. Although the system might
seem to function fine even if some of these tasks are overlooked, it is recommended that
all tasks are verified before adding devices.
Configuring Communications Manager for Maximum
Performance
Communications Manager is the heart of the call-processing system, and it is important
to ensure that it runs at peak performance. A number of processes can run on the
Communications Manager, but not all of them are always necessary. The following sections explore the various processes that might run on a Communications Manager and
discuss which can be safely disabled to preserve more processing power for other
Communications Manager functions.
The following sections look at the services that can be enabled to run on the
Communications Manager. There are two types of services: feature and network. For the
most part, the average administrator will not need to deal with the network services, so
these sections focus on the feature services.
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Configuring Cisco Unified Communications Manager and Unity Connection: A Step-by-Step Guide
Activating Communications Manager Services
Since the release of Communications Manager 3.3, all Communications Manager services
are deactivated by default. Before Communications Manager 3.3, the user determined
which services would run by choosing to install them during the installation process. In
the current versions of Communications Manager, all services are loaded but none are
activated. This section discusses each of these services and explores the proper way to
activate them.
Before you can have devices register, you must activate services using the Service
Activation screen within the Communications Manager Serviceability interface. You can
access this interface by entering https://[CM_IP Address]/ccmservice. If you are already
logged in to the Communications Manger administration interface, you can access the
Serviceability interface by selecting Cisco Unified Serviceability from the Navigation
drop-down menu in the upper-right corner. After you are in the Serviceability interface,
navigate to Tools > Service Activation.
Figure 2-1 shows the Service Activation screen as it appears the first time it is accessed.
Figure 2-1
Service Activation Screen
To activate the Services screen, follow these steps:
Step 1.
Enter https://[CM_IP Address]/ccmservice in the address bar of your browser and press Enter.
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Chapter 2: Preparing CUCM for Deployment
Step 2.
Enter the administrative username and password and click Login.
Step 3.
Navigate to Tools > Service Activation.
Step 4.
Select the desired server from the Server drop-down list.
Step 5.
Select the check box next to each service that you want to activate and
click Save.
Note The Set to Default button is next to the Refresh button. If the Set to Default button
is clicked, all the services that are required for the functioning of Communications
Manager will be activated. In most cases, you do not activate all the same services on all
the servers. Only use this button in a single-server environment. You should not select all
services on all the Communications Manager servers because doing so would activate services that might not be needed, thereby consuming CPU cycles that could be used for other
functions.
Table 2-1 lists all Communications Manager services, with a brief explanation of their
functions and activation recommendations.
After these services are activated, they can be deactivated if desired. To deactivate any of
these services, follow these steps.
Step 1.
Enter https://[CM_IP Address]/ccmservice in the address bar of your browser and press Enter.
Step 2.
Enter the administrative username and password and click Login.
Step 3.
Navigate to Tools > Services Activation.
Step 4.
From the Server drop-down list, select the server on which you want to deactivate service and click Go.
Step 5.
Deselect the check box next to each service that you want to deactivate and
click Save.
After all the services are configured properly on all the servers in the cluster, you can
move on to defining Communications Manager Enterprise settings. The next sections
explore the various parameters that should be defined before devices are deployed.
Configuring Communications Manager’s Enterprise
Settings
After services are active, Communications Manager is functioning. However, before
devices are added, a few more tasks must be accomplished. The following sections examine the importance of removing Domain Name System (DNS) reliance and the system’s
enterprise parameters.
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Configuring Cisco Unified Communications Manager and Unity Connection: A Step-by-Step Guide
Table 2-1
Communications Manager Services
Service
Function
Cisco Communications
Manager
In the control center, netProvides call-processing,
signaling, and call control func- work services ensure that the
Cisco Real-Time Information
tions.
Serve (RIS) Data Collector
service and Database Layer
Monitor service are running
on the node.
Cisco Trivial File Transfer
Protocol (TFTP)
Provides TFTP services for
device configuration files. It
is also responsible for
building the configuration files.
This service should be activated on at least one
Communications Manager.
This server is responsible for
servicing TFTP requests.
Cisco Messaging Interface
Provides Simplified Message
Desk Interface (SMDI)
connectivity for traditional
voicemail systems.
This does not need to be
activated if Unity is the
voicemail solution.
Cisco Unified Mobile Voice
Access Service
For mobile voice access to
Allows Cisco Unified
work, you must activate this
Mobility users to perform
service on the first node in
the following tasks:
the cluster and configure the
• Make calls from a cellular
H.323 gateway to point to
phone as if the call
the first Voice Extensible
originated from the desk
Markup Language (VXML)
phone. Depending on the
page. In addition, make sure
device that is used, a user
that the Cisco CallManager
might have to call a local
PSTN number owned by the and the Cisco TFTP services
run on one server in the cluscompany, enter a PIN, and
ter, not necessarily the same
then establish an
outbound toll call through the server where the Cisco
gateway. Some devices such Unified Mobile Voice Access
as the iPhone do not require Service runs.
this.
• Turn Cisco Unified Mobility
on.
• Turn Cisco Unified Mobility
off.
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Recommendations and
Dependencies
Chapter 2: Preparing CUCM for Deployment
Table 2-1
Communications Manager Services
Service
Function
Recommendations and
Dependencies
Cisco IP Voice Media
Streaming App
Provides service such as Music
on Hold (MoH), conferencing,
and Media Termination Points
(MTP).
You do not need to activate
this service on all
Communications Managers.
Activate it only on
Communications Managers
that you want to provide these
services. This should not be
activated in the Publisher.
Cisco Extension Mobility
Used to define time limits for
Extension Mobility. Activates
the XML service for login function.
This service should be activated on all Communication
Mangers in environments for
which Extension Mobility is
planned.
Cisco Extended Functions
Provides support for Cisco
Unified Communications
Manager voice-quality features,
including Quality
Report Tool (QRT). Also enables
the callback feature.
The Cisco RIS Data Collector
must be running on servers
that are running this service.
The CTI manager service
must be loaded on at least
one server.
Cisco Dialed Number
Analyzer
Supports Cisco Unified
Communications Manager
Dialed Number Analyzer.
If you are planning to use
Cisco Unified
Communications Manager
Dialed Number Analyzer, activate this service. This service
can consume many resources,
so only activate this service on
the node with the least amount
of call-processing activity or
during off-peak hours. It is further recommended that this
service only be activated while
troubleshooting.
Cisco DHCP Monitor Service Provides Dynamic Host
Configuration Protocol (DHCP)
service for phones
and monitors IP address changes
for IP phones in the database
tables.
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Activate this service on the
Communication Managers
that have DHCP server
enabled.
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Configuring Cisco Unified Communications Manager and Unity Connection: A Step-by-Step Guide
Table 2-1
Communications Manager Services
Service
Function
Recommendations and
Dependencies
Cisco IP Manager Assistant
(IPMA)
This is required to enable the
IPMA feature. This feature
allows managers additional
features such as divert and
DND.
This service should be active
on servers that service these
features.
Cisco WebDialer Web Service Allows users to dial from a
web page or desktop
application.
Typically, this service is
loaded on one server in the
cluster.
Cisco SOAP - CDRonDemand Receives Simple Object
Service
Access Protocol (SOAP)
requests for Call Detail
Records (CDR) filename and
returns a list of filenames that
fit the time duration that is
specified in the request.
You can activate the Cisco
SOAP - CDRonDemand
Service only on the first
server.
Cisco CAR Web Service
You can activate the Cisco
Loads the user interface for
CAR Web Service only on
CDR Analysis and Reporting
the first server.
(CAR), a web-based reporting
application that generates either
CSV or PDF reports by using
CDR data.
Cisco AXL Web Service
This service allows you to
modify database entries and
execute stored procedures
from client-based applications
that use Administrative XML
(AXL).
Cisco UXL Web Service
Activate if the Cisco IP
This service performs
Phone Address Book
authentication and user
Synchronizer is being used.
authorization checks. The
TabSync client in Cisco IP
Phone Address Book
Synchronizer uses the Cisco UXL
Web Service for queries to the
Cisco Unified Communications
Manager database.
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Activate on the first node
only. Failing to activate this
service causes the inability
to update Cisco Unified
Communications Manager
from client-based applications that use AXL.
Chapter 2: Preparing CUCM for Deployment
Table 2-1
Communications Manager Services
Service
Function
Recommendations and
Dependencies
Cisco Bulk Provisioning
Service
Supports the Bulk
Administration Tool (BAT).
You can activate the Cisco
Bulk Provisioning Service
only on the first node. If you
use the BAT to administer
phones and users, you must
activate this service.
Cisco TAPS Service
Activate this service on the
This service supports the
Cisco Unified Communications first node if TAPS is going to
Manager Auto-Register Phone be used.
Tool, which allows a user to
upload a customized
configuration on an autoregistered phone after a user
responds to Interactive Voice
Response (IVR) prompts.
Cisco Serviceability
Reporter
Generates a report once daily.
The generated reports include
statistics on servers, devices,
services, call activities, and
alerts. These reports can be
accessed from the
Communications Manager
Serviceability Tools menu.
This service is activated on
the Publisher. Be careful
when running reports,
because the service can
impact call processing. It is
best to run reports after
hours.
Cisco CallManager
SNMP Service
This service, which implements
the CISCO-CCM-MIB,
provides SNMP access to
provisioning and statistics
information that is available
for Cisco Unified
Communications Manager.
Activate this service on all
servers in the cluster if
Simple Network
Management Protocol
(SNMP) is going to be used.
Cisco CTL Provider
If enabled, this changes
the security mode from
unsecured to mixmode.
This service works in concert
with the Cisco CTL Client.
This service should be
activated on all servers.
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Configuring Cisco Unified Communications Manager and Unity Connection: A Step-by-Step Guide
Table 2-1
Communications Manager Services
Service
Function
Recommendations and
Dependencies
Cisco Certificate Authority
Proxy Function
Activate on only the first
Working in conjunction with
node.
the CAPF application, the
Cisco Certificate Authority
Proxy Function (CAPF)
service can perform the
following tasks, depending
on your configuration:
• Issue locally significant certificates to supported Cisco
Unified IP Phone models.
• Using SCEP, request
certificates from third-party
certificate authorities on
behalf of supported Cisco
Unified IP Phone models.
• Upgrade existing certificates
on the phones.
• Retrieve phone certificates
for troubleshooting.
• Delete locally significant
certificates on the phone.
Cisco DirSync
Ensures that the Cisco Unified Activate on only the first
node.
Communications Manager
database stores all user information. If you use an integrated
corporate directory, for example,
Microsoft Active Directory or
Netscape/iPlanet Directory, with
Cisco Unified Communications
Manager, the Cisco DirSync
service migrates the user
data to the Cisco Unified
Communications Manager
database.
Removing DNS Dependencies
It is recommended that Communications Manager be configured so that the devices that
register to it do not have to rely on DNS to resolve the Communications Manager. This
affects only devices such as phones and Media Gateway Control Protocol (MGCP) gateways
and is accomplished quite easily, as you see in the steps provided later. You might be
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Chapter 2: Preparing CUCM for Deployment
wondering why it is recommended to remove DNS reliance. Primarily there are three reasons
for this. First, it allows devices to be able to register without querying a DNS server. Second,
a DNS server failure will not cause devices such as phones to fail registering. Finally, because
the device makes a call directly to the IP address of the Communications Manager, it can
register faster. This is because an IP address can be resolved at Layer 3 of the Open Systems
Interconnection (OSI) model, whereas a DNS request has to go up to Layer 7.
Note You are removing the DNS reliance from the devices, not DNS from your network.
Removing DNS from your network can cause your network to stop functioning properly.
Removing DNS reliance simply requires that you change the name of each
Communications Manager server to the IP address. This is done within the web-based
Communications Manager Administration tool. Throughout the Communications
Manager chapters in this book, this interface is referred to as CM Administrator or
CCMAdmin. To access CM Administrator, point your web browser to
https://[CM_IP_address]/ccmadmin. For example, if the IP address of the
Communications Manager is 10.3.3.3, the address you enter is
https://10.3.3.3/ccmadmin. When you are at the login screen, enter an administrator
username and password and click Login.
Note Because CM Administrator uses Hypertext Transfer Protocol Secure (HTTPS), you
must make sure that you enter HTTPS in the address bar and not HTTP.
To change the names of the servers, follow these steps:
Step 1.
Step 2.
From within CM Administrator, open the System drop-down menu by placing
the mouse over System at the top of the screen.
Select Server from the drop-down menu.
A search criteria screen appears. You can search based on the name or the
description of the Communications Manager. You can also leave the search
criteria blank, and all servers will be listed. If you want, enter the criteria to
narrow the search and click the Find button.
Step 3.
Select the first server in the list, and click its name.
A screen similar to that shown in Figure 2-2 appears. In the Host Name/IP
Address field, the current name of the server appears. In Figure 2-2, you can
see that the name is being changed to 192.168.0.180.
Step 4.
If you use DNS that supports IPv6, enter the IPv6 name in the IPv6 Name
field. Otherwise, enter the non-link-local IP address.
Step 5.
The MAC Address field needs to be populated only if you plan to move the
server from time to time. Entering the MAC address aids in helping the other
devices identify the server. Enter the MAC address of the server in this field if
you plan to move the server.
Step 6.
In the Description field, enter a description that will aid in determining the
location and function of this server.
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Configuring Cisco Unified Communications Manager and Unity Connection: A Step-by-Step Guide
Figure 2-2 Server Configuration Screen
Step 7.
Click the Save button to save the changes.
Step 8.
Repeat Steps 1–7 for all Communications Managers in the cluster.
Now that the reliance on DNS has been removed, the enterprise parameters should be
defined. The next sections discuss the function and proper settings for each of these
parameters.
Defining Enterprise Parameters
Enterprise parameters are settings that apply to the entire cluster. There are more than 70
enterprise parameters divided into more than 20 categories. Many of these settings might
never need to be changed. The following sections explore the settings that might be of
interest to an administrator. In some cases, the recommended setting depends on your
environment, and in these cases, an explanation of the setting is offered to help you
determine the proper setting for your environment. It is important to understand that
although some of these settings can be left alone, others (such as the path entries) should
be changed in most environments. Path entries are settings that point to a path that
includes the server name. Because it is recommended that the server name be changed to
the IP address, any field that points to the old name must be changed to point to the new
name (the IP address). One example of such a field is the URL directories path. Because
there are such a large number of parameters, an overview of each set will be offered. This
will provide a more detailed look at a few sets that might be of interest to the average
administrator.
General Parameters
The first category that you find within enterprise parameters has no special designation
and is often referred to as general parameters. The majority of these features are left at
their default.
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Chapter 2: Preparing CUCM for Deployment
The one that might be of interest is the Auto Registration Phone Protocol. This determines what protocol will be used for phones that autoregister. The default value is Skinny
Client Control Protocol (SCCP). If you happen to use Session Initiation Protocol (SIP) in
your environment and you use autoregistration, you need to change this value to SIP.
The following steps show how to set this field to SIP.
Note Some of these parameters require that the Communications Manager be restarted
before changes will take effect. You will be informed if you change a parameter that
requires a restart.
Step 1.
From within CM Administrator, open the System drop-down menu by placing
the mouse over System at the top of the screen.
Step 2.
Select Enterprise Parameters from the drop-down menu.
You are presented with a screen similar to that seen in Figure 2-3.
Figure 2-3 Enterprise Parameters Configuration
Step 3.
Select the Auto Registration Phone Protocol field.
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Configuring Cisco Unified Communications Manager and Unity Connection: A Step-by-Step Guide
Note Most field names appear to be hyperlinks. If you click the name of the field, a help
screen appears that offers details on the purpose of the field. When you want to change
the value of a field, you need to click in the field, not on the name of the field.
Step 4.
Select SIP from the drop-down menu that appears.
Step 5.
Click Save to save this change.
Communications Manager Administrator Parameters
The second category in enterprise parameters is CM Administrator Parameters. These
parameters affect the CM Administrator web-based interface. Because these parameters
affect how the system appears, all of them will be reviewed. The following steps show
how these parameters are configured and explain the function of each of them:
Step 1.
Navigate to the Enterprise Parameters page in CM Administrator.
Step 2.
The first field under the CM Administrator Parameters is labeled Max List
Box Items. It determines the maximum number of items that a list box can
contain. The default value is 250. This setting can be set in the range of 50 to
9999. The larger the number, the longer it takes for the page to load. It is best
to leave this value set to default. If the number of items is larger than the
value set, only the current value appears in the list. A button labeled “...”
appears next to the list. By clicking this button, you can search for the item
you want to select.
Note A list box is the list of items that appears when you click the down arrow at the
end of a field. In Figure 2-4, you see that a list box appears when the down arrow is
clicked in the Route Partition field.
Step 3.
The next field, the Max Lookup Items field determines how many items will
be returned to the browser when a lookup is done. The default for this setting
is 1000, which should be adequate for most installations. This field can be set
from 250 to 99999. Keep in mind that the higher this value is set, the longer it
takes to load the page.
Step 4.
The third field in this category, the Enable Dependency Records, field determines whether dependency records can be retrieved. Because this feature can
be tasking on the CPU, it is set to False (off) by default. It is best to leave this
value set to False. However, there are times when this feature can save hours.
For example, if you are trying to delete a Calling Search Space (CSS), you can
only delete it after you have removed it from all devices to which it is
assigned. When you try to remove the CSS, a message displays stating that
the CSS cannot be deleted because it is assigned to one or more devices. If
dependencies are not enabled, you must manually search to find to which
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Chapter 2: Preparing CUCM for Deployment
Figure 2-4 List Box Example
devices the CSS is assigned. You save time if dependencies are enabled,
because you can find out exactly the number and type of devices to which
the CSS is assigned. When this feature is needed, it is best to enable it (set the
value to True) after hours only and then disable it (set the value to False) when
you are finished using it.
Step 5.
The last field in this category is Auto Select DN on Any Partition. When this
field is set to False and a DN is added that already exists in the database, a
partition must be selected. If set to True and a DN that is already in the database is added, it will automatically be in the same partition as the existing
DN. The result is a shared line.
CCMUser Parameters
The third enterprise parameters category is called CCMUser Parameters. These settings
determine what options are available to a user from the user web pages. The user web
pages allow users to change many of their phone settings. Often you might not want the
user to change certain settings from the web pages, such as the Message Wait Indication
(MWI) policy. Each setting has two valid values: True or False. If an option is set to True,
it is available through the user web pages. If it is set to False, it is not available. This section simply explains what action the user can take from the user web page if the option is
set to True. If you do not want one or several of these features available from the user
web page, set the value to False.
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■
Show Ring Settings: Allows the user to set when and how a line rings, based on the
state of the line. This means that if the phone is idle, the user can choose to have the
phone ring normally, and when the phone is in use, the user can choose to have it
only flash the light. Each line can be configured to act differently. The ring options
available while the phone is idle are to ring normally, ring once, flash only, or do nothing. The options available while the phone is in use are to ring normally, ring once,
flash only, beep only, or do nothing.
■
Show Call Forwarding: Allows users to see whether call forwarding is currently set
on their phones and change it.
■
Show Speed Dial Settings: Allows users to see the current speed-dial settings for
their phones and change them.
■
Show Cisco IP Phone Services Settings: Allows users to view the services to which
the phone is currently subscribed. Also allows users to subscribe to new services.
■
Show Personal Address Book Settings: Allows users to view, edit, and add the
entries in their personal address books.
■
Show Message Waiting Lamp Policy Settings: Allows users to edit if and how each
line indicates that a new voicemail message has arrived for that line. They can choose
to have the light illuminate, have an icon appear next to the line number, both of
these options, or have no indication.
■
Show Line Text Label Settings: Allows users to configure the line text.
■
Show Locale for Phone Settings: Allows users to set the language on the phone.
■
Show Locale for Web Pages Settings: Allows users to set the language for the user
web pages.
■
Show Change Password Option: Allows users to change their passwords.
■
Show Change PIN Option: Allows users to change their PINs.
■
Show Download Plugin Option: Allows users to download plug-ins.
■
Show Online Guide Option: Adds a link to the user guide for the type of phone that
is associated with the user. The guide is in PDF format. In some cases, the guide
might not be available for certain model phones. This is normally the case if the
phone model is relatively new.
■
Show Directory: Allows users to search the directory from the web page.
■
Show Mobility Features Options: users to access remote destinations and access lists.
■
Show Manager Name in Directory: Displays the manager’s name in the directory
find list.
■
Show Extension in Directory: Displays the extension number in the directory find list.
■
Show LDAP Extension in Directory: Displays the Lightweight Directory Access
Protocol (LDAP) extension in the directory find list.
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CDR Parameters
The fourth enterprise parameters category is called CDR Parameters and contains only
the CDR File Time Interval parameter. This determines how many minutes of CDR data
that each CDR file will contain. The default is 1, which means that each CDR file will
contain one minute of data.
Localization Parameters
The fifth category within the enterprise parameters is called Localization Parameters and
deals with languages and tones. The language used depends on where the
Communications Manager solution is deployed. Just as there are different languages
around the world, there are also different tones and cadences for phones. For example,
when placing a call in England, the caller hears a double ringback instead of the single
ringback that is heard in the United States. The following steps show how to edit these
parameters and explains each of them:
Step 1.
Navigate to the Enterprise Parameters page in CM Administrator.
Step 2.
The first field under the Localization parameters is the Default Network
Locale field. This field is used to determine what types of tones and cadences
are used by the system. The default is United States. If needed, this parameter
may be changed by clicking the down arrow at the end of the field. A list of
valid choices display in the drop-down list. Choose the desired language from
the list.
Step 3.
The last field in this category is the Default User Locale field. This field
determines the default language used for the devices in the system. English
United States is the default value. If additional languages are installed on the
Communications Manager, they display in the drop-down list in this field.
Choose the appropriate language for your environment.
Multi-Level Precedence and Pre-Emption (MLPP) Parameters
The sixth category of enterprise parameters is the MLPP Parameters. MLPP adds the ability to grant higher priority to calls. This is useful in many government and military situations where it might be imperative that a call is successful. A call of a higher priority can
actually cause a call of a lower priority to be terminated so that the resources can be used
to complete the higher-priority call. The MLPP parameters deal with this feature. The following steps include an explanation of each parameter and the recommended settings:
Step 1.
Navigate to the Enterprise Parameters page in CM Administrator.
Step 2.
The first field under the MLPP Parameters is the MLPP Domain Identifier.
MLPP grants higher priority only from calls within the same MLPP domain.
For this reason, an MLPP identifier must be chosen. The default value is 0.
Step 3.
The second field in this category, which is called MLPP Indication Status,
determines whether tones and indications are presented when a precedence
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call is made. The precedence indication can be a special ringback or a display
notification, if the caller’s phone supports it, and a special ringer on the called
party’s side. By default, this parameter is set to MLPP Indication Turned Off.
To enable this feature, set this parameter to MLPP Indication Turned On.
Step 4.
The next field is MLPP Preemption Setting. This parameter determines
whether a higher-precedence call preempts a lower-precedence call. The
default value of No Preemption Allowed does not allow this to happen. To
cause a lower-precedence call to be terminated if a higher-precedence call
requires the resources, set this parameter to Forceful Preemption.
Step 5.
The next field, which is called Precedence Alternate Party Timeout, determines how long the system waits before sending a precedence call to the alternative party. The default is 30 seconds. Valid values for this field are 4 to 60.
Enter the desired value in the field.
Note This parameter is used when the called party has set an alternate party diversion.
This is separate from a standard call forward setting.
Step 6.
The last parameter in this category is labeled Use Standard VM Handling for
Precedence Calls and determines whether precedence calls are forwarded to
voicemail in the same manner as all normal calls. It is recommended that
precedence calls not be forwarded to voicemail, so the default value is False.
For a configuration that forwards precedence calls to voicemail, set the field
to True. It is recommended that the field be left at False.
Security Parameters
The seventh category within enterprise parameters is Security Parameters. These parameters are defined if security is enabled within the cluster and set as default security behavior for phones. There are four fields in this category:
■
Cluster Security Mode: This is the setting that displays whether the cluster is operating in a nonsecure or mixed-mode fashion. This field is read-only and can only be affected by running the CTL client plug-in. The valid values for this field are 0, which
indicates a nonsecure mode, and 1, which indicates a mixed mode.
■
CAPF Phone Port: This defines a port that listens for Cisco Authority Proxy
Function Service (CAPF) requests from a phone for a certificate. The valid values for
this field are 1023 through 55556. The default is 3804. Leave this set to 3804 unless
the default port is not used.
■
CAPF Operation Expires in (days): This parameter determines how many days a
CAPF operation has to complete. The valid values are 1 to 365. The default is 10.
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■
Enable Caching: Enable credentials can be stored in memory for up to 2 minutes.
This can save time for often-accessed credentials.
Prepare Cluster for Roll Back
This category has only one parameter, and it is only used if you are going to roll
Communications Manager back to a version earlier than 8.0. If you change this parameter,
users lose the use of features such extension mobility but still have basic call functions
until the parameter is set to False.
Phone URL Parameters and Secured Phone URL Parameters
The next two categories of enterprise parameters are Phone URL Parameters and Secured
Phone URL Parameters. These parameters deal with URLs that phones use for various
purposes. One example is the URL for services; this is the address that the phones point
to when the Services button is pressed. The difference between the two categories is that
the Phone URL Parameters are for the 7940, 7945, 7960, 7965, and 7970 phones, whereas
the Secured Phone URL Parameters are for secured models of Cisco phones. The following steps explain each parameter and its recommended configuration:
Step 1.
Navigate to the Enterprise Parameters page in CM Administrator.
Step 2.
The first field under the Phone URL parameters is called URL Authentication.
This field specifies the URL that certain phones use for authentication proxy
services between them and the LDAP directory. This value is set during the
installation and in most cases should not be changed. However, if you change
the name of the server to the IP address, make sure that you change that portion of the URL. For example, if the URL is http://DTW_CCM/CCMCIP/
authenticate.asp and you have changed the name of the server to its IP
address, which is 10.3.3.3, make sure that you change the URL to
http://10.3.3.3/CCMCIP/authenticate.asp.
Step 3.
The next field, which is called URL Directories, contains the URL that
phones use to access directory functions. This occurs when the Directories
button on the phone is pressed. Once again, if you changed the name of the
server to the IP address, make sure to change it in this URL.
Step 4.
The next field is labeled URL Idle. Some Cisco IP Phones have the capability
to display information on the screen after the phone is idle for a certain
amount of time. This parameter specifies the URL that the phone should call
to retrieve this information. A common use for this feature is to display the
company’s logo on the phone. By default this field is empty. Enter the URL
you want to use if you have configured one.
Tip For more information on how to create a graphic that can be used for this purpose,
refer to the document titled “Creating Idle URL Using Graphics on Cisco IP Phone” at
Cisco.com. The document ID is 42573.
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Step 5.
The fourth Phone URL parameter is URL Idle Time. This parameter determines how long a phone must be idle before the Idle URL is displayed. The
default is 0, which means that the URL is not displayed. Set the desired value
in seconds.
Step 6.
The next parameter is URL Information. This field specifies the URL that the
phone requests when the help function is invoked by pressing the i or ? button on the phone. Make sure that if you change the name of the server to the
IP address, you change it in this URL.
Step 7.
The next parameter, which is called URL Messages, contains the URL that
the phone requests when the Messages button is pressed. By default this field
is empty.
Step 8.
The next field, which is labeled IP Phone Proxy Address, specifies the
address of the proxy server that all phone URL requests use. If this field is
blank, the phone tries to make a direct request for URLs. By default this field
is empty. If this field is used, the address must contain the server name or IP
address and the port number.
Step 9.
The last parameter in this category is URL Services. This field contains the
URL that the phone requests when the Services button is pressed. Again, if
you change the name of the server to the IP address, make sure to change it in
this URL.
User Search Parameters
The ninth enterprise parameter category is User Search Parameters. Certain Cisco IP
Phones enable the user to search the phone directory from the LCD screen on the phone.
This feature is activated by pressing the Directories button on the phone. The newermodel phones have an icon that looks like an open book instead of the word Directories.
This gives the phones a more global look. These two parameters define the breadth of a
directory search and its results. These parameters also apply to searches that are made
from within the Communications Manager User Options web page.
The first parameter, which is labeled Enable All User Search, determines whether a search,
in which the user left all the fields blank, is allowed. By default, the value is True, which
means that a search of this nature would attempt to return all the names in the directory.
To disable this type of search, set the value to False.
The other parameter in this category determines the maximum number of results that a
search can return. This parameter is called User Search Limit. The default value is 64,
which means that no more than 64 results are returned. The valid values for this parameter range from 1 to 500. Keep in mind that the higher this value is, the longer it can take
to return the results. A number higher than 64 can have a negative effect on the performance of the Communications Manager.
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CCM Web Services Parameters
The tenth category of enterprise parameters is CCM Web Services Parameters. These
parameters limit the number of performance counters and device queries that can be
made. If too many requests are made, it can negatively affect performance. Other applications, such as the Voice Health Monitoring and Gateway Statistic Utility, might receive
delayed responses. The following are the four parameters with brief descriptions:
■
Allowed Performance Queries Per Minute: This parameter determines the maximum
Architecture for Voice, Video and Integrated Data (AVVID) XML Layer performance
counter queries that are allowed per minute. The default value is 50. Valid values for
this parameter are 1 to 80.
■
Allowed Device Queries Per Minute: This parameter determines the maximum AXL
device queries that are allowed per minute. The default value is 15. Valid values for
this parameter are 1 to 18.
■
Performance Queue Limit: This parameter determines the size of the queue for performance counter queries. If a query is made and the queue is full, the query is
dropped. The default is 100. Valid values for this parameter are 20 to 1000.
■
Maximum Performance Counters Per Session: This parameter sets the maximum
number of performance counters allowed per session.
Trace Parameters
The next set of parameters is the Trace Parameters. The first parameter is called File Close
Thread Flag. This parameter is set to False by default. If set to True, it enables separate
threads to be used to close trace files. This can improve performance when traces are run.
The last trace parameter is called FileCloseThreadQueueWatermark. This parameter
defines how many trace files the separate thread accepts before refusing trace files. The
trace file is then closed without the use of a separate thread. The default is 100. The valid
values are 0 through 500.
As you can see, there are a number of enterprise parameters. Most of them can be left at
the default. But others, such as those that contain the name of the Communications
Manager, might need to be changed. In most cases, the integrator makes these changes at
the time of installation. It is still a good idea to know the function of each parameter. If
one ever needs to be adjusted, you know where to find it, how to change it, and what the
impact of the change is.
At this point, the system is almost ready for the attachment and configuration of devices.
There are, however, a few more tasks that should be completed before device attachment
and configuration. The next sections examine the core tasks that should be accomplished
before devices are allowed to register to the system.
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User Management Parameters
This category has only one parameter, which is Effective Access Privileges for
Overlapping User Groups and Roles. The parameter determines whether the maximum or
minimum user privileges will be assigned when a user is a member of multiple groups
that contain overlapping rights.
Service Manager TCP Ports Parameters
The TCP ports that the Service Manager uses to send and receive data are defined in the
Service Manager TCP Ports Parameters category.
CRS Application Parameters
The two parameters in the CRS Application Parameters category simply define whether
the Auto Attendant or IPCC Express is installed. These parameters are read-only and are
set by the Customer Response Solutions (CRS) server.
Cluster Domain Configuration
The Cluster Domain Configuration parameters allow you to configure the top-level
domain and the fully qualified domain name. For example, the Organization Top Level
Domain field for Cisco would be cisco.com. The Cluster Fully Qualified Domain Name
parameter for Cisco might be something like cm2.dtw.cisco.com.
Denial-of-Service Protection
The Denial-of-Service Protection parameter should be set to True and helps prevent some
types of denial of service (DoS) attacks.
TLS Handshake Timer
The Transport Layer Security (TLS) Handshake Timer parameter determines the maximum amount of time (in seconds) that a TLS protocol handshake can be performed. It is
recommended that this be left at the default because this value helps prevent DoS attacks.
Cisco Support Use
The two parameters in the Cisco Support Use category are, as you might have guessed,
only for use by Cisco technical support.
IPv6 Configuration Modes
There are four parameters in the IPv6 Configuration Modes category. The first one,
Enable IPv6, determines whether Communications Manager and the phones are going to
participate in an IPv6 environment. The default is False and should only be set to True if
your environment is configured to support IPv6.
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The next two parameters, IP Addressing Mode Preference for Media and IP Addressing
Mode Preference for Signaling Required Field, determine whether IPv4 or IPv6 will be
used for media and signaling events. This only applies to dual-stack devices.
The last parameter in this category, Allow Auto-Configuration for Phones, determines
whether the phones will be allowed to use stateless autoconfiguration. If set to On, the
phone can use advertisements from the router to obtain an address; if set to Off, the
phone will use Dynamic Host Configuration Protocol version 6 (DHCPv6).
Cisco Syslog Agent
The Cisco Syslog Agent parameters allow you to define a syslog server that a syslog message can be sent to as well as the severity level of the messages that should be sent. Set
the name of the syslog server in the Remote Syslog Server Name field and the severity in
the Syslog Severity for Remote Syslog Messages field.
CUCReports Parameters
The Report Socket Connection Timeout parameter defines the maximum amount of time
(in seconds) that is allowed while establishing a connection to another server. The Reports
Socket Timeout defines the maximum amount of time (in seconds) that is allowed when
reading data from another server.
Logical Partitioning Configuration
The last category of parameters is the Logical Partitioning Configuration parameters.
Logical partitioning is used to permit or deny calls between the following pairs of
VoIP devices:
■
VoIP phone and a VoIP gateway
■
VoIP gateway and another VoIP gateway
■
Intercluster trunk and a VoIP phone
■
Intercluster trunk and a VoIP gateway
Certain areas have restrictions on the type of VoIP calls that are permitted. For example,
in India, it is illegal to have a call that passes through a PSTN gateway connect directly
by using a WAN to a VoIP phone or VoIP PSTN gateway in a different geographic location. Logical partitioning can prevent calls like this from happening.
Logical partitioning works with geolocation settings to determine the location of each
device and determine whether the call is allowed. The following four parameters are defined:
■
Enable Logical Partitioning: This parameter determines whether logical partitioning
is enabled. This is used to restrict certain types of features and calls.
■
Default Geolocation: This parameter determines the default geolocation for all
devices. The geolocation settings define up to 17 parameters that define the device’s
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geographical location. These parameters define the country, city, postal code, and so
on. More detail on all 17 parameters can be found in RFC 4119, “GEOPRIV
Location Object.” This setting can be overridden at the device level.
■
Logical Partitioning Default Policy: This parameter determines whether calls
between geolocations will be allowed or denied by default.
■
Logical Partitioning Default Filter: This parameter can help limit the number of
fields that appear in the geolocation configuration window. Geolocation filters must
be configured before you can modify this field.
Preparing Communications Manager for Device
Registration
When a device registers with Communications Manager, a number of things need to be
known about that device, such as the type of device and the IP address and MAC
addresses. However, determining other information is also necessary. This includes the
particular time zone that the phone is located in and with which Communications
Manager the device should register. Instead of defining this information at every device,
two components known as device pools and common device configuration are used.
Because most often a large number of devices require the same information, the needed
information is assigned to a device pool and a common device configuration. Then the
device pool and common device configuration are assigned to those devices. This way,
instead of defining multiple configuration values for each device, you assign the values
once to the device pool and common device configuration.
In the versions of Communications Manager prior to 4.2, there were only device pools. In
4.2, the device pool parameters were spilt into two components, now referred to as device
pools and common device configurations. The reason for this change was to add more
flexibility for mobility. There are certain parameters that should change if the device
moves and others that should stay the same regardless of where the device resides.
Parameters that might need to be changed are stored in the device pool, and parameters
that will always stay the same, regardless of location, are configured in the common
device configuration.
Device Pools
As of Communications Manager 8.0, the device pool contains more than 30 fields. Eight
of these fields are required to be configured when creating a new device pool. Although
configuring the others is not required, it is recommended. This section briefly explains
each field and explores how to configure the fields that are crucial to a proper deployment. The following is a list of fields found in a device pool:
■
Device Pool Name (Required): No doubt you can figure out what this is without any
description. It is simply the name of the device pool. The name you choose should
enable you to easily identify the function of this device pool. For example, you might
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name the device pool EST-DTW1-DTW2 for a device pool that is used by devices
that are in the Eastern time zone and use Communications Manager DTW1 as a primary Communications Manager and Communications Manager DTW2 as a secondary server.
■
Cisco Unified Communications Manager Group (Required): A Communications
Manager group defines which Communications Manager is used as the primary server, which is the secondary server, and which is the tertiary server. The next section
discusses how to create these servers.
■
Calling Search Space for Auto-Registration: When using autoregistration, it is a
good idea to restrict where the device can call until its legitimacy is confirmed. This
field allows you to specify a calling search space for all devices that autoregister.
■
Adjunct CSS: This field is used when cross-cluster Extension Mobility is deployed. It
is used to ensure that that the country-specific emergency dialing plan is supported.
■
Revert Call Focus Priority: This determines whether reverted (call returning after
being on hold) or incoming calls will be the focus of user action. In other words, if
there is an incoming call and a reverted call coming in at the same time, this field will
determine which call is answered when the phone goes off hook.
■
Local Route Group: Local route groups are used to streamline dial plan creation for
multisite clusters.
■
Intercompany Media Services Enrolled Group: IME allows calls outside an organization to be routed across the Internet through dynamic SIP trunks. If IME is
deployed in your organization, the appropriate IME enrolled group is required.
■
Date/Time Group (Required): Because it is possible that devices belonging to the
same cluster might be in different time zones, it is necessary to define the correct
time zone for each device. The date/time group is used to specify to which time
zone a device belongs. The date and time format is also defined by the date/time
group. The configuration of date/time groups is covered later in this chapter.
■
Region (Required): Because VoIP calls can traverse links of different bandwidth, it
is often necessary to use a codec that requires less bandwidth to make most efficient use of the link. A codec converts voice signals from their analog form to digital signals that are acceptable to modern digital transmission systems. Some codecs
use algorithms that compress the voice stream so that less bandwidth is required.
Using regions, you can define what codec one device should use when talking to
another device. Further discussion and the configuration of region are covered later
in this chapter.
■
Media Resource Group List: Media resources provide features such as Music on
Hold (MoH) and conferencing. A media resource group list defines device access
within a media resource.
■
Location: Locations are used to determine the amount of bandwidth that can be
used for VoIP calls. CUCM uses the value in this field in conjunction with the region
to determine whether bandwidth is available for a call.
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■
Network Locale: This field determines what locale is used for the devices. The field
impacts the tones and cadences used. The default network locale was defined in the
enterprise parameters. If a different value is selected here, this value takes precedence. Make sure that you select only a locale that has been loaded in the system;
otherwise, the device associated with this device pool fails. If this field is set to
None, the enterprise parameter setting is used.
■
Survivable Remote Site Telephony (SRST) Reference (Required): SRST is a feature
that allows phones that are remote to the Communications Manager to continue
functioning, even if they cannot communicate with the Communications Manager.
This is done by having the phones register with a local router that is acting as an
SRST box. The SRST reference is used to define which device the phone should try
to register with if it is unable to communicate with the Communications Manager.
Another use for SRST is when a small office has only a single Communications
Manager. Although it is always recommended to have two Communications
Managers for redundancy purposes, it is possible to use SRST for redundancy when
only one Communications Manager is installed.
■
Connection Monitor Duration: This setting defines how long a phone, which is in
SRST mode, waits to fail back to the Communications Manager after it detects that
Communications Manager is back online. This can help prevent the phone from
switching back and forth between the SRST server and the Communications
Manager when a Communications Manager connection is going up and down (also
know as flapping).
■
Single Button Barge: Determines whether barge or cBarge can be invoked by pressing a single button.
■
Join Across Lines: Determines whether a user can join calls on separate lines together
to create an ad hoc conference.
The following parameters relate to device mobility. Device mobility allows a device to
assume settings based on physical location. For example, perhaps you work in New
York but must relocate to the Chicago office for a month. You can take your phone with
you, and when you plug it in to the Chicago network, you would have the same extension but certain settings would change because you are in Chicago. This is accomplished
because the device pool can be selected based on the network into which you are
plugged. If the device pool that is selected is different than the device’s preprogrammed
device pool, the system checks the physical location. If the physical location is different,
the device is considered to be roaming, and the system assigns the Device Mobility
parameters to the device:
■
Physical Location: This field is used to set the physical location of the device. This is
useful when the Device Mobility feature is used. Because device pools can be assigned based on the subnet that a device is in, the correct physical location will be associated to the device automatically when it is moved.
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■
Device Mobility Group: This determines which device mobility group the device
belongs to.
Note If the following parameters are configured here and at the device level, the device
level overrides these parameters. If using Device Mobility, it is best to leave these set to the
defaults at the device level and set them here.
■
Device Mobility Calling Search Space: This Defines the CCS that the device will use
when the device is roaming.
■
AAR Calling Search Space: Automated Alternate Routing (AAR) allows a call to be
routed over the PSTN if the WAN is not available. Because a device’s CSS might
not allow it to place a call using the PSTN, the AAR CSS is used to allow the call to
be placed.
■
AAR Group: This Defines which AAR group the device belongs. This is used to determine how the dialed call must be transformed so that is can be placed over the PSTN.
■
Calling Party Transformation CSS: Calling party transformation allows you to
change the number that is presented as the caller ID. The CSS selected must have
access to the calling party transformation pattern that is assigned to the device pool.
■
Called Party Transformation CSS: Called party transformation allows you to
change the number that is dialed. The CSS selected must have access to the called
party transformation pattern that is assigned to the device pool.
■
Geolocation: Some features can require geolocation information. This information is
also referred to as a civic address. The geolocation information can be used to determine the logical partition of a device.
■
Geolocation Filter: There are 17 configurable geolocation fields. Geolocation filters
allow you choose which fields are used to create a geolocation identifier.
■
Incoming Calling Party Settings: These fields allow you to add prefix and strip digits for numbers that are defined as national, international, or subscriber in the Calling
Party Number Type field. This field is configured when you create a pattern.
■
Incoming Called Party Settings: These fields allow you to add prefix and strip digits
for numbers that are defined as national, international, or subscriber in the Called
Party Number Type field. This field is configured when you create a pattern.
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Common Device Configuration
Based on the device’s physical location, some of the parameters might need to change.
The parameters found in the Common Device Configuration section remain the same
regardless of the physical location of the device:
■
Network Hold MOH Audio Source: There are two kinds of hold: user hold and network hold. A user hold occurs when the Hold button on the phone is pressed. Any
other time a call is placed on hold, for example, if the call is being transferred, it is
considered a network hold.
The Network Hold MOH audio source determines what audio stream a caller hears
when placed on a network hold.
■
User Hold MOH Audio Source: The User Hold MOH audio source determines what
audio stream a caller hears when placed on a user hold.
■
User Locale: This parameter determines the language and fonts used for the devices
in this device pool. The default user locale, which is set in the enterprise parameters,
will be used if this field is set to None.
■
MLPP Indication (Required): This field determines whether tones and indications
are presented when a precedence call is made to or from a device in this device pool.
If this field is set to Default, the MLPP Indication setting in the enterprise parameters is used.
■
MLPP Preemption (Required): This parameter determines whether a higher-precedence call preempts a lower-precedence call. If this field is set to Default, the MLPP
Indication setting in the enterprise parameters is used.
■
MLPP Domain: This field determines which MLPP domain the devices in this device
pool belong to. If this field is left blank, the MLPP Domain Identifier defined in the
enterprise parameters is used.
A number of these fields point to other objects that might not have been configured
yet. Throughout this book, many of these components are discussed in more detail.
From this point on, we are going to focus on the required components that most
often need to be defined before devices are deployed.
Creating Communications Manager Groups
A Communications Manager group defines which Communications Managers a given
device points to as its primary, secondary, and tertiary Communications Manager. The
concept is quite simple. The Communications Manager group contains a list of
Communications Managers; the order in which they appear determines their function
(that of either a primary, secondary, or tertiary Communications Manager). A
Communications Manager can be in more than one group, and its function can be different in each group. For example, Table 2-2 shows two Communications Manager groups.
Each group contains the same three servers, but they appear in a different order. The
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result is that the DTWA Communications Manager is the primary in Group DTWABC,
but it is the tertiary in group DTWCBA. The only thing that defines the Communications
Manager’s function in regard to primary/secondary/tertiary is the order in which they
appear in the Communications Manager Group.
Table 2-2
Two Communications Manager Groups
Communications Manager Group
DTWABC
Communications Manager Group
DTWCBA
DTWA
DTWB
DTWC
DTWC
DTWB
DTWA
Creating Communications Manager groups is a simple task, but before creating them, you
need to spend some time determining how many Communications Manager groups you
need. The number of Communications Managers you have will impact the number of
Communications Manager groups you have. For example, if you have only two
Communications Managers, you have only two Communications Manager groups at
most. However, having six Communications Managers by no means indicates that there
will be six Communications Manager groups. When determining the number of
Communications Manager groups, keep two goals in mind: redundancy and load balancing. By keeping these goals in mind and using common sense, you can easily figure out
the proper number of Communications Manager groups for your environment. For further information on Communications Manager groups and cluster sizes, refer to the Cisco
IP Telephony Network Design Guide, which can be found on Cisco.com by searching for
“Cisco IP Telephony Network Design Guide.”
Communications Manager groups are configured using CM Administrator. The process is
quite simple, as you will see in the following steps. Before you start, make sure that all
Communications Managers are running and able to communicate with each other. To add
Communications Manager groups, follow these steps:
Step 1.
From within CM Administrator, open the System menu by mousing over
System at the top of the window and selecting Cisco Unified CM Group.
Step 2.
Click the Add New link in the upper-left corner of the screen.
The screen will display a figure similar to that shown in Figure 2-5.
Note A default Communications Manager group already exists. If you choose to use the
default, make sure that all the desired Communications Managers appear in this group in
the desired order. This can be done using the steps listed here as well as by clicking the
Find button and then selecting the Default group instead of selecting Add a New
Communications Manager Group.
Step 3.
Enter the desired name for this Communications Manager group.
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Give it a name that it is easily identified. For example, in the example shown
in Figure 2-5, the name CMGAB is used to show that Communications
Managers DTWA and DTWB belong to that group in that order.
All the available Communications Managers are displayed in the Available
Cisco Unified Communications Managers box.
Figure 2-5 Communications Manager Group Configuration
Step 4.
Highlight the Communications Manager that you want to act as the primary
Communications Manager in this group.
Between the two boxes on the screen are two arrows. To move the selected
server, click the arrow that points to the Selected Cisco Unified
Communications Managers box.
Step 5.
Repeat Step 4 to move the remainder of the desired Communications
Managers into the Selected Cisco Unified Communications Managers box.
The order in which the Communications Managers appear in the Selected
Cisco Unified Communications Managers box determines whether they are
considered primary, secondary, or tertiary. You can change the order in which
they appear by highlighting a Communications Manager and clicking the up
and down arrows to the right of the box.
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Step 6.
After all the Communications Managers have been added and are displayed in
the desired order, click the Save button.
Use these steps to configure all Communications Manager groups that you need. After
that is accomplished, you need to define data/time groups.
Defining Date/Time Groups
Imagine that you have asked someone to call you at 10:00 a.m. and that person calls you
at 11:00 a.m. You answer the phone frustrated from the wait, only to find that your friend
is calling from a different time zone and thinks she’s on time. Just as we need be aware of
time zones in our daily lives, we need to make sure that Communications Manager knows
the time zone of its registered devices. This is accomplished by configuring a date/time
group for each time zone that devices will be in.
The default date/time group, called CMLocal, is created when Communications Manager
is installed, and it is set to the zone selected during the installation. If all the devices are
in the same time zone, you can use the CMLocal date/time group without adding any
others. However, if your devices are in different time zones, you can use these steps to
guide you through the process of creating a date/time group for each time zone:
Step 1.
From within CUCM Administrator, open the System menu by mousing over
System at the top of the window and selecting Date/Time Group.
Step 2.
From the screen that displays, either search for an existing date/time group or
add a new one. Click the Add New link in the upper-left corner of the screen.
A screen similar to that shown in Figure 2-6 appears. All fields on this screen
are required. The first field is Group Name.
Step 3.
Give the group a name that is easily identified.
For example, the date/time group that is being created in Figure 2-6 is for the
devices that are in Chicago, and Chicago is in the Central time zone, so it has
been named Chicago-CST.
Step 4.
In the Time Zone field, select the correct time zone from the drop-down list
that is displayed when you click the down arrow at the end of the field.
Step 5.
The next field, called Separator, is used to define the symbol that is used as
a separator when the date is displayed. The valid values for this field are a
dash (-), forward slash (/), or dot (.). Choose the desired value from the drop
down list.
Step 6.
Choose the desired date format from the Date Format drop-down list.
This field determines how the date is presented. The choices are month-dayyear, day-month-year, or year-month-day.
Step 7.
Choose the desired time format from the Time Format drop-down list.
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Figure 2-6 Date/Time Group Configuration
This field determines whether the time will be displayed in 12-hour or 24hour format.
Step 8.
Click the Save button.
Repeat these steps for each date/time group needed.
Configuring Regions
When calls traverse lower-bandwidth IP links, such as a WAN link between offices, it
might be necessary to compress the audio stream so that the link is used efficiently. The
compression is accomplished by using a codec that has a lower bandwidth requirement,
such as G729, for the call. When using a codec that requires lower bandwidth, the quality
of the call is affected. For this reason, it is recommended that these codecs are used only
when bandwidth is at a premium. With this in mind, you can understand that the codec
used should depend on the call destination, rather than the device placing the call.
Regions allow this.
The two most often-used codecs in a Cisco VoIP environment are G.711 and G.729.
G.711 requires approximately 80 Kbps when overhead is included. G.729 requires only
24 Kbps including overhead. Because most of the newer-generation Cisco IP Phones
support both codecs, G.729 is often used when traversing a WAN link.
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By creating and assigning regions, you define which codec is used between regions.
Figure 2-7 shows three regions and the codecs that are used when calls are placed within
the region and to other regions. As you can see, if someone from Detroit calls another
party in Detroit, the G.711 codec is used, but when a party in Detroit calls someone in
Chicago, the G.729 codec is used.
Detroit
Chicago
G.729
Within Chicago = G.711
9
72
G.
G.
72
9
Within Detroit = G.711
San Jose
Within San Jose = G.711
Figure 2-7
Use of Regions Example
Note In newer versions on Communications Manager, you configure the maximum bit
rate that is used between regions instead of the codec. The end effect is the same it is just
displayed differently. For example if you want to use G.729 between regions you would set
the Max Audio Bit Rate to 8 kbps. The bit rate is selected from a drop-down list. Next to
the bit rate the codec that aligns with the bit rate displays.
A default region named Default is created during the installation of Communications
Manager. If calls in your environment never traverse lower-bandwidth links, additional
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regions might not be needed. Regions are created from within CM Administrator. You
can create the required regions by following these steps:
Step 1.
From within CM Administrator, open the System menu by mousing over
System at the top of the window and selecting Region.
From the displayed screen, you can either search for an existing region or add
a new one.
Step 2.
Click the Add New link in the upper-left corner of the screen.
Step 3.
On the next screen, enter the name of the region in the Name field.
It is recommended that you enter a name that is easily identified. In this example, because the region will be used for devices that are in Detroit, the region
is named Detroit.
A figure similar to that shown in Figure 2-8 appears.
At this point, notice that the Max Audio Bit Rate is set to Keep Current
Setting. The default for intraregion calls is G.722 if supported; if not, G.711 is
used. For inter-region calls G.729 is used by default. Based on these defaults,
to have calls over the WAN use G.729, simply create new regions and assign
them to device pools. In previous versions of Communications Manager you
had to manually set the codec. To change the bit rate/codec, select the desired
bit rate from the Max Audio Bit Rate dropdown list. For example, in Figure
2-8, the Detroit region is being configured. To change the region used
between the Detroit region and the Default region, highlight the Default
region in the Regions box and select the desired bit rate from the Max Audio
Bit Rate drop-down list.
You might want to configure a few more things before creating device pools. One is softkey templates. There is a default softkey template, so if you plan on using just that one,
you can move on to creating device pools. Because softkey templates are specific to
phones, they are covered in the next chapter with phone button templates. For the purpose of the next section, which covers building device pools, we use the default values
available for all other required objects.
Building Device Pools
Now that the device pool’s required components have been configured, you can create
the device pools. The first thing you have to determine is how many device pools are
needed. This is determined by looking at the number of device pool components you
have created and how they tie together. Following is a simple example.
Imagine that you have configured two Communications Manager groups, three regions,
and two date/time groups, as shown in Table 2-3.
You can tell that because you have three regions, there need to be at least three device
pools. If the devices in each region are part of the same date/time groups, and use the
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Figure 2-8 Adding a Region
same Communications Manager group, only three device pools are needed. However, in
this example, you want half the phones in each region to use Communications Manager
group DTWABC and the other half to use Communications Manager group DTWCBA,
so three device pools won’t be enough. It would take six device pools to accomplish this,
as shown in Table 2-4.
From this simple example, you can see how the number of device pools could easily grow
if not managed properly. So, what is the right number of device pools? There is no definitive answer. That depends on your specific requirements, but make sure that you take the
time to properly plan the number of device pools that your environment will require.
Table 2-3
Determining the Number of Device Pools Needed
Communications Manager Groups
DTWABC
DTWCBA
Regions
Detroit
Grand Rapids
Chicago
Date/time groups
Eastern
Central
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Table 2-4
Device Pool Example
Device Pool
Communications
Manager Group
Region
Date/Time Group
DTWABC_DP
DTWABC
Detroit
Eastern
DTWCBA_DP
DTWCBA
Detroit
Eastern
GRRABC_DP
DTWABC
Grand Rapids
Eastern
GRRCBA_DP
DTWCBA
Grand Rapids
Eastern
ORDABC_DP
DTWABC
Chicago
Central
ORDCBA_DP
DTWCBA
Chicago
Central
After all the components that belong in a device pool are created, you have planned out
how many device pools you need, and planned what components will belong to each
pool, creating the device pools is quite simple. The following steps show how to create a
device pool. The steps will not detail each component because each was discussed previously in this chapter.
Step 1.
From within CM Administrator, open the System menu by mousing over
System at the top of the window and selecting Device Pool.
From the displayed screen, you can either search for an existing device pool
or add a new one.
Step 2.
Click the Add New link in the upper-right corner of the screen.
A screen similar to that shown in Figure 2-9 appears.
Step 3.
In the Device Pool Name field, enter the name of the new device pool.
Remember to give the device pool an easily identifiable name.
Step 4.
Now simply select the desired object for each of the remaining fields.
If you are uncertain of the function of any object, refer to the earlier section
in this chapter that discussed each field. The fields that are marked with an
asterisk (*) are required, so you must select a value. The other fields are not
required, but it is recommended that a value be set. Many of these values are
discussed in more detail in future chapters.
Step 5.
After you select an object for all required and desired fields, click the Save
button.
That’s all there is to creating a device pool. Creating a device pool is simple because the
pool’s only function is to point to other objects. The real work, as you learned in this
chapter, is making sure that all the objects contained in the device pool have been created.
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Figure 2-9 Device Pool Configuration
Summary
This chapter examined various Communications Manager predeployment tasks. First, the
Windows and Communications Manager services that are required were discussed, in
addition to those that can be disabled to increase the performance of the server. Next,
the DNS requirements and enterprise parameters were covered, and finally the device
pools and components of the device pools were explored. At this point, you are ready to
start adding devices to the system. The next chapter discusses what tasks must be performed to add phones and gateways and describes the various ways that these devices
can be added.
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Chapter 3
Deploying Devices
After all predeployment tasks are completed, devices can be added to the system. This
chapter discusses the tasks required to add certain devices to the Communications
Manager cluster.
A number of types of devices can be added to the Communications Manager. The
devices discussed in this chapter fall into one of two categories: clients or gateways.
Other devices, such as gatekeepers, are discussed in future chapters.
You start by looking at clients, more specifically phones. There are a number of different
models of Cisco IP Phones, but the task of adding each is similar. Next, gateways are covered. Gateways allow connectivity to another system such as the public switched telephone network (PSTN) or another PBX. In Communications Manager 8.0, there are more
than 40 different types of gateways, and although the function of each is similar, the configuration varies. It is not possible to provide a step-by-step guide for the configuration
of each gateway, but this chapter includes detailed steps on how to configure the most
popular types.
Note In all the steps in this book, I have tried to cover all the parameters that appear on
the screen. Because each version of Communications Manager adds fields, all the fields
that you see in these steps might not appear on your screen, or additional fields might be
present, because of the version of Communications Manager you are running. Regardless
of the version you run, the steps in this chapter can help you walk through the process.
Not all the parameters must be configured. Some can remain at default, whereas others are
not required. The end of the names of required parameters are marked with an asterisk (*).
In some cases, you might need to configure only the required parameters for a device to
function. It is a good idea, however, to review each parameter so that you can be certain
that the device is configured exactly the way you want it.
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Adding Clients
Depending on the environment, adding a phone can be as simple as plugging it into a port
that has connectivity to the Communications Manager. Although you can configure a
Communications Manager to allow phones to be automatically added simply by connecting them to the network, it is not always wise or desired. The following sections explore
four ways that phones can be added to the system. Before adding phones to a system,
there are a few more components to configure.
You are probably thinking, “Wait a minute; the last chapter discussed predeployment
tasks.” That’s right, it did, but those were the general or global settings that should be
configured. These sections look at settings that are specific to phones.
Note Which came first, the chicken or the egg? This is a question that has plagued the
minds of great thinkers from the beginning of time. Okay, maybe it hasn’t, but a similar
issue sometimes occurs when teaching new technologies. When two separate concepts are
interdependent, it might be difficult to grasp either concept until you understand both of
them. This being said, you will notice throughout this chapter that new concepts and components are mentioned that are not discussed in detail until later in the book. In these
cases, a brief description is offered for each new concept, as is a reference to the chapter
that lends greater detail. You might choose to jump ahead to gain a better understanding or
just accept the brief description, knowing that more detail is offered later.
Defining Device Settings
Before adding phones, it is recommended, but not required, that some device settings be
configured. Configuring the device setting first will most likely save time. These settings
include phone button templates, softkey templates, and device defaults. The following
section explains the function of each of these and describes how to configure them.
Phone Button Templates
Some Cisco IP Phones have buttons that can be configured for various functions.
Although these buttons can be configured to support a number of different functions,
the most popular functions are lines and speed dials. When configured as lines, extension
numbers can be assigned to the buttons. Each phone must have at least one button configured as a line. When a button is configured as speed dial, administrators or users can
assign a speed-dial number to it, and because most Cisco phones have only a few of
these buttons (one to eight), the number of speed dials that can be defined in this way is
limited. The user can access additional speed dials by using a feature known as abbreviated dial, which is discussed later in this chapter.
In addition to lines and speed-dial functions, these buttons can be configured for other
functions, depending on the phone model. Although the majority of phones sold today
have configurable buttons, you need to know that not all phones have that capacity.
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When you create a new phone button template, it will be quite clear which buttons on
certain phones are not configurable.
The exact function that can be assigned to buttons varies depending on the model of the
phone. The buttons on most phones can be defined to support a number of features, but
the most common are as follows:
■
Line: Extension number is assigned.
■
Speed dial: Assigned number is dialed when button is pressed.
■
Service URL: A phone service is accessible by pressing the button.
■
Privacy: Makes a call private when Call Barge is enabled.
■
BLF/Speed dial: In addition to acting as a speed dial, the status of the extension assigned can be monitored.
The other features that can be assigned to buttons are typically assigned to a softkey, so
they do not need to be assigned to a button. Softkeys are discussed later in this chapter.
When a phone is added to the system, a phone button template is associated to the
phone. The template is used to determine the function that each button will serve. If the
phone is added through autoregistration, it uses the template defined under device
defaults. Device default settings are discussed later in this chapter. The creation of these
templates is fairly simple.
To create phone button templates follow these steps:
Step 1.
From within CCMAdmin, select Device > Device Settings > Phone Button
Template.
Step 2.
Click the Add New link.
Step 3.
On the next page, you must select an existing phone button template to copy.
This is because a new phone button template must be based on an existing
template. From the drop-down list in the Phone Button Template field, select
the standard template for the correct model of phone. For example, if you are
creating a phone button template for a 7960, select Standard 7960 SCCP.
Step 4.
Click the Copy button.
Step 5.
In the Button Template Name field, enter a descriptive name and click Save.
Step 6.
A screen similar to that shown in Figure 3-1 displays. At first glance, you
might think that you have selected the wrong phone type because it shows 34
buttons instead of the six you were expecting. Because the 7960 can support
two 7914 expansion modules, which support 14 buttons apiece, 34 total buttons are possible.
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Configuring Cisco Unified Communications Manager and Unity Connection: A Step-by-Step Guide
Figure 3-1
Phone Button Template Configuration
Note It is recommended that the button template name identify the configuration of this
template. For example, if the template is configured with two lines, three speed dials, and a
privacy button, a good name would be “7960 2-3 w/Privacy.”
Step 7.
Assign the proper function to each line by selecting the function from the
drop-down list next to the button number.
Step 8.
Next to the Feature field is a Label field. Enter a descriptive label in this field,
such as Speed Dial 2 or Line 3.
Step 9.
Click Save to save the new template.
Softkey Template
In addition to the buttons that can be used for lines and speed dials, Cisco IP Phones
have buttons that are referred to as softkeys. In Figure 3-2 you can see where softkeys
appear on the phone. These buttons (keys) allow the user to access features of the phone
such as hold, transfer, conference, as well as many others. The function of each key
changes depending on the state of the call. Because there are often more available functions configured than physical softkeys, the last softkey functions as a toggle key, allow-
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Chapter 3: Deploying Devices
ing the user to scroll through all the available options. The softkey template allows you to
determine what functions are available on the phone and in what order they display.
Much like phone button templates, the softkey templates can be associated directly with
a phone. However, as you saw in the last chapter, the softkeys templates are also associated with device pools, which are in turn associated with phones. By associating the softkey template to a device pool, you can easily and quickly assign the softkey template to a
large number of phones. If different softkey templates are associated with a phone at
both the phone level and the device pool, the one at the phone level takes precedence.
Phone Button (line, speed dial, etc)
Softkeys
Figure 3-2 Softkeys and Phone Buttons
The following steps take you through the process of creating a softkey template.
Step 1.
From within CCMAdmin, select Device > Device Settings > Softkey
Template.
Step 2.
Click the Add New link.
Step 3.
On the next page, you must select an existing softkey template for the basis
of the new template. Select a template from the drop-down list in the Create
a Softkey Template Based On field.
Step 4.
Click the Copy button.
Step 5.
On the next screen, enter the name and description for the new template. The
description should help identify the features associated with the template.
The Application field lists the applications that are available in this template
and cannot be changed on this screen.
Step 6.
Click Save to add the new template.
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Step 7.
After the template is added, you must configure it. If you want to add applications that are found on other softkey templates, click the Add Application
button. If you do not want to add applications from an existing template, skip
this and the next step. For example, you would not add an application if you
are simply moving or adding standard softkeys to the template.
Step 8.
If you selected Add Application in the new window that displays, select the
standard softkey template that contains the application you want to add to
the new template and click Save and then Close.
Step 9.
In the upper-right area of the screen, you can find the Related Links dropdown list. Make sure that Configure Softkey Layout is selected and click Go.
A screen similar to that found in Figure 3-3 displays. The first field is a dropdown list of all the possible states of a phone. Because the softkeys change
depending on the state of the phone, each state must be configured separately. In Figure 3-3, the new template allows access to the callback feature. In this
example, the callback softkey must be configured for all states from which it
might be accessed, which are On Hook and Ring Out.
Figure 3-3 Softkey Layout Configuration
Step 10. Determine which call state you want to modify and select it from the list. For
example, if you wanted to add a softkey that would appear on the phone
while you were on a call, you would select Connected.
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Step 11. The softkey currently assigned displays in the box on the right side of the
screen labeled Selected Softkey. The softkeys that can be added display in the
box on the left side labeled Unselected Softkeys. To add unselected softkeys,
click once on the softkey and then click the top arrow of the two arrows that
display between the two boxes.
Note Because there are a limited number of softkeys on the phone, it is common to have
more softkeys assigned than buttons on the phone. In this case, the last button becomes a
More button that acts as a toggling mechanism to allow users to access the other features.
You need to understand that the More button automatically appears on the phone and is
not something you configure in the template.
Step 12. To remove a current softkey, click the softkey you want to remove, and click
the bottom arrow of the two arrows that display between the boxes.
Step 13. The softkeys display on the phone in the same order that they display in the
Selected Softkeys box. To change the order in which they display, highlight
the desired softkey, and click the up or down arrow that displays on the left
side of the Selected Softkey box.
Note If softkeys are used in more than one call state, for example, the callback softkey is
available in the On Hook and Ring Out call states; pay attention to the position of the softkey in each of the call states. It’s a good idea to keep the softkey in the same position for
each call state if possible. If it is desirable to leave one or more softkey positions unused,
the Undefined softkey can be used as a placeholder.
Step 14. After you modify all the desired call states, click Save.
Note When modifying an existing softkey template, you need to reset the devices associated with the templates. To do this, click the Restart Devices button on this page. Take
care when resetting devices, because resetting causes the phones to be unusable while
they reset. This does not affect phones currently on a call. The reset occurs when the
phone is idle. It is always recommended that whenever a reset must be performed on a
large number of phones, it be done off hours to minimize the impact on the end user.
Device Defaults
When a phone boots, it requests a configuration file from the Communications Manager
TFTP server. The file that it requests is associated with the phone’s MAC address. If the
phone has not previously been registered with a Communications Manager on the system
and has not been manually added, no configuration file exists. When autoregistration is
being used, a phone uses a default configuration file that defines how it is to attach to the
system and register. The Device Defaults settings define the values used.
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Three settings are defined under the Device Defaults settings: load information, device
pool, and phone template.
Load Information
This is the ID of the firmware load that the device should run. When Communications
Manager is shipped, it includes a current version of the firmware loaded for each device.
From time to time, the firmware is upgraded to offer additional features or patches. By
editing this field, you can specify the ID of the new load. The next time the device boots,
it downloads the new firmware load.
Device Pool
This field enables you to define which device pool is used when autoregistration takes place.
Phone Template
This field allows you to define what phone button template devices use when autoregistration takes place.
The following steps show how the device defaults are configured:
Step 1.
From within CCMAdmin, select Device > Device Settings > Device
Defaults.
Step 2.
A screen similar to that shown in Figure 3-4 displays. Most often, the load
information can remain at its default. However, from time to time, new loads
need to be deployed. To update the load, you must first locate the name of
the device. Enter the new load ID in the Load Information field. Make sure
that you enter the ID correctly and that the ID is for the correct device.
Entering an invalid or incorrect ID in this field can cause the device to fail.
Note When a new load is to be used, it must first be copied to the TFTP server so that
the device can download it.
Step 3.
Next, edit the device pool. To change the default device pool for a device,
click the down arrow in the Device Pool field associated with the device and
select the desired device pool from the displayed list.
Step 4.
The last item that can be selected on this page is the phone button template.
To change this item for a device, click the down arrow in the Phone Template
field associated with the device and select a template from the displayed list.
Adding Phones
When a phone is added to the system, information about the phone is entered into the
SQL database. This information defines nearly every aspect of the device. Phones can be
added to the system in a number of ways, but the net effect is the same.
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Figure 3-4 Device Defaults Configuration
Four methods for adding phones are explored in this chapter, beginning with the method
known as autoregistration. Before delving too deeply into this method, take a quick look
at all four:
■
Autoregistration: This allows phones to be plugged into the system and automatically register. An extension is assigned to the phone from a range defined by the
administrator.
■
Manual: All information for the phones is manually entered before the phone is
plugged in.
■
Bulk Administration Tool (BAT): This method is used when installing information
for a large number of phones. This is done by entering the information into a
comma-separated values (CSV) file and then using BAT to insert the information into
the Communications Manager database.
■
Tools for Autoregistration Phone Support (TAPS): This is similar to BAT, except that
the MAC address is not entered in the CSV file. Additional steps are required when
the phone is plugged into the network. However, even with the added steps, it is
quicker than just using BAT when performing a large installation.
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Autoregistration
As the name implies, using this method allows phones to register to the system by merely
being plugged into the Communications Manager network. Although this method has the
advantage of allowing you to quickly add phones, it has disadvantages as well. When a
phone is added by this method, it is assigned an extension number from a range of numbers that you define. It assigns these extensions in a first-come-first-served fashion. For
example, if you defined the range to be from 2000 to 2100, the first phone to register
would receive the extension 2000, the second 2001, and so on. You can see how this
might be undesirable. If you are fortunate enough to be performing a deployment in
which you can assign extensions as you want, this might not be a concern. However, in
situations such as when you are replacing an existing PBX, you will most likely want to
reuse the existing extensions; hence the autoassignment of extensions might not be desirable. Autoregistration can still be used in these environments; you just need to modify
the extensions that were automatically assigned.
Rogue phones are another anomaly that can arise when autoregistration is used. If
improperly configured, it is possible for a phone to be added to the system without your
knowledge. The addition would, of course, have to be made by someone who has a Cisco
IP Phone and can plug it into your network. Although this might seem less than likely, it
is still important to ensure that the dialing capabilities be restricted for any rogue phones
that might find their way onto your system. This is done by defining a Calling Search
Space (CSS) for autoregistration in the device pool. CSS defines a device’s calling privileges. This concept is discussed in greater detail in Chapter 5, “Configuring Class of
Service and Call Admission Control.” Choose a CSS that has limited access, perhaps only
internal or local access. This, however, also has a drawback. Limiting the CSS of autoregistered phones means that all phones that are added using autoregistration will have a limited CSS. This means that for a phone that was added through autoregistration to have
greater calling privileges, the CSS must be changed. Often autoregistration is used during
the initial deployment and then turned off.
As you can see, using autoregistration has both benefits and drawbacks. The choice of
whether to use it depends on the environment and is ultimately up to you.
If all the previously discussed predeployment tasks have been completed, little additional
configuration is required to implement autoregistration. Next, assign an extension range
and enable autoregistration at the Communications Manager group. If you choose to use
autoregistration, the steps that follow help walk you through these tasks:
Step 1.
From within CCMAdmin, select System > Cisco Unified CM.
Step 2.
Enter search criteria in the search field, and click Find. You might also leave
the search field blank and click Find. This results in all Communications
Managers being displayed.
Step 3.
From the list that displays, select the Communications Manager on which you
want to enable autoregistration.
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Step 4.
A screen similar to that shown in Figure 3-5 displays. Enter the starting extension number in the Starting Directory Number field and the ending extension in the Ending Directory Number field.
Figure 3-5 Communications Manager Configuration for Autoregistration
Step 5.
You should notice that in the Auto-Registration Information section, the
Auto-Registration Disabled check box is deselected at this point. This check
box automatically becomes deselected when you enter a starting and ending
extension number. Do not select this check box. Selecting this box resets the
starting and ending extension numbers.
Step 6.
Click Save to save your settings.
Note In the next section, you select a Communications Manager group for autoregistration. If the Communications Manager group that you select contains more than one
Communications Manager, you should enter the starting and ending extension numbers for
each of the Communications Managers in the group.
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After an extension range is defined, you need to enable autoregistration for the
Communications Manager group. The steps that follow show how this is done:
Step 1.
From within CCMAdmin, select System > Cisco Unified CM Group.
Step 2.
Enter search criteria in the search field and click Find. You can also leave the
search field blank and click Find. This results in displaying all
Communications Manager Groups.
Step 3.
From the list that displays, select the Communications Manager group on
which you want to enable autoregistration.
Step 4.
A screen similar to that shown in Figure 3-6 displays. Select the Auto-registration Cisco Unified Communications Manager Group check box.
Figure 3-6 Communications Manager Group Configuration for
Autoregistration
Step 5.
Click Save to save your changes.
Note Only one Communications Manager group can be selected as the autoregistration
Communications Manager group. If you select this check box and there is already an existing autoregistration Communications Manager group, a window displays stating, “You
have selected this Cisco Communications Manager Group to be the Auto-registration
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Cisco Communications Manager Group. The old Auto-registration Cisco Communications
Manager Group will be deselected.” Only select that check box for the Communications
Manager group that is responsible for autoregistration.
Manually Adding Phones
In addition to using autoregistration to add phones, you can manually add them. You
often add phones manually for a small number of phones (such as 20). If all the predeployment tasks have been completed, the process to manually add phones is quite simple.
This section takes you through the steps of the process.
Manually adding phones has few drawbacks, aside from the fact that it can be somewhat
time-consuming. You must enter the MAC address and other information, such as device
pool and directory numbers, when you add each phone. Because CCMAdmin is a webbased interface, you must wait for the page to reload when you update information.
Although a second or two might not seem that long, when you have to wait for multiple
page reloads, it can start to add up. That is just the nature of this type of interface and little can be done about it.
The steps that follow take you through the process of manually adding a phone and
explore each field that can be populated for each phone. A brief description is included
for any field that has not yet been discussed in this book. The phone used in these steps
is a 7960, but the process is similar for most of the more popular Cisco IP Phone models.
When you add other phone models, a field might not display. This simply means that the
model does not require or support that particular field. Because there are a number of
steps to this process, section headings are used to mark the point at which each new set
of parameters begins. These same headings display on the configuration screen as well,
which should help you keep track of where you are.
Step 1.
From within CCMAdmin, select Device > Phone.
Step 2.
Click the Add New link.
Step 3.
A new page displays. From the drop-down list, select the type of phone you
want to add.
Step 4.
Click Next.
Step 5.
Select the desired protocol and click Next.
Device Information
Step 6.
A screen similar to that shown in Figure 3-7 should display. The first field that
must be entered is the MAC address of the phone. The MAC address can be
found on the back of the phone as well as on the box in which the phone was
packaged.
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Figure 3-7 Phone Configuration
Note It seems that over the years the numbers on the back of the phone have been getting smaller in type size. Then again, it seems that everything I am reading nowadays is getting smaller, so maybe it’s not the phone. If you find the same to be true, you can also get
the MAC address to display on the LCD of the phone. The method for this differs among
the various models. On the 7960 and 7940, press the Settings and then the 3 keys on the
keypad. Of course, power must first be supplied to the phone.
Step 7.
The next field is the Description field. Enter a description that will help you
quickly identify the phone later.
Note If you do not enter anything in the Description field, a default description, which is
the MAC address preceded by SEP, is entered automatically. SEP stands for Selsius
Ethernet Phone. Selsius was the name of the company that made Communications
Manager before Cisco purchased the company and the product. A description I like to
apply uses the following format: Last name, First name Extension number (for example,
Smith, John 2012). This format allows you to search by name and easily see the extension
number from the search page.
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Step 8.
From the Device Pool drop-down list, select the device pool that the phone
will use.
Note Although there are a number of required fields, most of them contain default values. The device pool, MAC address, phone button template, and device security profile (on
certain models) are the only required fields that do not contain default values. You can add
a Skinny Client Control Protocol (SCCP) phone without configuring any other parameters.
This is useful if you need to quickly add phones and you are certain that the default system
values, and those set in the device pool, are adequate for this phone.
Step 9.
From the Common Device Configuration drop-down list, select the common
device configuration that the phone will use.
Step 10. In the next field, a phone button template is selected for this phone. From
the drop-down list in the Phone Button Template field, select the desired
template.
Note Not all the templates will display in this field, only those that apply to the model of
phone you are adding.
Step 11. From the drop-down list in the Softkey Template field, select a softkey template for the phone.
Step 12. The Common Phone Profile field contains user-specific service and feature
attributes such as Do Not Disturb (DND) options, phone personalization
information, and product-specific configuration information. Often the default
profile is adequate. Select the appropriate profile from the drop-down list.
Step 13. The CSS determines the destinations that can be dialed from the phone. CSSs
are discussed in Chapter 5. Choose a CSS from the Calling Search Space
drop-down list. If this field is left at None, the dial privileges of the phone
could be limited.
Step 14. Automated Alternate Routing (AAR) is used to provide an alternate route if a
call fails because of insufficient bandwidth. The AAR CSS can be used to
limit the paths a call can use when it is rerouted. ARR is covered in Chapter 6,
“Configuring CUCM Features and Services.” Select an AAR CSS from the
AAR Calling Search Space drop-down list.
Step 15. The next field is Media Resource Group List. It determines the accessibility of
media resources to the phone. Media resources are discussed in further detail
in Chapter 5. From the Media Resource Group List drop-down list, select the
desired group. If no media resource group list is chosen, the list defined in the
device pool is used.
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Step 16. The next two fields allow you to configure what audio source is heard when a
call is placed on hold. The first field, which is labeled User Hold Audio
Source, determines what is heard when the call is placed on hold by pressing
the Hold button. The second, Network Hold Audio Source, determines what
audio is heard when the call is placed on hold by pressing the Transfer, Call
Park, or Conference button. Select the desired audio source from the dropdown list for each field. If no audio source is chosen, the source defined in
the enterprise parameters is used.
Step 17.
Information entered in the Location field is used to prevent WAN links from
becoming oversubscribed in centralized deployments. These locations are discussed more in Chapter 5. If you have defined locations, select the appropriate one for this phone from the drop-down list.
Step 18. The AAR Group field determines the appropriate association of this device
with an AAR group. An AAR group defines the prefix that is assigned when a
call fails because of insufficient bandwidth. AAR is discussed in further detail
in Chapter 6. Select an AAR group if AAR is being used. If this field is set to
None, the AAR group associated with the device pool or line is used.
Step 19. The User Locale field determines the language and fonts used for the phone.
The default user locale, which is set in the enterprise parameters, is used if
this field is left set to None. If the phone needs to use a different locale than is
defined by its device pools, or the enterprise parameters, select the proper
one from the drop-down list.
Step 20. The Network Locale field determines what locale is used for the phone. This
impacts the tones and cadences used. The default network locale was defined
in the enterprise parameters. If a different value is selected here, this value
takes precedence. If this field is set to None, the enterprise parameter setting
is used. If this phone needs to use a different locale than is defined by its
device pools or the enterprise parameters, select the proper locale from the
drop-down list.
Step 21. The Built-In Bridge field is used to enable and disable the built-in bridge. This
bridge can be used when the Barge feature is invoked. Select the desired state
of the built-in bridge from the drop-down list.
Note Barge is a feature that allows a phone to join an active call on another phone if the
two phones have a shared line.
Step 22. The Privacy field is used to determine whether the phone can enable privacy
for calls on a shared line. Select the desired state for this field from the dropdown list.
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Step 23. Device Mobility allows a device to use different settings for the following
parameters based on the physical location of the device:
■
Cisco Unified Communications Manager Group
■
Roaming Device Pool
■
Location
■
Region
■
Network Locale
■
AAR Group
■
AAR Calling Search Space
■
Device Calling Search Space
■
Media Resource Group List
■
Survivable Remote Site Telephony (SRST)
The Device Mobility Mode determines whether device mobility is enabled on
this device. Set this field to On to enable device mobility and Off to disable
it. If left at Default, the mode is based on the service parameter device mobility setting.
Step 24. Using the Owner User ID drop-down list, assign a user ID to this device. The
ID will be recorded in the Call Detail Records (CDR).
Step 25. Phone Load Name is the name of the firmware load that the device should be
running. When Communication Manager is shipped, it includes a current version of the firmware loaded for each device. From time to time, the firmware
is upgraded to offer additional features or patches. By editing this field, you
can specify the name of the new load. The next time the device boots, it
downloads the new firmware load. This field allows the load to be changed on
one phone so that it can be tested. To change the load on all phones, the load
information should be updated in the Device Defaults settings.
Step 26. The Single Button Barge field determines whether a Barge or cBarge can be
initiated using a single button. Set this field to Barge to configure it for singlebutton barge or to cBarge to configure single-button cbarge. If left at Default,
the service parameter setting is used.
Step 27.
The Join Across Lines fields determine whether this feature is enabled. To
enable the feature, set this field to On. To disable this feature, set this field to
Off. The service parameter setting is used if the field is set to Default.
Step 28. The Use Trusted Relay Point field determines whether a relay point such as a
Media Termination Point (MTP) or a transcoder must be labeled trusted to
be used by this device. This field is typically changed only in virtualized
environments.
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Step 29. The BLF Audible Alerts Setting (Phone Idle) field determines whether an
audible alert is played for the BLF DN when there is no current call. When set
to On, an alert is played. When set to Off, an alert is not played. The service
parameters setting is used when set to Default.
Step 30. The BLF Audible Alerts Setting (Phone Busy) field determines whether an
audible alert is played for the BLF when there is an active call on the BLF DN.
When set to On, an alert is played. When set to Off, an alert is not played.
The service parameters setting is used when set to Default.
Step 31. The Always Use Prime Line field determines which line is activated when the
phone is taken off hook. If this field is set to Off, whichever line is ringing will
be used when the phone goes off hook. When set to On, the primary line will
always be used even if an incoming call is ringing another line of the phone.
To answer any line other than the primary line, the user must manually select
the line. The service parameter setting is used when this field is set to Default.
Step 32. If Always Use Prime Line for Voice Messages is set to On, the voicemail
account associated with the primary line is dialed when the Messages button
is pressed. If set to Off, the account associated with the line that has voicemail will be dialed. If set to Default, the service parameter setting is used.
Step 33. Calling party transformation enables you to change the caller ID. Select a
Calling Party Transformation CSS that contains the called party transformation pattern that is assigned to the device. You can also leave this set to None
and use the Calling Party Transformation CSS assigned to the device pool by
selecting the Use Device Pool Calling Party Transformation CSS check box.
Step 34. The geolocation information can be used to determine the logical partition of
a device. If you are using features that depend on geolocation, select the
desired geolocation from the drop-down list.
Step 35. If the Retry Video Call as Audio check box is selected, a video call will try
to connect as an audio call if it cannot connect as a video call.
Step 36. If the Ignore Presentation Indicators (internal calls only) check box is selected, internal caller ID restrictions are ignored. This means that if an internal call
is configured to block caller ID, the caller ID will still show up on the device.
Step 37.
Select the Allow Control of Device from CTI check box to enable computer
telephony integration (CTI) control.
Step 38. The Logged into Hunt Group check box allows an administrator to log the
device out of a hunt group by deselecting this box.
Step 39. The Remote Device check box causes Communications Manager to allocate a
buffer for the phone that allows SCCP messages to be bundled. This should
only be selected if the device is a remote device and is experiencing delayed
connect times.
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Step 40. If the Protected Device check box is selected, a two-second tone is played
when a call is encrypted. This tone informs the callers that the call is
“protected.”
Step 41. A feature known as Private Line Auto Ringdown (PLAR) allows a call to be
placed by simply picking up the handset. Hot Line Device is an extension of
PLAR. If this check box is selected, the device can connect only with other
Hot Line devices.
Note If Hot line Device is enabled, a custom softkey template without supplementary
services must be created and assigned to this device.
Protocol-Specific Information
Step 42. The next two fields, Packet Capture Mode and Packet Capture Duration,
are for troubleshooting purposes only and should not be configured when
adding a new phone.
Step 43. Presence is a feature that allows a user to monitor the status of another
party’s line. The Presence Group that a device belongs to determines what
lines it is allowed to monitor. Select the Presence group this device should
belong to. If no Presence group is selected, the device can monitor all devices
that are not in a Presence group.
Step 44. The next field is labeled Device Security Profile. This field defines Certificate
Authority Proxy Function (CAPF) information such as authentication mode
and key size. A security profile must be selected. If you choose to not enable
security on a phone, the nonsecure profile can be selected.
Step 45. The SUBSCRIBE Calling Search Space determines which partitions the
device can access when invoking a presence request. Select a CSS that
includes the partitions assigned to the lines the device can monitor.
Step 46. The Unattended Port check box is used to indicate that the device has unattended ports. This is normally implemented if the port is used to send calls to
an application such as a voicemail server. In most cases, this box should be
left deselected.
Step 47.
The Require DTMF Reception field is often required for remote phones. This
is often needed for remote phones connected through a SIP trunk because it
allows dual-tone multifrequency (DTMF) signals to be received out of band.
Step 48. RFC 2833, “RTP Payload for DTMF Digits, Telephony Tones and Telephony
Signals,” deals with DTMF relay. To disable this feature, select the RFC2833
Disabled check box. In most cases, this box should be left deselected.
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Certificate Authority Proxy Function (CAPF) Information
Step 49. The next set of parameters deals with CAPF. These settings are used to configure certificate-specific information. Certificates are used to help prevent
the tampering of call signaling and media streams. For more information on
CAPF, refer to the Cisco IP Phone Authentication and Encryption for Cisco
Communications Manager guide at Cisco.com. This guide can be found by
searching for “Cisco IP Phone Authentication and Encryption” at Cisco.com.
If CAPF is not being used, these fields do not need to be configured. The
Certificate Operation field is used to install, upgrade, or delete a certificate.
The available options are
■
No Pending Operation: Displays when a certificate operation is currently inactive.
■
Install/Upgrade: Select this when you want to install or upgrade a
certificate.
■
Delete: When selected, the current certificate will be deleted.
■
Troubleshoot: Allows certificate information to be viewed in a CAPF
trace file.
Step 50. The Authentication Mode field determines the method that will be used by
the phone to authenticate with CAPF. Choose one of the following:
■
By Authentication String
■
By Null String
■
By Existing Certificate (Precedence to LSC)
■
By Existing Certificate (Precedence to MIC)
Step 51. The Authentication String is used only when By Authentication String is chosen in the previous step. Enter the string you want to use. It must be between
4 to 10 digits.
Step 52. The Key Size field determines the key size for the certificate. The valid choices are 512, 1024, and 2048. Select the desired value.
Step 53. The Operation Completes By field specifies when the install, upgrade, or
delete must be complete. Enter the desired date and time in this field.
Step 54. The Certificate Operation Status field displays the progress of the certificate
operation. Nothing is entered in this field; it is read-only.
Expansion Module Information
Step 55. The fields labeled Module 1 and Module 2 are used when expansion modules
(7914, 7915, or 7916) are used with the phone being added. If the phone has
expansion modules, select the modules from the drop-down list.
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Step 56. The Module 1 Load Name and Module 2 Load Name fields are used to
define which firmware load ID the expansion modules use. In most cases,
these fields should be left blank. When left blank, the load ID specified on the
Device Defaults Configuration page is used. If the need ever arises to set a
specific phone to a specific load ID, the load ID should be entered in this field.
Cisco IP Phone—External Data Locations
Step 57.
The next set of fields is used to define data locations for the phone. This
information is used to determine where the phone should search for certain
data, such as help screens and phone services. In most cases, these fields can
be left blank and the system defaults will be used. Table 3-2 lists these fields
and a brief description of each. If values other than the system defaults need
to be used by this phone, enter them in the appropriate fields. Any values
entered in these fields will be used for the device and will override the values
found on the Enterprise Parameters page.
Note The fields labeled Secure are similar to those listed in Table 3-2. The only difference
is that they point to a secure server. Secured servers might be desired if you want this
transmitted data to be encrypted.
Table 3-1
Cisco IP Phone—External Data Locations Parameters
Parameter
Description
Information
The URL the phone uses when the (i) is pressed on the phone.
Directory
The location of the directory the phone uses.
Messages
The URL that is used when the Messages button is pressed.
Because you normally want a number to be dialed when this button is pressed, this field should be left blank.
Services
The URL where services can be found.
Authentication Server
URL of the authentication server for requests made to the phone
web server.
Proxy Server
The proxy server used by the phone.
Idle
The URL that is displayed on the phone after the idle timer expires.
Idle Timer
The amount of time in seconds that the phone must remain idle
before the idle URL is displayed.
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Extension Information
Step 58. Extension mobility allows users to log in to phones and have that phone take
on the characteristics such as extension number, speed dials, and so on of
their phone. If extension mobility is used on this device, select the Enable
Extension Mobility check box.
Step 59. The Log Out Profile field determines what device profile will be assigned to
this phone when a user logs out of extension mobility. Select the desired profile from the drop-down list.
Step 60. The Log In Time field displays the time the user logged in to the device using
extension mobility.
Step 61. The Log Out Time field displays the time a user logged off of the device
using extension mobility.
Multilevel Precedence and Preemption Information
Step 62. The next three fields define Multilevel Precedence and Preemption (MLPP)
characteristics of the phone. If these fields are left blank or set to default, the
values set in the device pool are used. If MLPP is not being used, these fields
can be left blank. The first MLPP field is the MLPP Domain. MLPP grants
only higher priority from calls within the same MLPP domain. For this reason,
an MLPP domain is needed.
Step 63. The second field in this category, which is called MLPP Indication, determines whether tones and indications are presented when a precedence call is
made. The precedence indication can be a special ringback or a display if the
caller’s phone supports it, and a special ringer on the called party’s side.
Step 64. The third MLPP field is MLPP Preemption. This parameter determines
whether a higher-precedence call preempts a lower-precedence call. The value
of Disabled does not allow this to happen. To cause a lower-precedence call
to be terminated if a higher-precedence call requires the resources, set this
parameter to Forceful.
Do Not Disturb
Step 65. The Do Not Disturb check box allows an administrator to enable DND on
this device. To enable it, select the check box.
Step 66. From the DND Option drop-down list, you can select how the phone will
behave when DND is enabled and an incoming call arrives. When Ringer Off
is selected, the phone will not ring, but the caller information is displayed on
the phone. When Call Reject is selected, caller information is not displayed,
and based on the DND Incoming Call Alert (see the next parameter), a beep
can be played or the light can flash. The DND Option set in the Common
Phone Profile is used when Use Common Phone Profile Setting is selected.
Step 67.
The DND Incoming Call Alert works in concert with DND Option. This
parameter determines whether the phone beeps or the light flashes when the
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DND Option is set to Call Reject or Ringer Off. When this setting is set to
None, the setting in the Common Phone Profile is used. To turn the flash and
beep off, set this to Disable. To have a beep played, set this parameter to
Beep Only. When set to Flash Only, the light on the phone will flash.
Secure Shell Information
Step 68. Certain phones support Secure Shell (SSH). This is normally used by Cisco
TAC when troubleshooting an issue. Unless instructed by TAC, you should
leave the Secure Shell User and Secure Shell Password fields empty.
Product-Specific Information
Step 69. The last set of fields, labeled Product Specific Configuration, are specific to
the model of phone you are configuring. To see an explanation of each of
these fields, consult the online help.
Step 70. After all settings have been defined, click the Save button at the top of the
screen.
Add a Line to a Phone
After a phone is added, a line must be configured for it. The following steps illustrate the
process of adding a line to a phone. Because there are a number of steps to this process,
section headings are used to mark the point at which a new set of parameters begins.
Step 1.
If you are adding a new phone and have used the steps in the preceding section, you should see a screen similar to that shown in Figure 3-8. To add a line
to an existing phone, follow Steps 2 though 4 to reach this screen. If you are
already at this screen, skip to Step 5.
Step 2.
From within CCMAdmin, select Device > Phone.
Step 3.
Enter search criteria in the search field to limit the results and click the Find
button.
Step 4.
From the list that is generated, select the phone to which you want to add a line.
Step 5.
On the left side of the Phone Configuration screen, the available lines are listed. Choose a line that has the label Add new DN.
Note If all lines have an extension number already assigned, you cannot add any additional lines to this device. If the phone button template assigned to this phone has some of
the buttons defined as speed dial, you might add more lines by changing the phone button
template.
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Figure 3-8 Directory Number Configuration
Directory Number
Step 6.
The first and only field you must complete is the Directory Number. Enter
the extension number in this field.
Step 7.
The Route Partition field defines the partition to which this directory number
is assigned. The partition is used to determine the devices that can call this
extension. Partitions are discussed in more detail in Chapter 5.
Step 8.
The next field is the Description field. Enter a description that can help you
quickly identify the line later.
Step 9.
In the Alerting Name field, enter the name that should be displayed on the
caller’s phone while the called party’s phone is ringing.
Step 10. The ASCII Alerting Name field is used on devices that do not support
Unicode characters.
Step 11. Select the Allow Control of Device from CTI check box to enable CTI control of this line.
Step 12. The Associated Devices box lists all the devices that this line is assigned to.
To remove the line from a listed device, highlight the device in the Associated
Devices box and click the down arrow that is between the Associated
Devices box and the Dissociated Devices box. The device should now
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appear in the Dissociated Devices box. Click Save and the line is removed
from the selected device.
Note The Allow Control of Device from CTI and Associated Devices fields do not
appear on the screen until the DN has been saved.
Directory Number Settings
Step 13. The Voice Mail Profile field determines which voicemail profile the directory
number uses. The voicemail profile defines the number that is dialed when
the Messages button on the phone is pressed. Voicemail profiles are discussed in further detail in Chapter 5. Select the voicemail profile from the
drop-down list.
Step 14. The next field allows a CSS to be assigned as the line level. This determines
what destinations can be reached when calling from this line. Select Calling
Search Space from the drop-down list.
Note It is important to understand what happens when a CSS is assigned to the line and
the device. In short, the two CSSs are combined; however, there is a little more to it. For a
detailed explanation, refer to Chapter 5.
Step 15. The Presence group that a line belongs to determines what devices are
allowed to monitor this line. Select the Presence Group this device should
belong to.
Step 16. The next two fields allow you to configure what audio source is heard when a
call is placed on hold. The first, which is labeled User Hold Audio Source,
determines what is heard when the call is placed on hold by pressing the Hold
button. The second, Network Hold Audio Source, determines what audio is
heard when the call is placed on hold by pressing the Transfer, Call Park, or
Conference button. Select the desired audio source from the drop-down list
for each field. If no audio source is chosen, the source defined at the device
level is used, and if None is chosen, the source set in the device pool is used.
Step 17.
The Auto Answer field determines whether the line automatically answers
incoming calls without the handset being lifted. This parameter can be set to
auto-answer using the speakerphone or headset. If you want the line to autoanswer, select either Auto Answer with Headset or Auto Answer with
Speakerphone from the drop-down list. If auto-answer is not desired on the
line, leave this field set to Auto Answer Off.
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AAR Settings
Step 18. Automated Alternate Routing (AAR) is used to provide an alternate route if a
call fails because of insufficient bandwidth. To have inbound calls that fail to
reach this line because of insufficient bandwidth sent to voicemail, select the
check box under Voice Mail. If you do not choose to send the calls to voicemail, you must choose an AAR group and you might need to configure the
AAR destination mask. By default, AAR uses the external phone number
mask to determine the fully qualified number of the destination. If you do
not want to use the external phone number mask, enter the correct mask in
the AAR Destination Mask field. The AAR Group field determines the AAR
group with which the device is associated. An AAR group defines the prefix
that is assigned when a call fails because of insufficient bandwidth. Select the
appropriate AAR group from the drop-down list.
Call Forward and Pickup Settings
Step 19. The next ten fields deal with call forwarding. These fields determine the forwarding destination, which depends on the reason for the forward. Table 3-2
provides ten forwarding types and associated actions.
You can configure each type of “forward” to forward calls to voicemail or a
specific extension. To forward to voicemail, select the Voice Mail check box.
To accomplish this, a voicemail profile must be defined for the line. To forTable 3-2
Forward Types
Forward Type
Forward Action
Forward All
Forwards all incoming calls
Forward Busy Internal
Forwards calls from internal callers when the line is busy
Forward Busy External
Forwards calls from external callers when the line is busy
Forward No Answer Internal
Forwards calls from internal callers that are not answered
Forward No Answer External
Forwards calls from external callers that are not answered
Forward No Coverage Internal
Forwards calls from internal callers when a line group has
no coverage
Forward No Coverage External
Forwards calls from external callers when a line group has
no coverage
Forward on CTI Failure
Forwards calls when a CTI route point or CTI port fails
Forward Unregistered Internal
Forwards calls from internal callers when the line is not registered
Forward Unregistered External
Forwards calls from external callers when the line is not
registered
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ward calls to another extension, enter the extension number in the
Destination field. When a destination is entered into any of the internal forwards, the number is automatically entered into the corresponding external
forward. If you want the external calls to be forwarded to a different destination, simply enter the desired destination in the appropriate external forward
field. A calling search space can be applied to each forward type, which limits
the destinations to which a call can be forwarded. This is useful when you
want to restrict a line from forwarding calls to numbers that are long distance,
but still want long-distance calls to be placed from the line. The CSS selected
must contain the partition of the number in the destination field. Enter the
appropriate destinations and calling search spaces for each forward type.
Step 20. In the No Answer Ring Duration field, enter the number of seconds that the
line will ring before forwarding to the Forward No Answer destination. If this
field is left blank, the value configured in the Communications Manager service parameter is used.
Step 21. The Call Pickup Group field determines which call pickup group this directory number belongs to. Call pickup groups allow a user to redirect an incoming
call on another phone to the user’s phone. Select the desired call pickup
group from the drop-down list. Call pickup groups are covered in more detail
in Chapter 6.
Park Monitoring
Step 22. The Park Monitoring Forward No Retrieve Destination External field determines where external calls are forwarded to when they are placed on park and
not retrieved. To send them to voicemail, select the check box below Voice
Mail. To send them to an alternative number, enter that number in the destination field. If the parker’s CSS does not have rights to dial the alternate destination, you need to select a CSS from the Calling Search Space drop-down list.
Step 23. The Park Monitoring Forward No Retrieve Destination Internal field determines where internal calls are forwarded to when they are placed on park and
not retrieved. To send them to voicemail, select the check box below Voice
Mail. To send them to an alternative number, enter that number in the destination field. If the parker’s CSS does not have rights to dial the alternate destination, you need to select a CSS from the Calling Search Space drop-down list.
Step 24. The Park Monitoring Reversion Timer field determines how many seconds a
call can be parked before Communications Manager sends an alert to the
parker phone. If left blank, the value set in the service parameters is used. The
default is 60 seconds.
MLPP Alternate Party Settings
Step 25. The next set of parameters deals with MLPP alternative party settings. These
settings allow you to configure an alternate destination for precedence calls
that are not answered on this line or the forwarded number assigned to this
line. If MLPP is not being used, these parameters can be left empty. In the
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first field, which is labeled Target (Destination), enter the number to which
unanswered precedence calls should be forwarded.
Step 26. In the MLPP Calling Search Space field, select the appropriate search space
from the drop-down list. This calling search space limits the destinations to
which precedence calls can be forwarded.
Step 27.
In the MLPP No Answer Ring Duration field, enter the number of seconds
that the phone will ring when it receives a precedence call before forwarding
to the Forward No Answer destination if unanswered.
Line Settings for all Devices
Step 28. The Hold Reversion Ring Duration field determines how many seconds a call
can be on hold before Communications Manager rings the phone that placed
the call on hold. The call will ring until it is answered or the maximum hold
duration expires. Enter the desired amount of seconds in this field. If left
blank, the value set in the service parameters is used.
Step 29. The Hold Reversion Notification Interval field determines the intervals at
which the holding parties receive an alert reminding them that a call is on
hold. Enter the desired amount of seconds in this field. If left blank, the value
set in the service parameters is used.
Step 30. The Party Entrance Tone field determines whether a tone is played when a
new caller joins a call. When this parameter is set to Default, the value set in
the service parameters is used. To ensure that a tone is played, set this to On.
To ensure that no tone is played, set this to Off.
Line Settings for this Device
Step 31. The set of parameters under Line Settings for this Device define caller ID,
Message Waiting Indicator (MWI), and ring settings. The first two fields,
labeled Display (Internal Call ID), are used to configure which caller ID is
displayed when calls placed to other internal callers are connected. Enter up
to 30 characters in this field. Both letters and numbers are allowed. If this field
is left blank, the line’s directory number will be used.
Note There two fields for the Internal Caller ID. The second, which is labeled ASCII
Display, is used for devices that do not support Unicode character display. This is also true
for the next two fields, which are labeled Line Text Label.
Step 32. The next two fields, which are labeled Line Text Label, are used to define
how the line displays on the phone. If you want the extension number to display next to the line button, leave this field blank. To display a label other
than the directory number next to the line button, enter that label in this field.
Step 33. The External Phone Number Mask field can be used to modify the external
caller ID for calls placed from this line. An example mask might be
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408370XXXX. The extension number is used to fill in the XXXX portion.
In this example, if the directory number is 1401, the external phone mask
would cause the external caller ID number to be 4083701401. The external
phone mask on the first line creates the fully qualified directory number that
is displayed above the first extension on certain IP phones. The configuration
also makes the fully qualified number appear on the top bar of the phone’s
display. The external phone number mask is also used with AAR. Masks are
explored in further detail in Chapter 4, “Implementing a Route Plan.”
Step 34. The Visual Message Waiting Indicator Policy field determines whether the
light on the phone is turned on when a new message is left for this extension.
In most cases, this value should be left at Use System Policy. The available
choices are as follows:
■
Light and Prompt: The light turns on and the envelope icon next to the
line displays.
■
Prompt Only: Only the envelope icon next to the line displays.
■
Light Only: Only the light is turned on.
■
None: No indication is used.
■
Use System Policy: Uses the setting selected in CCM services parameters.
Step 35. The Audible Message Indicator Policy field determines whether a stutter dial
tone is heard when a message is waiting. Select Off to disable this. Select On
to enable this.
Step 36. The next two settings determine whether the phone rings when incoming calls
are being received on this directory number. In most cases, this value should
be left at Use System Default. The available choices are as follows:
Step 37.
■
Disable: Phone does not ring.
■
Flash Only: The light flashes. No ring.
■
Ring Once: Rings once and then stops.
■
Ring: Normal ringing.
■
Beep Only: A beep is played (only valid for the Phone Active setting).
■
Use System Policy: Uses the setting selected under services parameters.
Select the desired value for Ring Setting (Phone Idle) and Ring Setting
(Phone Active).
Step 38. The Call Pickup Group Audio Alert Setting (Phone Idle) parameter determines whether members of a call pickup group will hear a tone while their
phone is on hook and a call comes in to another member of the call pickup
group. To ensure that an alert is not sent, set this parameter to Disable. To
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have a short ring sent, set the parameter to Ring Once. To use the service
parameters settings, set the parameter to Use System Default.
Step 39. The Call Pickup Group Audio Alert Setting (Phone Active) parameter determines whether members of a call pickup group will hear a tone while they are
on a call and a call comes in to another member of their call pickup group. To
ensure that an alert is sent, set the parameter to Disable. To have a short ring
sent, set the parameter to Ring Once. To use the service parameters settings,
set the parameter to Use System Default.
Step 40. The Monitoring Calling Search Space field is used when silent monitoring is
used. A supervisor must have a CSS that can access the lines that are to be
monitored.
Multiple Call / Call Waiting Settings
Step 41. The next field, which is labeled Maximum Number of Calls, determines how
many active calls can be on the line. The maximum is 200 active calls per
phone. Enter the maximum number of calls in this field. The default of 4
should be adequate for most phones.
Step 42. The field labeled Busy Trigger determines how many active calls are required
before the line is considered busy. The default is 2. This means that if the maximum number of calls on the line is 4 and the busy trigger is 2, the third call
will receive a busy indication. However, two more calls could be placed from
this phone because the maximum number of calls is four.
Forwarded Call Information Display
Step 43. The Forwarded Call Information Display section determines what information is sent when a call is forwarded. Select the information to be sent by
selecting the check box next to each desired field.
Users Associated with Line
Step 44. To associate an end user to a line, click the Associate End Users button.
When the new window appears, click Find. Check the box next to the user
you want to associate to the line, and click the Add Selected button.
Step 45. Click the Save button to complete the configuration of this line.
That’s all there is to it! It is a simple task after you are familiar with all the parameters that
need to be configured. However, if you are new to Communications Manager, a number
of these parameters might seem confusing. Rest assured that these parameters will be
explained in greater detail throughout the remainder of this book.
Using BAT to Add Devices
Imagine having to add more than a hundred phones and you must choose between
adding them manually or using autoregistration. At first glance, you would most likely
choose to add them using autoregistration. Normally, this would be a good choice;
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however, you discover that each phone needs to have a specific directory number. To
make things more complicated, a number of the phones need to use different phone button templates. You could still use autoregistration, but after the phones were added, you
would have to change the directory numbers and phone button templates of each phone.
It would be more efficient to add these phones and not have to go back and modify each
one. This is where a utility called Bulk Administration Tool (BAT) comes in. BAT allows
you to prepopulate Communications Manager with all the information for the phones
before they are connected to the system. By adding the phones’ information, a configuration file is created that the phones download from the TFTP server when it boots up.
Note Although this section portrays only BAT as a tool to add phones, it can be used to
add gateways, client matter codes, forced authentication codes, pickup groups, and other
information to the Communications Manager database. It can also be used to update
device information and export information.
BAT adds devices to Communications Manager by importing a comma-separated values
(CSV) file that contains the required information about the phones. The CSV file contains
a number of fields that must be populated. The easiest way to create this CSV file is to
use an Excel template that is included with BAT. The template is an Excel XTL file that is
used to generate the CSV files. It offers an easy, well-formatted interface to add all the
required and optional information.
Activating the BAT Service
In early versions of Communications Manager, BAT had to be installed before it could be
used, but now it is installed by default and accessible from Communications Manager
Administration. However, the Cisco Bulk Provisioning Service must be activated before
you can use BAT. To activate the service, follow these steps:
Step 1.
Enter https://[CM_IP Address]/ccmservice in the address bar of your browser and press Enter.
Step 2.
Enter the administrative username and password and click Login.
Step 3.
Navigate to Tools > Service Activation.
Step 4.
Select the desired server from the Server drop-down list.
Step 5.
Select the check box next to the Cisco Bulk Provisioning Service and click
Save.
Step 6.
A window will appear informing you that activating or deactivating service
can take a while. Click OK and wait for the page to refresh.
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BAT CSV and Template Overview
After the service is activated, CSV files and templates must be created. The order in
which you create these is not too important. However, it is necessary that both are created before the next steps can be performed. To create a CSV file, it is recommended that
you use the Excel template that was mentioned previously. The Excel template is stored
on the Communications Manager server. You need to download the template to your PC
before you can use it. To do this, follow these steps:
Step 1.
From within Communications Manager Administrator, select Bulk
Administration > Upload/Download Files.
Step 2.
Click Find.
Step 3.
Select the check box next to the file named bat.xlt and click Download
Selected.
Step 4.
Save the file in a location of your choosing using a descriptive name such as
7960BAT.xls.
The number of available fields in the template depends on the model of the phone you are
importing. For example, the 7960 can contain more than 30 fields. The top row displays
what information should be entered into the field and whether it is a required field. The
fields include, but are not limited to, information such as the user’s name, MAC address
of the phone, directory numbers, and speed-dial numbers. When you first open the template, it displays only a few fields. The template then enables you to add other fields as
you want. This feature enables you to avoid dealing with unneeded fields. After the information is added to the template, a CSV file is created and must be uploaded to the
Communications Manager server. The necessary steps are presented later in this section.
It might appear that the hardest part of creating the CSV file is the entry of all the information. However, it is often more difficult to acquire the information that must be
entered into the template. Make sure that you take the proper amount of time beforehand
to ensure that you have obtained all the required information. The best way to get this
information is to perform station reviews. A station review is a formal process that is used
to record information about each phone that is to be added to the system. The information acquired during this process includes the number of lines (and directory numbers
assigned to each), number of speeds, required features (conference calls, voicemail, and so
on), and the type of calls that can be placed by the phone (local, national, international,
and so on). This, of course, is only a sample of the information that is gathered during the
station review process. It is important that you gather as much information as possible
during station reviews. After this information is collected, you can create the CSV files.
In addition to the CSV file that is created, templates within the BAT application must also
be created. These templates define the characteristics of the phones that you are going to
add. These templates contain parameters for things such as phone type, phone button
template, softkey templates, calling search spaces, number of lines, and partitions, just to
name a few. After these templates are configured, they are used in concert with the CSV
file that has been created to insert phones into the system. Because these templates contain many parameters, it might be necessary to create a number of templates depending
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upon your environment. For each set of phones that require different settings on any
parameter defined within a template, a new template must be created. Now look at a simple example so that you can see how easily the number of these templates can grow.
Consider that you have 70 7940s and 50 7960s to deploy. Right off the bat (no pun
intended), you know that you need at least two templates because there are two types of
phones. For this example, assume that half the 7940s are going to be two-line phones and
the other half are going to be one-line phones. This means that at least two templates are
needed for the 7940s alone. As for the 7960s, assume that through the station review
process, you determine that you need two-, three-, and four-line phones, which means
that at least three templates are needed for the 7960s. If all the phones with the same
number of lines do not differ in any other way, five templates in all are needed. However,
just to make things interesting, consider that half of the three-line phones and half of the
four-line phones require different calling search spaces. This adds the need for two more
templates, bringing the total to seven. You can see how minor changes can cause the
number of required templates to grow quite quickly. It is important to understand that
because each template is used with a CSV file, you need to create the same number of
CSV files as there are templates.
Note Make sure that you take time to determine the number of templates that will be
needed. Sometimes after determining how many templates are needed, you might feel it is
not worth the effort. I can remember an installation of 3000 phones that required more
than 230 templates. Although this was a large number of templates, days if not weeks of
time were saved on that deployment by creating the templates and CSV files. Keep in mind
that this is an extreme example. Another deployment of 300 phones required only one
template.
Before you can determine how many templates you need, you must be familiar with all
the parameters that are set in the BAT templates. Each model of phone has unique parameters, but there are also a number of parameters that are common among all models. The
following is an example of some of the parameters that can be defined in the template for
most models of phones:
■
Protocol
■
Device Pool
■
Calling Search Space
■
AAR Calling Search Space
■
User and Network Hold Audio Source
■
Location
■
User Locale
■
Network Locale
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■
Phone Button Template
■
Phone Load Name
■
External Data Locations
■
Multilevel Precedence and Preemption Information
■
Number of Lines
By taking the number of different models of phone that will be deployed and determining how many of the parameters defined in the template will be different among like
models, you can determine approximately how many templates and CSV files that must
be created.
Note After you have set up a template, you can often reuse it with only minor modifications.
Now that you have a good overview of what is required for BAT to work, take a look at
the steps required to add phones to the system using BAT.
After BAT is installed, you can begin the process of creating the CSV files and the templates. Before creating the templates and CSV files, you should have conducted a detailed
analysis of the phones that are to be deployed. Based on the disparity of the phones in
your system, you should have a good idea of how many templates and CSV files you
need. The analysis should also have supplied the information that will be needed to create
the templates and files.
You can create either the template or the CSV file first. In the steps that follow, the creation of the CSV file is described first.
Note It’s a good idea to use same naming convention for both the template and the CSV
file that will be used together. This way, when you are selecting the CSV file, it is easy to
determine which file goes with which template.
Creating a CSV File for BAT
Follow these steps to create the CSV file:
Step 1.
Open the BAT template in Excel. You might get a security warning about
macros. Macros must be enabled for this template to function properly, so
you might need to change Excel’s settings to allow this.
Step 2.
Select the Phones tab. A template similar to that shown in Figure 3-9 displays.
As you can see, by default, only two fields display in the template.
Step 3.
Click the Create File Format button to add fields.
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Figure 3-9 Excel BAT Template
Step 4.
A screen similar to that shown in Figure 3-10 should display. Select the device,
line, and intercom fields that you want to add to the template by highlighting
the field in the box on the left side and clicking the double arrows (>>).
Step 5.
To remove the device, line, and intercom fields from the template, highlight
the field in the box on the right side and click the double arrows (<<).
Step 6.
After you move all the desired fields to the box on the right side, click the
Create button. When asked whether you want to overwrite the existing CSV
format, click Yes.
Step 7.
Proceed to the far right of the template, and in the appropriately labeled fields
(refer to Figure 3-8), enter the number of lines and speed dials these phones
will have. Then click another field so that the new values will register. When
you enter information in these fields, the template adds the appropriate fields.
Step 8.
At this point, the template is ready for the data to be entered.
Step 9.
After all the data is entered, click the Export to BAT Format button, which is
located at the upper-right portion of the template.
Step 10. You are prompted for a place to store this file. BAT creates a default location
and filename. An example is C:\XlsDataFiles\Phones-02102010135805. BAT
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Figure 3-10 Excel BAT Template File Format
always selects c:\XlsDataFiles as the default file location. The filename is
based in part on the tab you have selected to export, such as Phones,
PhoneUsers, and UserDeviceProfiles, followed by a hyphen. The second part
of the filename is the date and time that the file was exported. Note that in
the preceding example, 02102010135815 represents February 10, 2010 at
1:58 p.m. and 15 seconds. If you prefer a different file location or name, you
can change it before clicking OK.
After the CSV file is created, you need to upload it to the Communications Manager
server. The following steps illustrate how to upload a file to the Communications
Manager server:
Step 1.
From within Communications Manager Administrator, select Bulk
Administration > Upload/Download Files.
Step 2.
Click the Add New link.
Step 3.
A window similar to the one shown in Figure 3-11 will appear. Click Browse,
and select the CSV file you created.
Step 4.
Select the type of data in the CSV file from the Select The Target drop-down
list. For example, if the CSV contained the information for phones that you
added, select Phones.
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Figure 3-11 File Upload
Step 5.
Select the type of transaction that you want to complete, such as Insert Phones.
Note If a file of the same name already exists, you need to select the Overwrite File
check box.
Step 6.
Click Save and the file is uploaded.
Adding Phones Using BAT
The process of inserting phones using BAT is actually a four-stage process, and each
stage has its own set of steps. The first stage of adding phones in BAT is creating the templates. A template is used to define the settings for devices added by BAT. The following
steps walk you through this process. A 7965 template is created in this example.
Step 1.
From within Communications Manager Administrator, select Bulk
Administration > Phones > Phone Template.
Step 2.
Click the Add New link.
Step 3.
Select the model of the phone you are adding and click Next.
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Step 4.
Now select the proper protocol and click Next.
Step 5.
A screen similar to that shown in Figure 3-12 displays. On this screen, you
define the parameters for this template. In the Template Name field, enter a
name for this template. Try to keep the naming convention the same as for the
CSV file that will be used with this template.
Figure 3-12 BAT Phone Template Configuration
Step 6.
The remainder of this page contains all the parameters that can be configured
for the device. Because all these parameters are covered earlier in this chapter,
they are not discussed in this section. If necessary, refer to Steps 7–70 in the
“Manually Adding Phones” section, earlier in this chapter, for details of each
parameter. Enter the appropriate information in each field.
Step 7.
After the parameters have been configured, click the Save button at the top of
the screen. When the save is complete, a window similar to the one shown in
Figure 3-13 appears.
Step 8.
After the template is inserted, line details need to be configured. Click Line[1]
- Add a New DN.
Step 9.
A screen similar to that shown in Figure 3-14 displays. This screen is used to
define all the parameters for this line. A name must be assigned to each line
template. Because all the remaining parameters are covered earlier in this chap-
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ter, they are not discussed in this section. If necessary, refer to Steps 6–44 in
the “Add a Line to a Phone” section, earlier in this chapter, for details on each
parameter. Enter the appropriate information in each field and click Save.
Figure 3-13
Phone Template
Step 10. To configure additional lines, select Configure Device from the Related Links
drop-down menu and click Go. Now repeat Steps 8 through 9 for each additional line.
Step 11. When configuring templates for certain models of phones, you can also add
speed dial and services to the template. To add speed dials to the template,
click one of the lines labeled Add a New SD or select Add/Update Speed
Dials from the Related Links drop-down menu and click Go.
Step 12. A screen displays that allows you to add two types of speed dials. The first
type is associated with a button on the phones. The user accesses the second
type by pressing the two-digit speed-dial number (which has the number he
wants to reach assigned to it) and by pressing the AbbrDial softkey. Enter the
desired speed dials and click Save. After you have entered all speed dials,
click Close.
Step 13. To add services to the template, select Subscribe/Unsubscribe Services from
the Related Links drop-down menu and click Go. Services are cover in more
detail in Chapter 6.
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Figure 3-14 BAT Phone Line Configuration
Step 14. A window displays that allows you to select the available services. Select the
desired service and click Next. Enter any additional information that the service might require and click Subscribe.
Step 15. After all services have been added, the template is complete. Click the Save
button and close the window.
Step 16. Now click the Save button on the Phone Template configuration screen.
Now that the template and CSV file are created, it is time to validate that the information
in the CSV file will align with the template that was created. For example, if the CSV file
has two directory numbers defined but the template has only one, the validation process
will alert you. The following steps show you how to validate the data.
Step 1.
From within Communications Manager Administrator, select Bulk
Administration > Phones > Validate Phones.
Step 2.
Make sure that the Validate Phones Specific Details radio button is selected.
From the File Name drop-down list, select the CSV file that you created using
the Excel template. Only the CSV file uploaded will display in this list.
Step 3.
Select the corresponding template that you created from the Phone Template
Name drop-down list.
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Note The Validate Phone All Details radio button and corresponding File Name field
are used to validate from an exported phones file that was generated by using the All
Details option.
Step 4.
Click the Submit button, and the validation request will be sent to the job
scheduler.
After the job is submitted, you will want to check to confirm the completion and review
the results. The following steps illustrate this process:
Step 1.
From within Communications Manager Administrator, select Bulk
Administration > Job Scheduler.
Step 2.
Click Find.
Step 3.
A list of jobs will appear. If the one you are interested in is complete (check the
status column), click the job ID and the results of the validation will appear.
If everything was configured properly, the report should show that no errors were found.
If errors were found, click the name in the Log File Name column to review the log. When
you have determined why it failed, correct the problem and repeat the validation process.
These last steps examine how to insert the phones into the Communications Manager
database:
Step 1.
From within Communications Manager Administrator, select Bulk
Administration > Phones > Insert Phones.
Step 2.
Make sure that the Insert Phones Specific Details radio button is selected.
From the File Name drop-down list, select the CSV file that you created using
the Excel template. Only the uploaded CSV file will display in this list.
Step 3.
Select the corresponding template that you created from the Phone Template
Name drop-down list.
Step 4.
Leave the Create Dummy MAC Address check box deselected. Dummy
MAC addresses are with Tool for Auto-Registered Phones Support (TAPS),
which is discussed later in this chapter.
Note The Insert Phone All Details radio button and corresponding File Name fields are used
to insert from an exported phones file that was generated by using the All Details option.
Step 5.
The next section is Override Options. If you are inserting phones that are
already in the database, you might need to select the override option so that
the new data is inserted. Select the check boxes associated with the data you
want to overwrite.
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Step 6.
In the Job Information section, enter a descriptive name for this transaction
and then select the Run Immediately radio button. This will cause the job to
run as soon as you click the Submit button. If you want to wait and run the
job later, select the Run Later radio button.
Step 7.
Click Submit, and the job will be submitted to the scheduler.
After the job is submitted, you will want to confirm the completion and check the results.
The following steps describe this process:
Step 1.
From within Communications Manager Administrator, select Bulk
Administration > Job Scheduler.
Step 2.
Click Find.
Step 3.
A list of jobs will appear. If the one you are interested in is complete (check the
status column), click the job ID, and the results of this transaction will appear.
If everything was done properly, the report should show that no errors were found. If
errors were found, click the name in the Log File Name column to review the log. When
you have determined why it failed, correct the problem and repeat the process.
Adding Phones Using TAPS
A more advanced method of adding phones, called Tool for Auto-registered Phone
Support (TAPS), is also available. This tool allows you to prepopulate Communications
Manager’s database with all the information about the phones, except for the MAC
address. After the information is entered, the phones are plugged into the system and
autoregistered. Because the MAC addresses have not been entered into the system, the
TFTP server cannot deliver a configuration file to the phone based on the MAC address.
This causes a default configuration to be issued and the phone to be registered with basic
configuration. After the phone is registered, a predetermined number is dialed that routes
the call to an Interactive Voice Response (IVR) server. The IVR server asks what language
you would like to use and the extension number that should be assigned to the phone.
After this is completed, the MAC address of the phone is associated with the configuration that matches the extension number entered, and a configuration file is created. The
phone gets this file from the TFTP server when it reboots.
Integrators often use this method of adding phones during large deployments. Because
this method allows the integrator to populate the system without having to add MAC
addresses, the deployment of the phones is simpler. Integrators don’t have to make sure
that a certain phone with a certain MAC address is plugged into a certain jack. The integrator need only be concerned that the proper model is placed in each location. Previous
to the introduction of this tool, the MAC addresses had to be entered before the phones
were deployed, and this required the integrator to label the box of each phone with the
extension number for which it was configured. This also meant that no information could
be added to the system until the phones were in hand.
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The detailed steps for TAPS are outside the scope of this book, but further information
can be found in the Cisco Bulk Administration Tool documentation on the Cisco website.
You can find the latest copy of this by searching for “bulk administration tool guide” at
Cisco.com.
Adding Gateways
After phones are configured, calls can be placed to other phones within the cluster.
Although this is useful, it has its limitations. Imagine if the phone system were allowed to
reach only other phones within the cluster. Most likely, not many of these systems would
be sold. Therefore, you need to consider how to allow calls outside the cluster. This is
accomplished by configuring gateways that connect the cluster to other systems. The
function of a gateway is to allow connectivity between dissimilar systems, that is, to
allow your phone system to make calls to other phone systems.
In most cases, the first system you should connect to your system is the public switched
telephone network (PSTN). This allows callers on your system to call nearly any other
phone in the world. In addition to connecting to the PSTN, you might also want to connect your system to other systems in your company and bypass the PSTN. In either case,
the steps are similar.
The following sections explore six types of gateways. The main difference in these gateways is the protocol that they use to communicate with the Communications Manager.
Their functions are the same: to provide connectivity to another system.
Table 3-3 describes the six types of gateways and gives a brief description of each.
Adding H.323 Gateways
The first type of gateway to be discussed is an H.323 gateway. H.323 is best described as
a suite of protocols. It is also often referred to as an umbrella under which a number of
other protocols exist. The specifics of what H.323 is and how it works are far beyond the
scope of this book. This section focuses on the process of configuring an H.323 gateway.
The configuration steps required for an H.323 gateway are somewhat more advanced than
those for other types of gateways. This is because much of the configuration has to be
done using the command-line interface (CLI) of the gateway itself. For example, when
configuring a Media Gateway Control Protocol (MGCP) gateway, only a few commands
must be entered in the CLI, and Communications Manager handles all the call routing.
An H.323 gateway does not depend on the Communications Manager for call routing; it
contains a dial plan of its own. This means that the dial plan must be entered using the
CLI, which increases the difficulty of configuring an H.323 gateway. To properly configure an H.323 gateway, you must have a high level of experience in configuring IOS
devices for VoIP deployments. Because it is not possible to provide you with all the
required knowledge to configure the CLI portion of an H.323 gateway in a book of this
size, we focus on the steps required in CCMAdmin. The CLI configuration varies in each
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environment and can sometimes become quite complex. For more information on the CLI
configuration for an H.323 gateway, refer to the Cisco IOS Voice, Video, and Fax
Configuration Guide, which can be found at Cisco.com by searching “Cisco IOS Voice,
Video, and Fax Configuration Guide.”
Table 3-3
Gateway Descriptions
Gateway
Connectivity
Requires Own Dial Amount of
Plan Configuration? Configuration
Required
H.323
Used to connect to the Yes
PSTN
Requires a significant
amount of configuration on the gateway
MGCP
Used to connect to the No
PSTN
Requires minimal configuration on the
gateway
Non-IOS MGCP
Used to connect to the No
PSTN
Requires almost no
configuration on the
gateway itself
SCCP
Used to connect to the No
analog port gateways
Requires minimal configuration on the
gateway
Intercluster Trunk
Connects
Communications
Manager clusters
together
SIP
Yes
Connects
Communications
Manager to SIP trunks
and devices
No
All configuration is
done on the
Communications
Manager
Requires some configuration on the gateway
After the entire CLI configuration of the gateway is completed, it can be configured in
Communications Manager. There are a number of Cisco devices that can be configured
as H.323 gateways. Because the list of supported devices is always evolving, it is best to
check the Cisco website to see what gateways currently support H.323. You can find a
list of supported gateways by searching “Understanding Voice Gateways” at Cisco.com.
You can further refine the search by including the Communications Manager version with
this search. For example, by searching “Understanding Voice Gateways 4.1(2),” the first
result is a link to the correct page. Because there are a variety of devices that can be configured as an H.323 gateway, it is impossible to include a step-by-step guide on how to
configure each of them. Because the configurations are similar, you should be able to use
the following steps to configure a Cisco 3620 as an H.323 gateway and install the H.323
gateway you choose.
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The following steps are required to configure a Cisco 3825 as an H.323 gateway. Because
there are a number of steps to this process, section headings are used to mark the point at
which each new set of parameters begins. These same headings display on the configuration screen as well, which should help you keep track of where you are.
Step 1.
From within CCMAdmin, select Device > Gateway.
Step 2.
Click the Add New link.
Step 3.
Select H.323 Gateway from the Gateway Type drop-down list and click Next.
Device Information
Step 4.
The Gateway Configuration screen, as shown in Figure 3-15, displays. Enter
the DNS name or IP address of the gateway in the Device Name field.
Figure 3-15 H.323 Gateway Configuration
Step 5.
In the Description field, enter a description that will help make this device
easily identifiable.
Step 6.
From the Device Pool drop-down list, select the desired Communications
Manager group for this gateway.
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Step 7.
From the Common Device Configuration drop-down list, select the common
device configuration (CDC) the gateway will use. If you have not configured a
CDC, leave this set to <none>.
Step 8.
From the Call Classification drop-down list, select whether incoming calls
on this device should be considered OnNet or OffNet. This parameter is
used to determine whether calls can be transferred and forwarded to help
prevent fraud.
Step 9.
The next field is Media Resource Group List. This determines the accessibility of media resources to the phone. Media resources are discussed in further
detail in Chapter 6. From the Media Resource Group List drop-down list,
select the desired group. If no media resource group list is chosen, the one
defined in the device pool is used.
Step 10. The Packet Capture Mode parameter is only used during troubleshooting
and is beyond the scope of this book. Leave this set to None.
Step 11. The Packet Capture Duration parameter is also only used for troubleshooting and is beyond the scope of this book. Leave this set to 0.
Step 12. Information entered in the Location field is used to prevent WAN links from
becoming oversubscribed in centralized deployments. These locations are discussed more in Chapter 5. If you have defined locations, select the appropriate one for this phone from the drop-down list.
Step 13. The AAR Group field determines the appropriate association of this device
with an AAR group. An AAR group provides the prefix that is assigned when
a call fails because of insufficient bandwidth. AAR is discussed in further
detail in Chapter 6. Select an AAR group if AAR is being used. If this field is
set to None, AAR is, in effect, disabled on this device.
Step 14. The Tunnel Protocol parameter allows non-H.323 protocol information to be
within H.323 signaling information. To enable this, select QSIG from the
drop-down list. In most cases, this can be left at None.
Step 15. The QSIG Variant parameter is only configurable if QSIG is selected as the
tunnel protocol. Leave this parameter alone unless Cisco TAC instructs you to
change it.
Step 16. The ASN.1 ROSE OID Encoding parameter is only configurable if QSIG is
selected as the tunnel protocol and is beyond the scope of this book.
Step 17.
The Use Trusted Relay Point field determines whether a relay point such as a
Media Termination Point (MTP) or a transcoder must be labeled trusted to
be used by this device. This field is typically changed only in virtualized
environments.
Step 18. The Signaling Port field defines the H.225 signaling that the gateway uses.
The default of 1720 can be used in most cases.
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Step 19. The Media Termination Point Required check box needs to be selected if the
H.323 device does not support features such as hold and transfers. This is the
case with H.323 version 1 gateways. These gateways cannot support such features because H.323v1 does not have the capability to modify a channel,
which means that it cannot transfer a call or place a call on hold. That would
require modifying the channel.
Step 20. If the Retry Video Calls as Audio check box is selected, Communications
Manager sets up a voice call if a video call fails to set up.
Step 21. When the Wait for Far End H.245 Terminal Capability Set check box is
selected, Communications Manager expects to receive the far end’s capabilities before sending its own.
Step 22. Path Replacement Support is used in conjunction with QSIG tunneling and is
automatically selected when QSIG is selected as the tunnel protocol. Leave
this set at the default.
Step 23. If Transmit UTF-8 for Calling Party Name is left deselected, the user locale
setting in the device pool will be used to determine whether Unicode information is sent and translated. Typically, this can be left at the default.
Step 24. If you want to allow both secure and nonsecure calls on this gateway, you
must select the SRTP Allowed check box. If this is not selected, only nonsecure calls are allowed.
Step 25. If the H.235 Pass Through Allowed check box is selected, the shared-secret
key will be able to pass through a Communications Manager, allowing H.323
endpoints to set up a secure connection.
Step 26. If this gateway is used to connect to the PSTN, make sure that the PSTN
Access check box is selected.
Step 27.
The only MLPP setting that can be configured is the MLPP Domain. Enter
the domain in this field. If left blank, the settings found in enterprise parameters are used.
Call Routing Information—Inbound Calls
Step 28. The next set of fields deals with inbound calls. The Significant Digits field
determines the number of digits of an incoming dialed number that
Communications Manager uses. Communications Manager counts from right
to left, so if the number entered in this field is 4 and the digits received are
8105559090, 810555 would be removed and only 9090 would be used to
determine the destination of this call.
Step 29. A Calling Search Space (CSS) determines the accessibility and destination of
an inbound call to the gateway CSS, as discussed in Chapter 5. Choose a CSS
from the Calling Search Space drop-down list. If this field is left at None, the
dial privileges of this device could be limited.
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Step 30. The Automated Alternate Routing (AAR) is used to provide an alternate route
if a call fails because of insufficient bandwidth. The AAR CSS can be used to
limit the paths a call can use when it is rerouted. Select an AAR CSS from the
AAR Calling Search Space drop-down list.
Step 31. The Prefix DN field defines what digits will be added to the front of the
incoming destination number. This is applied to the number after
Communications Manager truncates the number based on the Significant
Digits setting.
Step 32. If your voicemail system supports redirecting number IE, select the
Redirecting Number IE Delivery–Inbound check box. Otherwise, leave this
box deselected.
Step 33. Select the Enable Inbound FastStart check box if you want to support H.323
FastStart. H.323 FastStart requires only two message exchanges to open logical channels, whereas normal setup requires 12. However, if FastStart is selected, both ends must support and be configured for FastStart.
Call Routing Information—Outbound Calls
Step 34. The next set of fields examines outbound calls. The Calling Party Selection
field determines what number is sent to outbound calls. The Calling Party
Selection choices are as follows:
■
Originator: The directory number of the device that placed the call
■
First Redirect Number: The directory number of the first device to forward the call
■
Last Redirect Number: The directory number of the last device to forward the call
■
First Redirect Number (External): The external directory number of
the first device to forward the call
■
Last Redirect Number (External): The external directory number of
the last device to forward the call
Select the desired value for this field.
Step 35. The Calling Party Presentation field determines whether Communications
Manager sends caller ID information. To send caller ID information, select
Allowed from the drop-down list. To block caller ID, select Restricted from
the drop-down list.
Step 36. Cisco recommends that the next four fields remain set to the default of Cisco
CallManager:
■
Called party IE number type unknown
■
Calling party IE number type unknown
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■
Called Numbering Plan
■
Calling Numbering Plan
These fields deal with dial plan issues and should be changed only when
advised to do so by Cisco or an experienced dial plan expert. The need to
change these usually occurs when installing Communications Manager
internationally.
Step 37.
The Caller ID DN field is used to determine what caller ID is sent out this
gateway. A mask or a complete number can be entered in this field. For example, if the mask 55536XX is entered in this field, Communications Manager
sends 55536 and the last two digits of the directory number (DN) that is placing the call.
Step 38. If the Display IE Delivery check box is selected, the calling and called party
name information is included in messages.
Step 39. The Redirecting Number IE Delivery–Outbound check box should be
selected when integrating with a voicemail system that supports redirecting
number IE. Otherwise, leave it deselected.
Step 40. Select the Enable Outbound FastStart check box if you want to support
H.323 FastStart. H.323 FastStart requires only two message exchanges to
open logical channels, whereas normal setup requires 12. However, if FastStart
is selected, both ends must support and be configured for FastStart.
Step 41. If FastStart Outbound is enabled, you must select a codec to be used for
FastStart. Select the desired codec from the Codec for Outbound FastStart
drop-down list.
Step 42. Called party transformation allows you to change the number that is dialed.
The CSS selected must have access to the calling party transformation pattern
that is assigned to the device. You can also leave this set to None and use the
Called Party Transformation CSS assigned to the device pool by selecting the
Use Device Pool Called Party Transformation CSS check box.
Step 43. Calling party transformation allows you to change the caller ID. Select a
Calling Party Transformation CSS that contains the called party transformation pattern that is assigned to the device. You can also leave this set to None
and use the Calling Party Transformation CSS assigned to the device pool by
selecting the Use Device Pool Calling Party Transformation CSS check box.
Geolocation
Step 44. Some features might require geolocation information. This information is
also referred to as a civic address. The geolocation information can be used
to determine the logical partition of a device. If you are using the geolocation feature, select the appropriate geolocation from the Geolocation dropdown list.
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Step 45. There are 17 configurable geolocation fields. Geolocation filters allow you
choose which fields are used to create a geolocation identifier. If you are
using the geolocation feature, select the appropriate geolocation filter from
the Geolocation Filter drop-down list.
Intercompany Media Engine
Step 46. E.164 Transformation Profiles are used when Intercompany Media Engine
(IME) is used. IME allows different companies to automatically learn routes
that allow calls to travel across the Internet instead of the PSTN.
Incoming Calling/Called Party Settings
Step 47.
The Incoming Calling Party settings and Incoming Called Party settings are
used to globalize numbers. Each calling and called party number has a number type assigned to it. The incoming calling/called part settings are based on
the number type assigned. There are four number types: national, international, unknown, and subscriber. There are four settings for each of these
number types:
■
Prefix: Digits entered are added to the beginning of the number after
the number of strip digits specified are removed.
■
Strip Digits: The number of digits that should be stripped from the
number before the prefix is applied.
■
Calling Search Space: The CSS that is used after transformation has
occurred.
■
Use Device Pool CSS: When this check box is selected, the device pool
CSS is used.
If your environment requires the manipulation of incoming called and calling
numbers, configure the appropriate settings for each of these fields.
Step 48. Click Save.
Adding MGCP Gateways
Cisco recommends that you use a Media Gateway Control Protocol (MGCP) gateway
whenever possible. MGCP is a protocol that, as its name implies, is used to control gateways. Unlike H.323, MGCP does not handle the routing of calls; it depends on
Communications Manager for this. Therefore, very little needs to be configured at the CLI.
MGCP gateways that are used in a Cisco Communications Manager solution are separated into two categories: Internetworking Operating System (IOS) and non-IOS MGCP
gateways. The difference is whether the device that is acting as the gateway is running
IOS or not. For example, a 3825, which runs IOS, is considered an IOS gateway, whereas a
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Catalyst 6500 running native IOS is considered a non-IOS gateway. The following section
focuses on IOS MGCP Gateways. Non-IOS gateways are discussed in a later section.
Adding IOS MCGP Gateways
As with an H.323 gateway, some configuration must be done from the CLI of the gateway. Figure 3-16 shows a sample of the commands that previously had to be entered in
the CLI to configure the gateway to communicate through MGCP. As you can see, the
CLI configuration is quite simple. At present, it is even easier now that the MGCP gateway can be configured to download its configuration from the Communications Manger.
Only two commands are required to configure this. They are
■
ccm-manager config server 10.1.1.1
■
ccm-manager config
mgcp
mgcp call-agent 172.10.10.1
mgcp dtmf-relay codec all mode out-of-band
mgcp sdp simple
!
ccm-manager switchback graceful
ccm-manager redundant-host 172.10.10.10
ccm-manager mgcp
!
voice-port 1/1/1
!
dial-peer voice 4 pots
application MGCPAPP
port 1/1/1
Figure 3-16 MGCP Gateway CLI Configuration Example
An MGCP gateway does not require as much configuration on the gateway device
because it relies on Communications Manager for all call routing functions. For this reason, it is recommended that an MGCP gateway be used whenever possible.
For more information on the CLI configuration for an MCGP gateway, refer to the Cisco
IOS Voice, Video, and Fax Configuration Guide, which can be found at Cisco.com by
searching for “Cisco IOS Voice, Video, and Fax Configuration Guide.”
After the CLI configuration is completed, the MGCP gateway must be configured in
CCMAdmin. There are a number of Cisco routers that can act as MGCP gateways, and it
is not possible to include a step-by-step guide for each type in this section. However,
because their configuration is similar, you should be able to use the following steps that
show how to configure a Cisco 3825 as an MGCP gateway. You can use these steps as a
guide to help you install the MGCP gateway you are using.
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The following steps are required to configure a Cisco 3825 as an MGCP gateway.
Because there are a number of steps to this process, section headings are used to mark
the point at which each new set of parameters begins. These same headings display on
the configuration screen as well, which should help you keep track of where you are.
Step 1.
From within CCMAdmin, select Device > Gateway.
Step 2.
Click the Add New link.
Step 3.
On the next page, select the desired gateway from the Gateway Type dropdown list. In this example, the 3825 is selected. Click Next.
Step 4.
Select MCGP from the Protocol drop-down list and click Next.
Step 5.
A page similar to that shown in Figure 3-17 displays. In the first field, enter
the DNS name of the gateway in the Domain Name field, if DNS is configured to resolve this name. If DNS is not used, enter the host name of the gateway. It is important to note that if an IP domain name is configured on the
router, it must be included in the Domain Name field. For example, if the host
name is MGCPGateway and the IP domain name is configured as cisco.com,
the Domain Name field should read MGCPGateway.cisco.com. The name is
case-sensitive. The gateway will not register to Communications Manager if
the name is not entered correctly.
Figure 3-17 MGCP Gateway Configuration
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Step 6.
In the Description field, enter a description that will help make this device
easily identifiable.
Step 7.
From the Cisco Unified Communications Manager Group drop-down list,
select the desired Communications Manager group for this gateway.
Installed Voice Cards
Step 8.
The next set of fields defines what type of voice ports are installed in the
gateway. This example assumes that two FXS and two FXO ports are
installed. Select the voice module(s) from the drop-down list next to each slot.
For this example, the NW-4VWIC-MBRD is selected.
Step 9.
Click the Save button. New fields display next to each configured slot. From
the Subunit drop-down list, select the device that is installed in this slot. In
this example, VIC-2FXS was selected for subunit 0 and a VIC-2FXO was
selected for subunit 1.
Step 10. Click the Save button. Endpoint identifiers display next to each configured
subunit, as seen in Figure 3-18.
Figure 3-18 MCGP Gateway Configuration with Endpoints
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Gateway Information
Step 11. The endpoint identifiers must be configured. Select an endpoint to display the
Endpoint Configuration page. In this example, the 0/0/0 FXS port is used.
Select the 0/0/0 port by clicking the icon that looks like a phone jack with a
question mark in the upper-right corner. Depending on the type of port
selected, the fields you work with can differ.
Step 12. On the next screen, select the port type from the drop-down list and click
Next. In this example, POTS is selected.
Step 13. A description for this endpoint is prepopulated in the Description field. You
can modify it if you like.
Step 14. From the Device Pool drop-down list, select the device pool that this endpoint will use.
Step 15. From the Common Device Configuration drop-down list, select the common
device configuration the device will use.
Step 16. From the Call Classification drop-down list, select whether incoming calls on
this device should be considered OnNet or OffNet. This parameter is used to
determine whether calls can be transferred and forwarded. This should help
prevent fraud.
Step 17.
The next field is the Media Resource Group List. It determines the accessibility of media resources to a device. Media resources are discussed in further
detail in Chapter 6. From the Media Resource Group List drop-down list,
select the desired group. If no media resource group list is chosen, the one
defined in the device pool is used.
Step 18. The Packet Capture Mode parameter is only used during troubleshooting
and is beyond the scope of this book. Leave this set to None.
Step 19. The Packet Capture Duration parameter is also only used for troubleshooting and beyond the scope of this book. Leave this set to 0.
Step 20. The Calling Search Space (CSS) field determines where the device (in this
case, the gateway) will be able to dial. The CSS affects only incoming calls
through the gateway. CSSs are discussed in Chapter 5. Choose a CSS from
the Calling Search Space drop-down list. If this field is left at None, the dial
privileges of this device could be limited.
Step 21. The z (AAR) is used to provide an alternate route if a call fails because of
insufficient bandwidth. The AAR CSS can be used to limit the paths a call
can use when it is rerouted. Select an AAR CSS from the AAR Calling Search
Space drop-down list.
Step 22. In centralized deployments, locations are used to prevent WAN links from
becoming oversubscribed. These are discussed more in Chapter 5. If you have
defined locations, select the appropriate one for the phone from the Location
drop-down list.
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Step 23. The AAR Group field determines the appropriate association of the device
with an AAR group. An AAR group provides the prefix that is assigned when
a call fails because of insufficient bandwidth. AAR is discussed in further
detail in Chapter 6. If AAR is being used, select an AAR group. If this field is
set to None, AAR is, in effect, disabled on this endpoint.
Step 24. The Network Locale field determines what locale is used for this endpoint.
This impacts the tones and cadences used. The default network locale was
defined in the enterprise parameters. If a different value is selected here, this
value takes precedence. If this field is set to None, the enterprise parameter
setting is used. If the phone requires a different locale than is defined by its
device pools, or the enterprise parameters, select the proper one from the
drop-down list.
Step 25. Some features might require geolocation information. This information is
also referred to as a civic address. The geolocation information can be used
to determine the logical partition of a device. If you are using the geolocation feature, select the appropriate geolocation from the Geolocation dropdown list.
Step 26. If Transmit UTF-8 for Calling Party Name is left deselected, the user locale
setting in the device pool will be used to determine whether Unicode information is sent and translated. Typically, this can be left at the default.
Step 27.
Calling party transformation allows you to change the caller ID. Select a
Calling Party Transformation CSS that contains the called party transformation pattern that is assigned to the device. You can also leave this set to None
and use the Calling Party Transformation CSS assigned to the device pool by
selecting the Use Device Pool Calling Party Transformation CSS check box.
Step 28. Hot Line Device is an extension of PLAR. If this check box is selected, the
device is restricted to connect only with other hot line devices.
Multilevel Precedence and Preemption (MLPP) Information
Step 29. The next three fields define the Multilevel Precedence and Preemption
(MLPP) characteristics of the endpoint. The only field that applies to the
device in this example is the MLPP Domain. MLPP grants higher priority
only from calls with the same MLPP domain. For this reason, an MLPP
domain is needed if MLPP is being used.
Port Information
Step 30. The Port Direction parameter determines whether this port is to be used for
inbound or outbound calls. Select Inbound to set the port for inbound calls
only. Select Outbound for outgoing calls only. To allow inbound and outbound calls, select Bothways.
Step 31. The Prefix DN field defines what digits will be added to the front of an incoming destination number. This is applied to the number after Communications
Manager truncates the number based on the Num Digits setting.
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Step 32. The Num Digits field determines the number of digits of an incoming dialed
number that Communications Manager will use. Communications Manager
counts from right to left, so if the number entered in this field is 4 and the
digits received are 8105559090, 810555 would be removed and only 9090
would be used to determine the destination of this call.
Step 33. The Expected Digits field defines how many digits are expected on inbound
calls. This field is rarely used and can be left at the default of 0 in most cases.
Step 34. The last field, which is the SMDI Port Number, is used only when integrating
with a traditional voicemail system. The port number should be set to the
same port number as on the voicemail system.
Step 35. The Unattended Port check box is used to indicate that the device has unattended ports. This is normally applied if the port is used to send calls to an
application such as a voicemail server. In most cases, this box should be left
deselected.
Step 36. Click Save. The endpoints now display on the left side of the page. After you
have configured all the endpoints, directory numbers can be assigned.
Step 37.
To add a directory number to an endpoint, click the Add a new DN link on
the left side of the screen.
Step 38. The Directory Number Configuration page displays, which is just like the
page used when adding a line to a phone. For detailed information on how to
configure this page, refer to Steps 6–44 in the “Add a Line to a Phone” section, earlier in this chapter.
Step 39. Navigate to the Gateway Configuration page. To do this, select the Configure
Device option from the Related Links drop-down list and click Go. Now,
select Back to MGCP Configuration from the Related Links drop-down list
and click Go.
Step 40. Configure the remaining ports by following the previous steps.
Step 41. After all configurations are completed, return to the Gateway Configuration
page and click the Reset button.
Step 42. A window will appear from which you need to click Reset. When the status
changes to Reset request was sent successfully, click Close.
Adding Non-IOS MGCP Gateways
As mentioned in the previous section, some non-IOS devices can be configured as MGCP
gateways. Just as with the IOS MGCP gateway, these devices do not handle call-routing
tasks. They depend on Communications Manager to handle all call-routing needs. Unlike
the IOS MGCP gateway, there is virtually no configuration required on the gateway itself.
The most widely used non-IOS gateways are the Catalyst 6000 T1/E1 blades. The T1
blades are Cisco 6608s and have eight T1 ports. These ports must be configured in the
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Catalyst switch with the appropriate IP configuration, including the TFTP server address
and VLAN information. To complete the configuration of the T1 port in CCMAdmin,
you need to know the MAC address of the port.
While the 6608 is no longer sold it is still in use in some environments so the following
configuration steps have been provided. These steps take you through the configuration
of a single T1 port of a 6608 as a non-IOS MGCP gateway. Because there are a number
of steps to this process, section headings are used to mark the point at which each new
set of parameters begins. These same headings also display on the configuration screen,
which should help you keep track of where you are.
Step 1.
From within CCMAdmin, select Device > Gateway.
Step 2.
Click the Add New link.
Step 3.
On the next page, select the desired gateway from the Gateway Type dropdown list. In this example, the Cisco Catalyst 6000 T1 VoIP Gateway is
selected. Click Next.
Step 4.
Select either Digital Access PRI or Digital Access T1 from the Protocol dropdown list and click Next. In this example the Digital Access PRI is used.
Device Information
Step 5.
The Gateway Configuration page displays. Enter the MAC address of the T1
port in the MAC Address field. The MAC address can be found by using the
show port command at the CLI of the Catalyst 6000.
Step 6.
In the Description field, enter a description that will help make this device
easily identifiable.
Step 7.
From the Device Pool drop-down list, select the device pool that this gateway
or device will use.
Step 8.
From the Common Device Configuration drop-down list, select the common
device configuration to be used by the gateway.
Step 9.
From the Call Classification drop-down list, select whether incoming calls on
this device should be considered OnNet or OffNet. This parameter is used to
determine whether calls can be transferred and forwarded. This should help
prevent fraud.
Step 10. From the Network Locale drop-down list, select the locale for this gateway. If
it remains set to None, the locale selected in the device pool is used.
Step 11. The next field is Media Resource Group List. It determines the accessibility
of media resources to a device. Media resources are discussed in further
detail in Chapter 6. From the Media Resource Group List drop-down list,
select the desired group. If no media resource group list is chosen, the one
defined in the device pool is used.
Step 12. Locations are used to prevent WAN links from becoming oversubscribed in
centralized deployments. These are discussed more in Chapter 5. If you have
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defined locations, select the appropriate one for this device from the
Locations drop-down list.
Step 13. The AAR Group field determines the appropriate association of this device
with an AAR group. An AAR group defines the prefix that is assigned when a
call fails because of insufficient bandwidth. AAR is discussed in further detail
in Chapter 6. If AAR is being used, select an AAR group. If this field is set to
None, AAR is disabled on this gateway.
Step 14. The next field defines which firmware load ID the gateway uses. In most
cases, this field should be left blank. When left blank, the load ID specified
on the device defaults configuration page is used for this gateway. If the need
ever arises to set a specific gateway to a specific load ID, the load ID should
be entered in this field.
Step 15. The Use Trusted Relay Point field determines whether a relay point such as a
Media Termination Point (MTP) or a transcoder must be labeled as trusted to
be used by this device. This field is typically changed only in virtualized
environments.
Step 16. If Transmit UTF-8 for Calling Party Name is left deselected, the user locale
setting in the device pool will be used to determine whether Unicode information is sent and translated. Usually this can be left at the default.
Step 17.
If this gateway is used to connect to the PSTN, make sure that the PSTN
Access check box is selected.
Multilevel Precedence and Preemption (MLPP) Information
Step 18. The next three fields define the Multilevel Precedence and Preemption
(MLPP) characteristics of this gateway. If these fields are left blank or set to
default, the values set in the device pool are used. The first MLPP field is the
MLPP Domain. MLPP grants higher priority only from calls with the same
MLPP domain. For this reason, an MLPP domain is needed.
Step 19. The second field in this category, which is called MLPP Indication, determines whether tones and indications will be presented when a precedence call
is made. If the is field set to Off, no precedence indication is presented. If this
field is set to On, indication is used for a precedence call.
Step 20. The third MLPP field is MLPP Preemption. This parameter determines
whether a higher-precedence call preempts a lower-precedence call. The value
of Disabled does not allow this to happen. To cause a lower-precedence call
to be terminated if a higher-precedence call requires the resources, set this
parameter to Forceful.
Interface Information
Step 21. The next set of fields defines characteristics of the interface. The first of these
fields, labeled PRI Protocol Type, defines the protocol used. The value placed
in this field depends on the type of equipment the T1 is connected to on the
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other side. Your T1 provider should be able to supply this information. Select
the appropriate protocol from the drop-down list.
Step 22. The QSIG Variant parameter is only configurable if PRI ISO QSIG T1 is
selected as the PRI protocol type. Leave this parameter alone unless Cisco
TAC instructs you to change it.
Step 23. The ASN.1 ROSE OID Encoding parameter is only configurable if PRI ISO
QSIG T1 is selected as the PRI protocol type. It is beyond the scope of this
book.
Step 24. The Protocol Side field determines whether the T1 connects to a network
device or a user device. The easiest explanation is that one side must be User
and the other side Network. If connecting to the phone company’s central
office (CO), you will most likely choose User. Select the appropriate protocol
side from the drop-down list.
Step 25. The Channel Selection Order field determines in what order the ports are
used, either starting at the first, which is referred to as TOP_DOWN, or starting at the last, which is referred to as BOTTOM_UP. Select the desired value
from the drop-down list.
Step 26. The next field, which is labeled Channel IE Type, determines the channel
selection method. The selection in this field depends on the connection on the
other side.
Step 27.
The PCM Type field determines the type of encoding format that is being
used. For the United States, Japan, Taiwan, and Hong Kong, select “μ-law”
(mu-law). In the rest of the world, select “a-law.”
Step 28. The Delay for First Restart (1/8 sec ticks) field determines how long this port
waits before restarting when instructed. This allows you to stagger the restart
times when a large number of PRIs are installed.
Step 29. The next field, labeled Delay Between Restarts (1/8 sec ticks), determines
the length of time between restarts of the PRI when PRI RESTART is sent.
Step 30. The Inhibit Restarts at PRI Initialization check box determines whether a
RESTART message is sent when the D-channel successfully connects. The
default is to leave this box selected, which does not cause a RESTART message to be sent.
Step 31. When the Enable Status Poll check box is selected, the Change B-Channel
Maintenance Status is enabled, which allows individual B-Channels to be
taken out of service. The default is to leave this box deselected.
Step 32. The Unattended Port check box is used to indicate that the device has unattended ports. This is normally used if the port is used to send calls to an
application such as a voicemail server. In most cases, this box should be left
deselected.
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Call Routing Information—Inbound Calls
Step 33. The next set of fields deals with inbound calls. The Significant Digits field
determines the number of digits of an incoming dialed number that will be
used by Communications Manager. Communications Manager counts from
right to left, so if the number entered in this field is 4 and the digits received
are 8105559090, 810555 will be removed and only 9090 will be used to
determine the destination of this call.
Step 34. A Calling Search Space (CSS) determines the accessible destinations of
inbound calls to the gateway. CSS is discussed in Chapter 5. Choose a CSS
from the Calling Search Space drop-down list. If this field is left at None, the
dial privileges of the calls coming in to the gateway could be limited.
Step 35. Automated Alternate Routing (AAR) is used to provide an alternate route if a
call fails because of insufficient bandwidth. The AAR CSS can be used to
limit the paths a call can use when it is rerouted. Select an AAR CSS from the
AAR Calling Search Space drop-down list.
Step 36. The Prefix DN field defines what digits are added to the front of an incoming
destination number. This is applied to the number, after Communications
Manager truncates the number, based on the Significant Digits setting.
Call Routing Information—Outbound Calls
Step 37.
The next set of fields deals with outbound calls. The Calling Line ID
Presentation field determines whether Communications Manager sends caller
ID information. To send caller ID information, select Allowed from the dropdown list. To block caller ID, select Restricted from the drop-down list.
Step 38. The Calling Party Selection field determines what number is sent to outbound calls. The choices are
■
Originator: The directory number of the device that placed the call
■
First Redirect Number: The directory number of the first device to forward the call
■
Last Redirect Number: The directory number of the last device to forward the call
■
First Redirect Number (External): The number of the first device to
forward the call using the external phone mask
■
Last Redirect Number (External): The number of the last device to forward the call using the external phone mask
Select the desired value for this field.
Step 39. Cisco recommends that the following four fields remain set to the default of
Cisco CallManager:
■
Called party IE number type unknown
■
Calling party IE number type unknown
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■
Called Numbering Plan
■
Calling Numbering Plan
These fields deal with dial plan issues and should be changed only when
advised to do so by Cisco or an experienced dial plan expert. The need to
change these usually occurs when installing Communications Manager
internationally.
Step 40. The next field, which is labeled Number of Digits to Strip, determines how
many digits should be stripped for outbound calls. For example, if this value
is set to 2 and the number 995551001 is dialed, the 99 is stripped before the
call is sent.
Step 41. The Caller ID DN field is used to determine what caller ID is sent out this
gateway. A mask or a complete number can be entered in this field. For example, if the mask 55536XX is entered in this field, Communications Manager
sends 55536 and the last two digits of the calling number.
Step 42. The SMDI Base Port field is used only when integrating with a traditional
voicemail system. Enter the first T1 port being used for voicemail functions.
Step 43. Called party transformation allows you to change the number that is dialed.
The Called Party Transformation CSS selected must have access to the calling party transformation pattern assigned to the device. You can also leave
this set to None and use the Called Party Transformation CSS assigned to the
device pool by selecting the Use Device Pool Called Party Transformation
CSS check box.
Step 44. Calling party transformation allows you to change the caller ID. Select a
Calling Party Transformation CSS that contains the called party transformation pattern that is assigned to the device. You can also leave this set to None
and use the Calling Party Transformation CSS assigned to the device pool by
selecting the Use Device Pool Calling Party Transformation CSS check box.
PRI Protocol Type–Specific Information
Step 45. The PRI Protocol Type Specific Information fields determine what information elements are sent in messages. For most installations, this field can remain
at the default.
UUIE Configuration
Step 46. The Passing Precedence Level Through UUIE check box determines whether
the MLPP precedence is sent through the use of the User-to-User Information
Element (UUIE). This is only configurable if the PRI protocol type is 4ESS. To
enable this, select the check box and set the security access level.
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Intercompany Media Engine
Step 47.
E.164 transformation profiles are used when Intercompany Media Engine
(IME) is used. IME allows different companies to automatically learn routes,
which allow calls to travel across the Internet instead of the PSTN.
Incoming Calling Party Settings
Step 48. The Incoming Calling Party Settings are used to globalize numbers. Each calling and called party numbers has a number type assigned to it. The incoming
calling party settings are based on the number type assigned. There are four
number types: national, international, unknown, and subscriber. There are
four settings for each of these number types:
■
Prefix: The digits entered are added to the beginning of the number
after the number of strip digits specified are removed.
■
Strip Digits: This is the number of digits that should be stripped from
the number before the prefix is applied.
■
Calling Search Space: The CSS that is used after transformation has
occurred.
■
Use Device Pool CSS: When this check box is selected, the device pool
CSS is used.
If your environment requires the manipulation of incoming calling numbers,
configure the appropriate settings for each of these fields.
Product-Specific Configuration
Step 49. The last set of fields, which is labeled Product Specific Configuration, is specific to the gateway you are configuring. To see an explanation of each of these
fields, click the i icon located to the right of the category title. The information
needed to determine the settings of these fields comes from the T1 carrier.
Enter the information provided by your carrier in the appropriate fields. Tables
3-4 through 3-7 list these fields and supply a brief description of each field.
Step 50. After the required and desired parameters have been entered, click the Insert
button at the top of the page. After the gateway is added, an alert window
informs you that the gateway needs to be reset. Click the OK button.
Geolocation Configuration
Step 51.
Some features might require geolocation information. This information is also
referred to as a civic address. The geolocation information can be used to
determine the logical partition of a device. If you are using the geolocation feature, select the appropriate geolocation from the Geolocation drop-down list.
Step 52. Geolocation filters allow you choose which fields are used to create a geolocation identifier. If you are using the geolocation feature, select the appropriate geolocation filter from the Geolocation Filter drop-down list.
Step 53. Click Save.
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Table 3-4
Product-Specific Configuration Detail
Parameter
Description
Clock Reference
This parameter specifies the location from which the master clock is
derived.
Required field.
Default: Network
TX-Level CSU
This parameter determines the transmit value based on the distance
between the gateway and repeater.
Default: 0dB
FDL Channel
This determines what type of Facility Data Link (FDL) is supported.
Default: ATT 54016
Framing
This parameter specifies the multiframe format of the span.
Required field.
Default: ESF
Audio Signal
Adjustment into IP
Network
This parameter specifies the gain or loss applied to the received
audio signal relative to the port application type.
Required field.
Default: NoDbPadding
Audio Signal
Adjustment from IP
Network
This parameter specifies the gain or loss applied to the transmitted
audio signal relative to the port application type.
Required field.
Default: NoDbPadding
Yellow Alarm
This parameter determines how a remote alarm is coded.
Default: Bit2
Zero Suppression
This parameter specifies how the T1 or E1 span electrically codes
binary 1s and 0s on the wire (line coding selection).
Required field.
Default: B8ZS
Digit On Duration
(50–500 ms)
This parameter specifies the duration in milliseconds of dual-tone
multifrequency (DTMF) digits generated by the gateway. Valid durations range from 50 to 500 ms.
Required field.
Default: 100
Minimum: 50
Maximum: 500
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Table 3-4
Product-Specific Configuration Detail
Parameter
Description
Interdigit Duration
(50–500 ms)
This parameter specifies the duration to pause between digits when
sequences of DTMF digits are generated by the gateway. This parameter is in milliseconds. Valid durations range from 50 to 500 ms.
Required field.
Default: 100
Minimum: 50
Maximum: 500
SNMP Community
String
This parameter specifies the community string to be used for accessing the Simple Network Management Protocol (SNMP) agent for this
interface.
Default: public
Disable SNMP Set
Operations
This parameter specifies whether all SNMP set operations are disabled.
Required field.
Default: false
Debug Port Enable
This parameter specifies whether the developer (debug) port should
be enabled.
Required field.
Default: true
Hold Tone Silence
Duration
This parameter specifies, in milliseconds, the intertone (off) duration
of the hold tone. If this value is set to 0, the default duration of 10
seconds is used.
Required field.
Default: 0
Minimum: 0
Maximum: 65535
Port Used for Voice
Calls
This parameter should be selected if this port is used for voice calls.
If this port is used only for fax or modem calls, do not select this
option.
Required field.
Default: true
Port Used for Modem
Calls
This parameter should be selected if this port is used for modem
calls. If this port is used only for fax or voice calls, do not select this
option.
Required field.
Default: true
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Table 3-4
Product-Specific Configuration Detail
Parameter
Description
Port Used for Fax Calls This parameter should be selected if this port is used for fax calls. If
this port is used only for voice or modem calls, do not select this
option.
Required field.
Default: true
Table 3-5 Product-Specific Configuration Detail for Fax and Modem Parameters
Parameter
Description
Fax Relay Enable
This parameter specifies whether Cisco fax relay encoding should
be negotiated when a fax tone is detected.
Required field.
Default: true
Fax Error Correction
Mode Override
This parameter specifies whether error correction mode (ECM)
should be disabled for fax transmissions using Cisco fax relay.
Required field.
Default: true
Maximum Fax Rate
This parameter specifies the maximum fax rate to negotiate. This
parameter applies only to fax transmissions using Cisco fax relay.
Required field.
Default: 14400bps
Fax Payload Size
This parameter specifies the fax payload size. The default is 20
bytes.
Required field.
Default: 20
Minimum: 20
Maximum: 48
Non-Standard Facilities
Country Code
This parameter specifies the nonstandard facilities country code.
A value of 65535 will leave this field unused. This parameter
applies only to fax transmissions using Cisco fax relay.
Required field.
Default: 65535
Minimum: 0
Maximum: 65535
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Table 3-5 Product-Specific Configuration Detail for Fax and Modem Parameters
Parameter
Description
Non-Standard Facilities
Vendor Code
This parameter specifies the nonstandard facilities vendor code. A
value of 65535 will leave this field unused. This parameter applies
only to fax transmissions using Cisco fax relay.
Required field.
Default: 65535
Minimum: 0
Maximum: 65535
Fax/Modem Packet
Redundancy
This parameter specifies whether packet redundancy (RFC 2198)
should be enabled for modem calls and fax calls not using Cisco
fax relay.
Required field.
Default: false
Named Service Event
(NSE) Type
This parameter specifies the NSE type to be used for peer-to-peer
messaging.
Required field.
Default: Non-IOS Gateways
Table 3-6
Product-Specific Configuration Detail for Playout Delay Parameters
Parameter
Description
Initial Playout Delay
This parameter specifies the initial delay introduced by the jitter
buffer in milliseconds.
Required field.
Default: 40
Minimum: 20
Maximum: 150
Minimum Playout Delay
This parameter specifies the minimum delay introduced by the jitter buffer in milliseconds.
Required field.
Default: 20
Minimum: 20
Maximum: 150
Maximum Playout Delay
This parameter specifies the maximum delay introduced by the
jitter buffer in milliseconds.
Required field.
Default: 150
Minimum: 20
Maximum: 150
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Table 3-7
Product-Specific Configuration Detail for Echo Canceller Configuration
Parameter
Description
Echo TailLength (ms)
This parameter defines the tail length duration
(in ms) to be used by the echo canceller.
Supported values are 24, 32, 48, and 64 ms.
Required field.
Default: 32ms
Minimum Echo Return Loss (ERL) (db)
This parameter defines the minimum ERL
value that can be handled by the echo canceller. Available options are 0db, 3db, and 6db.
Required field.
Default: 6db
Adding Intercluster Trunks
The last type of gateway to be covered is known as an intercluster trunk (ICT). This type
of gateway is configured to allow calls to be placed between Communications Manager
clusters across some type of IP connectivity. Intercluster trunks are normally used to
connect clusters, within the same organization, together across an exiting WAN link, thus
utilizing the WAN link for both data and voice.
The concept is simple. In each cluster, a gateway is configured that points to the IP
address(es) of the Communications Manager in the other cluster. It can be confusing at
first because unlike other types of gateways, there is no real gateway “device” used. The
easiest way to think of it is that an intercluster trunk is a process that runs on the
Communications Manager cluster.
The following steps show how to configure an intercluster trunk and explain the various
settings. Because there are a number of steps to this process, section headings are used to
mark the point at which each new set of parameters begins. These same headings display
on the configuration screen as well, which should help you keep track of where you are.
Step 1.
From within CCMAdmin, select Device > Trunk.
Step 2.
Click the Add New link and then click Next.
Step 3.
On the next page, select Inter-Cluster Trunk (Non-Gatekeeper Controlled)
from the Trunk Type drop-down list.
Note Typically nongatekeeper-controlled intercluster trunks are used only when bandwidth is not an issue. Gatekeepers and gatekeeper-controlled intercluster trunks are discussed in Chapter 5. The configuration of intercluster trunks is very similar for both types.
Step 4.
The Device Protocol field can be left at Inter-Cluster Trunk. No other option
is available. Click the Next button.
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Device Information
Step 5.
The Trunk Configuration screen, as shown in Figure 3-19, displays. Enter a
functional name for the device in the Device Name field.
Figure 3-19 Trunk Configuration
Step 6.
In the Description field, enter a description that makes this device easily
identifiable.
Step 7.
From the Device Pool drop-down list, select the desired device pool for
this gateway.
Step 8.
From the Common Device Configuration drop-down list, select the common
device configuration the phone will use.
Step 9.
From the Call Classification drop-down list, select whether incoming calls on
this device should be considered OnNet or OffNet. This parameter is used to
determine whether calls can be transferred and forwarded. This should help
prevent fraud.
Step 10. The next field is the Media Resource Group List. It determines the accessibility of media resources to a device. These are discussed further in Chapter 6.
Step 11. Information entered in the Location field is used to prevent WAN links from
becoming oversubscribed in centralized deployments. These are discussed
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more in Chapter 5. If you have defined locations, select the appropriate one
for this device from the drop-down list.
Step 12. The AAR Group field determines the appropriate association of this device
with an AAR group. An AAR group provides the prefix that is assigned when
a call fails because of insufficient bandwidth. AAR is discussed in further
detail in Chapter 6. Select an AAR group if AAR is being used. If this field is
set to None, AAR is, in effect, disabled on this device.
Step 13. The Tunneled Protocol drop-down list allows you to select Q Signaling
(QSIG), which enables intercluster trunk (ICT) to transport non-H.323 protocol information by tunneling it through H.323. Leave this set to None, unless
you know that this type of tunneling is required.
Step 14. The QSIG Variant parameter is only configurable if QSIG is selected as the
tunnel protocol. Leave this parameter alone unless Cisco TAC instructs you to
change it.
Step 15. The ASN.1 ROSE OID Encoding parameter is only configurable if QSIG is
selected as the tunnel protocol. This is beyond the scope of this book.
Step 16. The next two fields, Packet Capture Mode and Packet Capture Duration,
are for troubleshooting purposes only and should not be configured when
adding a new phone.
Step 17.
The Media Termination Point Required check box needs to be selected if the
H.323 device does not support features such as hold and transfers.
Step 18. If the Retry Video Call as Audio check box is selected, Communications
Manager sets up a voice call if a video calls fails to set up.
Step 19. The Path Replacement Support check box is automatically selected if you
select QSIG from the Tunneled Protocol drop-down list. Otherwise it is left
deselected.
Step 20. If the Transmit UTF-8 for Calling Party Name check box is left deselected, the user locale setting in the device pool will be used to determine
whether Unicode information is sent and translated. Typically, this can be
left at the default.
Step 21. The Unattended Port check box is used to indicate that the device has unattended ports. This is normally used if the port is used to send calls to an
application such as a voicemail server. In most cases, this box should be left
deselected.
Step 22. If you want to allow both secure and nonsecure calls on this gateway, you
must select the SRTP Allowed check box. If this is not selected, only nonsecure calls are allowed.
Step 23. If the H.235 Pass Through Allowed check box is selected, the shared-secret
key will be able to pass through a Communications Manager, allowing H.323
endpoints to set up a secure connection.
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Step 24. Service Advertisement Framework (SAF) is a network service that allows clusters to exchange information and build dynamic route plans. To allow SAF
information to be sent across this trunk, you must select the Enable SAF
check box.
Step 25. The Use Trusted Relay Point field determines whether a relay point such as a
Media Termination Point (MTP) or a transcoder must be labeled as trusted to
be used by this device. This field is typically changed only in virtualized
environments.
Step 26. If this gateway is used to connect to the PSTN, make sure that the PSTN
Access check box is selected.
Intercompany Media Engine
Step 27.
E.164 transformation profiles are used when Intercompany Media Engine
(IME) is used. IME allows different companies to automatically learn routes,
which allows calls to travel across the Internet instead of the PSTN.
Incoming Calling/Called Party Settings
Step 28. The Incoming Calling Party Settings and Incoming Called Party Settings are
used to globalize numbers. Each calling and called party numbers has a number type assigned to it. The incoming calling/called part settings are based on
the number type assigned. There are four number types: national, international, unknown, and subscriber. There are four settings for each of these
number types:
■
Prefix: The digits entered are added to the beginning of the number
after the number of strip digits specified are removed.
■
Strip Digits: This is the number of digits that should be stripped from
the number before the prefix is applied.
■
Calling Search Space: This is the CSS that is used after transformation
has occurred.
■
Use Device Pool CSS: When this check box is selected, the device pool
CSS is used.
If your environment requires the manipulation of incoming called and calling
numbers, configure the appropriate settings for each of these fields.
Multilevel Precedence and Preemption (MLPP) Information
Step 29. The next two fields define the Multilevel Precedence and Preemption (MLPP)
characteristics of this gateway. If these fields are left blank or set to default,
the values set in the device pool are used. The first MLPP field is the MLPP
Domain. MLPP grants higher priority only from calls with the same MLPP
domain. For this reason, an MLPP domain is needed.
Step 30. The second field is called MLPP Indication. It determines whether tones and
indications will be presented when a precedence call is made. If the is field set
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to Off, no precedence indication is presented. If this field is set to On, indication is used for a precedence call.
Call Routing Information—Inbound Calls
Step 31. The next set of fields relates to inbound calls. The Significant Digits field
determines the number of digits of an incoming dialed number that
Communications Manager uses. Communications Manager counts from right
to left, so if the number entered in this field is 4 and the digits received are
8105559090, 810555 would be removed and only 9090 would be used to
determine the destination of this call.
Step 32. A Calling Search Space (CSS) determines the accessible destinations of
inbound calls to the gateway CSS as discussed in Chapter 5. Choose a CSS
from the Calling Search Space drop-down list. If this field is left at None, the
dialing privileges of this gateway could be limited.
Step 33. Automated Alternate Routing (AAR) is used to provide an alternate route if a
call fails because of insufficient bandwidth. The AAR CSS can be used to
limit the paths a call can use when it is rerouted. Select an AAR CSS from the
AAR Calling Search Space drop-down list.
Step 34. The Prefix DN field defines what digits are added to the front of an incoming
destination number. This is applied to the number, after Communications
Manager truncates the number, based on the Significant Digits setting.
Step 35. The Redirecting Number IE Delivery–Inbound check box should be used if
your voicemail system supports redirecting number IE. Otherwise, leave this
box deselected.
Step 36. If the Enable Inbound FastStart check box is selected, FastStart will be used.
H.323 FastStart requires only two message exchanges to open logical channels, whereas normal setup requires 12. However, if FastStart is selected, both
ends must support and be configured for FastStart.
Call Routing Information—Outbound Calls
Step 37.
Called party transformation allows you to change the number that is dialed.
The Called Party Transformation CSS selected must have access to the calling party transformation pattern assigned to the device. You can also leave
this set to None and use the Called Party Transformation CSS assigned to the
device pool by selecting the Use Device Pool Called Party Transformation
CSS check box.
Step 38. Calling party transformation allows you to change the caller ID. Select a
Calling Party Transformation CSS that contains the called party transformation pattern that is assigned to the device. You can also leave this set to None
and use the Calling Party Transformation CSS assigned to the device pool by
selecting the Use Device Pool Calling Party Transformation CSS check box.
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Step 39. The Calling Party Selection field determines what number is sent for outbound calls. The choices are
■
Originator: The directory number of the device that placed the call
■
First Redirect Number: The directory number of the first device to
redirect the call
■
Last Redirect Number: The directory number of the last device to redirect the call
■
First Redirect Number (External): The external directory number of
the first device to redirect the call
■
Last Redirect Number (External): The external directory number of
the last device to redirect the call
Select the desired value for this field.
Step 40. The Calling Line ID Presentation field determines whether Communications
Manager sends caller ID information. To send caller ID information, select
Allowed from the drop-down list. To block caller ID, select Restricted from
the drop-down list.
Step 41. Cisco recommends that the next four fields remain set to the default of Cisco
CallManager. The four fields are
■
Called party IE number type unknown
■
Calling party IE number type unknown
■
Called Numbering Plan
■
Calling Numbering Plan
These fields deal with dial plan issues and should be changed only when
advised to do so by Cisco or an experienced dial plan expert. The need to
change these usually occurs when installing Communications Manager
internationally.
Step 42. The Caller ID DN field is used to determine what caller ID is sent out this
gateway. A mask or a complete number can be entered in this field. For example, if the mask 55536XX is entered in this field, Communications Manager
sends 55536 and the last two digits of the calling number.
Step 43. If the Display IE Delivery check box is selected, the calling and called party
name information is included in messages.
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Step 44. The Redirecting Number IE Delivery–Outbound check box should be
selected when integrating with a voicemail system that supports redirecting
number IE. Otherwise, leave it deselected.
Step 45. If the Enable Outbound FastStart check box is selected, FastStart will be
used. H.323 FastStart requires only two message exchanges to open logical
channels, whereas normal setup requires 12. If FastStart is selected, both ends
must support and be configured for FastStart.
Step 46. If the check box is selected, you must select the codec that is to be used. This
is selected from the Codec for Outbound FastStart drop-down list.
Remote Cisco Communications Manager Information
Step 47.
In the next three fields, the IP addresses of the Communications Manager in
the destination cluster are entered. Enter the IP addresses of the remote
Communications Manager group. Enter the remote primary Communications
Manager IP address in the Server 1 field, the remote backup Communications
Manager IP address in the Server 2 field and, if remote tertiary
Communications Manager exists, enter its IP address in the Server 3 field. If
there are more than three Communications Managers in the remote cluster,
additional intercluster trunks might need to be configured.
UUIE Configuration
Step 48. The Passing Precedence Level Through UUIE check box determines whether
the MLPP precedence is sent through the use of the User-to-User Information
Element (UUIE). To enable this, select the check box and set the security
access level.
Geolocation Configuration
Step 49. Some features might require geolocation information. This information is
also referred to as a civic address. The geolocation information can be used
to determine the logical partition of a device. If you are using the geolocation feature, select the appropriate geolocation from the Geolocation dropdown list.
Step 50. Geolocation filters allow you choose which fields are used to create a geolocation identifier. If you are using the geolocation feature, select the appropriate geolocation filter from the Geolocation Filter drop-down list.
Step 51. Click Save.
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Summary
This chapter covered all the required tasks needed to add phones and gateways to a
Communications Manager system. Four methods to add phones to the system were discussed: manual registration, autoregistration, BAT, and TAPS.
After phones were added, the addition of four types of gateways was discussed. A
description and an installation example of H.323, MGCP, non-MGCP, and intercluster
trunk gateways were provided.
The chapter should help you feel comfortable with the task of adding phones and gateways. Because it is not possible to include step-by-step instructions for every possible
type of phone and gateway in this book, the examples given were based on widely
deployed models. These instructions should serve as a good guide to help you deploy
most other models as well.
Now that the phones and gateways are deployed, a dial plan is needed. A dial plan is used
to determine how calls are routed. The next chapter explores dial plans.
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Chapter 4
Implementing a Route Plan
After phones and gateways have been added to the cluster, a dial plan must be created to
allow calls to reach destinations outside the cluster. As soon as the first directory number
is entered in Communications Manager, the creation of a dial plan begins. By default,
calls that are placed to destinations within the same cluster can be successfully routed.
This is because all directory numbers that are registered to the Communications Manager
cluster become part of that cluster’s dial plan. However, for Communications Manager to
route a call to a destination that is outside its cluster, additional information must be programmed that will make up Communications Manager’s external dial plan. Although
directory numbers do belong to the dial plan, it is common to refer to the external dial
plan as “the dial plan.”
A dial plan has five components: a numbering plan, path selection, digit manipulation,
class of service, and call coverage. This chapter focuses on the first three, which are often
referred to as a route plan. It is not uncommon for a route plan to be called a dial plan,
but it is important to understand that there is more to a dial plan then just the route plan
components. Chapter 5, “Configuring Class of Service and Call Admission Control,”
focuses on class of service. Call coverage is covered in Chapter 12, “Maximizing CUCM
and Unity/Connection.”
A dial plan can become quite complex, and many unforeseen issues can arise, so you
should outline the plan on paper before any devices are deployed. Someone that has
proven experience in this field should create the dial plan. After the dial plan is created on
paper, configuring it in the Communications Manager can begin. If you are new to dial
plans, seek out an individual with dial plan experience for assistance.
Before discussing how to configure a dial plan, take a closer look at the components that
are required to make a call. The next section provides an overview of the flow of a typical call.
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Note Because the components within a route plan have to be created in the opposite
order of the call flow, it is recommended that you read the entire chapter before trying to
implement any of the specific tasks covered. This can help you achieve a higher understanding of each component and its role in the call flow process.
Understanding Call Flow
A call begins when someone picks up the handset and dials a number. Although this
seems rather simple, much more is happening than the entry of numbers on one end and
the magical ringing of the phone on the other end. For the purposes of this chapter, you
are only concerned with how the call is routed within the cluster. After the call is handed
off to a device outside the cluster, the Communications Manager has no control on the
path that the call takes.
Before examining an example call flow, you need to be familiar with a dial plan’s required
components. The following is a list of components that are used to make up a route plan,
along with a brief description of each. Later, sections of the chapter supply more detail
on each component:
■
Route pattern: Patterns are made up of numbers, wildcards, and special characters.
Wildcards allow a single pattern to match multiple dialed numbers. The most popular
wildcard is the “X.” This wildcard matches any single digit 0–9. For example, a pattern of 91248XXXXXXX matches any dialed number that begins with 91248 and is
followed by seven more digits, such as 912485554123. Wildcards are covered in detail later in this chapter.
■
Device: This is a gateway that connects the Communications Manager system to
another system such as the public switched telephone network (PSTN). In some documentation, this is also referred to as a route group device.
■
Route group: This is an ordered list of gateways to which a call can be sent. The call
is sent to the first device in the group, and if the call is unable to use that gateway
for any reason, the route group sends the call to the next device in its list.
■
Route list: This prioritized list of route groups is used to determine to which route
groups an outbound call is sent. The call is sent to the first route group in the list. If
that route group refuses the call for any reason, the route list sends the call to the
next route group in its list.
The following steps outline what happens within the cluster when a call is placed to a
destination outside the cluster. For example, assume that the call is being placed to the
pizza parlor down the street.
Step 1.
The caller dials the phone number of the pizza parlor.
Step 2.
Communications Manager looks at those digits and finds a pattern that
matches it. If it finds multiple patterns that match, it uses the closest (the one
with the fewest possible matches).
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Step 3.
The pattern that the dialed number matches points to a route list that in turn
points to one or more route groups.
Step 4.
The route list sends the call to the first route group in its list.
Step 5.
The route group points to one or more gateways (or trunks) and sends the call
to the first gateway in the group.
Step 6.
If the gateway cannot handle the call, the route group sends the call to the
next gateway in its list if one is available.
Step 7.
If no other gateway exists in the group or if the last gateway in the group cannot route the call, the call is returned to the route list and the route list sends
the call to the next route group in the list.
Step 8.
The next route group sends the call to the first gateway in the group.
Step 9.
After the call reaches a gateway that can handle the call, the call is sent out of
the system using that gateway.
Step 10. If no gateway is available, the call fails.
Figure 4-1 offers an example of the preceding steps. This simple example shows that the
processing of a single call can be quite complicated. For calls to be delivered to the proper destination, each of the components discussed in this example, such as patterns, route
list, route groups, and gateways, must be configured in Communications Manager. This
section has offered a broad overview of these components. The following sections build
on what you have learned here, they offer more detail on each component, and they
explain how each component is configured.
1
2
Dialed Number
912485554123
Pattern:
912478XXXXXXX
Route List
3
4
5
First Route Group
in Route List
First Gateway
in Route Group
Second Route Group
7
in Route List
Second Gateway
6
in Route Group
8
First Gateway
in Route Group
Figure 4-1 Call Flow
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Second Gateway
9
in Route Group
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Note The flow of the call starts at the pattern and ends at the gateway. However, the creation of the required components is in the opposite order. When a route group points to a
gateway, the gateway in which it’s pointed to must exist. Therefore, the gateway must be
configured first, followed by the route group, then the route list, and last but not least, the
pattern. Because gateway configuration is discussed in the last chapter, you look at the
route groups and lists first.
Understanding Route Groups and Route Lists
The job of a route group is to send the call to the gateway or gateways to which it points.
The route group sends the call to the first gateway in the group, and if that gateway cannot handle the call, the route group sends the call to the next gateway in the group. This
process is repeated until a gateway in the list is identified that is able to handle the call, or
there are no more gateways in the group. If the route group is unable to find an available
gateway, the call is returned to the route list.
The job of a route list is to send the call to a route group, which in turn sends the call to a
gateway. In the example shown in Figure 4-2, the call is first routed across the WAN. If
that path is unable to accommodate the call, it is routed to the PSTN. As you can see,
depending on which path the call takes, a different number of digits need to be sent. If
the call is sent across the intercluster trunk (ICT), only five digits are needed, assuming
that the remote cluster is using five-digit directory numbers. However, if the call goes
across the PSTN, 11 digits are needed. To accomplish this, digit manipulation must take
place. Digit manipulation occurs when a called (dialed) or calling (caller ID) number is
changed. In this case, the digit translation strips the required number of digits from the
called number so that the call can be routed across the chosen path. Because the path the
call takes is not known until it reaches the route group, the digit manipulation should
take place there.
Dialed Number
912485554123
Pattern:
912478XXXXXXX
Route List
WAN Route Group
PSTN Route Group
InterCluster Trunk
(5 digits required)
T1 Gateway to PSTN
(11 digits required)
Figure 4-2 Digit Manipulation Required
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Digit manipulation can be accomplished in a number of ways. Each of the various methods used for digit manipulation is covered in this chapter, as are examples of when each
type might be used. First look at what type of digit manipulation can take place at the
route group level.
After the call is sent to a route group and an available gateway is found, six types of digit
manipulation can be performed. Three affect the calling number (caller ID), and three
affect the called party number. Digit manipulation can be performed by applying a mask
to digits. Communications Manager applies the mask to digits by right-justifying both
the digits and the mask, placing the mask directly under the digits, so to speak. Where
there are Xs in the mask, Communications Manager will pass the digits. Where there are
numbers in the mask, Communications Manager will replace the digits with the number
in the mask. For example, a mask of 612555X2XX applied to the digits 4321 would look
like this:
4321 - Digits
612555X2XX - Mask
6125554221 – End Result
The following is a list of the six methods of digit manipulation and a brief explanation
for each:
■
Calling party’s external phone mask: If configured under the directory number, this
mask changes the caller ID information. When adding a route group to a route list,
you can choose to enable or disable this mask. For example, this mask can be used to
send a phone’s Direct Inward Dial (DID) number for caller ID. If you apply an external phone mask of 408370XXXX to a 4-digit extension number and you choose to
enable this mask, Communications Manager can pass the DID number for caller ID.
For example:
4112 – Extension number
408370XXXX – External Phone Mask
4083704112 – Caller ID
With this type of mask, Communications Manager passes the actual fully qualified
directory number as caller ID.
■
Calling party transformation mask: Sometimes it is not desirable to pass the
phone’s DID number as caller ID, or the phone DN is not a DID number. The calling
party transformation mask can be used to further manipulate digits passed for caller
ID. It can also be used to pass a DID number for caller ID if the external phone mask
is not used. The actual digits supplied depend on whether the external phone number mask is applied. If the external phone mask is not used, the calling party transformation mask is applied to the extension number as follows:
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4112 – Extension Number
408370XXXX – Calling Party Transformation Mask
4083704112 – End Result
If the external phone mask is used, the calling party transformation mask is applied
to the end result of the external phone mask transformation as follows:
4083704112 – Result of external phone mask transformation
4083701200 – Calling party transformation mask
4083701200 – End result (perhaps the company’s main number)
As you can see, transformation masks are powerful tools.
■
Prefix digits: These digits prepend the calling party number to which they are
applied. An example might be to use 40837 as prefix digits in a 5-digit internal dial
plan. For example, if the extension number is 58321, Communications Manager adds
40837 in front of 58321, which results in 4083758321.
■
Discard digits: This determines whether any dialed digits are discarded. Discard digits are often used to remove the 9 from the front of outbound calls. The specific
options of this field are covered later in this chapter.
■
Called party transformation mask: This mask changes the called number (dialed
digits). This could be used in an environment that requires callers to dial only five
digits to reach a phone through an ICT. However, if the ICT cannot accommodate
the call, it is routed to the PSTN. The PSTN needs the full phone number, so a transformation mask is used. A mask of 1408370XXXX could be used in this example.
This mask tells Communications Manager to apply 1408370XXXX to the dialed
number of 58321 as follows:
58321 - Dialed number
1408370XXXX – Called party transformation mask
14083708321 – Digits passed to the PSTN
The assumption in this example is that 14083708321 is the called party’s phone
number.
■
Prefix digits: These digits are added to the front of a called number (dial digits). An
example might be 140837. This mask tells Communications Manager to place 140837
in front of the extension. For example, if the extension number is 58321,
Communications Manager adds 140837 in front of 58321, which results in
14083758321.
Even though digit manipulation can be done at the group level, it is applied to the group
at the route list level. This might sound confusing at first, but it makes more sense after
you have seen it applied. This is examined in the section “Creating a Route List,” later in
this chapter.
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Now that you understand the task of the route list and the route groups, take a look at
how they are created. Because they must be created in the opposite order of the call
flow, the route groups must be created first.
Creating Route Groups
Because route groups point to gateways, the gateways must be configured before creating route groups. For information on gateway configuration, refer to Chapter 3,
“Deploying Devices.” The steps presented in this section assume that gateways already
exist and that you have a written dial plan, which you can refer to while configuring the
route groups.
The following steps show how to create a route group that can point to multiple gateways:
Step 1.
From within CCMAdmin, select Route Plan > Route/Hunt > Route Group.
Note In earlier versions of Communications Manager, the path to reach route groups is
slightly different, such as Route Plan > Route Group.
Step 2.
Click the Add New link.
A screen similar to that shown in Figure 4-3 displays.
Figure 4-3 Route Group Configuration
Step 3.
Enter a descriptive name in the Route Group Name field.
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Step 4.
From the Distribution Algorithm drop-down box, choose how Communications
Manager will distribute the calls. If you want Communications Manager to
send the call to the first available gateway in the list, choose Top Down.
When Circular is chosen, the calls are load balanced across multiple gateways.
Step 5.
From the Available Devices box, highlight the gateway that you want to add to
this route group. If there are many gateways, you can limit the ones that appear
in this box by entering search criteria in the Device Name Contains field.
Step 6.
Certain gateways allow you to choose which ports on the gateway you
want to use for this route group. In the Port(s) field, select the ports on this
gateway that should be added to this route group. If the gateway you are
configuring does not allow the selection of ports, leave this field set to the
default.
Step 7.
Click the Add to Route Group button. Repeat Steps 5 through 7 for gateways
that you want to add to this group.
Step 8.
After you have selected all the desired gateways, they appear in the Selected
Devices box. The order in which they appear in this box determines the order
in which calls are distributed. To change the order, highlight the gateway you
want to move and click the up or down arrow to the right of the box.
Step 9.
You can remove a gateway from the route group by highlighting the gateway
and clicking the down arrow below the Selected Devices box. The gateway
then appears in the Removed Devices box.
Step 10. Click the Save button to complete the configuration of this route group.
After the route group is created, it must be added to a route list. As mentioned previously, digit manipulation can be done at the route group level, but is configured when route
groups are added to a route list. The next section discusses how to create and configure
route lists.
Creating a Route List
Before proceeding, ensure that you have created all required route groups. Because a
route list points to route groups, the route groups must exist. The steps provided in this
section guide you through the creation and configuration of a route list, in addition to
configuring route-group-level digit manipulation.
Step 1.
From within CCMAdmin, select Call Routing > Route/Hunt > Route List.
Note In earlier versions of Communications Manager, the path to reach route groups is
slightly different, such as Route Plan > Route List.
Step 2.
Click the Add New link.
A screen similar to that shown in Figure 4-4 displays.
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Chapter 4: Implementing a Route Plan
Figure 4-4 Route List Configuration
Step 3.
Enter a descriptive name in the Route List Name field.
Step 4.
In the Description field, enter a description that helps to easily identify this
route list.
Step 5.
From the Cisco Unified Communications Manager Group drop-down list,
select the Communications Manager group that will be used to determine to
which Communications Manager this route list will register.
Step 6.
Click the Save button.
Step 7.
A screen similar to that shown in Figure 4-5 displays. Notice that below the
Communications Manager group is a check box labeled Enable this Route
list. By default, this is selected, which means that the route group is active. If
during testing or troubleshooting you need to disable this route group, deselect this check box. For now, leave it at the default.
Step 8.
To add a route group to this list, click the Add Route Group button.
A screen similar to that shown in Figure 4-6 displays.
Step 9.
From the Route Group drop-down menu, select the desired route group.
Step 10. This is the point at which you can configure digit manipulation. If you want
to affect caller ID information, configure the three fields found under the
Calling Party Transformations heading. These fields are discussed earlier in
this chapter. The first field is labeled Use Calling Party’s External Phone
Number Mask. This field determines whether the mask configured on the
directory number is used for calls that are routed through this route group.
Step 11. In the Calling Party Transform Mask field, enter any mask that you want to
affect the caller ID.
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Figure 4-5 Route List Configuration—Adding Route Groups
Step 12. In the Prefix Digits (Outgoing Calls) field, enter any digits that you want to
have added to the front of the caller ID.
Step 13. By default, the Calling Party Number Type and the Calling Party Number
Plan fields are set to Cisco Unified Communications Manager. Only an individual that has extensive knowledge and experience using the dialing plans
such as the North American Numbering Plan (NANP) or the European dialing
plan should change this field. In many cases, these can be left to the default,
but some service providers might require these values to be set based on the
type of call being made. Check with your service provider to see whether this
is required in your environment.
Step 14. The fields under the Called Party Transformations heading affect the dialed
number. This is where you can manipulate the number sent to the gateway. An
example of how this might be used was shown in Figure 4-2. When the call
goes across the WAN, only five digits are needed, but if the WAN cannot
handle the call and it is sent to the PSTN, 11 digits are needed for the PSTN
to properly route the call. From the Discard Digits drop-down list, select the
digit discard instructions that should apply to calls that are sent out this gateway. A discard instruction determines which digits are removed from the
dialed digits before the call is sent out the gateway. There are more than 30
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different discard instructions for this field. However, most of them are combinations of one another. Table 4-1 lists the seven core discard instructions.
Figure 4-6 Adding Route Groups to the Route List
Note If the @ wildcard is not used, the only valid discard digit instructions are PreDot
and None.
Note These instructions are combined to create more than 30 instructions that you will
find in the list. Here’s an example. The discard instruction PreDot 10-10-Dialing removes
any number before the dot in the pattern, the 10-10, and carrier code. When the number
910103215835551212 is dialed, it matches the 9.@ pattern. The PreDot instruction causes the 9 to be removed. The 10-10 and carrier code of 321 are removed because of the
10-10 Dialing instruction. The final result is 5835551212. Applying what you learned
here, you can quite easily determine what effect the other discard instruction will have on
a dialed number.
Note Some of the discard instructions depend on the pattern that is matched when the
number is dialed. The following examples make reference to these patterns, and the next
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section explains these patterns in more detail. So, if you are unfamiliar with patterns
shown in these examples, fear not—you will find explanations for them very soon.
Table 4-1
Digit Discard Instructions
Discard Instruction
Description
Example
10-10-Dialing
Removes 1010 and the carrier 1010-321-585-555-5555
code
becomes 585-555-5555.
11/10D->7D
Changes 11 and 10 digits to 7 1-246-555-1212 or 246-5551212 becomes 555-1212.
11D->10D
Changes 11 digits to 10
1-246-555-1212 becomes 246555-1212.
Intl TollBypass
Removes international access
and country codes
011-64-3214322 becomes
3214322.
PreAt
Removes all numbers before When 912485551212 matches
the @ in the matching pattern the pattern 9@, the 9 is
stripped, resulting in
12485551212.
PreDot
Removes all digits before the When 912485551212 matches
dot (.) in the matching pattern the pattern 9.@, the 9 is
stripped, resulting in
12485551212.
Trailing-#
Removes the # from the end of 12465551212# becomes
the dialed digits
12465551212.
Step 15. In the Called Party Transform Mask field, enter the mask that you want to
use for calls going out this gateway.
Step 16. In the Prefix Digits (Outgoing Calls) field, enter any digits that you want to
add to the front of the dialed number before the call is sent out the gateway.
Note Because a number of transformations can be applied, it is important to understand
the progression in which the transformations take place. The order for calling party transforms is external phone mask first, followed by calling party transform mask, and finally
prefix digits. The order for called party transforms is discard instructions, followed by
called party transform mask, and finally prefix digits.
Step 17.
By default, the Called Party Number Type and Called Party Number Plan
fields are set to Cisco Unified Communications Manager. Only an individual
that has extensive knowledge and experience using the dialing plans such as
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the NANP or the European dialing plan should change this field. It is recommended that this field be left at the default.
Step 18. Click the Save button.
Step 19. An informational window displays stating that the route group has been
added and that the route list must be reset for the changes to take effect.
Click OK.
Step 20. The page shown in Figure 4-5 redisplays with the addition of the route group
that you just added. Repeat Steps 8 through 16 for any additional route
groups that you want to enter.
Step 21. After all route groups are added, they display in the Selected Groups box.
The order in which they display in this box determines the order in which
calls are distributed. To change the order, highlight the route group you want
to move and click the up or down arrow to the right of the box.
Step 22. You can remove a route group from the route list by highlighting the route
group and clicking the down arrow below the Selected Groups box. This
route group displays in the Removed Groups box.
Step 23. Click the Save button to complete the configuration of this route list.
Step 24. Depending on the version of Communications Manager, the route list might
need resetting before the change takes effect. Some versions of
Communications Manager include a note that states “change effective on
update; no reset required.” If a message appears stating that a reset is
required, click OK and you will be returned to the Route List Configuration
page. From this page, click the Reset button.
Step 25. You need to complete the reset by clicking the Reset button in the Device
Reset window that appears. Close the Device Reset window.
After all route groups and route lists are created, patterns need to be configured so that
when someone dials a number, there is something to match it against and route it to the
proper route list.
Understanding Route Patterns
When dialed digits are sent to the Communications Manager, the Communications
Manager must be able to match those digits with a pattern. A pattern is simply a set of
numbers or wildcards that Communications Manager matches against a dialed number. If
there is no pattern configured in Communications Manager that matches the dialed digits, the call fails. You have already seen some examples of patterns in previous sections.
However, this section supplies more detail and the configuration process on patterns.
Now that you know that every dialed number must have a matching pattern configured in
Communications Manager, you might think that it is impossible to enter a pattern for
every number that might ever be dialed. If not for wildcards, you would be correct.
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Patterns are made up of numbers, wildcards, and special characters. Wildcards allow a
single pattern to match multiple dialed numbers. The most popular wildcard is the “X.”
This wildcard matches any single digit of 0 through 9. You could use it, for example, if
you want to configure a pattern that matches the five-digit extensions of another cluster.
Perhaps the extension range of the other cluster is 52000 through 52999. In this case, a
pattern of 52XXX can be used because it matches all numbers within this range.
The “X” is only one of a number of useful wildcards. Table 4-2 list all the wildcards that
can be used.
Note You can extend the power of patterns by using multiple wildcards in the same pattern. This can also lead to unexpected behavior. For example, the pattern 532X? results in
matching nearly any dialed number that begins with 532. Take time to test the patterns
that you create to make sure that they behave as expected.
Table 4-2
Pattern Wildcards
Pattern
Definition
Example
X
This wildcard matches any single digit
0–9.
52XXX matches 52000 through
52999.
@
9.@ matches a number that can be
The “at” sign matches any number in
NANP. A simple way to think of this pat- dialed from North America and is proceeded by a 9.
tern is that it matches any number that
can be dialed from a phone in North
America.
!
The exclamation point (!) matches one or 55! matches any number that begins
more digits.
with a 55. The possible matches that
result from using the (!) are almost limitless.
[]
The digits found within the bracket
represent a range of numbers that a
single digit can match.
55[2-5] matches 552, 553, 554, and
555.
[^]
The digits found within brackets that
include a caret (^) represent a range of
numbers that are to be excluded when
matching a single digit.
55[^2-5] matches 550, 551, 556, 557,
558, 559, 55*, and 55#.
+
The plus sign (+) matches one or more
instances of the preceding character or
range in the pattern.
91X+ matches all numbers in the range
910 through
91999999999999999999999.
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In addition to wildcards, there are other characters that can be part of a pattern. These
are sometimes referred to as special characters and are listed as follows:
Table 4-2
Pattern Wildcards
Pattern
Definition
Example
?
The question mark (?) matches zero or
more instances of the preceding
character in the pattern.
57?2 matches 572, 5772, 5772,
577772, and 5777772. In short, it
matches any pattern that begins with a
5 followed by any number of 7s that is
followed by a 2 as the last number.
.
The dot (.) is used with digit manipulation. It is used to determine what digits
are to be stripped.
If the discard instruction of PreDot is
used with the pattern 9.@, the 9 is
stripped because it is in front of (pre)
the dot (.). If the number
912485551212 is dialed, the 9 is
stripped leaving, 12485551212.
*
The asterisk (*) is a valid digit that can be 543* matches only 543*.
dialed from a phone, so it can be part of
a pattern. It is important to realize that
this is not a wildcard. The asterisk is used
as a wildcard in many applications, but
not when it is part of a Communications
Manager pattern.
#
542342# matches only 542342# and
The octothorpe (#), which is often
routes the call as soon as the (#) is
referred to as the pound sign or hash
pressed. This is useful when the (!) is
mark, is also a valid digit that can be
dialed. However, it is most often used to used in a pattern. Typically the (#) is
signify that the caller has finished dialing. used after the (!) in a pattern.
The octothorpe should only be used as
the last digit of a pattern. Discard instructions can be used with this pattern to
remove the (#) before sending the call out.
Now that you know what a pattern does and what characters can be used to make up a
pattern, you can create some. The following section takes you through the steps that are
required to add a basic route pattern, and to point that pattern to a route list.
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Creating Basic Route Patterns
Before creating any route patterns, route lists should be created. This is because route
patterns point to route lists and can’t point to something that doesn’t exist. The following
steps show how to create a route pattern and point the route pattern to a route list. For
example, you can create a route pattern that matches all 11-digit calls and uses 9 as the
PSTN access code.
A number of route patterns are needed to create an efficient dial plan. The exact number
of route patterns needed varies. There are many things to consider when creating dial
plans. One example is the number of digits that must be dialed for each type of call.
Note While configuring a route pattern, keep in mind that, as with many components
that are configured in Communications Manager, not all parameters need to be configured.
In some cases, you might need to configure only a few parameters. To allow you to fully
understand the purpose of each, these steps cover each parameter.
Step 1.
From within CCMAdmin, select Call Routing > Route/Hunt > Route Pattern.
Step 2.
Click the Add New link. A screen similar to that shown in Figure 4-7 displays.
Enter the route pattern in the Route Pattern field. In this example, you want
to match 11 digits; this includes the 9 that is used as a PSTN access code. The
route pattern is 9[2-9]XX[2-9]XXXXXX. The [2-9] is used instead of an X
because in the NANP area codes and exchanges, you should never start with
a 1 or 0.
Step 3.
The Route Partition field determines what devices can access this route pattern. Partitions are discussed in more detail in Chapter 5. Select the desired
partition from the Partition drop-down list.
Step 4.
In the Description field, enter a description that helps identify the purpose
of this route pattern.
Step 5.
In the Numbering Plan field, choose the appropriate numbering plan.
Step 6.
From the Route Filter drop-down list, select the route filter that is to be
applied to this route pattern. Route filters are used to limit what digits match
the route pattern. These are explained in more detail in Chapter 5.
Note The Numbering Plan and Route Filter fields are only configurable if the pattern has
an @ in it, such as 9.@.
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Figure 4-7 Route Pattern Configuration
Step 7.
Determine the precedence level to be assigned to this route pattern from the
MLPP Precedence drop-down list (MLPP stands for Multilevel Precedence
and Preemption). Table 4-3 shows the available options.
Step 8.
The Resource Priority Namespace Network Domain field is used in environments that utilize Voice over Secured IP (VoSIP) using SIP trunks. If your
organization is utilizing this, select the appropriate domain from the dropdown list.
Step 9.
The Route Class parameter should be left at the default. This is usually used
only when the class route of an inbound T1-CAS needs to be translated to a
CM route class.
Step 10. From the Gateway/Route List drop-down list, select the route list to which
calls matching this route pattern are sent.
Note Typically a route list is selected here; however, there might be times that you want to
send a call directly to a gateway. This usually occurs when troubleshooting route plan issues.
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Table 4-3
MLPP Precedence Levels
Precedence
Description
Flash Override
Highest level of precedence, also known as level 0
Flash
Second-highest level of precedence, also known as level 1
Immediate
Third-highest level of precedence, also known as level 2
Priority
Fourth-highest level of precedence, also known as level 3
Routine
Lowest level of precedence, also known as level 4
Default
Leaves the incoming precedence unchanged
Step 11. To allow calls matching this route pattern to be routed, select the Route this
pattern radio button.
Note So far, we have discussed only using route patterns to route calls. However, route
patterns can also be used to block calls by selecting the Block This Pattern radio
button. For example, if you want to block all 900 calls, you can add a route pattern of
91900[2-9]XXXXXX and then select the Block This Pattern radio button.
Step 12. From the Call Classification drop-down list, select whether incoming calls on
this device should be considered OnNet or OffNet. This parameter is used to
determine whether calls can be transferred and forwarded. This is to help prevent fraud.
Step 13. The Allow Device Override check box allows the call classification setting
configured on the gateway or trunk to be used. It is recommended that this
be left deselected.
Step 14. To have a secondary dial tone played after the first digit is dialed, select the
Provide Outside Dial Tone check box.
Note One caveat to the Provide Outside Dial Tone option is that dial tone is provided
only when all possible matching patterns have this option selected. This can result in a dial
tone being provided after multiple digits have been dialed.
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Step 15. If the device you connect to requires that each digit is sent to it one at a time,
select the Allow Overlap Sending check box. While this is sometimes used in
Europe, in most cases, this box can be left deselected.
Step 16. To route a call as soon as it matches this route pattern, select the Urgent
Priority check box. Typically, a call is not routed if there are other patterns
that might also match, for example, if there is a 911 route pattern and a
91XXXXXXXXXX route pattern. When a caller dials 911, Communications
Manager matches the 911 route pattern but could also match the
91XXXXXXXXXX route pattern. If Urgent Priority is not selected,
Communications Manager waits for the interdigit timeout to expire (15 seconds by default) before routing the call. You can see how this is undesirable in
many cases. By selecting Urgent Priority, you ensure that Communications
Manager does not wait for any other digits to be dialed and instead routes the
call immediately.
Step 17.
In some versions of Communications Manager, you have the ability to choose
the Require Forced Authorization Codes (FAC) and Client Matter Codes
(CMC) fields. FAC forces the user to dial a code to be able to place the call.
CMC allows the user to enter a code so that the call can later be billed back
to a client. If you are implementing these features, select the appropriate
check boxes. If you are implementing FAC, enter an authorization level. FAC
and CMC are discussed in greater detail in Chapter 6, “Configuring CUCM
Features and Services.”
Step 18. If you want to affect caller ID information, configure the fields found under
the Calling Party Transformations heading. These fields are discussed earlier
in this chapter. The first field is labeled Use Calling Party’s External Phone
Number Mask. It determines whether the mask configured on the directory
number is used for calls that are routed through this route group.
Step 19. In the Calling Party Transform Mask field, enter any mask that you want to
affect the caller ID.
Step 20. In the Prefix Digits (Outgoing Calls) field, enter any digits that you want to
have added to the front of the caller ID.
Step 21. The Calling Line ID Presentation field determines whether caller ID information is to be blocked for outbound calls that match this route pattern. To
block caller ID, select Restricted from the drop-down list. To allow caller ID,
select Allowed from the drop-down list.
Step 22. The Calling Name Presentation field determines whether caller name information is to be blocked for outbound calls that match this route pattern. To
block calling name ID, select Restricted from the drop-down list. To allow
calling name ID, select Allowed from the drop-down list.
Step 23. By default, the Calling Party Number Type and the Calling Party Number
Plan fields are set to Cisco Unified Communications Manager. Only an indi-
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vidual that has extensive knowledge and experience using the dialing plans
such as the NANP or the European dialing plan should change this field. It is
recommended that this field be left at the default.
Step 24. The Connected Line ID Presentation field determines whether the connected
(destination) party’s ID information should display on the calling party’s
phone. To block the connected party’s ID, select Restricted from the dropdown list. To allow the connected party’s ID, select Allowed from the dropdown list.
Step 25. The Connected Name Presentation field determines whether the connected
party’s name information is displayed on the calling party’s phone. To block
the connected party’s name, select Restricted from the drop-down list. To
allow the connected party’s name, select Allowed from the drop-down list.
Step 26. From the Discard Digits drop-down list, select the digit discard instruction
that should be applied to calls that match this route pattern. These instructions are explained earlier in this chapter. Refer to Table 4-1 for additional
information.
Step 27.
In the Called Party Transform Mask field, enter the mask that you want to
use for calls that match this route pattern. If no mask will be used, leave this
field empty.
Step 28. In the Prefix Digits (Outgoing Calls) field, enter any digits that you want to
have added to the front of a dialed number before it is sent to the route list.
Step 29. By default, the Called Party Number Type and the Called Party Number
Plan fields are set to Cisco Unified Communications Manager. Only an individual that has extensive knowledge and experience using the dialing plans
such as the NANP or the European dialing plan should change this field. It is
recommended that this field be left at the default.
Step 30. ISDN network service can be invoked by configuring the ISDN NetworkSpecific Facilities Information Element fields. To do this, first select the
appropriate protocol from the Network Service Protocol drop-down list.
Your carrier provides this information.
Step 31. Enter the carrier code in the Carrier Identification Code field. Your carrier
provides this information if required.
Step 32. Based on the protocol used, various services can be selected and configured
using the last three fields, which are Network Service, Service Parameter
Name, and Service Parameter Value. These are rarely used and are left blank
in most cases.
Step 33. Click the Save button at the top of the screen to save this route pattern.
Now that you know how to add route patterns, move on and take a look at what a basic
route plan might look like.
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Chapter 4: Implementing a Route Plan
Using Pattern Wildcards to Create a Basic Dial Plan
Many factors have to be taken into consideration when implementing a route plan, such
as types of calls that will be allowed, the path that each type of call will take, and the
dialing characteristics of your local carrier. To help explain these factors in more detail,
take a look at a mock company. This illustration touches on the different factors to consider and offers a sample route plan for the company.
Figure 4-8 shows an overview of what must be accomplished for this company.
Dialing Rules for Chicago
• Calls to San Jose use 7 as a leading digit followed by an extension number.
• Most 10-digit calls and all 7-digit calls are local.
• Toll charges apply to area codes 249 and 348.
• All long-distance calls are allowed.
• International calls are allowed.
• 900 numbers are to be blocked.
• 911 calls are to be given priority.
• Must dial 9 to access the PSTN.
Chicago
PSTN
San Jose
WAN
20XX Extensions
14XX Extensions
Figure 4-8 Example Company Overview
The goal is to create a route plan for the Chicago office that satisfies the following dialing rules:
■
To reach the San Jose office, callers should dial 7 followed by the extension number.
■
All calls routed to the PSTN should be preceded by a 9.
■
The user should be able to dial all local calls.
■
Ten-digit calls that are not local should be allowed. These numbers require a pattern
different from the one that matches ten-digit calls that are local. This is done so that
when calling search spaces and partitions are added, these calls can be restricted.
■
Long-distance calls should be allowed.
■
International calls should be allowed.
■
900 calls should be blocked.
■
911 calls should be routed immediately.
At first glance, you might think that the 9.@ pattern can accomplish these goals.
However, it will not because it does not allow calls to the San Jose office and it does not
block 900 calls. So, as you can see, a single pattern will not suffice. Using the 9.@ could
allow calls that you had not intended. 9.@ matches any number that can be dialed from
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North America as long as a 9 is dialed first. The @ wildcard is equal to adding hundreds
of patterns. Because you often want to restrict the destinations that can be reached from
certain devices, it is sometimes desirable to use patterns more specific than 9.@.
Now look at the requirements one at a time to determine what patterns need to be created. Table 4-4 lists the goals and the patterns needed to accomplish this goal.
This example offers you a small sample of the types of patterns that need to be created
for an efficient and effective dial plan. This is only an example and should by no means
be interpreted as a recommended dial plan. Remember that dial plans vary based on the
environment. It is up to you to design a proper route plan from scratch and thoroughly
test it before using it in a production environment. Chapter 5 builds on this example to
show how to restrict certain devices from dialing certain numbers.
Note There is a 911 and a 9911 pattern. This is done to ensure that all attempts to reach
emergency services are successful. It is not possible to know whether people dialing 911
will know or remember that they must first dial a 9 and then 911. By having both patterns,
the call will be successful regardless of which pattern is used.
Table 4-4
Example Dial Plan
Goal
Pattern
7.14XX
To reach the San Jose office, callers
should be able to dial 7 followed
by the extension number.
All calls routed to the PSTN
should be preceded by a 9.
—
Notes
PreDot Discard
Add 9. to the front of all patterns that point to the PSTN.
Ensure that local calls can be
dialed.
9.[2-9]XXXXXX
9.773[2-9]XXXXXX
9.847[2-9]XXXXXX
PreDot Discard PreDot
Discard
Allow nonlocal 10-digit calls that
use a pattern different from that
used for 10-digit calls that are
local.
9.249[2-9]XXXXXX
9.348[2-9]XXXXXX
PreDot Discard PreDot
Discard
Allow long-distance calls.
9.1[2-9]XX[2-9]XXXXXXPreDot Discard
Allow international calls.
9.011!
9.011!#
PreDot Discard
PreDot Trailing-# Discard
Block 900 calls.
9.1900[2-9]XXXXXX
Block Pattern
Route 911 calls immediately.
911
9.911
Urgent Priority
PreDot Discard and Urgent
Priority
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Now that you understand what a route plan is and how to create one, move on and take a
look at some additional components that can be used when creating complex dial plans.
Advanced Route Plan Components and Behavior
The following sections examine the function and configuration steps for three additional
components that are found in dial plans:
■
Route filter
■
Translation pattern
■
Computer Telephony Integration (CTI) route points
These components are used when creating a more complex dial plan.
You start with a brief overview of each component and then see a more detailed exploration of the configuration of each. The first component is a route filter. Route filters are
applied to route patterns to limit the matches of the pattern. As mentioned before, the @
wildcard is equal to hundreds of different patterns. Route filters allow you to reduce the
number of patterns that match the @. For example, you can create a pattern of 9.@ and
apply a route filter that disallows 900 numbers from matching this pattern. The next section takes a look at how these are configured.
The next component that is discussed is the translation pattern. Translation patterns allow
a dialed number to be changed to another number. This is another place that digit manipulation can be accomplished. You might wonder why you would need another place for
digit manipulation because it can be done at both the route group and the route pattern.
There are times when digit manipulation might need to be done before matching a route
pattern and for incoming calls. Translation patterns allow the manipulation of both the
dialed number and the caller ID of the calling party. In addition, translation patterns can
be used to grant a different Calling Search Space and MLPP precedence. In the “Creating
Translation Patterns” section, later in this chapter, these are explored in greater detail.
The last route plan component examined in this chapter is CTI route points. CTI route
points are used to route calls to external service or devices. For example, CTI route
points are used to direct calls to a Cisco IP Interactive Voice Response (IVR) system and
the Cisco Personal Assistant system. After a CTI route point is created, a device or service registers with it and Communications Manager learns its IP address. All calls that
match the CTI route point are then sent to the registered IP address. More discussion and
configuration examples are offered in the “Creating CTI Route Points” section, later in
this chapter.
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Creating Route Filters
Route filters can make the @ wildcard much more manageable. It is often tempting to just
add the @ pattern to your route plan to make sure that all calls are allowed to be placed,
and if your company is unconcerned with who is calling where, you might get away with
this. However, most companies today want to make sure that their resources are not
being misused. By applying route filters to route patterns, you can limit which calls match
the pattern. Route filters allow you to use the @ and still ensure that only intended types
of calls are successful. Route filters are only used with route patterns that contain the @.
Note It is important to ensure that calls that should not be placed cannot be placed.
“When in doubt, don’t let it out.” This simply means that it is better for a route plan to be
too restrictive than too liberal. If callers can place calls that normally get placed, they will
be sure to let you know. However, if they discover they can place calls that they normally
cannot place, don’t expect them to tell you. This being said, remember above all else, that
you must ensure that certain types of calls, such as emergency services, can be placed
from all phones.
To make route filters work, you create one or more mandatory conditions for the filter to
allow the call to match the pattern. Figure 4-9 shows how a filter can affect which calls
match a pattern. As you see, the pattern used in this example is 9.@, and the condition in
the filter is that the area code matches 248. All the dialed digits match the pattern, but
only 912485559093 match the filter conditions. You might think that the 92485559090
should match because the area code is 248, but if it is not preceded by a 1, it is considered
a local area code and does not meet the conditions of the filter. There is a condition that
can be set on a route filter matching local area codes, but it is not included in this example.
Dialed Digits
Pattern Matches
Filter
Area Code = 248
912485559093
91900550000
92485559090
95551212
9@
912485559093
91900550000
92485559090
95551212
Pattern
File Matches
912485559093
Figure 4-9 Route Filter Example
Now create a route filter and apply it to a pattern. The following steps show how to create a route filter:
Step 1.
From within CCMAdmin, select Call Routing > Route Filter.
Step 2.
Click the Add New link.
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Step 3.
Select the appropriate numbering plan from the Numbering Plan drop-down
list and click Next.
Step 4.
A page similar to that shown in Figure 4-10 displays. In the Route Filter
Name field, enter a descriptive name for this filter. For example, if this filter is
going to match an area code of 248, you might want to name it AC248.
Whatever naming convention you choose, make sure that it helps you recognize the function of the route filter.
Figure 4-10 Configuring a Route Filter
Step 5.
Choose what conditions must be met.
Table 4-5 lists each tag and gives a brief description. The example in the description shows the portion of the number to which the tag is referring in bold.
Step 6.
To enable any of these tags, you have to select an operator from the dropdown list to the right of the tag. The available operators are
■
NOT-SELECTED: Ignores this tag.
■
EXISTS: Requires that this portion of the number exist.
■
DOES-NOT-EXIST: Requires that this portion of the number does
not exist.
■
==: Requires that this portion of the number match the value entered
in the box that appears to the right of this operator. The value box
does not appear unless this operator is selected.
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Table 4-5
Route Filter Tags
Tag
Description
Area-Code
Three digits that follow the 1 when dialing 11digit numbers.
[1-248-555-9093]
Country-Code
Each country has a one-, two-, or three-digit
country code.
[011-63-2-475983#]
End-of-Dialing
Character used to signify the end of a dialing
string. This is the # in the NANP.
[011-62-2-475983#]
International-Access
Two digits that signify that an international
number is being dialed. 01 is used in the
United States.
[011-62-2-475983#]
International-Direct-Dial
The digit that follows the International-Access
code. 1 is used in the United States.
[011-62-2-475983#]
International-Operator
The digit that is used when accessing the international operator. 0 is used in the United
States.
[00]
Local-Area-Code
The first three digits of a 10-digit call.
[248-555-9093]
Local-Direct-Dial
The single-digit code dialed when dialing a
local call directly. 1 is used in the NANP.
[1-555-9093]
End-of-Dialing
Character used to signify the end of a dialing
string. This is the # in the NANP.
[011-62-2-475983#]
Local-Operator
The digits used to reach the operator. 0 is used
in the NANP.
[0]
Long-Distance-Direct-Dial
The single-digit code dialed when dialing a
long-distance call directly. 1 is used in the
NANP.
[xxx]
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Table 4-5
Route Filter Tags
Tag
Description
Long-Distance-Operator
One- or two-digit code that requests operator
assistance for the call.
[xxx]
National-Number
The nation-specific portion of an international
call.
[xxx]
Office-Code
The first three digits of a 7-digit call.
[555-9093]
Satellite-Service
Single digit used to access satellites for international calls.
[xxx]
Service
Three-digit codes used for services.
[911]
Subscriber
The last four digits of a 7-digit number.
[555-9093]
Transit-Network
The 4-digit carrier code.
[1010321-1-248-555-9093]
Transit-Network-Escape
The 3-digit code entered before the carrier
code
[1010321-1-248-555-9093]
Step 7.
Select the operator for each tag that you want to enable, and enter the value
for any tag you select (==) as the operator.
Step 8.
Click the Save button.
Figure 4-11 shows what a page looks like for a filter that matches on a number that does
not include a country or area code and has 248 as the local area code. The filter would
allow the number 248-555-9093 through, but not 1-248-555-9093.
Now that the route filter is created, it must be applied to a route pattern. Typically route
filters are applied when the route pattern is created. The steps for creating a route pattern
showed how to apply a route filter. The following steps show how to add a route filter to
a route pattern that already exists:
Step 1.
From within CCMAdmin, select Calling Plan > Route/Hunt > Route Pattern.
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Figure 4-11
Step 2.
Route Filter Configuration Example
Search for the route pattern that you want to add to the route filter. You can
search by pattern, description, or partition by selecting one of these from the
first drop-down list. In the next drop-down list, select the search match operator and enter the search criteria in the field to the left of the Find button.
Click the Find button. Leaving the search criteria empty results in returning
all route patterns. You can use the drop-down list to the left of Items Per Page
to select how many results you want to have returned on a page.
Note It is important to remember that route filters only work with patterns that include
the @ wildcard.
Step 3.
From the results that display, select the route pattern that you want to add to
the route filter.
Step 4.
The next page that appears shows the properties of the route pattern you
selected. From the Route Filter drop-down list, select the route filter that you
want to apply to this pattern.
Step 5.
Click the Save button to save the changes.
As you can see, route filters have a significant effect on the route patterns to which they
are applied. Make sure that after you have created and applied route filters to route
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patterns, the pattern behaves as you expect. The Dialed Number Analyzer is a good tool
for verifying the results. This tool can be found on, and installed from, the Install Plugins
page. The Dialed Number Analyzer allows you to enter an origination and destination
number, and shows you how the call will be handled based on the current dial plan. For
more information on this tool, refer to the Dialed Number Analyzer guide, which can be
found at Cisco.com by searching for “Dialed Number Analyzer.”
Creating Translation Patterns
At times, digit manipulation might be necessary other than at the route pattern or route
group level. This is where translation patterns can be used. Translation patterns allow
called and calling digit manipulation to be performed. Translation patterns can also
change the calling search space and MLPP precedence of a call.
The issues that you can address by using translation patterns are limited only by your
imagination. Often, I come across interesting problems that are resolved using translation
patterns. The change in extension numbers represents one of the more common issues
that translation patterns can resolve. Often when a company moves, it is forced to get a
new range of DID numbers from the phone company. More times than not, the new
range is completely different than the old range. When the last few digits of a person’s
DID is also his or her extension number (which is normal), the person ends up with a new
extension number after the move. Employees are used to dialing the old extensions,
which are no longer valid, so the call fails. By creating a translation pattern that matches
the old extension and transforms it to the new extension, you can reduce the frustration
of the users while they get used to the new extension range.
Now take a closer look at how this would work. Assume that the old extension range was
5000–5999, but the new DID range is 7000–7999. The only part of the extension number
that changes is the first digit, so a translation pattern of 5XXX is created that changes
5XXX to 7XXX. When a user dials 5050, it matches the 5XXX translation pattern and
changes the called number to 7050. Because 7050 is the new extension, the call is
extended to the phone.
Another use for a translation pattern is to change the calling search space of a call. An
example of this is used with Cisco Personal Assistant (PA). PA has the capability to intercept calls on behalf of a PA user and route the call based on rules that the user creates.
This is done by creating a CTI route point that matches the extension of all PA users’
phones. The phones are then placed in a partition to which only the PA has access. This
works wonderfully until the PA is not functioning. At this point, all calls to the PA users’
phones fail. By creating a translation pattern that also matches the PA users’ phones and
assigning a calling search space that has access to the PA users’ phones, calls are allowed
directly to the phone if the PA fails. More complete information on the configuration can
be found in the PA installation guide, but the previous summary gives you an idea of how
a translation pattern can help resolve this issue by granting a call in the required calling
search space.
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The following steps show how to create and configure a translation pattern:
Step 1.
From within CCMAdmin, select Call Routing > Translation Pattern.
Step 2.
Click the Add New link.
Step 3.
A screen similar to that shown in Figure 4-12 displays. In the Translation
Pattern field, enter the pattern that you want to match. For example, if you
are trying to match all calls to the extensions from 2000 to 2999, enter 2XXX.
Figure 4-12
Translation Pattern Configuration
Step 4.
The Partition field determines what devices can access this pattern. Partitions
are a grouping of patterns and are discussed in more detail in Chapter 5.
Select the desired partition from the drop-down Partition list.
Step 5.
In the Description field, enter a description that helps identify the purpose
of this pattern.
Step 6.
In the Numbering Plan field, choose the appropriate numbering plan.
Step 7.
From the Route Filter drop-down list, select the route filter that is to be
applied to this pattern. Route filters are used to limit what digits match the
pattern and are only used when the pattern contains the @.
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Step 8.
Determine the precedence level that will be assigned to this pattern from the
MLPP Precedence drop-down list. Refer to Table 4-3, earlier in this chapter,
for the available options.
Step 9.
The Resource Priority Namespace Network Domain field is used in environments that utilize Voice over Secured IP (VoSIP) using SIP trunks. If your
organization is utilizing this, select the appropriate domain from the dropdown list.
Step 10. From the Calling Search Space drop-down list, select the calling search space
that the translation pattern searches for the newly created number.
Step 11. In certain version of Communications Manager, you can use an adjunct route
server to make routing decisions. If you are using this feature, select the
desired profile from the External Call Control Profile drop-down list.
Step 12. To allow calls to be routed that match this pattern, select the Route This
Pattern radio button. To disallow calls that match this pattern, select the
Block This Pattern radio button. When you select the button that blocks the
pattern, you must select a reason from the drop-down list to the right of the
radio button.
Step 13. To provide a secondary dial tone, select the Provide Outside Dial Tone
check box.
Step 14. To route a call as soon as it matches this route pattern, select the Urgent
Priority check box. Typically, a call is not routed if there are other patterns
that might also match, for example, if there is a 911 route pattern and a
91XXXXXXXXXX route pattern. When a caller dials 911, Communications
Manager matches the 911 route pattern, but could also match the
91XXXXXXXXXX route pattern. If Urgent Priority is not selected,
Communications Manager waits for the interdigit timeout to expire (15 seconds by default) before routing the call. You can see how this is undesirable
in many cases. By selecting Urgent Priority, you ensure that CallManager
does not wait for any other digits to be dialed and instead routes the call
immediately.
Note In earlier versions of Communications Manager, the Urgent Priority box cannot be
deselected. This means that as soon as a pattern matches a translation pattern, the call is
sent through the translation pattern. Communications Manager does not wait for additional digits even if other possible matches exist.
Step 15. If call screening that is based on the calling party number is being used
between clusters, you should select the Route Next Hop by Calling Party
Number check box. However, in most cases, you can leave this box deselected.
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Step 16. The Use Calling Party’s External Phone Number Mask field determines
whether the mask configured on the directory number is used for calls that
are routed through this route group.
Step 17.
In the Calling Party Transform Mask field, enter any mask that you want to
affect the caller ID.
Note This is where most people make mistakes. Often administrators enter the pattern
here to which they want the number changed. This is incorrect. This field affects caller ID,
not the number dialed. To affect a change on the dialed number, change the called party
transform mask. This is a common mistake because the Called Party Transform Mask field
cannot be seen on the screen unless you scroll down.
Step 18. In the Prefix Digits (Outgoing Calls) field, enter any digits that you want to
have added to the front of the caller ID.
Step 19. The Calling Line ID Presentation field determines whether caller ID information is to be blocked for outbound calls that match this pattern. To block
caller ID, select Restricted from the drop-down list. To allow caller ID, select
Allowed from the drop-down list.
Step 20. The Calling Name Presentation field determines whether caller name information is to be blocked for outbound calls that match this pattern. To block
calling name ID, select Restricted from the drop-down list. To allow calling
name ID, select Allowed from the drop-down list.
Step 21. By default, the Calling Party Number Type and the Calling Party Number
Plan fields are set to Cisco Unified Communications Manager. Only an individual that has extensive knowledge and experience using the dialing plans
such as the NANP or the European dialing plan should change this field. It is
recommended that this field be left at the default.
Step 22. The Connected Line ID Presentation field determines whether the connected
party’s ID information is to be displayed on the calling party’s phone. To
block the connected party’s ID, select Restricted from the drop-down list. To
allow the connected party’s ID, select Allowed from the drop-down list.
Step 23. The Connected Name Presentation field determines whether the connected
party’s name information is displayed on the calling party’s phone. To block a
connected party’s name, select Restricted from the drop-down list. To allow a
connected party’s name, select Allowed from the drop-down list.
Step 24. The next set of fields determines whether digit manipulation is performed on
the dial digits. From the Discard Digits drop-down list, select the digit discard instruction that should be applied to calls that match this pattern. These
instructions are explained earlier in Table 4-1.
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Step 25. In the Called Party Transform Mask field, enter the mask you want to use
for calls that match this pattern. This field determines how the dialed digits
will be transformed.
Step 26. In the Prefix Digits (Outgoing Calls) field, enter any digits that you want to
have added to the front of the dialed number before it is sent to the route list.
Step 27.
By default, the Called Party Number Type and the Called Party Number
Plan fields are set to Cisco Unified Communications Manager. Only an individual that has extensive knowledge and experience using the dialing plans
such as the NANP or the European dialing plan should change this field. It is
recommended that this field be left at the default.
Step 28. Click the Save button at the top of the page to add this translation pattern.
At this point, you should feel comfortable with translation patterns and understand how
to create them. When trying to create custom solutions, make sure that you keep translation patterns in mind because they can be used in a limitless number of ways.
Creating CTI Route Points
The last route plan component that is covered in this chapter is CTI route points. CTI
route points can be viewed as virtual ports that allow connectivity to other services and
devices. CTI route points are also sometimes used as “dummy” phones when you have a
DID that is associated with a voice mailbox, but not assigned to a physical phone. A CTI
route point can be created to match the DID number. This number is then configured to
forward all calls directly to the voicemail system.
The creation and configuration of CTI route points are similar to those of a phone. The
following steps show how to create and configure a CTI route point:
Step 1.
From within CCMAdmin, select Device > CTI Route Point.
Step 2.
Click Add New link.
Step 3.
A screen similar to that shown in Figure 4-13 displays. The first field that
must be configured is the device name for this route point. Use a name that
helps identify the function of this route point. For example, if this route point
is to be used for a contact center system, a name such as “CC_Route_Point”
would be a good choice.
Step 4.
The next field is the Description field. Enter a description that helps you
quickly identify the CTI route point.
Step 5.
From the Device Pool drop-down list, select the device pool that this CTI
route point will use.
Step 6.
From the Common Device Configuration drop-down list, select the common
device configuration that the CTI route point will use.
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Figure 4-13
CTI Route Point Configuration
Step 7.
A Calling Search Space (CSS) determines what destination the CTI route
point will be able to reach. CSS is discussed in Chapter 5. Choose a CSS from
the Calling Search Space drop-down list. If this field is left at None, the dial
privileges of this CTI route point could be limited.
Step 8.
Locations are used to prevent WAN links from becoming oversubscribed in
centralized deployments. These are discussed more in Chapter 5. If you have
defined locations, select the appropriate one for this CTI route point from the
Location drop-down list.
Step 9.
The User Locale field determines the language and fonts used for the CTI
route point. The default user locale, which is set in the enterprise parameters,
is used if this field is left at None. If this CTI route point needs to use a different locale than is defined by its device pools or the enterprise parameters,
select the proper one from the drop-down list.
Step 10. The Media Resource Group List field determines to which media resources
this CTI route point will have access. Media resources are discussed in further
detail in Chapter 6. From the Media Resource Group List drop-down list,
select the desired group. If no media resource group list is chosen, the one
defined in the device pool is used.
Step 11. The next two fields allow you to configure what audio source is heard when a
call is placed on hold. The first of the two, which is labeled User Hold Audio
Source, determines what is heard when the call is placed on hold by the
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application. The Network Hold Audio Source field determines what audio is
heard when the call is placed on hold by pressing the Transfer, Call Park, or
Conference button. Select the desired audio source from the drop-down list
for each field. If no audio source is chosen, the one defined in the device
pool is used.
Step 12. The Use Trusted Relay Point field determines whether a relay point such as a
Media Termination Point (MTP) or a transcoder must be labeled trusted to
be used by this device. This field is typically changed only in virtualized
environments.
Step 13. Calling party transformation allows you to change the caller ID. Select a
Calling Party Transformation CSS that contains the calling party transformation pattern assigned to the device. You can also leave this set to None and
use the Calling Party Transformation CSS assigned to the device pool by
selecting the Use Device Pool Calling Party Transformation CSS check box.
Step 14. The geolocation information can be used to determine the logical partition of
a device. If you use features that depend on geolocation, select the desired
geolocation from the drop-down list.
Step 15. Click the Save button to create the CTI route point.
Adding a Line to a CTI Route Point
Follow these steps to add a line to a CTI route point:
Step 1.
If you are adding a new CTI route point and have used the previous steps,
click Line [1] - Add a New DN. You should see a screen similar to that shown
in Figure 4-14. To add a line to an existing CTI route point, follow Steps 2
through 5 to reach this screen. If you are already at this screen, skip to Step 6.
Step 2.
From within CCMAdmin, select Device > CTI Route Point.
Step 3.
Enter search criteria in the search field to limit the results and click the Find
button.
Step 4.
Select the CTI route point to which you want to add a line from the list that is
generated.
Step 5.
At the bottom of the CTI Route Point Configuration screen, the available
lines are listed. Choose a line that has the label Add New DN.
Directory Number Information
Step 6.
The first and only field that you must complete is the Directory Number
field. Enter the extension number in this field.
Step 7.
The Route Partition field defines the partition to which this directory number
is assigned. The partition is used to determine what devices can call this
extension. Partitions are discussed in more detail in Chapter 5.
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Figure 4-14
CTI Route Point Directory Number Configuration
Step 8.
In the Description field, enter a description that will help you quickly identify the line later.
Step 9.
In the Alerting Name field, enter the name that should be displayed on the
caller’s phone while it is ringing.
Step 10. The ASCII Alerting Name field is the used on devices that do not support
Unicode characters.
Step 11. The Associated Devices box lists all the devices to which this line is
assigned. To remove the line from a listed device, highlight the device in the
Associated Devices box and click the down arrow that is between the
Associated Device box and the Dissociate Devices box. The device should
not appear in the Dissociate Devices box. Click Save and the line is removed
from the selected device.
Directory Number Settings
Step 12. The Voice Mail Profile field determines which voicemail profile the directory
number will use. The voicemail profile defines the number that is dialed when
the Messages button on the phone is pressed. Voicemail profiles are discussed in further detail in Chapter 5. Select the voicemail profile from the
drop-down list.
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Step 13. The next field allows a Calling Search Space (CSS) to be assigned as the line
level. This determines what destinations can be reached when calling from this
line. Select the Calling Search Space from the drop-down list.
Note It is important to understand what happens when a CSS is assigned to the line and
the device. In short, the two CSSs are combined; however, there is a little more to it. For a
detailed explanation, refer to Chapter 5.
Step 14. The presence group to which a line belongs determines which lines are allowed
to monitor. Select the presence group to which this device should belong.
Step 15. The next two fields allow you to configure what audio source is heard when a
call is placed on hold. The first of the two, which is labeled User Hold Audio
Source, determines what is heard when the call is placed on hold by pressing
the Hold button. The second field, Network Hold Audio Source, determines
what audio is heard when the call is placed on hold by pressing the Transfer,
Call Park, or Conference button. Select the desired audio source from the
drop-down list for each field. If no audio source is chosen, the source defined
at the device level is used, and if None is chosen, the source set in the device
pool is used.
AAR Settings
Step 16. Automated Alternate Routing (AAR) is used to provide an alternate route if a
call fails because of insufficient location bandwidth. To have failed calls reach
this line because of insufficient bandwidth sent to voicemail, select the check
box under Voice Mail. If you do not choose to send the calls to voicemail,
you must choose an AAR group and you might need to configure the AAR
destination mask. By default, AAR uses the external phone number mask to
determine the fully qualified number of the destination. If you do not want to
use the external phone number mask, enter the correct mask in the AAR
Destination Mask field. The AAR Group field determines the AAR group
with which the device is associated. An AAR group defines the prefix that is
assigned when a call fails because of insufficient bandwidth. Select the appropriate AAR group from the drop-down list.
Call Forward and Pickup Settings
Step 17.
The next seven fields deal with call forwarding. These fields determine the
forwarding destination, which depends on the reason for the forward.
Table 4-6 lists the ten types of forwards.
You can configure each type of “forward” to forward calls to voicemail or a
specific extension.
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Table 4-6
Forward Types
Forward Type
Forward Action
Forward All
Forwards all incoming calls
Forward Busy Internal
Forwards calls from internal callers when the
line is busy
Forward Busy External
Forwards calls from external callers when the
line is busy
Forward No Answer Internal
Forwards calls from internal callers that are not
answered
Forward No Answer External
Forwards calls from external callers that are
not answered
Forward No Coverage Internal
Forwards calls from internal callers when the
application that controls the directory number
fails
Forward No Coverage External
Forwards calls from external callers when the
application that controls the directory number
fails
Forward on CTI Failure
Forwards calls when a CTI route point or CTI
port fails
Forward Unregistered Internal
Forwards calls from internal callers when the
line is not registered
Forward Unregistered External
Forwards calls from external callers when the
line is not registered
To forward to voicemail, select the Voice Mail box. For this to work, a voicemail profile must be defined for the line.
To forward calls to another extension, enter the extension number in the
Destination field. When a destination is entered into any of the internal forwards, the number is automatically entered into the corresponding external
forward.
If you want the external calls to be forwarded to a different destination, simply enter the desired destination in the appropriate external forward field. A
calling search space can be applied to each forward type, which limits the
destinations to which a call can be forwarded. This is useful when you want
to restrict a line from forwarding calls to numbers that are long distance, but
still want long-distance calls to be placed from the line. Enter the appropriate
destinations and calling search spaces for each forward type.
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Step 18. In the No Answer Ring Duration field, enter the number of seconds that the
line will ring before forwarding to the Forward No Answer destination. If this
field is left blank, the value configured in Communications Manager service
parameters is used. The default value is 12 seconds.
Step 19. The Call Pickup Group field determines to which call pickup group this
directory number belongs. Call pickup groups allow a user to redirect an
incoming call on another phone to the user’s phone. Select the desired call
pickup group from the drop-down list. Call pickup groups are covered in
more detail in Chapter 6.
Park Monitoring
Step 20. The Park Monitoring Forward No Retrieve Destination External field determines where external calls are forwarded to when they are placed on park and
not retrieved. To send them to voicemail, select the check box below Voice
Mail. To send them to an alternate number, enter that number in the
Destination field. If the parker’s CSS does not have rights to dial the alternate
destination, you need to select a CSS from the Calling Search Space dropdown list.
Step 21. The Park Monitoring Forward No Retrieve Destination Internal field determines where internal calls are forwarded to when they are placed on park and
not retrieved. To send them to voicemail, select the check box below Voice
Mail. To send them to an alternative number, enter that number in the
Destination field. If the parker’s CSS does not have rights to dial the alternate
destination, you need to select a CSS from the Calling Search Space dropdown list.
Step 22. The Park Monitoring Reversion Timer field determines how many seconds a
call can be parked before CM sends an alert to the parked phone. If left blank,
the value set in the service parameters is used. The default is 60 seconds.
MLPP Alternate Party Settings
Step 23. The next set of parameters deals with MLPP alternate party settings. These
settings allow you to configure an alternate destination for precedence calls
that are not answered on this line or the forwarded number assigned to this
line. If MLPP is not being used, these parameters can be left empty. In the
first field, which is labeled Target (Destination), enter the number to which
unanswered precedence calls should be forwarded.
Step 24. In the MLPP Alternate Party Calling Search Space field, select the appropriate search space from the drop-down list. This calling search space limits the
destinations to which precedence calls can be forwarded.
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Step 25. In the MLPP No Answer Ring Duration field, enter the number of seconds
that the phone will ring when it receives a precedence call before forwarding
to the Forward No Answer destination if unanswered.
Line Settings for All Devices
Step 26. The Hold Reversion Ring Duration field determines how many seconds a call
can be on hold before CM rings the phone that placed the call on hold. The
call will ring until it is answered or the maximum hold duration expires. Enter
the desired amount of seconds in this field. If left blank, the value set in the
service parameters is used.
Step 27.
The Hold Reversion Notification Interval field determines the intervals at
which the holding party will receive an alert reminding him or her that a call is
on hold. Enter the desired amount of seconds in this field. If left blank, the
value set in the service parameters is used.
Step 28. The Party Entrance Tone field determines which tone is played when a new
caller joins a call. When this parameter is set to Default, the value set in the
service parameters is used. To ensure that a tone is played, set this to On. To
ensure that no tone is played, set this to Off.
Line Settings for This Device
Step 29. The next two fields, labeled Display (Internal Call ID), are used to configure
which caller ID is displayed when calls are placed to other internal callers.
Enter up to 30 characters in this field. Both letters and numbers are allowed. If
this field is left blank, the line’s directory number will be used.
Note There two fields for the internal caller ID. The second, which is labeled ASCII
Display, is used for devices that do not support Unicode character display. This is also true
for the next two fields labeled Line Text Label.
Step 30. The External Phone Number Mask field can be used to modify the external
caller ID for calls placed from this line. An example mask might be
408370XXXX. The extension number is used to fill in the XXXX portion.
In this example, if the directory number is 1401, the external phone mask
would cause the external caller ID number to be 4083701401. The external
phone mask on the first line creates the fully qualified directory number that
is displayed above the first extension on certain IP phones.
Multiple Call / Call-Waiting Settings
Step 31. The next field, which is labeled Maximum Number of Calls, determines how
many active calls can be on the line. The maximum is 10,000 active calls per
phone. Enter the maximum number of calls in this field.
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Step 32. The Busy Trigger field determines how many active calls are required before
the line is considered busy. The default is two. This means that if the maximum number of calls on the line is four and the busy trigger is two, the third
inbound call will receive a busy indication. The user can place additional outbound calls up to the maximum number of four.
Forwarded Call Information Display
Step 33. The Forwarded Call Information Display section determines what information
is sent when a call is forwarded. Select the information to be sent by selecting
the check box next to each desired field.
Step 34. Click the Save button at the top of the screen to add this line.
The number of CTI route points you need depends on their individual requirements and
number of applications needed for connection. Refer to the specific application installation guide when configuring CTI route points to ensure that you set all the parameters
correctly.
Summary
This chapter has covered the tasks required to implement a basic dial plan. The call flow
was explored to give the reader a good overview of the components that make up a dial
plan. After that, each individual component was discussed, including route groups, route
lists and route patterns, and step-by-step configurations. Additional components that are
used in more advanced dial plans, such as route filters, translation patterns, and CTI route
points, were also covered. After completing this chapter, you should feel comfortable
with how a basic route plan works and how to configure one. The next chapter covers
components that can be added to a route plan to restrict certain devices from placing
calls to certain destinations, and discusses ways to ensure that WAN links are not oversubscribed, which in turn helps maintain good voice quality.
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Configuring Class of Service
and Call Admission Control
Now that you have created a basic dial plan, it is time to build on that and create a more
complete dial plan. Often you want to allow and disallow access to certain destinations.
For example, you might want only a certain group of callers to dial international numbers. This is done by creating a telephony class of service (CoS). In addition, if there are
calls traversing limited-bandwidth links, some type of Call Admission Control (CAC)
should be deployed to help ensure voice quality. This chapter examines the various concepts associated with CoS and CAC and describes how to configure the required components for each.
Rights and Restrictions
After the dial plan is created and users can place calls to destinations outside the cluster,
you might think that you are all set and can sit back and relax. Not quite. After the system is configured to enable calls to be placed outside of the system, you need to start
working on how to prevent certain calls from being placed. This chapter touches on how
you can use route patterns to block certain destinations, and now you need to move
beyond that and discuss how certain destinations can be reachable by some devices, but
not by others. To accomplish this, you need to configure Calling Search Spaces (CSS) and
partitions. The following sections explain what these are and how they work.
Understanding Call Search Spaces and Partitions
Of all the concepts within a Communications Manager environment, it is believed that
the CSS and partitions cause the most confusion. This is rather odd because they are not
complex. Simply put, the partition assigned to the destination affects what devices can
reach it, and the CSS determines which destinations can be reached. Locks and key rings
are good analogies. Think of the partition as a lock and the CSS as the key ring. To place
a call to a destination, you must have a key that matches the device’s lock. The key ring
contains all the keys and therefore determines which destinations you can reach.
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Of course, there is more to it than just locks and keys, but by using this analogy, you
begin to understand how they work. You take a closer look at this analogy. Figure 5-1
shows five phones. The first four phones have partitions (locks). It is important to point
out that the partitions (locks) are not assigned to devices, but rather to patterns and directory numbers (DN). For this example, assume that each phone has only a single line and
that the partition (lock) is assigned to that line. Below each phone is a CSS (key ring) that
shows to which partitions (locks) the phone has access. CSS (key rings) can be assigned
to the device or the line. In this example, assume that they are assigned to the device.
A
B
C
D
E
IP Phone
IP Phone
IP Phone
IP Phone
IP Phone
Figure 5-1 Calling Search Spaces and Partitions Analogy
The locks in this figure have different-shaped keyholes, which means that to open a lock,
you must have a key ring that has the correct-shape key. Keeping in mind that the locks
represent partitions and that the key rings represent CSS, answer the following questions:
■
What phones can phone A reach?
■
What phones can phone D reach?
■
What phone can reach all other phones?
■
What phones can reach phone E?
■
What phones can phone E reach?
The answers to these questions are as follows:
Q. What phones can phone A reach?
A. To determine what phones phone A can reach, you need to look at its CSS (key ring).
Phone A has a circular key and a square key on its key ring, which means that it can call
itself and phone B. However, because phone E has no lock (partition) assigned to it, any
phone can reach it, just as a door with no lock can be opened by anyone.
Q. What phones can phone D reach?
A. Because phone D has only a square key, it can dial phone A and, of course, phone E
because it has no lock (partition).
Q. What phone can reach all other phones?
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A. Because phone B has a key ring (CSS) that contains all the keys, it can reach all the
devices.
Q. What phones can reach phone E?
A. Because phone E has no lock (partition), all phones can reach it.
Q. What phones can phone E reach?
A. Because phone E has no keys, it can only reach devices that have no locks. In this example, phone E can only dial itself.
Figure 5-1, along with these questions and answers, should help you begin to understand
how partitions and CSS work. Of course, as with any simple concept, it has the potential
to become more complicated as the number of CSS and partitions grows. This is where
some people begin to become confused, because of an inaccurate base understanding of
the concepts. Now look at some of the more interesting aspects of CSS and partitions.
The first misconception that should be dispelled is this: If two devices have the same partition, they can call each other. Having the same partition alone is not enough. Going
back to the lock and key ring analogy, if two people have the same locks, keyed the same
way on their houses, but they have no keys, can they access each other’s houses? Of
course they can’t, and they cannot even access their own houses. This demonstrates that
a device’s partition (lock) has no effect on where the device can call. However, if two
devices that have the same partition also have a CSS that enables them access to their
partitions, they can dial each other.
The next important point is the order of CSS. As demonstrated in the earlier example,
CSS can enable access to more than one partition. Now, imagine that a device has a CSS
that enables it to match two devices with the same number, but in different partitions.
Figure 5-2 offers an example of this situation.
A
1001
B
1002
C
1003
D
1001
E
1004
IP Phone
IP Phone
IP Phone
IP Phone
IP Phone
Figure 5-2 CSS Matches Multiple Destinations
In this example, phones A and D have the same extension of 1001. Phones B and C can
reach both phones because their CSS enables access to both the square and triangle partition. So the question is, which phone rings when phone C dials 1001? Often people
answer this question with, “It takes the closer match.” Because 1001 matches 1001 exactly, both phones are the closest matches. Others assume that both phones ring because
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phone C has access to both partitions. What actually happens is that when a search for a
match is conducted, multiple closest matches are found. Because there are multiple closest matches, the order in which objects appear in the CSS comes into play. When you create a CSS, you prioritize the order in which partitions should be searched. This order
determines which partition is used if there are two closest matches. In the example,
Figure 5-2 shows that the order of the keys for phone C is square followed by triangle,
meaning that when phone C dials 1001, it would first match the 1001 that has the square
partition, which is phone A.
To add a little more complexity to this, it is possible to have a CSS on both the device
and the line. For example, the phone can have a CSS that grants access to the square partition, and a line on the phone can have a CSS that grants access to the triangle partition.
In such a case, the line CSS takes priority. Figure 5-3 shows an example of this. This
example moves away from the locks and keys analogy to focus more on the actual terms.
A
1001 CSS
1010 CSS
None
Executive
B
1004 Partition
Employee
C
1003 Partition
Lobby
D
1004 Partition
Executive
Device's CSS = Lobby, Employee
Figure 5-3 Line/Device CSS Example
In Figure 5-3, phone A has two lines, 1001 and 1010. 1001 has no CSS, and line 1010 has
a CSS that grants access to devices in the executive partition. Phone A also has a CSS at
the device level, which enables access to devices in the lobby and employee partitions.
Because the 1001 line does not have a CSS of its own, it has access only to devices that
can be reached using the device’s CSS. Because line 1010 has a CSS of its own, it has
access to devices that can be reached using its CSS and the device’s CSS. This means that
when dialing from line 1001, only devices in the lobby and employee partitions are accessible; but when dialing from line 1010, devices in the lobby, employee, and exec partitions are accessible.
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Now, take a look at the other three phones. Phone B has the extension of 1004, and that
line is in the employee partition. Phone C has the extension of 1003, and that line is in the
lobby partition. Phone D has the extension of 1004, and that line is in the exec partition.
Using what you have learned, answer the following three questions:
■
What is the result if 1004 is dialed from line 1001?
■
What is the result if 1004 is dialed from line 1010?
■
Can line 1010 reach line 1003?
The answers to these questions are as follows:
Q. What is the result if 1004 is dialed from line 1001?
A. Because line 1001 has no CSS of its own, it relies solely on the device’s CSS. The
device’s CSS has access to the employee and lobby partitions, so line 1001 can reach
only the 1004 on phone B because it is in the employee partition.
Q. What is the result if 1004 is dialed from line 1010?
A. Because line 1010 has a CSS, it has access to all devices to which the line and device’s
CSS grants access. Because it can reach phones B and D and both of them match 1004,
the line’s CSS takes priority and phone D rings.
Q. Can line 1010 reach line 1003?
A. Line 1010 can reach any device to which the line and/or the device’s CSS grants access.
Because the device’s CSS has access to the lobby partition, which is the partition that
1003 is in, it can reach it.
At this point, you should have a good idea of how partitions and CSS work. We now take
a look at a real-world, practical example of how CSS and partitions can be used.
The BGD Company has deployed a Communications Manager solution and has configured the route patterns that are shown in Table 5-1. As you can see, the route patterns
enable callers to reach anywhere they need to dial with the exception of 1-900 numbers,
which are blocked. The problem is that these patterns also enable some callers to make
unauthorized calls that the company disapproves of. For example, if a person’s job does
not require the placement of international calls, the dial plan should not enable the
employee’s phone to place them.
In this example, BGD has decided that it actually has four classes of users. The first class,
the executives, can make any calls they want, other than 1-900 calls. The second class,
the administrative assistants, are not allowed to make 1-900 calls or international calls.
The third class, standard users, can only reach internal extensions, local numbers, and
emergency services. The fourth class, lobby phones, for example, can only make calls
internally and to emergency services. To accomplish this, partitions and CSS must be
configured and assigned to patterns and devices.
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Table 5-1
BGD’s Route Patterns
Pattern
Notes
Matches
9.[2-9]XXXXXX
9.810[2-9]XXXXXX
PreDot Discard
PreDot Discard
Local 7- and 10-digit calls
9.810586XXXX
9.810587XXXX
PreDot Discard
PreDot Discard
10-digit calls that are not
local
9.1[2-9]XX[2-9]XXXXXX PreDot Discard
Long-distance calls
9.011!
9.011!#
PreDot Discard
PreDot Trailing-# Discard
International calls
9.1900[2-9]XXXXXX
Block Pattern
1-900 numbers
911
9.911
Urgent Priority
PreDot Discard and Urgent
Priority
Emergency service calls
When creating partitions, a common practice is to name them so that the name describes
to what the partition is assigned. For example, a partition that is going to be assigned to a
pattern that matches a local number can be called Local PT.
Tip The PT at the end of the name helps identify this object as a partition. Because it is
possible to have partitions and CSS with the same name, it is recommended that you add a
PT to the end of partition names and CSS to the end of CSS names.
In this example, five types of calls are allowed: internal, local, long-distance, international, and emergency. The following is the list of partitions that are needed, and to which
patterns they are assigned:
■
Internal_PT: Patterns that match internal numbers
■
Local_PT: Patterns that match local numbers
■
LD_PT: Patterns that match long-distance numbers
■
International_PT: Patterns that match international numbers
■
Emergency_PT: Patterns that match emergency service numbers
After the partitions are created, CSS are needed. Because BGD has defined four classes
of users, four CSS are needed. Just as with partitions, it is recommended that CSS are
named so that the name helps identify to which partitions the CSS have access. Table 5-2
shows the CSS and the partitions to which each has access and that are needed for BGD.
Now that partitions and CSS are defined, take a look at what each is assigned to. First,
examine the partitions. You need to understand that partitions are assigned to patterns
of DNs, not devices. This means that if you want to prevent a device from making
long-distance calls, you assign a partition to the patterns that match long-distance numbers,
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and make sure that the device’s CSS does not have access to the partition. Table 5-3 shows
the five partitions that have been created and the patterns to which each is assigned.
Table 5-2
CSS and Associated Partitions
CSS
Partitions
Internal_CSS
Internal_PT
Emergency_PT
Internal_Local_LD_CSS
Internal_PT
Local_PT
Emergency_PT
Internal_CSS
Internal_PT
Local_PT
LD_PT
Emergency_PT
Unlimited_CSS
Internal_PT
Local_PT
LD_PT
International_PT
Emergency_PT
Table 5-3
Partitions and Patterns
Partitions
Patterns
Internal_PT
DNs of the phones
Local_PT
9.[2-9]XXXXXX
9.810[2-9]XXXXXX
LD_PT
9.810586XXXX
9.810587XXXX
9.1[2-9]XX[2-9]XXXXXX
International_PT
9.011!
9.011!#
Emergency_PT
911
9.911
You might notice that the 9.1900[2-9]XXXXXX pattern has not been assigned to a partition but will still work. Remember, if a pattern does not have a partition explicitly assigned,
it falls into the null partition, and all devices have access to the null partition. Because the
9.1900[2-9]XXXXXX pattern is set up so that it blocks all calls that match it, you want all
devices to have access to it so that no one can place these types of calls. However, it is recommended to apply partitions to all patterns to ensure that no calls can be placed by
phones that do not have the proper CSS. With this is mind, the Internal_PT partition can be
applied to the 9.1900[2-9]XXXXXX pattern because all devices can reach that partition.
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Now look at how the CSS should be assigned. Remember that CSS can be assigned at
both the device and line. For this example, they are assigned at the device level only.
Table 5-4 shows the CSS and the types of device to which each is assigned.
Table 5-4
CSS and Assigned Devices
CSS
Devices
Unlimited_CSS
Executive phones
Internal_Local_LD_CSS
Administrative assistant phones
Internal_Local_CSS
Standard users
Internal_CSS
Lobby phones
Now that you understand which CSS and partitions are needed for BGD, and where each
is applied, take a look at the big picture. Table 5-5 shows which CSS is assigned to each
of the four different classes of phones. Under the CSS heading is a list of the partitions
that can be accessed. Under the partitions is a list of patterns. Using this table, it is easy
to see what destinations various phones can reach.
Table 5-5
CSS Assigned to Phones and the Patterns They Can Reach
Device
CSS>Partition>Patterns
Executive phones
Unlimited_CSS
Internal_PT
All Internal Phones
Local_PT
9.[2-9]XXXXXX
9.810[2-9]XXXXXX
LD_PT
9.810586XXXX
9.810587XXXX
9.1[2-9]XX[2-9]XXXXXX
International_PT
9.011!
9.011!#
Emergency_PT
911
9.911
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Table 5-5
CSS Assigned to Phones and the Patterns They Can Reach
Device
CSS>Partition>Patterns
Administrative assistant phones
Internal_Local_LD_CSS
Internal_PT
All Internal Phones
Local_PT
9.[2-9]XXXXXX
9.810[2-9]XXXXXX
LD_PT
9.810586XXXX
9.810587XXXX
9.1[2-9]XX[2-9]XXXXXX
Emergency_PT
911
9.911
Standard user phones
Internal_Local_CSS
Internal_PT
All Internal Phones
Local_PT
9.[2-9]XXXXXX
9.810[2-9]XXXXXX
Emergency_PT
911
9.911
Lobby phones
Internal_CSS
Internal_PT
All Internal Phones
Emergency_PT
911
9.911
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Up to this point, only the assigning of CSS to phones and lines has been discussed.
CSS are assigned to devices, which include gateways. A CSS is assigned to a gateway
so that inbound calls can reach internal destinations. In the example of BGD, all the
internal phones are placed in the Internal_PT partition. If the gateways do not have
access to this partition, no incoming calls are allowed. So you can see that not only
must phones have CSS, but gateways require them as well. In the case of BGD, the
Internal_CSS can be assigned to the gateways, which would grant outside calls access
to all internal phones.
Note CSS and partitions are only locally significant. This means, for all intents and purposes, that after a call leaves the local system, the CSS and partitions no longer exist.
In the BGD example, all internal phones were in the Internal_PT partition, meaning that
because all devices had a CSS that granted access to the Internal_PT partition, all phones
could be reached. In some cases, this is undesirable. Sometimes there are certain numbers
that should be reached only by certain devices. An example often used is that of an executive’s phone. Often it is desired that only the executive’s assistant be able to reach the
executive. To accomplish this, the executive’s phone is placed in a separate partition, to
which only the assistant’s phone has access.
Now that you have a good idea of what CSS and partitions are, move on to discussing
how they are created and configured.
Creating Calling Search Spaces and Partitions
Creating CSS and partitions is much easier than understanding and properly applying
them. Before you move on to the process of creating them, you should make sure that
you have taken the time to determine the different classes of users your environment
has, and what destinations each user will be allowed to call. After you have done this,
create a list of the partitions that are required. Next, create a CSS that defines what
partitions are accessible. After you have created this, you can begin to create the
partitions and CSS. Because CSS are created by choosing partitions to which they
will have access, the partitions must be created first. The following steps show how
to create partitions:
Step 1.
From within CCMAdmin, select Call Routing > Class of Control > Partition.
Step 2.
Click the Add New link.
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Step 3.
A screen displays that offers an area in which you can enter the name of the
partition followed by a description. You must place a comma (,) between the
name and description. If you do not enter a description, the name of the partition will be used as the description. You can create up to 75 partitions at a
time on this screen by placing each on a new line. Figure 5-4 shows an example of adding five partitions at one time.
Figure 5-4 Creating Partition Configurations
Step 4.
After you enter all the desired partitions, click Save.
As you can see, the creation of partitions is a simple task. Now that partitions are added,
you can start to create CSS by working through the following steps:
Step 1.
From within CCMAdmin, select Call Routing > Class of Control > Calling
Search Space.
Step 2.
Click the Add New link.
Step 3.
A screen similar to that shown in Figure 5-5 displays.
Step 4.
Enter a name in the Calling Search Space Name field. Remember that the
name should help identify the purpose of this CSS.
Step 5.
Enter a description in the Description field.
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Figure 5-5 Creating CSS
Step 6.
A list of partitions displays in the Available Partitions box. If you have a
large number of partitions, you can limit the partitions that display in this list
by entering the appropriate search criteria in the Find Partitions Where fields
and clicking Find.
Step 7.
Highlight the first partition to which you want the CSS to have access, and
click the Down Arrow below this box. This causes that partition to display in
the Selected Partitions box.
Step 8.
Repeat Step 7 for each partition to which you want the CSS to have access.
These should be added in the order you want to have them searched.
Step 9.
After all the partitions have been added, you can change the order in which
they display. Remember, the order in which they display determines which
partition is used if multiple partitions within the same CSS contain exact
matches for a dialed number. To change the order, highlight the partition you
want to move, and click the Up or Down Arrow to the right of the box.
Figure 5-6 shows what the screen looks like when adding the
Internal_Local_CSS, which was used in the previous example.
Step 10. After all desired partitions are listed in the correct order in the Selected
Partitions box, click Save.
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Figure 5-6 Example CSS
You need to repeat these steps to add all the CSS that your environment requires. After
all the partitions and CSS are added, it is time to apply them. Adding partitions and CSS
have absolutely no effect on call processing until they are applied to patterns and devices.
Applying Calling Search Spaces and Partitions
You are now ready to start applying the partitions and CSS to devices and patterns. After
a partition is added to a pattern, only devices that have the correct CSS can reach that
pattern. For this reason, you might want to assign CSS to the devices before assigning
partitions. Assigning partitions before assigning CSS is similar to putting a lock on a door
and not giving anyone a key. Until the keys are handed out, no one can get in.
Note When adding partitions and CSS to a system, it is best to apply them during nonproduction times. After they are added, thorough testing should be done. If it is not possible to add them off-hours, be certain to apply CSS before applying partitions. A good tool
that can be used to verify the results is called the Dialed Number Analyzer. This tool can
be installed from the Install Plugins page. This tool enables you to enter an origination and
destination number and shows you how the call will be handled, based on the current dial
plan. For more information on this tool, refer to the Dialed Number Analyzer guide, which
can be found at Cisco.com by searching for Dialed Number Analyzer.
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CSS are applied to devices and lines. When applied to both, the line’s CSS has priority
but does not nullify the devices. This means a line that has its own CSS has access to partitions that both the line’s CSS and the device’s CSS allow.
Note Often people want their assistants to answer their lines for them. To do this, you
must put the directory number on the assistant’s phone. The CSS assigned to the line stays
with the line, no matter which phone the line is on. This means that if the boss’s line has
the rights to call international numbers, the assistant can do so as well, if the boss’s line is
on the assistant’s phone. To deal with this, it is recommended that the more generous CSS
be applied to the device, not the line.
Now take a look at how a CSS is assigned to a phone, a line on the phone, and a gateway.
Assigning a CSS to a Phone
The steps that follow show how to assign a CSS to a phone:
Step 1.
From within CCMAdmin, Select Device > Phone.
Step 2.
Enter search criteria in the search field to limit the results and click Find.
Step 3.
Select the phone to which you want to assign a CSS from the list that is
generated.
Step 4.
The Phone Configuration screen displays. To assign a CSS to the phone,
select a CSS from the Calling Search Space drop-down list, as shown in
Figure 5-7.
Step 5.
Click Save.
Step 6.
A window displays to inform you that you must click the Apply Config button for the change to take affect. Click OK.
Step 7.
Click Apply Config.
Step 8.
A window displays to warn you that when you apply the configuration, the
device might go through a restart. Click OK.
Assigning a CSS to a Line
The steps that follow show how to assign a CSS to a line on a phone:
Step 1.
From within CCMAdmin, select Device > Phone.
Step 2.
Enter search criteria in the search field to limit the results and click Find.
Step 3.
Select the phone that contains the desired line from the list of phones that is
generated.
Step 4.
Click the desired line on the left side of the screen.
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Figure 5-7
Assigning a CSS to a Phone
Step 5.
On the Directory Number Configuration page, select the desired CSS from
the Calling Search Space drop-down list, as shown in Figure 5-8.
Step 6.
Click Save.
Note When you make a change to the directory number configuration and click Update,
the line will be reset on all the phones that appear on this line. If a caller is currently on a
call, the line resets after the call is ended.
Assigning a CSS to a Gateway or Trunk
The steps that follow show how to assign a CSS to a gateway or a trunk. Because the
steps are so similar for both components, they have been combined.
Step 1.
From within CCMAdmin, select Device > Gateway or Device > Trunk.
Step 2.
Enter search criteria in the search field to limit the results, and click Find.
Step 3.
From the list that is generated, select the gateway/trunk to which you want to
assign a CSS.
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Figure 5-8 Assigning a CSS to a Line
Note When configuring CSS for Session Initiation Protocol (SIP) trunks, additional CSSs
are configured. In the Inbound Call section, select the desired AAR CSS from the AAR
Calling Search Space drop-down list. Under the Outbound Calls section, select the desired
CSS from the Called Party Transformation CSS and Calling Party Transformation CSS
drop-down lists. Under the SIP Information section, there are additional CSSs that should
be configured. These included the Rerouting Calling Search Space, Out-Of-Dialog Refer
Calling Search Space, and SUBSCRIBE Calling Search Space.
Step 4.
Select the CSS from the Calling Search Space drop-down list.
Note For some gateways such as Media Gateway Control Protocol (MGCP), you need to
navigate to the subunit configuration page to assign a CSS.
Step 5.
Click Save.
Step 6.
A window displays informing you that you must click Apply Config for the
change to take affect. Click OK.
Step 7.
Click Apply Config.
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Step 8.
A window displays warning you that when you apply the configuration, the
device might go through a restart. Click OK.
Now that you have assigned CSS, you can assign partitions. Partitions are assigned to patterns of directory numbers. Examples of how to assign them to CSS and partitions follow.
Assigning a Partition to a Line (Directory Number)
The following steps show how to assign a partition to a line:
Step 1.
From within CCMAdmin, select Device > Phone.
Step 2.
Enter search criteria in the search field to limit the results and click Find.
Step 3.
Select the phone that contains the desired line from the list of phones that is
generated.
Step 4.
Click the desired line on the left side of the screen.
Step 5.
On the Directory Number Configuration page, select the desired partition
from the Route Partition drop-down list, as shown in Figure 5-9.
Figure 5-9 Assigning a Partition to a Line
Step 6.
Click Save.
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Assigning a Partition to a Pattern
Step 1.
From within CCMAdmin, select Call Routing > Route/Hunt > Route Pattern.
Step 2.
To limit the results, enter search criteria in the search field and click Find.
Step 3.
Select the route pattern from the list that displays.
Step 4.
Select the partition from the Route Partition drop-down list, as shown in
Figure 5-10.
Figure 5-10
Step 5.
Assigning a Partition to a Route Pattern
Click Save.
Note Two alert windows can appear. If you have not assigned authorization codes to this
pattern, an alert will inform you of this. You can click OK. The other alert informs you that
any update to a route pattern automatically resets the route list or gateway; click OK.
After the partitions are applied, you can begin testing the system to ensure that allowed
calls can be placed, and those that are not allowed cannot be placed.
Adding CSS and partitions after the system is in place can require a lot of work.
Remember that you can use the the Bulk Admin Tool (BAT) to quickly apply or change a
CSS or partition on a large number of objects.
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Implementing Call Admission Control
After you have set up your system to allow calls to be placed to outside destinations and
have applied CSS and partitions to restrict access, you need to configure the system to
ensure the quality of the calls. Although there are many things that can affect the quality
of the call, this book deals only with things that can be configured directly in
Communications Manager. The following sections discuss what must be configured to
ensure that the Voice over Internet Protocol (VoIP) link is not over subscribed.
When calls are placed between sites using an IP link as the transport, the quality of the
call can be affected if more calls are allowed than what the link can support. To prevent
this, some type of Call Admission Control must be deployed. How this is accomplished
depends on the environment. If the calls are being sent across intercluster trunks, a gatekeeper is required. If calls are being placed to remote sites that are part of the same cluster, locations are used. If both types of calls are taking place, both solutions must be
deployed.
Locations are objects that are configured within Communications Manager. A location
for each site is created that contains available bandwidth for calls. A closer look at the
configuration of locations is offered later in this chapter. Before looking at locations,
gatekeepers are examined.
Configuring CAC for a Distributed Deployment
A gatekeeper is a process that runs on a Cisco IOS router. It keeps track of the active
calls between clusters and determines whether a call can be placed across an intercluster
trunk. In most cases, only one gatekeeper is needed because each can support more than
a 100 sites. It is recommended, however, to have a redundant gatekeeper. This can be
accomplished by having a second router running Hot Standby Routing Protocol (HSRP)
or by implementing gatekeeper clustering. The only requirement for the physical location
of a gatekeeper is that all clusters must reach it through an IP path.
When a call is placed across a gatekeeper-controlled H.225 or intercluster trunk, the
Communications Manager on the originating side asks the gatekeeper whether the call
can be placed. If there is enough bandwidth, the gatekeeper grants admission. If admission is granted, call setup begins, and the Communications Manager on the other side of
the call must request admission. If the gatekeeper determines that there is enough bandwidth, admission is granted and the call setup is complete.
A gatekeeper grants admission based upon availability of configured bandwidth. The
gatekeeper is configured with the amount of bandwidth that can be used for calls. Each
time a call is placed, the gatekeeper removes a certain amount from the available bandwidth. When the call is over, it returns the bandwidth to the available pool.
The amount of bandwidth required for each call depends on which codec is being used.
The gatekeeper has the preconfigured amount of bandwidth that each codec requires,
and this number cannot be changed. This figure might not be the actual bandwidth the
call needs, but is used to ensure that enough bandwidth is available. A gatekeeper
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running IOS 12.2(2)XA or later assumes that 128 kbps is needed for G7.11 calls and 16
kbps is required for G.729 calls. Although it might seem odd that the gatekeeper might
request more or less bandwidth than it needs, it isn’t a problem because the amount of
available bandwidth is a setting that you configure in the gatekeeper. The gatekeeper
does not have the ability to monitor the link and decide whether there is available bandwidth. It relies totally on the number that is configured. It is best to determine the
amount of calls you want to allow on the link and the codec that will be used. Then simply multiply the amount of bandwidth the gatekeeper uses for that codec by the number
of calls. The result is the amount of bandwidth that should be configured. For example, if
the gatekeeper is running IOS version 12.2(2)XA or later and you want to allow ten calls
all using the G.729 codec, the formula is 10 x 16 (10 calls x 16 kbps), which means that
160 kbps will be needed.
The gatekeeper can also be configured to provide the destination IP address to which the
call should be sent. This feature is sometimes referred to as an anonymous device. An
anonymous device is preferred in many environments, especially those with multiple
intercluster trunks. When more than two clusters are connected, an intercluster trunk
must be created between each cluster if an anonymous device is not used. Figure 5-11
shows that when connecting four clusters, 12 intercluster trunks are required.
Figure 5-11 Intercluster Trunks
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The formula used to determine how many intercluster trunks are required is N x (N–1).
That is the number of clusters times the number of clusters minus 1. In Figure 5-11, there
are four clusters. This means that the total number of intercluster trunks is 4 x (4–1) or 4
x 3, which equals 12. As the number of clusters increases so does the number of required
trunks. For example, with four clusters, 12 intercluster trunks are needed, but with eight
clusters, 56 intercluster trunks are needed. This is where an anonymous device becomes
extremely useful. Instead of creating all the intercluster trunks, just one gatekeeper-controlled intercluster trunk is created, and all calls destined to any of the other clusters are
sent to this trunk. When the gatekeeper responds to an admission request, it also provides the IP address of the destination.
Because the gatekeeper is going to provide destination information, it must know the
destination IP address. This is part of the configuration that must be done on the gatekeeper itself. The bandwidth allowed for calls must also be configured in the gatekeeper
to give you an idea of what needs to be configured. The following example shows a partial configuration:
gatekeeper
zone local DTW bgd.com 10.10.12.28
zone prefix DTW 4...
gw-type-prefix 1#* default-technology
bandwidth total zone DTW 256
no shutdown
A complete explanation of this configuration can be found in the article “Configuring
H.323 Gatekeepers and Proxies” on Cisco.com. However, to give you an idea of what this
configuration is achieving, the third command, “zone prefix DTW 4...,” denotes that calls
in the 4000 range can be handled by the DTW gatekeeper. The fifth command, “bandwidth total zone DTW 256,” means that the total amount of bandwidth available for calls
to and from DTW is 256 kbps.
Warning The gatekeeper should be configured only by an individual who is extremely
knowledgeable of IOS configurations and commands and who thoroughly understands
VoIP technologies. Because the gatekeeper might also be serving other routing functions,
incorrect configuration could negatively affect the network as a whole.
Configuring a Gatekeeper
In addition to the required configuration on the gatekeeper itself, the gatekeeper must
also be configured in the Communications Manager. Adding a gatekeeper in
Communications Manager is quite simple. The following steps show how this is done:
Step 1.
From within CCMAdmin, select Device > Gatekeeper.
Step 2.
Click the Add New link.
Step 3.
A screen similar to that shown in Figure 5-12 displays. Enter the IP address of
the gatekeeper in the Host Name/IP Address field.
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Figure 5-12
Gatekeeper Configuration
Step 4.
In the Description field, enter a description that helps to identify this gatekeeper.
Step 5.
The Registration Request Time to Live field should be left at the default.
Change this field only if the Cisco Technical Assistance Center (TAC) tells you
to do so. This field determines how often the Communications Manager must
send a registration keepalive to the gatekeeper.
Step 6.
The Registration Retry Timeout field should be left at the default. Change
this field only if TAC tells you to do so. This value determines how long
Communications Manager waits before trying to register after a registration
attempt fails.
Step 7.
Typically, the Enable Device check box should be left selected. This allows
the gatekeeper to register with Communications Manager. When you need to
gracefully unregister the gatekeeper, deselect this box.
Step 8.
Click Save.
After a gatekeeper is configured, you must create a gatekeeper-controlled intercluster
trunk so that the calls placed across the intercluster trunk request admission from the
gatekeeper. This also allows you to take advantage of the anonymous device features
if the gatekeeper is configured to provide call-routing information. Creating a
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gatekeeper-controlled intercluster trunk is similar to creating a nongatekeeper-controlled
intercluster trunk.
Configuring a Gatekeeper-Controlled Trunk
The steps required to configure a gatekeeper-controlled H.225 trunk and a gatekeepercontrolled intercluster trunk are similar. The following steps show how to configure a
gatekeeper-controlled intercluster trunk and explain its various settings:
Step 1.
From within CCMAdmin, select Device > Trunk
Step 2.
Click the Add New link.
Step 3.
On the next page, select Inter-Cluster Trunk (Gatekeeper Controlled) from the
Trunk Type drop-down list.
Step 4.
The Device Protocol field can be left at Inter-Cluster Trunk. No other option
is available. Click the Next button.
Step 5.
The Trunk Configuration screen, as shown in Figure 5-13, displays. Enter a
functional name for the gateway in the Device Name field.
Figure 5-13 Trunk Configuration
Step 6.
In the Description field, enter a description that makes this device easily
identifiable.
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Step 7.
From the Device Pool drop-down list, select the desired device pool for
this gateway.
Step 8.
From the Common Device Configuration drop-down list, select the common
device configuration that the trunk will use.
Step 9.
From the Call Classification drop-down list, select whether incoming calls on
this device should be considered OnNet or OffNet. This parameter is used to
determine whether calls can be transferred and forwarded. This is to help prevent fraud.
Step 10. The next field is the Media Resource Group List. It determines the accessibility of media resources to a device. These are discussed further in Chapter 6,
“Configuring CUCM Features and Services.”
Step 11. Information entered in the Location field is used to prevent WAN links from
becoming oversubscribed in centralized deployments. If you have defined
locations, select the appropriate one for this device from the drop-down list.
Step 12. The AAR Group field determines the appropriate association of this device
with an AAR group. An AAR group provides the prefix that is assigned when
a call fails because of insufficient bandwidth. AAR is discussed in further
detail in Chapter 6. Select an AAR group if AAR is being used. If this field is
set to None, AAR is, in effect, disabled on this device.
Step 13. The Tunneled Protocol drop-down list allows you to select Q Signaling
(QSIG), which enables intercluster trunk (ICT) to transport non-H.323 protocol information by tunneling it through H.323. Leave this set to None, unless
you know that this type of tunneling is required.
Step 14. The QSIG Variant parameter is only configurable if QSIG is selected as the
tunnel protocol. Leave this parameter alone unless Cisco TAC instructs you to
change it.
Step 15. The ASN.1 ROSE OID Encoding parameter is only configurable if QSIG is
selected as the tunnel protocol and is beyond the scope of this book.
Step 16. The next two fields, Packet Capture Mode and Packet Capture Duration,
are for troubleshooting purposes only and should not be configured when
adding a new trunk.
Step 17.
The Media Termination Point Required check box needs to be selected if the
H.323 device does not support features such as hold and transfers.
Step 18. If the Retry Video Call as Audio box is selected, Communications Manager
sets up a voice call if a video calls fails to set up.
Step 19. The Path Replacement Support check box is automatically selected if you
select QSIG from the Tunneled Protocol drop-down list. Otherwise it is left
deselected.
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Step 20. If the Transmit UTF-8 for Calling Party Name check box is left deselected, the user locale setting in the device pool will be used to determine
whether Unicode information is sent and translated. Typically this can be
left at the default.
Step 21. The Unattended Port check box is used to indicate that the device has unattended ports. This is normally used if the port is used to send calls to an
application such as a voicemail server. In most cases, this box should be left
deselected.
Step 22. If you want to allow both secure and nonsecure calls on this gateway, you
must select the SRTP Allowed check box. If this is not selected, only nonsecure calls are allowed.
Step 23. If the H.235 Pass Through Allowed check box is selected, the shared-secret
key can pass through a CM, allowing H.323 endpoints to set up a secure
connection.
Step 24. The Use Trusted Relay Point field determines whether a relay point such as a
Media Termination Point (MTP) or a transcoder must be labeled trusted to
be used by this device. This field is typically only changed in virtualized
environments.
Step 25. If the Cisco Intercompany Media Engine feature is being used and calls
through this trunk might reach the PSTN, make sure the that PSTN Access
check box is selected.
Intercompany Media Engine
Step 26. E.164 transformation profiles are used when Intercompany Media Engine
(IME) is used. IME enables different companies to automatically learn routes,
which enables calls to travel across the Internet instead of the PSTN.
Incoming Calling/Called Party Settings
Step 27.
The Incoming Calling Party Settings and Incoming Called Party Settings
are used to globalize numbers. Each calling and called party number has a
number type assigned to it. The incoming calling/called part settings are
based on the number type assigned. There are four number types: national,
international, unknown, and subscriber. The four settings for each of these
number types are
■
Prefix: Digit enters are added to the beginning of the number after the
specified number of strip digits are removed.
■
Strip Digits: This is the number of digits that should be stripped from
the number before the prefix is applied.
■
Calling Search Space: This is the CSS that is used after transformation
has occurred.
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■
Use Device Pool CSS: When this box is selected, the device pool
CSS is used.
If your environment requires the manipulation of incoming called and calling
numbers, configure the appropriate settings for each of these fields.
Step 28. The next two fields define the Multilevel Precedence and Preemption (MLPP)
characteristics of this gateway. If these fields are left blank or set to default,
the values set in the device pool are used. The first MLPP field is the MLPP
Domain. MLPP grants higher priority only from calls with the same MLPP
domain. If MLPP is used, an MLPP domain is needed; otherwise, this field
can be left at None.
Step 29. The second field in this category, MLPP Indication, determines whether tones
and indications will be presented when a precedence call is made. If the is
field set to Off, no precedence indication is presented. If this field is set to
On, indication is used for a precedence call.
Call Routing Information—Inbound Calls
Step 30. The next set of fields refers to inbound calls. The Significant Digits field
determines the number of digits of an incoming dialed number that
Communications Manager uses. Communications Manager counts from right
to left. So if the number entered in this field is 4 and the digits received are
8105559090, 810555 would be removed. Only 9090 would be used to determine the destination of this call.
Step 31. A Calling Search Space (CSS) determines the accessible destinations of
inbound calls. Choose a CSS from the Calling Search Space drop-down list. If
this field is left at None, the dialing privileges of this gateway could be limited.
Step 32. Automated Alternate Routing (AAR) is used to provide an alternate route if a
call fails because of insufficient bandwidth. The AAR CSS can be used to
limit the paths a call can use when it is rerouted. Select an AAR CSS from the
AAR Calling Search Space drop-down list.
Step 33. The Prefix DN field defines what digits are added to the front of an incoming
destination number. This is applied to the number, after Communications
Manager truncates the number, based on the Significant Digits setting.
Step 34. The Redirecting Number IE Delivery–Inbound check box should be selected
if your voicemail system supports redirecting number IE. Otherwise, leave
this box deselected.
Step 35. If the Enable Inbound FastStart check box is selected, FastStart will be used.
H.323 FastStart requires only two message exchanges to open logical channels, whereas normal setup requires 12. However, if FastStart is selected, both
ends must support and be configured for FastStart.
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Call Routing Information—Outbound Calls
Step 36. Called party transformation enables you to change the number that is dialed.
Select the Called Party Transformation CSS that contains the called party
transformation patterns that should be applied to calls routed through the
trunk. You can also leave this set to None and use the Called Party
Transformation CSS assigned to the device pool by selecting the Use Device
Pool Called Party Transformation CSS check box.
Step 37.
Calling party transformation enables you to change the caller ID. Select a
Calling Party Transformation CSS that contains the calling party transformation pattern that is assigned to the device. You can also leave this set to None
and use the Calling Party Transformation CSS assigned to the device pool by
selecting the Use Device Pool Calling Party Transformation CSS check box.
Step 38. The Calling Party Selection field determines what number is sent for outbound calls. The choices are
■
Originator: The directory number of the device that placed the call.
■
First Redirect Number: The directory number of the first device to
redirect the call.
■
Last Redirect Number: The directory number of the last device to
redirect the call.
■
First Redirect Number (External): The external directory number of
the first device to redirect the call.
■
Last Redirect Number (External): The external directory number of
the last device to redirect the call. Select the desired value for this field.
Step 39. The Calling Line ID Presentation field determines whether Communications
Manager sends caller ID information. To send caller ID information, select
Allowed from the drop-down list. To block caller ID, select Restricted from
the drop-down list.
Step 40. Cisco recommends that the next four fields remain set to the default of Cisco
CallManager. The four fields are
■
Called party IE number type unknown
■
Calling party IE number type unknown
■
Called Numbering Plan
■
Calling Numbering Plan
These fields deal with dial plan issues and should be changed only when
advised to do so by Cisco or an experienced dial plan expert. The need to
change these usually occurs when installing Communications Manager
internationally.
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Step 41. The Caller ID DN field is used to determine what caller ID is sent out of this
gateway. A mask or a complete number can be entered in this field. For example, if the mask 55536XX is entered in this field, Communications Manager
sends 55536 and the last two digits of the calling number.
Step 42. If the Display IE Delivery check box is selected, the calling and called party
name information is included in messages.
Step 43. The Redirecting Number IE Delivery-Outbound check box should be selected when integrating with a voicemail system that supports redirecting number
IE. Otherwise, leave it deselected.
Step 44. If the Enable Outbound FastStart check box is selected, FastStart will be
used. H.323 FastStart requires only two message exchanges to open logical
channels, whereas normal setup requires 12. However, if FastStart is selected,
both ends must support and be configured for FastStart.
Step 45. If the Enable Outbound FastStart check box is selected, you must select the
codec that is to be used. This is selected from the Codec for Outbound
FastStart drop-down list.
Gatekeeper Information
Step 46. From the Gatekeeper Name drop-down list, select the desired gatekeeper.
Step 47.
The Terminal Type field specifies the type of devices this trunk controls.
Choose Gateway for normal trunks.
Step 48. The Technology Prefix field enables you to assign a prefix that matches the
prefix in the gatekeeper. By assigning a matching prefix, you can avoid having
to add the IP address of each Communications Manager in the gatekeeper on
the gw-type-prefix line. It is recommended that you use 1#* in both this field
and the gatekeeper configuration. The value entered in this field must exactly
match what is configured in the gatekeeper.
Step 49. The Zone field determines which zone this Communications Manager registers with on the gatekeeper. If this field is left blank, the gatekeeper’s zone
subnet command is used to determine to what zone the Communications
Manager registers. If you enter a zone name in this field, it must match exactly
with what is configured in the gatekeeper (this includes capitalization).
Geolocation Configuration
Step 50. The geolocation information can be used to determine the logical partition of
a device. If you are using the geolocation feature, select the appropriate
geolocation from the Geolocation drop-down list.
Step 51. There are 17 configurable geolocation fields. Geolocation filters enable you to
choose which fields are used to create a geolocation identifier. If you use the
geolocation feature, select the appropriate geolocation filter from the
Geolocation Filter drop-down list.
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Step 52. Click Save.
After the gatekeeper-controlled intercluster trunk is configured, you can add it to a route
group. Then configure a pattern that matches calls that should be routed over this trunk.
The pattern should point to a route list that contains the route group of which this trunk
is a member.
Configuring CAC for a Centralized Deployment
To accomplish CAC for environments that have remote sites, locations are configured in
Communications Manager. Locations define the amount of bandwidth that can be used
to place calls to and from the remote sites. After locations are configured, they must be
assigned to devices such as phones, trunks, and gateways. You can accomplish this by
assigning them to a device pool. This process enables the phones, trunks, and gateways’
device pool to determine the location. When a call is placed across the IP WAN,
Communications Manager uses the location information to determine whether there is
enough available bandwidth for the call. By deducting available bandwidth for each call
that is active on the WAN, Communications Manager can determine availability. When
using locations, Communications Manager assumes that the following bandwidth is
required for each codec:
■
A G.711 call uses 80 kbps.
■
A G.722 call uses 80 kbps.
■
A G.723 call uses 24 kbps.
■
A G.728 call uses 16 kbps.
■
A G.729 call uses 24 kbps.
■
A GSM call uses 29 kbps.
■
A wideband call uses 272 kbps.
To better understand locations, now look at the steps required to create and apply them.
Creating Locations
The following steps show how to create a location:
Step 1.
From within CCMAdmin, select System > Location.
Step 2.
Click the Add New link.
Step 3.
A screen similar to that shown in Figure 5-14 displays. Enter the name of the
location in the Name field.
Step 4.
In the Audio Bandwidth field, enter the amount of bandwidth available for
voice calls to and from this location. If you select the Unlimited radio button,
no limit is placed on voice calls. To determine the value to enter here, take the
bandwidth that Communications Manager used for each call based on the
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Figure 5-14
Location Configuration
codec that is being employed, and multiply it by the number of calls that you
know can safely traverse the link. For example, if you use G.729 and you
know that ten calls can traverse the link, multiply 24 kbps by 10. This tells you
that 240 should be entered in this field. The bandwidth that Communications
Manager assumes for each codec is listed earlier in this section.
Step 5.
In the Video Bandwidth field, enter the amount of bandwidth available for
video calls to and from this location. If you select the Unlimited radio button,
no limit is placed on video calls. You can also select the None radio button,
which prohibits video calls.
Note Video bandwidth is in increments of the video codec. If video codec is 384 and
two video calls are allowed, the bandwidth should be set to 768.
Step 6.
Resource Reservation Protocol (RSVP) can be used to reserve bandwidth for
calls. The RSVP settings can be determined by highlighting a location in the
Modify Setting(s) to Other Locations section and selecting the desired RSVP
setting from the RSVP Setting drop-down list.
Step 7.
Click the Save. The location has been added when the status line reads Insert
Completed.
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Assigning a Location to Devices
After locations are added, you must assign them to devices. As stated earlier, it is recommended that the location information be assigned to the device pool. The following steps
show how to assign a location to a device pool:
Step 1.
From within CCMAdmin, navigate to System > Device Pool.
Step 2.
To limit the results, enter search criteria in the search field, and click Find.
Step 3.
Select the device pool to which you want to assign a location from the list
that is generated.
Step 4.
Select a location from the Location drop-down list.
Step 5.
Click Save.
Step 6.
The device pool must be reset. Click Reset.
Step 7.
A new window will appear; click Reset in the new window.
Step 8.
Close the Device Reset window.
If you want to assign the location at the device level, follow these steps. The steps to add
a location to a phone, trunk, or gateway are all similar. The following steps can be used to
add a location to any of these devices:
Step 1.
The path you select from within CCMAdmin depends on which type of
device you are assigning a location. To assign a location to a phone, select
Device > Phone. To assign a location to an ICT, select Device > Trunk. To
assign a location to a gateway, select Device > Gateway.
Step 2.
To limit the results, enter search criteria in the search field, and click Find.
Step 3.
Select the device to which you want to assign a location from the list that is
generated.
Step 4.
If configuring an MGCP gateway, select the endpoint to which you want to
assign a location. If you are not configuring an MGCP gateway, skip this step.
Step 5.
The Device Configuration screen displays. Select a location from the
Location drop-down list, as shown in Figure 5-15. Figure 5-15 shows a phone
configuration screen, but the screen should be similar regardless of the device
that you are configuring.
Step 6.
Click Save.
Step 7.
A window displays informing you that you must click the Apply Config button for the change to take affect. Click OK.
Step 8.
Click Apply Config.
Step 9.
A window displays warning you that when you apply the configuration, the
device might go through a restart. Click OK.
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Figure 5-15
Assigning a Location to a Device
That’s all there is to it. Unlike gatekeeper, no additional configuration is required outside
of Communications Manager. Communications Manager handles all the CAC functions
itself when locations are used for remote sites.
Special Services Configuration
There are certain types of calls that should always be given priority and availability to be
dialed from all phones. The first call of this type is 911. When a 911 call is placed, it is
important that the call gets through. Not only is it necessary to make the call possible,
but you also need to ensure that it goes to the right destination. The following sections
discuss some of the issues that can arise with these services.
Special Services Overview
Depending on your local service, various special services might be available. For example, the following is a list of special service numbers that are commonly available in
North America. Check with your local phone company to see which of these are valid in
your area.
■
311: Nonemergency police services
■
411: Directory assistance
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■
511: Travel information
■
611: Phone equipment repair
■
711: Telecommunications Device for the Deaf (TDD) operator
■
911: Emergency
After you determine which services are available, you must configure route patterns that
will match these calls. The most important of these calls is 911. Because in an emergency, a
person might not think to dial 9 before dialing 911; patterns should be created that enable
the call to go out regardless of whether 9 is dialed first. This means that two patterns need
to be created, 911 and 9.911. PreDot discard instructions must be applied to the 9.911 pattern so that only 911 is sent out the public switched telephone network (PSTN).
When there are remote locations, things become a little more complicated. Imagine that
you have an office in San Jose and a remote office in San Francisco. When callers dial
911 from San Francisco, the call must be routed to the local emergency service, not the
service in San Jose. Although this seems obvious, it is sometimes overlooked. To accomplish this, multiple 911 and 9.911 patterns must be created. Partitions and CSS are used
to allow phones in each location to match only the pattern that routes the call to the correct location.
For all other special services, the 9.X11 pattern should be sufficient. Once again, be sure
to create patterns for each remote location so that the call is routed to the local PSTN.
Another concern when dealing with 911 calls is that some local legislation requires that
more detailed location information be sent than just the street address. These laws normally apply to buildings that are over a certain size. Typically the floor and room number
are required in addition to the street address. This requirement is referred to as an E911 or
enhanced 911. Imagine that someone dialed 911 from a 20-story building and all that was
sent was the street address. This would make it difficult to determine which floor, let
alone which office, it came from. The solution is to have a database that contains the
detailed address information for each phone number in your company. This database is
typically maintained by an outside company and is accessible by the emergency service.
Another issue that arises with Communications Manager is that because a phone can be
moved so easily, the information in the database can become outdated rather quickly. In
addition to this, a feature known as extension mobility makes the Communications
Manager system even more nomadic. To deal with these issues, Cisco offers an
Emergency Responder product. This product ensures that the correct detailed information is sent when a 911 call is placed. For more details on this product, refer to the “Cisco
Emergency Responder Administration Guide” on Cisco.com.
Configuring Special Services Route Patterns
To ensure that special services numbers are accessible, you must create route patterns for
them. As mentioned previously, it is recommended that you create at least three patterns
for each location. The first two are for 911 services and should be 911 and 9.911. If your
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location does not use a leading 9 for PSTN access, the first 9 in the 9.911 pattern should
be replaced with whatever number is used for PSTN access. The third pattern is 9.X11.
This pattern will match all other special services numbers.
The 9.911 and 911 patterns should be marked Urgent Priority so that as soon as the number is dialed, it is sent. If this pattern is not marked Urgent Priority, delays could occur
before the call is sent, and this should never happen.
As often happens, one solution creates another problem. I have heard people say that
they do not use the 911 pattern because people often dial it by mistake. What happens is
that a person dials 9 for an outside line, then presses one to begin a long distance call,
and then mistakenly presses one again. This, of course, matches 911 and routes the call to
emergency services. It is never recommended that you not include the 911 pattern.
Although people misdialing 911 is problematic, it is gravely problematic if 911 cannot be
dialed during an emergency. I have heard of many ways people have fixed this problem,
but I would not recommend any of them because they all result in either the failure or
delay of the call.
An overview of the tasks required to create patterns to allow access to special services
numbers follows. Refer to Chapter 4, “Implementing a Route Plan,” for detailed steps on
how to create route patterns.
Step 1.
Create a 911 route pattern.
Step 2.
Assign a partition to this pattern that all phones in the location can dial. If
there are remote locations, a separate pattern must be created and placed in a
partition that only phones in that location can reach. This pattern must then
point to a route list that will send the call out the local PSTN. Figure 5-16
shows an example of this.
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Chicago
Public Safety
San Jose
Public Safety
PSTN
San Jose
Chicago
V
V
Chicago CSS
San Jose CSS
911 (Chicago 911 Partition)
9.911 (Chicago 911 Partition)
911 (SanJose 911 Partition)
9.911 (SanJose 911 Partition)
Figure 5-16 Routing 911 Calls for Multiple Locations
Warning When configuring 911 patterns in environments that include remote locations,
it is imperative that you create a route plan that will send the call to the proper 911 operator. For example, if the Communications Manager cluster is in Chicago and a remote location is in San Jose, you must make certain that when a user in San Jose dials 911, the call
reaches the San Jose 911 operator, not the Chicago 911 operator. The responsibility to
ensure that this works properly is solely upon you. If you are uncertain how to properly
implement this, you should seek additional help from an individual with proven experience
in this field.
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Step 3.
Select a gateway or route list that will send this pattern out the local PSTN
gateway.
Step 4.
Select the Urgent Priority and OffNet Pattern (and Outside Dial Tone)
check boxes.
Note Although it is not necessary to provide an outside dial tone for this pattern, if you
do not, the outside dial tone will not be played for any number that begins with 9, until
enough digits are dialed so that the number does not match 911. That means, in most
cases, when dialing a long-distance number, the outside dial tone is not heard until three
digits are dialed. For this reason, it is recommended that you select the OffNet Pattern (and
Outside Dial Tone) check box.
Step 5.
Create a 9.911 route pattern. If your location does not use a leading 9 for
PSTN access, the first 9 in the 9.911 pattern should be replaced with whatever
number is used for PSTN access.
Step 6.
Assign a partition to this pattern that all phones in the location can dial. If
there are remote locations, a separate pattern must be created and placed in a
partition that only phones in that location can reach. The pattern must then
point to a route list that will send the call out the local PSTN. Figure 5-16
shows an example of this.
Warning When configuring 9.911 patterns in environments that include remote locations, it is imperative that you properly create a route plan that will send the call to the
proper 911 operator. For example, if the Communications Manager cluster is in Chicago
and a remote location is in San Jose, you must make certain that when a user in San Jose
dials 9.911, the call reaches the San Jose 911 operator, not the Chicago 911 operator. The
responsibility to ensure that this works properly is solely upon you. If you are uncertain as
to how to properly implement this, you should seek additional help from an individual with
proven experience in this field.
Warning Special emergency call routing considerations must be made when deploying
device mobility or extension mobility. Since device mobility and extension mobility allow
the user to change physical locations you must ensure that the call will be routed based on
the physical location of the user. The techniques used to ensure this are beyond the scope
of this book. If you are unsure as to how this should be configured you should enlist the
assistance of a experienced Cisco voice engineer. The responsibility to ensure all call routing, which includes emergency call routing, works properly is solely upon you.
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Step 7.
Select a gateway or route list that will send the pattern out the local PSTN
gateway.
Step 8.
Select the Urgent Priority and OffNet Pattern (and Outside Dial Tone)
check boxes.
Step 9.
Set the discard digits to PreDot.
Step 10. Create a 9.X11 route pattern.
Step 11. Assign a partition to the pattern that all phones in the location can dial. If
there are remote locations, a separate pattern must be created and placed in a
partition that only phones in that location can reach. The pattern must then
point to a route list that can send the call out the local PSTN.
Step 12. Select a gateway or route list that will send the pattern out the local PSTN
gateway.
Step 13. Set the discard digits to PreDot.
Step 14. Select the OffNet Pattern (and Outside Dial Tone) check box.
It is essential that after you have created patterns for these services, you make test calls
to ensure that the call is routed properly. The steps provided previously are only general
practices; additional configuration might be required. There is no guarantee that the previous steps will work in each situation. It is your responsibility to make sure that you test
these services thoroughly before the system goes live.
Summary
This chapter explored how certain calls can be restricted by applying CSS and partitions
to devices and patterns. Because it is often required that different devices have access to
various destinations, the steps for creating and applying CSS and partitions are provided.
When deploying VoIP solutions, ensuring the quality of the call is essential. To accomplish this, CAC was discussed. Detailed steps were provided that show how to configure
a gatekeeper that provides CAC for calls between clusters. Steps were also included to
show how to configure locations for CAC, for calls to and from remote sites.
Finally, special services, such as 911, were discussed in this chapter. An overview of the
required steps for the proper configuration of these services was given.
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Chapter 6
Configuring CUCM
Features and Services
Communications Manager provides a number of features that users have come to expect,
and each new version of Communications Manager adds new features. Many features add
needed functionality, such as conference calls, whereas others simply help to make the
phone system more enjoyable, such as Music On Hold (MoH). Years ago, Communications
Manager was not as feature rich as other traditional systems. Lately Communications
Manager has begun to provide features that go beyond those offered by traditional systems. This chapter examines a wide range of features and the configuration tasks required
to implement them.
Configuring Features
You start by looking at some of the basic features that are most often implemented.
These are features that offer users extended functionality and are found on most modern
phone systems.
As one might expect, all these features are configured in CCMAdmin. Each of the following six sections introduce you to a feature, explain its function, and show you how to
configure these features.
Many of the features available to the users do not require any additional configuration.
Because configuration is not required, these features are not discussed in detail in this
chapter. The features that do not require configuration include Hold, Call Waiting, Mute,
Transfer, On-Hook Dialing, and Redial.
Creating Call Pickup Groups
Call pickup groups enable people to answer a line that is ringing on another phone from
their phone. For example, if extension 1005 is ringing, the call can be answered from a
phone that does not have 1005 as an extension. This can occur two ways. The first is
called call pickup, which enables a call to be picked up from another phone if both
phones are in the same pickup group. The second type of call pickup is called group call
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pickup, which enables a call to be picked up from another phone if both phones are not in
the same pickup group. To help clarify this, Figure 6-1 shows three phones. Phones A and
B are in the same call pickup group, whereas phone C is in a separate call pickup group.
Phone A
Pickup Group = 2800
Pickup phone B by
pressing the "pickup"
softkey.
Picks up phone C by
pressing the "Gpickup"
softkey and dialing 2850.
Phone B
Pickup Group = 2800
Phone C
Pickup Group = 2850
Figure 6-1 Call Pickup Example
If phone B is ringing, the call can be answered from phone A by going off-hook and
pressing the pickup softkey. After the softkey is pressed, phone A will start ringing, and
the call can be answered. However, if phone C is ringing and a user wants to answer the
call from phone A, the user would have to go off-hook, press the Gpickup softkey, and
enter phone C’s group pickup number. After this is done, Phone A starts to ring and the
call can be answered.
There are a couple of things to be aware of when deploying the call pickup feature. First,
a phone must be a member of a call pickup group to use this feature. If a phone has not
been assigned to a call pickup group, that phone cannot answer another phone’s incoming call. Second, for a user to answer a call ringing on a phone that is not in the same call
pickup group, the user must know the extension number. In addition, if the call pickup
number has a partition assigned to it, the phone or line must have a Calling Search Space
(CSS) with access to that partition.
To implement the call pickup feature, call pickup groups must be created and then
assigned to the phones. The following steps show how to perform both of these tasks.
Add a Call Pickup Number
Follow these steps to add a call pickup number:
Step 1.
From within CCMAdmin, select Call Routing > Call Pickup Group.
Step 2.
Click the Add New link.
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Step 3.
A screen similar to that shown in Figure 6-2 displays. Enter a name in the Call
Pickup Group Name field. Make sure that it is descriptive because the same
name is used when assigning a pickup group to a line.
Figure 6-2 Call Pickup Group Configuration.
Step 4.
In the Call Pickup Group Number field, enter the desired number. This number must be unique, in that it is not assigned to any other function or device.
Step 5.
Enter a description that helps identify the purpose of this call pickup group
in the Description field.
Step 6.
If you want to assign a partition to this call pickup group, select one from the
Partition drop-down list.
An alert can be sent to all the phones in a group if any phone in the group
does not answer an incoming call after a specified period of time. If you do
not want this configured, skip to Step 10.
Step 7.
To enable call pickup group notification, select the type of alert you want to
have sent from the Call Pickup Group Notification Policy drop-down list.
Step 8.
Next, in the Call Pickup Group Notification Timer field, enter the number of
seconds that should pass before the alert is sent.
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Step 9.
To include the calling and/or called party information in the visible alert,
select the appropriate check boxes in the Call Information Display for Call
Pickup Group Notification area.
Step 10. Click the Save button to add this call pickup group.
Assign a Call Pickup Group to a Line
Follow these steps to assign a call pickup group to a line:
Step 1.
From within CCMAdmin, Select Device > Phone.
Step 2.
Enter search criteria in the search field to limit the results and click the Find
button.
Step 3.
Select the desired phone from the list that displays.
Step 4.
Click the desired line on the left side of the screen.
Step 5.
On the Directory Number Configuration page, select the desired call pickup
group from the Call Pickup Group drop-down list, as seen in Figure 6-3.
Figure 6-3 Assigning a Call Pickup Group to a Line
Step 6.
Click Save at the top of the screen.
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After call pickup groups are created and assigned to lines, this feature can be used. A
common mistake that users make when trying to use this feature is that they press the
pickup softkey without going off-hook. This softkey is not available until the phone is
off-hook. After off-hook, the pickup softkey can be pressed and the call starts to ring on
the phone.
Note Because call pickup groups are assigned to lines, a different call pickup group can
be assigned to line 1, then to line 2, and so on. One variation that is successful for some
customers is having different pickup groups assigned to different lines instead of using
group call pickup (the Gpickup softkey). This way, the user need only remember which line
belongs to which pickup group instead of having to memorize the pickup group number.
Creating Meet-Me Patterns
Communications Manager supports two types of conference calls. The first is called an
ad hoc conference call. An ad hoc conference call is the type that takes place when two
people who are having a phone conversation decide to conference in another person. One
of the active callers initiates the conference call by pressing the Conference button, which
is the softkey labeled Confrn on the phone, and calling the third party.
The other type of conference call is called a meet-me. A meet-me conference call is
scheduled ahead of time, and all participants are told to call a certain number to join the
conference.
Both types of conference calls require some type of conference resources, which are
referred to as conference bridges. Communications Manager supports two types of conference bridges: hardware and software. When Communications Manager is installed, a
software conference bridge is installed on Communications Manager when the Cisco IP
Voice Media Streaming Application is activated. Configuring hardware conference
bridges is discussed later in this chapter.
Ad hoc conference calls require no configuration if a conference bridge is available. By
default, the Conference button is available on each of the standard softkey templates. By
pressing this key, an ad hoc conference can be established. You can limit the number of
participants allowed in a conference with the use of settings in the Communications
Manager services parameters.
Meet-me conferences require more configuration than just having a conference bridge
available. A meet-me pattern must also be created. This is the number that participants
will use for the call. Each active meet-me conference requires a unique number. This
means that the meet-me patterns you create will determine the maximum amount of
meet-me conferences that can take place at one time.
Meet-me patterns enable the use of wildcards, which means that you can create one pattern that will result in multiple meet-me numbers. For example, if the pattern 523X is
used, it defines ten numbers as meet-me conference numbers, 5230 through 5239.
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Creating meet-me patterns is accomplished through CCMAdmin. The following steps
take you through this process:
Step 1.
From within CCMAdmin, select Call Routing > Meet-Me Number/Pattern.
Step 2.
Click the Add New link.
Step 3.
A screen similar that shown in Figure 6-4 displays. In the Directory Number
or Pattern field, enter the number or pattern that you want to use as a meetme number.
Figure 6-4 Creating a Meet-Me Pattern
Step 4.
In the Description field, enter a description that helps identify the purpose
of the meet-me pattern.
Step 5.
To allow only certain people to join a conference, assign partitions to the pattern. This prevents callers who do not have access to this partition from establishing or joining a conference using this number. This ensures that unauthorized callers can not accidentally join a conference call. If you want to restrict
the callers who can reach this meet-me number, select the desired partition
from the Partition drop-down list. To allow all devices access to this meet-me
number, select None for the partition.
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Step 6.
Step 7.
The Minimum Security Level field determines whether nonsecured phones are
able to join the conference. Select one of the following options from the
Minimum Security Level drop-down list:
■
Authenticated: Prevents nonsecure phones
■
Encrypted: Prevents authenticated and nonsecure phones
■
Non Secure: Allows all phones
Click the Save button to create the meet-me number.
After the meet-me numbers are created, users can start to use this feature. To initiate a
meet-me conference, the Meet-Me button must be pressed. All standard softkey templates include the Meet-Me softkey. Limiting access to specific meet-me numbers by
using a partition was discussed previously. One way to restrict a device from initiating
meet-me conferences is by assigning it a softkey template that does not have a Meet-Me
softkey. Phones without a Meet-Me softkey can still join a meet-me conference by simply
dialing the meet-me number after the meet-me conference is active.
Note The most common error made with meet-me conference calls is when the user
presses the Meet-Me button when attempting to join an active conference call. Make sure
that users understand that the Meet-Me button is only used to initiate conference calls.
Creating Call Park Numbers
Calls can be placed on hold by pressing the Hold button; however, the call can only be
picked up from the same line. This means that when a call is placed on hold, it can only
be retrieved from a phone with the same directory number from which it was placed on
hold. Often, a call placed on hold needs to be retrieved on a different phone. To accomplish this, the Call Park feature is used.
Call Park places the call on hold and parks it on a virtual directory number. By dialing
that number, you can retrieve the call from a different phone. A common use for Call
Park is to have an attendant or operator park the call and then announce the call using an
overhead paging system.
For Call Park to work, call park numbers must be configured. As with meet-me numbers, wildcards are allowed. This means that a single pattern can result in multiple call
park numbers.
Call Park patterns are created through CCMAdmin. The following steps take you through
this process.
Step 1.
From within CCMAdmin, select Call Routing > Call Park.
Step 2.
Click the Add New link.
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Step 3.
A screen similar to that shown in Figure 6-5 displays. In the Call Park
Number/Range field, enter the number or pattern that you want to use as a
call park number.
Figure 6-5 Creating a Call Park Pattern
Step 4.
In the Description field, enter a description that helps identify the purpose
of the call park pattern.
Step 5.
To restrict the retrieval of a parked call, a partition can be assigned to the call
park number. Only devices that have access to the assigned partition can then
retrieve the parked call. To restrict access to this call park number, select the
desired partition from the Partition drop-down list. To allow all devices
access to this call park number, select None for the partition.
Step 6.
From the Cisco Unified Communications Manager drop-down list, select the
pattern in Communications Manager that registers with the call park.
Note For a call to be parked, call park numbers must be configured for phones that are
registered to the Communications Managers. For this reason, it is recommended that all
Communications Managers employ this process. This means that you must also configure
call park numbers for backup Communications Managers, so that the Call Park feature is
available if the primary Communications Manager fails.
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Step 7.
Click the Save button to create the call park number.
After call park numbers are created, the feature is available for use. It is important that
users understand how this feature works. This feature is activated when the Park softkey
is pressed. The call is then automatically parked. The phone displays the number at which
the call is parked. Often users think that they choose the location of the parked call, but
actually Communications Manager chooses the number automatically based on the available call park numbers. This means that the person parking the call must look at the
phone display to see the number where the call is parked. By default, a call only stays
“parked” for 60 seconds before it reverts to the phone that originated the park. This value
can be modified through the call park reversion timer found in the Communications
Manager Service parameters.
Creating Directed Call Park Numbers
Often users prefer to select the number that the call will be parked on. For this reason,
more recent versions of Communications Manager have added the Directed Call Park feature. To use this feature, the user transfers a call to the directed call park number by
pressing the Transfer softkey and the directed call park number that they want the call
parked at.
It is important to understand that directed call park numbers are different than call park
numbers and must be configured separately.
Directed call park patterns are created through CCMAdmin. The following steps take you
through this process:
Step 1.
From within CCMAdmin, select Call Routing > Call Directed Park.
Step 2.
Click the Add New link.
Step 3.
A screen similar to that shown in Figure 6-6 displays. In the Number field,
enter the number or pattern that you want to use as a call park number.
Step 4.
In the Description field, enter a description that helps identify the purpose
of this call park pattern.
Step 5.
To restrict who can retrieve a parked call, a partition can be assigned to the
call park number. Only devices that have access to the assigned partition can
then retrieve the parked call. To restrict access to this call park number, select
the desired partition from the Partition drop-down list. To allow all devices
access to this call park number, select None for the partition.
Step 6.
In the Reversion Number field, enter the extension that the parked call should
be forwarded to if it is not retrieved.
Step 7.
Select a Calling Search Space (CSS) from the Reversion Calling Search Space
drop-down list. Make sure to select a CSS that has access to the partition that
the reverted number is in.
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Figure 6-6 Creating a Directed Call Park Pattern
Step 8.
The Retrieval Prefix field is required so that Communications Manager
knows that the user is trying to retrieve a directed call park and not a standard call park. When retrieving a call that was parked using Directed Call
Park, the user enters this prefix first, followed by the number at which the call
was parked. Enter the desired prefix in the Retrieval Prefix field.
Step 9.
Click the Save button.
Creating Intercoms
In earlier versions of Communications Manager a true intercom function was not available. There was a work around for it which used speed dials and the auto-answer function
of certain phones. In more recent versions, a true intercom feature has been added. It
allows a user to press a button that is configured as an intercom. When pressed, a one
way audio stream is setup to another phone. The user at the phone can then accept the
call and a two way stream is setup.
There are basically four sets required to setup an intercom. These steps are
■
Create intercom partitions
■
Verify the automatically created intercom CSS
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■
Create intercom numbers
■
Assign intercom numbers
Creating Intercom Partitions
Intercom partitions are similar to standard partitions. They are assigned to intercom
directory numbers. The calling intercom line must have an intercom CSS that allows
access to the intercom partition assigned to the destination intercom directory number.
Creation of intercom partitions is very similar to the creation of standard partitions. The
following steps show how to create intercom partitions:
Step 1.
From within CCMAdmin, select Call Routing > Intercom > Intercom Route
Partition.
Step 2.
Click the Add New link.
Step 3.
A screen displays that offers an area in which you can enter the name of the
partition followed by a description. You must place a comma (,) between the
name and description. If you do not enter a description, the name of the partition will be used as the description. You can create up to 75 partitions at a
time on this screen by placing each on a new line.
Step 4.
After you have entered all the desired partitions, click the Save button.
Intercom Calling Search Spaces
An intercom CSS is similar to a standard CSS. It determines which intercom partitions an
intercom directory number has access to. When you create an intercom partition, an intercom CSS is automatically created. The name that is assigned to it is the same as the partition’s name with “_GEN” added to the end. For example, if you create an intercom partition and name it Sales_Intercom, an intercom CSS named Sales_Intercom_GEN is created.
Note Automatically generated calling search spaces can be modified, this may be
required if an intercom directory number needs to dial multiple intercom numbers.
Creating Intercom Numbers
Intercom DNs are similar to directory numbers as they are assigned to phone buttons.
They differ in the fact that normally they are not used to place calls to any destination
other than other intercom directory numbers. Intercom DN may be created two ways.
The first is to simply add it when you assign it to a phone. The other is to create them
under the call routing menu. The second way allows you to create multiple intercoms at
once which can save time since the parameters only have to be configured once. The following steps show how to configure a range of intercom DNs
Step 1.
From within CCMAdmin, select Call Routing > Intercom>Intercom
Directory Number.
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Step 2.
Click the Add New link.
Step 3.
To create a range on intercom DNs, enter a starting directory number in the
Intercom Directory Number field and then an ending number in the next field
to the right. By doing this you are able to add multiple intercom DNs at once.
Note You can create a single intercom number by entering the same DN in both fields.
Step 4.
The Route Partition field defines the intercom partition to which this intercom directory number is assigned. The partition is used to determine the
intercoms DNs that can reach this intercom. Select the desired partition.
Partitions are discussed in more detail in Chapter 5.
Step 5.
The next field is the Description field. Enter a description that will help you
quickly identify the intercom DN later.
Step 6.
In the Alerting Name field, enter the name that should be displayed on the
caller’s phone when this number is dialed and the phone is ringing.
Step 7.
The ASCII Alerting Name field is used on devices that do not support
Unicode characters.
Step 8.
The next field allows a CSS to be assigned to the intercom DN. This determines what intercom DNs can be reached when calling from this line. Select
Calling Search Space from the drop-down list.
Step 9.
The Presence group that an intercom DN belongs to determines what devices
are allowed to monitor this DN. Select the Presence Group this DN should
belong to.
Step 10. The Auto Answer field determines whether the speakerphone or headset is
used when the intercom uses auto answered. This field is typically set to Auto
Answer with Speakerphone.
Step 11. The Default Activated Device field defines the device that this intercom DN
is active on. If you are creating a range of intercom DNs leave this field blank
and configure it when you assign the intercom DN to a phone. If you are creating a single intercom DN select the desired device from the drop-down list.
Step 12. Click Save.
Once the intercom DNs are created they must be assigned to phones.
Assigning an Intercom DN to a Phone
The steps required to add an intercom DN to a phone are similar to those used to assign a
DN to a phone but there are fewer parameters that must be configured. The following
steps show how to do this.
Step 1.
From within CCMAdmin, Select Device > Phone.
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Step 2.
Enter search criteria in the search field to limit the results and click the Find
button.
Step 3.
Select the desired phone from the list that displays.
Step 4.
Click the Add a new Intercom on the left side of the screen.
Note If none of the buttons are configured as an intercom button, you need to create a
new phone button template that has an intercom button and assign it to the phone.
Step 5.
Enter the desired intercom DN in the Intercom Directory Number field.
Step 6.
The Route Partition field defines the intercom partition to which this intercom directory number is assigned. The partition is used to determine the
intercoms DNs that can reach this intercom. If an intercom DN that has
already been created is assigned, the partition that was defined when it was
created it is automatically assigned. Partitions are discussed in more detail in
Chapter 5, “Configuring Class of Service and Call Admission Control.”
Step 7.
The next field is the Description field. Enter a description that will help you
quickly identify the intercom DN later.
Step 8.
In the Alerting Name field, enter the name that should be displayed on the
caller’s phone when this number is dialed and the phone is ringing.
Step 9.
The ASCII Alerting Name field is used on devices that do not support
Unicode characters.
Step 10. The next three fields, calling search space, presence group and auto-answered
are automatically populated using the parameters that were assigned to the
intercom DN when it was created. In most cased these fields should not need
to be changed.
Step 11. The Default Activated Device field defines the device that this intercom DN
is active on. Select the device this DN is assigned to from the drop-down list.
Step 12. The next two fields, labeled Display (Internal Call ID) and ASCII Display
(Internal Call ID), are used to configure what caller ID is displayed when calls
to other internal callers are connected. Enter up to 30 characters in this field.
Both letters and numbers are allowed.
Step 13. The next two fields, which are labeled Line Text Label and ASCII Line Text
Label, are used to define how the line displays on the phone. Enter that label
in this field that you want displayed for this intercom DN.
Step 14. In the Speed Dial field enter the number that is automatically dialed when the
intercom button is pressed. This is used when configuring an intercom that is
only used between two phones, such as an executive and assistant. If this field
is left blank the intercom line can be used to call multiple intercom numbers.
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Step 15. The External Phone Number Mask can be set to the main number of the
phone.
Step 16. The Forwarded Call Information Display section determines what information is sent when a call is forwarded. Select the information to be sent by
selecting the check box next to each desired field.
Step 17.
Click Save.
Creating Forced Authorization Codes
Earlier in this book, you learned how CSS and partitions are used to allow and restrict
calls from being placed to certain destinations. The problem with CSS and partitions is
that they are based on the device from which the call is being placed, not from the person who is making the call. In new versions of Communications Manager, you can use
forced authorization codes (FAC) to allow calls to be placed based on a code that is
entered. This means that the call is permitted or prohibited based on who is calling and
not the device from which they are calling.
It works this way: When a call is placed that requires an authorization code, a double
beep or prompt is heard that alerts the caller to enter the code. After the code is entered,
the call is completed.
Enabling FACs requires that FACs exist and that the route patterns that are to be restricted are configured to require a FAC. The following steps show how to accomplish both of
these tasks.
Create a Forced Authorization Code
Follow these steps to create a FAC:
Step 1.
From within CCMAdmin, select Call Routing > Forced Authorization Codes.
Step 2.
Click the Add New link.
Step 3.
A screen similar to that shown in Figure 6-7 displays. In the Authorization
Code Name field, enter a name for this code. Choose a name that helps identify the code. This name displays in authorization code reports that can be
produced.
Step 4.
In the Authorization Code field, enter a numeric code. This is the code a
caller will have to enter to place calls that require authorization.
Step 5.
In the Authorization Level field, enter a numeric value between 0 and 255.
This value determines the authorization level of this code. For the code to
allow a call, the value in this field must be equal to or greater than the value
assigned to the pattern.
Step 6.
Click the Save button to add this code. After the code is created, the status
line displays “Add successful.”
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Figure 6-7
Creating Forced Authorization Codes
Assign a Forced Authorization Code to a Route Pattern
Follow these steps to assign a FAC to a route pattern:
Step 1.
From within CCMAdmin, select Call Routing > Route/Hunt > Route Pattern.
Step 2.
Enter search criteria in the search field to limit the results, and click the
Find button.
Step 3.
From the list that displays, select the route pattern to which you want to
add a FAC.
Step 4.
A screen similar to that shown in Figure 6-8 displays. Select the Require
Forced Authorization Code check box, and enter a numeric value in the
Authorization Level field. This value is used to determine which FACs have
permission to use this pattern. A FAC must have a value equal to or greater
than this value to be granted access.
Step 5.
Click the Save button to apply the FAC to this pattern.
Step 6.
Repeat these steps for all other route patterns on which you want to enable a
FAC.
After completing these two tasks, FAC is enabled. Make sure that you take the time to
properly plan the implementation of FAC. If it is not properly implemented, it can cause
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Figure 6-8 Require a Forced Authorization Code for a Route Pattern
adverse effects. Make certain that FAC is never applied to a pattern that allows access to
emergency services such as 911.
Configuring Client Matter Codes
Another commonly requested feature is client matter codes (CMC). This enables a caller
to enter a client code while placing a call so that the call is associated with a client.
Companies that bill customers for time spent on projects, such as lawyer firms, require
this feature.
The process to enable CMCs is similar to that of configuring FACs. First, CMCs must be
created and then route patterns need to be configured to require CMCs. The following
steps walk you through both processes.
Create a Client Matter Code
Follow these steps to create a CMC:
Step 1.
From within CCMAdmin, select Call Routing > Client Matter Code.
Step 2.
Click the Add New link.
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Step 3.
A screen similar to that shown in Figure 6-9 displays. In the Client Matter
Code field, enter the desired code.
Figure 6-9 Creating Client Matter Codes
Step 4.
In the Description field, enter a description that identifies the client or project associated with the code.
Step 5.
Click the Save button to add this code. After the code is created, the status
line displays “Add successful.”
Assign a Client Matter Code to a Route Pattern
Follow these steps to assign a CMC to a route pattern:
Step 1.
From within CCMAdmin, select Call Routing > Route/Hunt > Route Pattern.
Step 2.
Enter search criteria in the search field to limit the results, and click the
Find button.
Step 3.
From the list that displays, select the route pattern to which you want to
add a CMC.
Step 4.
A screen similar to that shown in Figure 6-10 displays. Select the Require
Client Matter Code check box.
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Figure 6-10
Require Client Matter Codes on a Route Pattern
Step 5.
Click the Save button to apply the CMC requirement to this pattern.
Step 6.
Repeat these steps for all other route patterns on which you want to require a
CMC.
After these two tasks are completed, CMC is enabled. Make certain that CMC is never
applied to a pattern that enables access to emergency services such as 911.
Configuring Voice Ports and Profiles
Another feature—voicemail—is covered in great detail in the second half of this book. To
integrate with a voicemail system, a number of feature configurations must be completed.
The voicemail feature menu has five different submenus. This section examines the function of each of these and describes how each is configured.
Table 6-1 lists the five voicemail submenu items with a brief explanation of each.
The first thing that must be configured is the voicemail ports. Voicemail ports can be created manually or by using the wizard. It is recommended that you use the wizard because
it walks you through the process. The wizard enables you to add ports for a new voice-
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Table 6-1
Voicemail Submenu Items
Item
Description
Cisco Voice Mail Port
A virtual port that enables communications
between Communications Manager and Unity
Cisco Voice Mail Port Wizard
A wizard for creating voicemail ports
Message Waiting
Directory numbers for MWI activation
Voice Mail Pilot
A directory number that defines voicemail
pilot numbers
Voice Mail Profile
A profile that defines voicemail parameters for
devices
mail server, add ports for an existing voicemail server, or delete ports. The following steps
walk you through adding ports of a new voicemail server:
Step 1.
From within CCMAdmin, select Advanced Features > Voice Mail > Cisco
Voice Mail Port Wizard. Select the Create a New Cisco Voice Mail Server
radio button, and add ports to it and click Next.
Step 2.
On the next page, enter the voicemail port name in the Add Ports to a New
Cisco Voice Mail Server Using This Name field. In most cases, you can leave
it at CiscoUM1. This field needs to be changed only if CiscoUM1 is already
being used for another voicemail integration. Click Next.
Step 3.
In the next field, from the drop-down list, select the number of voicemail
ports you want to create. Click Next.
Step 4.
A screen similar to that shown in Figure 6-11 displays. In the Description field,
enter a description that helps identify the purpose of these voicemail ports.
Step 5.
From the Device Pool drop-down list, select the desired device pool to be
used by the voicemail ports.
Step 6.
A CSS determines where the voicemail port can dial. Choose a CSS from the
Calling Search Space drop-down list. If this field is left at None, the dial privileges of this voicemail port could be limited. This can affect message notification and call transfers.
Step 7.
Automated Alternate Routing (AAR) is used to provide an alternative route if
a call fails because of insufficient bandwidth. The AAR CSS can be used to
limit the paths a call will use when it is rerouted. Select an AAR CSS from the
AAR Calling Search Space drop-down list.
Step 8.
Locations are used to help WAN links from becoming oversubscribed in centralized deployments. Select the appropriate one for these voicemail ports
from the Location drop-down list.
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Figure 6-11
Step 9.
Voice Mail Port Wizard Device Configuration
From the Device Security Mode drop-down list, select a security mode for
the ports. If authentication using a mutual certificate is desired, select
Authenticated Voice Mail Port. To have the audio stream encrypted, select
Encrypted Voice Mail Port. To configure the port without security, select
Non Secure Voice Mail Port.
Step 10. The Use Trusted Relay Point drop-down list enables you to enable or disable
this requirement. In most cases, it can be left at the default.
Step 11. Click Next.
Step 12. A screen similar to that shown in Figure 6-12 displays. In the Beginning
Directory Number field, enter the directory number to be assigned to the first
voicemail port. Each additional port is assigned the next consecutive number.
Note On the previous configuration page, the parameters for the device were configured.
In this section, you configure the parameters for the line. This is the reason why you see
some of the same fields as those on the previous configuration page.
Step 13. Select the desired partition for the voicemail port line from the Partition
drop-down list.
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Figure 6-12 Voice Mail Port Wizard Directory Number Configuration
Step 14. Select the desired CSS for the line from the Calling Search Space dropdown list.
Step 15. Select the AAR group for the line from the AAR Group drop-down list.
Step 16. In the Internal Display field, enter the name that is displayed on a phone
when connected to voicemail.
Step 17.
In the Internal Display field (ASCII format), enter the name that is displayed
on a phone when connected to voicemail.
Step 18. In the External Number Mask field, enter the mask that should be applied to
the voicemail port line number when it places outside calls.
Step 19. Click Next.
Step 20. You are now asked whether these lines should be assigned to a line group.
Think of a line group as a group of lines through which a call can be forwarded. If the first line doesn’t answer, it is sent to the next number, and so on. In
most new installations, choose Yes, Add Directory Numbers to a New Line
Group. Click Next.
Step 21. Enter a name that helps identify this line group, perhaps VoiceMailLG.
Click Next.
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Step 22. A summary screen displays listing the configuration of the voicemail ports. If
all the information is correct, click Finish.
A screen displays informing you that the voicemail ports have been created and that a
hunt list and a hunt pilot must be created now. Because these components have not been
discussed yet, take a moment to define what they are and how they work.
A hunt pilot is a number that points to a hunt list. A hunt list is a list that points to one or
more line groups. A line group is a group of directory numbers through which a call can
travel. These combined components create hunt groups in Communications Manager.
The flow is similar to how a route pattern points to a route list, which points to a route
group, which points to a gateway.
Because Unity Connections has a number of ports on which a call can enter, a hunt
group has to be created to allow the call to try the first number and, if unavailable, move
on to the first available port. The Voicemail Port Wizard first creates the lines. Next, the
wizard creates the line group. Unfortunately, this is where the wizard ends and you begin
the rest of the setup manually.
Because the voicemail line and line group are configured, the hunt list must be created
next. Just as with route patterns, lists, and groups and devices, these components must be
created in a flow opposite to the call travel flow.
The following steps show how to create a hunt list:
Step 1.
From within CCMAdmin, select Call Routing > Route/Hunt > Hunt List.
Step 2.
Click the Add New link.
Step 3.
On the screen that displays, enter the name of the hunt list in the Hunt List
Name field.
Step 4.
In the Description field, enter a description that helps identify this hunt list,
such as VoiceMailHL.
Step 5.
Select the Communications Manager group that this hunt group will use from
the Cisco Unified Communications Manager Group drop-down list.
Step 6.
Click the Save button.
Step 7.
From the screen similar to that shown in Figure 6-13, the Enable This Hunt
List check box is selected by default. Typically, this is left at the default. If
you want to disable this hunt group, deselect this box.
Step 8.
Because you plan on using this hunt list for voicemail, select the For Voice
Mail Usage check box.
Step 9.
Click the Add Line Group button. A new screen displays. Select the desired
line group from the Line Group drop-down list. Click Save.
Step 10. An informational window displays telling you that the hunt list must be reset
before changes will take effect. Click OK.
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Figure 6-13 Hunt List Configuration
Step 11. The Hunt List Configuration screen returns. Click the Reset button. An informational window informs you that calls to this hunt list will be refused while
the list is being reset. Click Reset.
Step 12. The informational window reappears. The status should read “Reset request
was successfully sent.” Click Close.
Now that the hunt list is configured, a hunt pilot needs to be created. The hunt pilot is
the number that users will dial to reach the voicemail system. The steps that follow
demonstrate how to complete this process:
Step 1.
From within CCMAdmin, select Call Routing > Route/Hunt > Hunt Pilot.
Step 2.
Click the Add New link.
Step 3.
A screen similar to that shown in Figure 6-14 displays. Enter the route pattern
in the Hunt Pilot field.
Step 4.
The Route Partition field determines what devices can access the hunt pilot.
Select the desired partition from the Route Partition drop-down list.
Step 5.
Enter a description that helps identify the purpose of the hunt pilot in the
Description field.
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Figure 6-14
Hunt Pilot Configuration
Step 6.
Because you are configuring an internal extension, the Numbering Plan and
Route Filter drop-down lists will not be accessible.
Step 7.
Determine the precedence level that will be assigned to the hunt pilot by
selecting it from the MLPP Precedence drop-down list.
Step 8.
From the Hunt List drop-down list, select the appropriate hunt list to determine which calls are sent to the hunt pilot.
Step 9.
In the Alerting Name field, enter the name that should be displayed on the
caller’s phone while the hunt group is ringing.
Step 10. The ASCII Alerting Name field is the used on devices that do not support
Unicode characters.
Step 11. Select the Route This Pattern radio button.
Step 12. To have a secondary dial tone played after the first digit is dialed, select the
Provide Outside Dial Tone check box. In most cases, this box should be
deselected.
Step 13. To route a call as soon as it matches this route pattern, select the Urgent
Priority check box. In most cases, this box should be deselected.
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Step 14. The next two fields determine where a call is sent if the call is unanswered or
if all ports are busy. The fields are Forward Hunt No Answer and Forward
Hunt Busy. Enter the appropriate destination number in the Destination field.
If desired, a separate CSS can be defined for forwarded calls.
Step 15. Call pickup groups allow a user to redirect an incoming call on another
phone to the user’s phone. Pickup groups should not be added to voicemail
hunt pilots.
Step 16. In the Maximum Hunt Timer field, enter the maximum number of seconds
that a call is allowed to remain in the hunt group before being classified as
unanswered.
Step 17.
The Park Monitoring Forward No Retrieve Destination field determines
where calls are forwarded to when they are placed on park and not retrieved.
To send them to an alternative number, enter that number in the Destination
field. Because Park should not be configured for voicemail components, these
fields should be left at the default.
Note In most cases, the remaining fields can be left at the default. The following steps
can help you understand the purpose of each field.
Step 18. If you want to affect caller ID information, configure the fields located under
the Calling Party Transformation heading. These fields are discussed earlier
in this chapter. Use Calling Party’s External Phone Number Mask is the
first field.
Step 19. In the Calling Party Transform Mask field, enter any mask that you want to
affect the caller ID.
Step 20. In the Prefix Digits (Outgoing Calls) field, enter any digits that you want to
have added to the front of the caller ID.
Step 21. The Calling Line ID Presentation field determines whether caller ID information is to be blocked for outbound calls from the hunt pilot. To block caller
ID, select Restricted from the drop-down list. To allow caller ID, select
Allowed from the drop-down list.
Step 22. The Calling Name Presentation field determines whether caller name information is blocked for outbound calls from the hunt pilot. To block calling
name ID, select Restricted from the drop-own list. To allow calling ID name,
select Allowed from the drop-down list.
Step 23. Cisco recommends that the following two fields remain set to the default of
Cisco CallManager:
■
Calling Party Number Type
■
Calling Party Numbering Plan
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Step 24. The Connected Line ID Presentation field determines whether the connected
party’s ID information is displayed on the calling party’s phone. To block the
connected party’s ID, select Restricted from the drop-down list. To allow the
connected party’s ID, select Allowed from the drop-down list.
Step 25. The Connected Name Presentation field determines whether the connected
party’s name information is displayed on the calling party’s phone. To block
the connected party’s name, select Restricted from the drop-down list. To
allow the connected party’s name, select Allowed from the drop-down list.
Step 26. From the Discard Digits drop-down list, select the digit discard instruction
that is applied to calls that match this route pattern.
Step 27.
In the Called Party Transform Mask field, enter the mask that you want to
use for calls that match this route pattern.
Step 28. In the Prefix Digits (Outgoing Calls) field, enter any digits that you want to
have added to the dialed number before it is sent to the route list.
Step 29. Cisco recommends that the following two fields remain set to the default of
Cisco CallManager:
■
Called Party Number Type
■
Called Party Numbering Plan
Step 30. The AAR Group field is responsible for determining the device associated
with the AAR group. An AAR group provides the prefix that is assigned
when a call fails because of insufficient bandwidth. Select an AAR group if
AAR is being used. If this field is set to None, AAR is, in effect, disabled on
this device.
Step 31. AAR uses the external phone number mask to determine the fully qualified
number of the destination. Assign a mask that will expand the hunt pilot
number into a fully qualified number.
Step 32. Click the Save button.
A hunt pilot is now configured that points to a hunt list, which routes the call to the line
group of the voicemail port lines.
By dialing the hunt pilot number, users can access Unity. This is, of course, assuming that
Unity is running and properly integrated with Communications Manager. The Unity side
of the integration is covered in the Unity portion of this book.
Now the Message Waiting Indicator (MWI) settings must be configured so that the MWI
on the phone will activate when a new message is received. The MWI setting is simply a
directory number that is configured to act as either MWI on or MWI off. The following
steps show how to create MWI directory numbers in Communications Manager:
Step 1.
From within CCMAdmin, select Advanced Features > Voice Mail >
Message Waiting.
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Step 2.
On the next page, select the Add New link.
Step 3.
A screen similar to that shown in Figure 6-15 displays. In the Message
Waiting Number field, enter the directory number that you want to apply to
activate MWI on or MWI off.
Figure 6-15 MWI Configuration
Step 4.
If you want to assign a partition to the MWI number, select the desired one
from the Partition drop-down list. To allow unrestricted access to this number, select None.
Step 5.
In the Description field, enter a description that helps identify the purpose
of the MWI number.
Step 6.
Select either the On or Off radio button to define whether the number will be
used to turn MWI on or off.
Step 7.
Assign a CSS that has access to all the phones on which the MWI will activate the MWI. Select the appropriate CSS from the Calling Search Space
drop-down list.
Step 8.
Click Save to create this MWI number.
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After you create an MWI on and off number, you can test them by dialing each number
from a Cisco IP Phone. When the MWI on number is dialed, the light should come on.
When the MWI off number is dialed, the light should go off.
Now a voicemail pilot must be created. A voicemail pilot is used to determine the number
dialed when users press the Messages button on their phones. This number is normally
the same as the hunt pilot number you defined for the voicemail hunt list. After the voicemail pilot number is created, it is assigned to a voicemail profile. Voicemail profiles are
associated with lines on phones and are used to define voicemail-related attributes.
Voicemail profiles will be discussed shortly, but first, we create a voicemail pilot:
Step 1.
From within CCMAdmin, select Advanced Features > Voice Mail > Voice
Mail Pilot.
Step 2.
On the next page, click the Add New link.
Step 3.
In the Voice Mail Pilot Number field, enter the hunt pilot number that points
to the voicemail hunt list.
Step 4.
Select the CSS for this voicemail pilot from the Calling Search Space dropdown menu.
Step 5.
In the Description field, enter a description that will help identify this voicemail pilot. For example, if there are multiple Unity servers connected to this
Communications Manager, use the name of the Unity server to which this
pilot points as part of the description.
Step 6.
If you want this number to be the default voicemail number for phones on the
system, select the Make This the Default Voice Mail Pilot for the System
check box.
Step 7.
Click the Save button to add the voicemail pilot.
The last thing that must be configured is a voicemail profile. Voicemail profiles are
assigned to lines and determine what voicemail pilot the line uses. A voicemail profile
also applies a voice mailbox mask, if one is defined. A voice mailbox mask is used to
change the extension number that is sent to Unity. This might be necessary if the extension number in Unity is different than the one assigned to the phone. However, some similarity is needed. For example, if the extension on the phone is 2001 and the extension in
Unity is 52001, a mask of 5XXXX could be used. This mask changes 2001 to 52001.
To create a voicemail profile, follow these steps:
Step 1.
From within CCMAdmin, select Advanced Features > Voice Mail > Voice
Mail Profile.
Step 2.
On the next page, click the Add New link.
Step 3.
In the Voice Mail Profile Name field, enter a name for this voicemail profile.
Step 4.
In the Description field, enter a description that helps identify this voicemail
profile. For example, if there are multiple Unity servers connected to this
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Communications Manager, use the name of the Unity server to which this
profile points as part of the description.
Step 5.
Select the voicemail pilot from the Voice Mail Pilot drop-down list.
Step 6.
In the Voice Mail Box Mask field, enter a mask if it is necessary to change
the extension number that is sent to Unity.
Step 7.
If you want this profile to be the default voicemail profile for phones on the
system, select the Make This the Default Voice Mail Profile for the System
check box.
Step 8.
Click the Save button to add the voicemail pilot.
All the configuration required within Communications Manager for a Unity integration
should now be complete. By default, all the phones on the system will use the default
voicemail profile that is defined as shown in Step 7 of the previous steps.
Creating Users
Some Communications Manager features require that users be defined on
Communications Manager. By default, the users added to Communications Manager
used to be stored in a Lightweight Directory Access Protocol (LDAP) directory on
Communications Manager. As of Communications Manager 5.0, the user is stored in the
Informix database.
Some of the features covered later in this chapter require that users be created, so we
now take a look at how this is done. This section demonstrates the primary principles of
how to create a user. Later in the chapter, advanced user configuration is discussed. This
section is also limited to covering how to add a user when using DC Directory, which is
installed on the Communications Manager, not AD or Netscape.
The following steps show how to create a user on Communications Manager:
Step 1.
From within CCMAdmin, select User Management > End User.
Step 2.
Click the Add New link.
Step 3.
A screen similar to that shown in Figure 6-16 displays.
Step 4.
In the User ID field, enter a username. This name is used when the user logs
in to Communications Manager’s user web pages. After the user is added, this
field cannot be changed.
Note If extension mobility is going to be used, you might want to make the user ID all
numeric. Extension mobility can be configured to accept only digits for user IDs. This
accelerates the login process for the users so that they don’t have to spell out user IDs.
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Figure 6-16
User Configuration
Step 5.
Enter the password for the user in the User Password field and confirm it in
the Confirm Password field. The user password is the same one that is used
when the user logs in to the user web pages.
Step 6.
Enter a PIN in the PIN field and confirm it by entering it again in the Confirm
PIN field. The PIN can only be numeric and is used to log in to services on
the phone.
Step 7.
Enter the first and last name in the appropriate fields.
Step 8.
Enter the telephone number of the user in the Telephone Number field.
Step 9.
Enter the user email address.
Step 10. Enter the manager of the user in the Manager User ID field. The user you
enter in this field must already exist in the directory.
Step 11. In the Department field, enter the name of the department in which the
user works.
Step 12. Select the locale for the user from the User Locale drop-down list. This information is used for features such as extension mobility.
Step 13. When a Cisco softphone is used on a PC, the IP address or the host name of
the PC must be enter in the Associated PC field.
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Step 14. When using third-party SIP phones, digest credentials are required. This is
an alphanumeric string that is used for authentication purposes. If this user
is using a SIP phone, enter the appropriate credentials in the Digest
Credential field.
Step 15. Enter the digest credentials again in the Confirm Digest Credential field.
Step 16. The Controlled Devices box lists the devices that are associated to the user.
To associate a device to the user, click the Device Association button. The
Device Association button only appears after the user is saved. Click the Find
button. Select the check box next to the device that you want to associate
and click the Save Selected/Changes button. Now select Back to User from
the Related Links drop-down list and click Go to return to the User
Configuration page.
Step 17.
In the Extension Mobility section, a list of available extension mobility profiles are listed in the Available Profile box. To assign a profile to the user,
highlight the desired profile in the Available Profile box, and click the down
arrow below the Available Profile box. This causes the profile to appear in the
Controlled Profile box.
Step 18. From the Default Profile drop-down list, select the extension mobility profile
that should be used as the default profile for this user.
Step 19. The presence group that a device belongs to determines the lines that are
allowed to be monitored. Select the presence group that the user should
belong to.
Step 20. The SUBSCRIBE Calling Search Space field determines which partitions this
device can access when invoking a presence request. Select a CSS that
includes the partitions assigned to the lines that this user can monitor.
Step 21. To allow control of the device from CTI applications, select the Allow
Control of Device from CTI check box.
Step 22. As of CM 8.0, extension mobility can be configured to work across multiple
clusters. To allow a user to use this feature, the Enable Extension Mobility
Cross Cluster check box must be selected.
Step 23. Because multiple phones and extensions can be assigned to a single user, you
can define the user’s primary extension by selecting it from the Primary
Extension drop-down list.
Step 24. To enable Mobile Connect, select the Enable Mobility check box.
Step 25. If you select the Enable Mobility check box, you must select a primary
device that will be associated to this user from Primary User Device dropdown list.
Step 26. To allow voice access to Mobile Connect, select the Enable Mobile Voice
Access check box.
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Step 27.
The Maximum Wait Time for Desk Pickup value determines how much time
is allowed for a desk phone to be answered when a call is transferred from the
user’s cell phone. This value is set in milliseconds.
Step 28. The Remote Destination Limit value determines the maximum number of
devices that a user can send calls to through Mobile Connect.
Step 29. A remote destination profile defines parameters such as calling search space,
MoH audio source, and the device pool that should be used with Mobile
Connect. Select the desired profile from the Remote Profile Destination
drop-down list.
Step 30. Click the Save button to add this user.
After users are added, the user can log in to Communications Manager’s user web pages
and change items, such as passwords and PINs. The user web pages also enable users to
subscribe to phone services, configure speed dials, and configure other features on the
phone, but this can only be done after a device is associated with the user. The following
steps demonstrate how to associate a device to a user.
Step 1.
From within CCMAdmin, select User Management > End User.
Step 2.
Enter the user’s last name in the Search field, and click Find.
Step 3.
From the displayed list, select the user you want to associate to the profile.
Step 4.
Click the Device Association link.
Step 5.
Enter search criteria to limit the results returned, and click Find.
Step 6.
A list of devices displays. Select the check box next to the device(s) that you
want to associate with this user and click Save Selected/Changes.
Configuring Advanced Services
In addition to the standard features that users expect a phone system to offer,
Communications Manager offers a number of advanced features. The first part of the following sections explores a number of Communications Manager features that offer additional flexibility. These features enable users to do things such as log in to phones other
than their own and have those phones take on all the attributes of their own phones,
including directory numbers. There are a number of other features that can be configured
that are accessible from a Cisco IP Phone by pressing the Services button. These sections
examine a few of these.
The second part of the following sections looks at media resources, which are required
for many standard features, such as conferencing and MoH. A number of media resources
are automatically installed when certain services are activated, such as the software conference bridge that was discussed earlier in this chapter. A closer look at these and hardware media resources is offered later in these sections.
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Implementing Advanced Features
All users expect a phone to enable them to place a call on hold, transfer a call, and possibly initiate a conference call from time to time. These types of features are considered
standard on most systems. But how about being able to look up stock quotes or check
the status of an airline flight on the display of your phone? Although many would not
consider this a standard feature, it is just one of the many advanced features that can be
performed on many Cisco IP Phones. As cool and useful as some of these features are,
not all phones need them, so they can be made available only to certain phones if you
want. However, before any of these features can be used, they must first be configured
on the Communications Manager. Now take a look at some of these features and the
configuration that each requires.
Configuring IP Phone Services
Most Cisco IP Phones have the capability to run services. These services enable them to
perform functions that range from checking the weather to serving as a time clock. The
services available are too numerous to mention. Each service is an XML script that the
phone downloads from a server. If you have programming experience, you can choose to
create your own services or purchase services from a variety of companies that sell
prepackaged phone services.
Before any phone can use a service, the system administrator must make the service available by configuring it on the Communications Manager. The steps required to configure
the services are similar for all services, but many services require specific parameters.
Each service includes instructions that are unique for that particular service.
To give you an idea of how these services are configured on Communications Manager,
the following steps take you through the process of adding a typical service:
Step 1.
From within CCMAdmin, select Device > Device Settings > Phone Services.
Step 2.
On the next page, click the Add New link.
Step 3.
A screen similar to that shown in Figure 6-17 displays. In the Service Name
field, enter a descriptive name. This field determines the name of the service.
The name displays on the phone under Services.
Step 4.
In the ASCII Service Name field, enter the service name to be displayed on
devices that do not support Unicode characters.
Step 5.
In the Service Description field, enter a description for the service. In this
example, use Personal Address Book.
Step 6.
In the Service URL field, enter the location of the service. In this example,
use “http://x.x.x.x/ccmpd/xmlExample.asp,” where x.x.x.x is the IP address
of the Communications Manager and the quotes are not used. Before entering
the URL, you can always check to make sure that it is correct by entering the
URL in your browser address box and pressing Enter. A good URL gives you
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a line-by-line display of the code; a bad URL gives you the following message: “This page cannot be displayed.”
Figure 6-17
Step 7.
IP Phone Service Configuration
The Secure-Service URL is similar to the service URL. However, when using
a secure-service URL, the connection between the server and the endpoint
will be encrypted.
Note The URL might be case-sensitive, so be sure to enter it exactly as the installation
notes state.
Step 8.
Cisco supports XML and Java for phone services. Depending on the type of
phones you are using, Java might not be supported. Select the service category (XML or Java) from the drop-down list.
Step 9.
The Service Type value determines whether the service will be associated
with the Services, Directories, or Messages button on the phone. Typically,
Standard IP Phone Services should be selected.
Step 10. The Service Vendor field allows you to define the vendor of this service. This
field is required when using Cisco-signed Java MIDlets.
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Step 11. You can enter the version of the server in the Service Version field, but it is
not mandatory.
Step 12. Select the Enable check box.
Step 13. If you want the service to be available to all devices without having to subscribe to it, select the Enterprise Subscription check box. This field cannot
be modified after the service is configured, so be sure to set it properly.
Step 14. Click Save.
Some services can require additional information to be entered. Make sure that you consult the configuration documentation for a service prior to adding it.
Extension Mobility
Extension mobility is another feature of Communications Manager. This allows users to
log in to other phones and have those phones take on all the attributes of their own
phones or user profiles. A user device profile includes all the required phone attributes,
including extension numbers. This can be useful in environments where users are often
working in different locations.
A number of items must be configured to implement extension mobility, and because
certain tasks depend on the completion of other tasks, the order in which these tasks are
done is important. First, you need to make sure that the Extension Mobility service is
activated. The following steps walk you through this process:
Step 1.
Enter https://[CM_IP Address]/ccmservice in the address bar of your browser and press Enter.
Step 2.
Enter the administrative username and password and click Login.
Step 3.
Navigate to Tools > Service Activation.
Step 4.
Select the desired server from the Server drop-down list.
Step 5.
Select the Cisco Extension Mobility Service check box and click Save.
Now it is necessary to configure the extension mobility service parameters. To configure
the extension mobility services on the Communications Manager, follow these steps:
Step 1.
From within CCMAdmin, select System > Service Parameters.
Step 2.
Select the Communications Manager name from the Server drop-down list.
Step 3.
Select Cisco Extension Mobility from the Service drop-down list.
Step 4.
A screen similar to that shown in Figure 6-18 displays. The Enforce Intracluster Maximum Login Time field determines whether the value in the
Maximum Login Time field is enforced. If set to True, phones automatically
log out when the time entered in the Intra-cluster Maximum Login Time field
has expired. If you do not want to log out phones after this time has expired,
select False.
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Figure 6-18
Extension Mobility Service Parameters
Step 5.
In the Intra-cluster Maximum Login Time field, enter the amount of time
after which phones will automatically log out when logged in locally. This
value is used only if the Enforce Intra-cluster Maximum Login Time field is
set to True.
Step 6.
In the Inter-cluster Maximum Login Time field, enter the amount of time
after which phones will automatically log out when logged in remotely.
Step 7.
From the Intra-cluster Multiple Login Behavior drop-down list, choose
whether multiple logins will be permitted and how. The choices are
Step 8.
■
Multiple Logins Allowed: A user can log in to multiple devices at the
same time.
■
Multiple Logins Not Allowed: A user can log in to only one device
at a time.
■
Auto Logout: If a user who is already logged in to one device logs in
to another device, he or she will be automatically logged out of the
first device.
The Alphanumeric User ID drop-down list defines whether user IDs will contain letters and numbers or numbers only. Selecting True means that the ID contains letters and numbers, and False means that the ID contains only numbers.
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Step 9.
The Remember the Last User Logged In field determines whether the phone
will remember the ID of the last user who logged in. For security reasons, it is
recommended that this be set to False.
Step 10. The Clear Call Logs on Intra-Cluster EM value determines whether missed,
placed, and received call history is removed from the phone when a user logs
in or out. To have the information cleared, set this value to True.
Step 11. If you have changed any of the values on this page, click Save.
We now look at the steps required to add the extension mobility IP phone service:
Step 1.
From within CCMAdmin, select Device > Device Settings > Phone Services.
Step 2.
On the next page, click the Add New link.
Step 3.
Enter Extension Mobility in the Service Name field.
Step 4.
Enter Extension Mobility in the ASCII Service Name field.
Step 5.
Enter Extension Mobility Service in the Service Description field.
Step 6.
Enter
http://x.x.x.x:8080/emapp/EMAppServlet?device=#DEVICENAME#,
where x.x.x.x is the IP address of the Communications Manager.
Step 7.
Select XML Service from the Service Category drop-down list.
Step 8.
In the Service Type field, select Standard IP Phone Service.
Step 9.
Enter Cisco in the Service Vendor field.
Step 10. Enter the version of extension mobility in the Service Version field.
Step 11. Select the Enable check box.
Step 12. If you want this service to be available to all devices without having to subscribe to it, select the Enterprise Subscription check box. This field cannot
be modified, after the service is configured, so be sure to set it properly.
Step 13. Click Save.
After these tasks are completed, user device profiles must be created for users who will be
using extension mobility. In most cases, the user device profile should have the same
attributes as the user’s phone. The following steps show how to create a user device profile:
Step 1.
From within CCMAdmin, select Device > Device Settings > Device Profile.
Step 2.
On the next page, click the Add New link.
Step 3.
From the Device Profile Type drop-down list, select the type of phone on
which this profile is based and click Next.
Step 4.
Select the protocol that the device will use from the Device Protocol dropdown list and click Next.
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Step 5.
A screen similar to that shown in Figure 6-19 displays. In the Device Profile
Name field, enter a name for this profile.
Figure 6-19
Device Profile Configuration
Step 6.
In the Description field, enter a description that helps identify this profile.
Step 7.
From the User Hold MoH Audio Source field, select the audio source to be
played when a call is placed on hold by pressing the Hold button.
Step 8.
The User Locale field determines the language and fonts used for the phone.
The default user locale, which is set in the enterprise parameters, is used if
this field is left set to None. If this phone needs to use a different locale than
is defined by its device pools or the enterprise parameters, select the proper
one from the drop-down list.
Step 9.
From the drop-down list in the Phone Button Template field, select the
desired template for this profile.
Step 10. From the drop-down list in the Softkey Template field, select a softkey template for the phone.
Step 11. The Privacy field is used to determine whether the device can enable privacy
for calls on a shared line. Select the desired state for this field from the dropdown list.
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Step 12. The Single Button Barge field determines whether a barge or cbarge can be
initiated using a single button. Set this field to Barge to configure it for
single-button barge or cBarge to configure single-button cbarge. If left at
Default, the service parameter setting is used.
Step 13. The Join Across Lines field determines whether this feature is enabled. To
enable this feature, set this field to On. To disable this feature, set this field to
Off. The service parameter setting is used if the field is set to Default.
Step 14. The Always Use Prime Line field determines which line is activated when the
device is taken off-hook. If this field is set to Off, whichever line is ringing
will be used when the phone goes off-hook. When set to On, the primary
line will always be used, even if an incoming call is ringing another line of the
phone. To answer any line other than the primary line, the user must manually select the line. The service parameter setting is used when this field is set
to Default.
Step 15. If the Always Use Prime Line for Voice Message field is set to On, the voicemail account associated to the primary line is dialed when the Messages button is pressed. If set to Off, the account associated to the line that has voicemail will be dialed. If set to Default, the service parameter setting is used.
Step 16. If the Ignore Presentation Indicators (internal calls only) check box is selected, internal caller ID restrictions are ignored. This means that if an internal call
is configured to block caller ID, the caller ID will still show up on this device.
Step 17.
The Do Not Disturb check box enables an administrator to enable DND on
this device. To enable it, select the check box.
Step 18. From the DND Option drop-down list, you can select how the device will
behave when DND is enabled and an incoming call arrives. When Ringer Off
is selected, the device will not ring, but caller information will display on the
phone. When Call Reject is selected, caller information is not displayed and,
based on the DND Incoming Call Alert (see next parameter) setting, a beep
might be played or the light might flash.
Step 19. The DND Incoming Call Alert field works in concert with DND Option. This
parameter determines whether the device will beep or the light will flash
when the DND option is set to Call Reject or Ringer Off. When this setting
is set to None, the setting in the common phone profile is used. To turn the
flash and beep off, set this to Disable. To have a beep played, set this parameter to Beep Only. When set to Flash Only, the light on the phone will flash.
Step 20. From the Extension Mobility Cross Cluster CSS drop-down list, select the
CSS that is to be used with this profile through the Extension Mobility Cross
Cluster feature.
Step 21. The next two fields, Module 1 and Module 2, are used when expansion modules (7914, 7915, or 7916) are used with this profile. If this profile is going to
use expansion modules, select the modules from the drop-down list.
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Step 22. The next three fields define MultiLevel Precedence and Preemption (MLPP)
characteristics of this phone. If these fields are left blank or set to default, the
values set in the device pool are used. The first MLPP field is the MLPP
Domain. MLPP only grants higher priority from calls with the same MLPP
domain. For this reason, an MLPP domain is needed.
Step 23. The MLPP Indication field determines whether tones and indications are presented when a precedence call is made. The precedence indication can be a
special ringback or a display, if the caller’s phone supports it, and a special
ringer on the called party’s side.
Step 24. The MLPP Preemption field determines whether a higher-precedence call
preempts a lower-precedence call. The value of Disabled does not permit this
to happen. To cause a lower-precedence call to be terminated if a higherprecedence call requires the resources, set this parameter to Forceful.
Step 25. In the Login User ID field, enter the user ID of the user associated to the
profile. If the user device profile is used as a logout profile, specify the login
user ID that is associated with the phone. After the user logs out from this
user device profile, the phone will automatically log in to this login user ID.
Step 26. Click the Save button to add this profile.
Step 27.
Now you must add lines to this profile. Select the first Add a New DN link on
the left side of the screen. There might be more than one link with this label.
Step 28. A screen that enables you to configure the parameters for the line displays.
In the Directory Number field, enter the extension number to be used for
this line.
Step 29. The remaining settings on this page are the same as those found on the directory line configuration of a phone. Because you should already be familiar
with those settings, individual steps are not included for each. However, if
you want to review the individual steps, refer to the “Adding a Line to a
Phone” section in Chapter 3, “Deploying Devices.”
Step 30. After you complete filling in the remaining fields, click Save to add the line. A
message displays stating that the changes have been made and that any device
using this profile must log out and log back in before the changes will take
effect. Click OK.
Step 31. Click the Configure Device Profile (name) link and add additional lines as
needed.
Note If the Enterprise Subscription check box were selected when the Extension
Mobility service was configured, the service is available to all devices, so the next four
steps are not required.
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Step 32. After all lines are configured, you must add the extension mobility service to
this profile. Select Subscribe/Unsubscribe Services from the Related Links
drop-down menu and click Go.
Step 33. A screen similar to that shown in Figure 6-20 displays. Select Extension
Mobility from the Select a Service drop-down list. Click Next.
Figure 6-20 Add Extension Mobility Service to a Profile
Step 34. Click the Subscribe button.
Step 35. After the window refreshes, close it.
Now the profile must be associated with a user as shown in the steps that follow. These
steps assume that the user already exists in the directory.
Step 1.
From within CCMAdmin, select User Management > End User.
Step 2.
Enter the user’s last name in the Search field and click Find.
Step 3.
From the list that displays, select the user that you want to associate with
the profile.
Step 4.
In the Extension Mobility section make sure that the Allow Control of Device
from CTI check box is selected. If it isn’t, select the box and click Save.
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Step 5.
Highlight the desired profile from the Available Profiles box, and click the
down arrow below that box to move the profile to the Controlled Profiles box.
Step 6.
Click Save.
After device profiles are configured and associated with a user, extension mobility must
be enabled on the phones. This requires that the phone is subscribed to the extension
mobility service and that the extension mobility parameters are configured on the phone.
The following steps show how to complete the tasks:
Note If the Enterprise Subscription check box were selected when the Extension
Mobility service was configured, the service is available to all devices, so these steps are
not required.
Step 1.
From within CCMAdmin, select Device > Phone.
Step 2.
Enter search criteria in the Search field to limit the results and click the
Find button.
Step 3.
From the list that is generated, select the phone that you want to subscribe to
the extension mobility service.
Step 4.
Figure 6-21 shows the extension mobility parameters that display on the
Phone Configuration page. To enable this feature, select the Enable Extension
Mobility check box.
Step 5.
The Log Out Profile field determines what profile the phone uses when no
one is logged in to it. To have the phone keep its current settings, select Use
Current Device Settings.
Step 6.
Click Save to save the changes. A message displays stating that you must click
the Apply Config button before the changes take effect. Click OK.
Step 7.
Click Apply Config. A window appears to inform you to save changes before
applying the configuration. Click OK.
Step 8.
Select Subscribe/Unsubscribe Services from the Related Links drop-down
menu and click Go.
Step 9.
A screen similar to that shown when adding the extension mobility server to a
device profile displays. Select Extension Mobility from the Select a Service
drop-down list. Click Next.
Step 10. Click the Subscribe button.
Step 11. After the window refreshes, close it.
At this point, users should be able to start to use the extension mobility feature.
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Figure 6-21 Phone Configuration Page—Extension Mobility
Creating and Managing Media Resources
To function, some of the features that are available on Communications Manager require
media resources. These resources can be processes that run on the Communications
Manager or separate hardware devices. The following sections explore various media
resources and describe how each is configured. A few of these are created automatically
when certain Communications Manager services are activated, such as annunciators,
MoH servers, software Media Termination Points (MTP), and software conference
bridges. For these types, these sections focus more on the function than the creation
because these media resources are automatically created.
Configuring an MOH Server
MoH requires an MoH server. This is a service that runs on Communications Manager
and is created when the IP Voice Media Streaming Application is activated. This service
provides the resources needed to play an audio source while a call is on hold. Because
this resource is automatically created, detailed configuration steps are not provided here.
However, certain parameters that are unique to this service are explained. Figure 6-22
shows the configuration screen for an MoH server. Navigate to Media Resources >
Music On Hold Server, click Find, and select the desired MoH server to reach this page.
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Figure 6-22 Music on Hold Server Configuration
The following parameters are unique to the MoH server:
■
Maximum Half Duplex Streams: Determines the number of devices to which the
server can provide a unicast audio stream at one time.
■
Maximum Multicast Connections: Defines the maximum number of devices to
which the server can provide a multicast music audio stream at one time.
■
Fixed Audio Source Device: Enables you to use a live audio source for MoH if a
sound card is installed in the Communications Manager that is acting as the MoH
server.
■
Run Flag: When set to No, the MoH server does not provide music on hold. Set this
field to Yes to have the MoH server provide music.
■
Base Multicast IP Address: The base address that multicast uses if multicast is
enabled. To enable multicast, the Enable Multi-cast Audio Sources on this MOH
Server check box must be selected, and the MoH server must be assigned to a multicast media resource group. Media resource groups are discussed later in this section.
■
Base Multicast Port Number: Determines the base port number that the multicast
stream uses.
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The multicast stream can increase, based on port number or IP address. This is configured by choosing the appropriate radio button on this page.
Additional audio files can be added to the MoH server. After they are added, devices can
be configured to play different audio sources. Adding new audio sources is fairly simple.
The audio file must be in WAV format. The following steps take you through the process
of adding an audio source file:
Step 1.
The file must first be uploaded to the Communications Manager that is functioning as the MoH server. From within CCMAdmin, navigate to Media
Resources > Music on Hold Audio Source Files.
Step 2.
Click the Add New link.
Step 3.
Click the Upload File button.
Step 4.
An Explorer window opens, enabling you to select the WAV file on your PC.
Step 5.
After the name of the desired file appears in the Upload File field, click
Upload File.
Step 6.
After the file is uploaded, you are returned to the Upload File window.
Click Close.
Step 7.
Up to 51 audio sources can be configured. Using the MOH Audio Stream
Number drop-down list, select the number that determines where this audio
source displays in the list of audio sources.
Step 8.
From the MOH Audio Source File drop-down list, select the desired audio
source. After you select a file, the detail of the file displays in the MOH
Audio Source File status box.
Step 9.
Enter a name in the MOH Audio Source Name field that helps identify this
audio source later.
Step 10. To ensure that the audio source repeats when it reaches the end, select the
Play Continuously check box.
Step 11. If you have enabled multicast for the MOH server, the Allow Multicast
check box must be selected for audio sources that you want to use as a multicast stream.
Step 12. Click the Save button to add this audio source.
The audio source is now available to be added to a device. Throughout this book, you
have seen that an MoH audio source file can be placed at the line level, the device level,
or the device pool level. If an audio source is not specified on any of these levels, the
default level defined on the Cisco Communications Manager service parameters configuration page is used. The steps that follow illustrate how to assign the user hold and network hold audio source fields at the device level for a phone.
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Assign an MOH Audio Source to a Phone
The steps that follow show how to assign an MoH audio source to a phone:
Step 1.
From within CCMAdmin, select Device > Phone.
Step 2.
Enter search criteria in the Search field to limit the results, and click the
Find button.
Step 3.
From the list that is generated, select the phone to which you want to assign
an MoH audio source.
Step 4.
The phone configuration screen displays. To assign a user hold audio source
to the phone, select an audio source from the User Hold Audio Source dropdown list.
Step 5.
To assign a network hold audio source to the phone, select an audio source
from the Network Hold Audio Source drop-down list.
Step 6.
Click Save.
Step 7.
A window displays informing you that you must click the Apply Config button for the change to take effect. Click OK.
Step 8.
Click the Apply Config button.
Step 9.
A window displays informing you to save the configuration before continuing. Click OK.
After all audio sources are added, assign them as desired. In most cases, audio source
files can be added at the device-pool level. If there is a need to have phones within the
same device pool use different audio sources, you can configure the audio sources at the
device level, which will override the device pool settings.
It is important to understand that a device’s audio source determines what the held party
hears when a call is placed on hold. A device’s audio source does not determine what that
device hears when it is placed on hold. To determine what outside callers hear, configure
the audio sources on the PSTN gateway.
Creating Conference Bridges
As discussed earlier, Communications Manager has the ability to accommodate conference calls. To do this, a conference bridge is required. Conference bridges come in two
types: hardware and software.
The software bridge runs on the Communications Manager and is created during the
installation of Communications Manger. Because the software conference bridge is a
process that runs on Communications Manager, it takes available CPU cycles away from
other functions of the Communications Manager. To avoid this, hardware conference
bridges are recommended.
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Hardware conference bridges run on a number of Cisco devices. These devices have digital signal processors (DSP) that can be used for conferencing purposes. The devices that
support hardware conference bridges are continuing to expand. It is recommended that
you check Cisco.com for the most current list of hardware. The Cisco ISR 2800 with a
DSP farm is an example of the type of equipment that can serve as a conference bridge.
Depending on the hardware that is being used, the configuration of a conference bridge
varies slightly. The main difference is the first field you need configure. It will either be a
MAC address or the name of the device. To give you an idea of the design process, the
following steps show how to configure a Cisco IOS Enhanced Conference Bridge. It is
important to understand that there is additional configuration required on the device that
functions as the conference bridge. This configuration is beyond the scope of this book,
but you can find more information by searching “configuring enhanced conferencing and
transcoding for voice gateway routers” at Cisco.com.
Step 1.
From within CCMAdmin, select Media Resource > Conference Bridge.
Step 2.
On the next page, click the Add New link.
Step 3.
From the Conference Bridge Type drop-down list, select Cisco IOS
Enhanced Conference Bridge.
Six different types of conference bridges can be configured. Table 6-2 provides a brief explanation of each bridge and examples of the type of hardware required.
Step 4.
Enter the name of the bridge as it is specified in the device’s configuration. To
find the name, you must access the CLI of the device that is functioning as
the bridge. The associate profile command contains the name.
Step 5.
In the Description field, enter a description that helps identify the purpose
of the conference bridge. In the description, you might want to include on
which Catalyst it is installed.
Step 6.
From the Device Pool drop-down list, select the device pool for the conference bridge.
Step 7.
From the Common Device Configuration drop-down list, select the common
device configuration that the device will use.
Step 8.
From the Location drop-down list, select a location for this conference bridge
if one is required.
Step 9.
From the Device Security Mode drop-down list, select Non Secure
Conference Bridge or Encrypted Conference Bridge. This setting must
match what is configured on the device functioning as the bridge.
Step 10. The Use Trusted Relay Point field determines whether a relay point such as a
Media Termination Point (MTP) or a transcoder must be labeled trusted to
be used by this device. This field is typically changed only in virtualized
environments.
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Table 6-2
Conference Bridges
Bridge Type
Feature
Hardware
Cisco Conference Bridge (WS- Up to 32 conference resources WS-SVC-CMM-ACT
SVC-CMM)
per ACT port adapter.
Codecs supported are G7.11
and G7.29.
Cisco Conference Bridge
Hardware
Up to 32 conference resources WS-X6608-T1 (EoS)
WS-X6608-E1 (EoS)
depending on codec.
Codecs supported are G7.11,
G7.29, G7.23, GSM FR, and
GSM EFR.
Cisco Conference Bridge
Software
Up to 64 ad hoc or 128 meet- Runs on Communications
me conference resources.
Manager. The Cisco IP Voice
Media Streaming App must be
Supports G7.11 codec.
activated.
NM-HDV
Cisco IOS Conference Bridge Total supported number of
participants per conference is NM-HDV-FARM
six.
Codecs supported are G7.11
and G7.29.
Cisco IOS Enhanced
Conference Bridge
NM-HD
Total supported number of
participants per conference is NM-HD-2V
NM-HD-1V/2V/2VE
eight.
Codecs supported are G7.11,
G7.29.
Cisco Video Conference
Bridge (IPVC-35xx)
The number of conferencing IP/VC 3511
resources varies based on hard- IP/VC 3540
ware and the type of conferencing.
Supports a wide variety of
audio and video codecs.
Step 11. Click the Save to add this bridge.
After the conference bridge is created, it is available for use. Sometimes it might be necessary to reset the T1 port on the Catalyst 6000 for it to register to Communications
Manager before the bridge can be used.
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Configuring MTPs
Media Termination Points (MTP) are typically thought of as a software process that runs
on the Communications Manager. Although MTPs can also be hardware devices, this section focuses on the function and configuration tasks for software MTPs. Hardware MTPs
are discussed in the next section.
MTPs serve two different functions. First, they provide what are called supplementary
services for calls coming from an H.323 (version 1) gateway. Supplementary services are
features such as hold, transfer, conferencing, and park. Each of these features requires
that the connection be modified, and H.323 (version 1) does not have the ability to modify connections. So, the MTPs serve as a termination point between the endpoint and the
gateway. This allows modification of the connection to take place between the MTP and
the endpoint while the connection between the MTP and the gateway remains intact.
MTPs can also be required for Session Initiation Protocol (SIP) calls. SIP sends dual-tone
multifrequency (DTMF) tones in-band, whereas Skinny Client Control Protocol (SCCP)
uses out-of-band. This means that DTMF tones need to be translated between the two.
MTPs can provide this function.
Software MTPs are created during the installation of Communications Manager. No
other configuration is needed except to change the MTP’s device pool. To change the
MTP’s device pool, follow these steps:
Step 1.
From within CCMAdmin, select Media Resources > Media Termination Point.
Step 2.
On the next page, enter search criteria to limit the results and click Find.
Step 3.
Select the MTP from the list that displays.
Step 4.
Select the desired device pool from the Device Pool drop-down list.
Step 5.
Click Save. A message displays stating that the change will take effect when
the streaming is idle. Click OK.
The activation of the Cisco IP Voice Media Streaming Application service initiates the
creation of MTPs. Therefore, nothing else needs to be executed other than assigning
MTPs to media resource groups and lists, which will be discussed later in this chapter.
Creating Transcoders
Transcoders are hardware devices that convert calls from one codec to another. This
needs to be done when a call is placed between two devices that cannot communicate
using the same codec. For example, if a device that could only use G.729 called a device
that could only use G.711, a transcoder is necessary to convert the codec for the call to
take place. Here is a real-world example. A phone in a remote branch, which is configured
to use G.729 when placing calls across the WAN, wants to participate in a conference call
that originated on the other side of the WAN, and a software conference bridge is being
used. Because software conference bridges support only G.711 and the remote branch
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can only support G.729 across the WAN, a transcoder is needed to convert G.729 to
G.711 and vice versa.
Transcoders can run on a number of Cisco devices. These devices have digital signal
processors (DSP) that can be used for transcoding purposes. The devices that support
hardware transcoding continue to expand. It is recommended that you check Cisco.com
for the most current list of hardware. The Cisco 2821 with an NM-HDV2 is an example
of the wide range of equipment that can serve as transcoders.
Depending on the hardware that is being used, the configuration of a transcoder will
vary. The main difference is the first field that you need configure. It will either be a
MAC address or the name of the device. To give you an idea of the configuration
process, the following steps illustrate how to configure a Cisco 2821XM with an
NM-HDV2 as a transcoder:
Step 1.
From within CCMAdmin, select Media Resource > Transcoder.
Step 2.
On the next page, click the Add New link.
Step 3.
From the Transcoder Type drop-down list, select Cisco Conference
Enhanced IOS Media Termination Point.
Four different types of transcoders can be configured. Table 6-3 provides a
brief explanation of each with examples of the type of hardware required.
Step 4.
Table 6-3
In the Description field, enter a description that helps identify the purpose
of the conference bridge. In that description, you might want to include on
which Catalyst it is installed.
Transcoders
Transcoder Type
Feature
Hardware
Transcodes G.711, G.729, and G.723
Cisco Media
Termination Point (WSSVC-CMM)
WS-SVCCMMACT
Cisco Media
Termination Point
Hardware
Transcodes G.711, G.729, G.723, GSM FR, and GSM
EFR
WSX6608-T1
WSX6608-E1
Cisco IOS Media
Termination Point
Transcodes G.711 and G.729
NM-HDV
Cisco IOS Enhanced
Transcodes G.711 and G.729
Media Termination Point
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NMHDV2
NM-HD1V/2V/2VE
Chapter 6: Configuring CUCM Features and Services
Step 5.
Enter name of the IOS device in which the transcoding resource is installed in
the Device Name field.
Step 6.
From the Device Pool drop-down list, select the device pool for the conference bridge.
Step 7.
From the Common Device Configuration drop-down list, select the common
device configuration that the device will use.
Step 8.
The Special Load field enables you to enter special load information. In most
cases, this field is left blank.
Step 9.
The Use Trusted Relay Point field determines whether a relay point such as a
Media Termination Point (MTP) or a transcoder must be labeled trusted to
be used by this device. This field is typically changed only in virtualized
environments.
Step 10. Click the Insert button to add the transcoder.
After the transcoder is created, it is available for use. Later in this chapter, you see how to
assign transcoders to media resource groups and lists.
Configuring Annunciators
Annunciator is a process that runs on a Communications Manager that plays recorded
announcements to devices. These announcements are triggered by specific events. An
example is when a user dials a number that is not valid. In the past, the user would receive
a reorder tone, which sounds like a fast busy signal. With the annunciator, a message
plays that states “Your call cannot be completed as dialed. Please consult your directory
and call again or ask your operator for assistance. This is a recording.”
The annunciator is also used to play other messages to inform the caller of events, such as
service disruption or the MLPP causing a call to fail.
An annunciator is created during the installation of Communications Manger. No other
configuration is needed other than to change the device pool. To change the device pool
of an annunciator, follow these steps.
Step 1.
From within CCMAdmin, select Media Resources > Annunciator.
Step 2.
On the next page, enter search criteria to limit the results and click Find.
Step 3.
Select the Annunciator from the list that displays.
Step 4.
Select the desired device pool from the Device Pool drop-down list.
Step 5.
Click Save.
Step 7.
Click the Apply Config button.
Step 9.
A window displays informing you to save the configuration before continuing. Click OK.
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Because an annunciator is created when the Cisco IP Voice Media Streaming Application
service is activated, the only other configuration that you might want to make is to add
the annunciator to a media resource group. Media resource groups and lists are discussed next.
Media Resource Management
After all the media resources are configured, they can be used. To make the most effective use of the resources, Communications Manager allows you to manage how resources
are used and what devices are allowed to use them. In addition, media resource management permits the resource to be shared throughout the cluster.
Media resource management is accomplished by creating media resource groups (MRG)
and media resource group lists (MRGL). Media resource groups contain various media
resources. A media resource group list contains one or more media resource groups. A
device is then assigned to a media resource group list that determines the accessibility of
the device to media resource groups. The order in which the groups display in the list
determines which group is queried first for a requested resource. Resources that are
assigned to an MRG are shared in a round-robin fashion, with the exception of conference bridges. Consequently, the first conference resource listed in an MRG wouldn’t necessarily be the one that is used. In the case of conference resources, Communications
Manager checks each conference bridge in the MRG to see how many resources are available and selects the bridge with the greatest number available, which gives the user the
greatest chance of successfully adding the number of audio streams to the conference.
The order in which the media resources appear in the group determines the order in
which requested resources are used. Figure 6-23 shows an example of this.
DTW-AZO Resource List
DTW Resource Group
DTWHWConfBr
GRRHWConfBr
DTWxcode
GRRxcode
DTWMTP
GRRMTP
AZO Resource Group
AZOHWConfBr
AZOxcode
AZOMTP
Figure 6-23 Media Resource Group List Example
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In this example, the phone is assigned the DTW-AZO resource list. When the phone
needs a conference bridge resource, all conferences in the MRG are queried to see which
one has the most available resources. When all resources are available on all conference
bridges, the first one is used. In this example, it would be the DTWHWConfBr because it
is the first one listed in the first group. If all the conference resources in the first group
were unavailable, the AZOHWConfBr would be used because it is in the second group.
In the event that a device requests a resource that is not in any of its resource groups, an
attempt is made to use a resource allocated to the default group. Any device that is not
associated with a specific group is part of the default group.
Creation and configuration of media resource groups and lists are very similar to those of
route groups and lists. We now take a look at how a media resource group is configured.
The following steps explain how to create a media resource group:
Step 1.
From within CCMAdmin, select Media Resources > Media Resource Group.
Step 2.
Click the Add New link.
Step 3.
A screen similar to that shown in Figure 6-24 displays. In the Name field,
enter a name for this group.
Figure 6-24 Media Resource Group Configuration
Step 4.
In the Description field, enter a description that helps identify this group.
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Step 5.
From the Available Media Resources box, select the resource that you want
to add to this group. Click the down arrow below this box.
Note You can select more than one resource at a time by holding down the Ctrl key
when selecting them.
Step 6.
The selected resources now display in the Selected Media Resources box.
You can remove a resource by selecting it and clicking the up arrow.
Step 7.
After all the desired resources are in the Selected Media Resources box, click
the Save button to add this group.
Now the groups need to be assigned to a media resource list. The following steps illustrate how to assign an MRG to an MRGL.
Step 1.
From within CCMAdmin, select Media Resources > Media Resource
Group List.
Step 2.
Click the Add New link.
Step 3.
A screen similar to that shown in Figure 6-25 displays. Enter a descriptive
name in the Name field.
Figure 6-25 Media Resource Group List Configuration
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Chapter 6: Configuring CUCM Features and Services
Step 4.
From the Available Media Resource Groups box, select the groups that you
want to add to this list. Click the down arrow below this box.
Note You can select more than one group at a time by holding down the Ctrl key when
selecting them.
Step 5.
The selected resources now display in the Selected Media Resource Groups
box. You can change the order in which a group displays in the list by highlighting the name and clicking the up and down arrows located to the right of
the Selected Media Resource Groups box. You can also remove a group by
selecting it and clicking the Up Arrow located above the box.
Step 6.
After all the desired groups are in the Selected Media Resource Groups box,
click the Save button to add this group.
After the resource lists are created, you must assign them to devices. This can be done at
the device level or at the device pool level. If a media resource list is not assigned to a
device, the device will use the one assigned to its device pool.
The following steps show how to assign a media resource group list to a phone and a
device pool.
Assign a Media Resource Group List to a Phone
Follow these steps to assign an MRGL to a phone:
Step 1.
From within CCMAdmin, select Device > Phone.
Step 2.
Enter search criteria in the Search field to limit the results, and click the
Find button.
Step 3.
From the list that is generated, select the phone to which you want to assign a
media resource group list.
Step 4.
The phone configuration screen displays. Select a media resource group list
from the Media Resource Group List drop-down list.
Step 5.
Click the Save button.
Step 6.
A window displays informing you that you must click the Apply Config button for the change to take effect. Click OK.
Step 7.
Click the Apply Config button.
Step 8.
A window displays informing you to save the configuration before continuing. Click OK.
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Assign a Media Resource Group List to a Device Pool
Follow these steps to assign an MRGL to a device pool:
Step 1.
From within CCMAdmin, select System > Device Pool.
Step 2.
To limit the results, enter search criteria in the Search field and click the
Find button.
Step 3.
From the list that is generated, select the device pool to which you want to
assign a media resource group list.
Step 4.
The device pool configuration screen displays. Select a media resource group
list from the Media Resource Group List drop-down list.
Step 5.
Click the Save button.
In most cases, it is recommended to assign the media resource group list at the device
pool level. This makes it easier to manage and advantageous if you are using extension
mobility. If a device requires a unique media resource, it should be assigned at the
device level.
Configuring Remote Site Failover
When deploying Communications Manager with remote sites, it is necessary to deploy
some type of remote site redundancy in the event that the link with the central site is
lost. Every time a phone places a call, it must be able to talk to Communications
Manager. No call can be placed if the phone is unable to communicate with a
Communications Manager. As mentioned earlier in this book, as soon as the
phone goes off-hook, it sends a request to Communications Manager. If the primary
Communications Manager fails, the secondary can respond to all requests. What happens
in a remote environment when the link to the Communications Manager fails? Because
the phone can’t reach the Communications Manager, the phone doesn’t know to provide
dial tone. The solution is to deploy a Survivable Remote Site Telephony (SRST) solution at
the remote site.
Another issue that can occur with remote sites is that the link can be operating properly
but all the available bandwidth is in use and additional calls cannot be sent across this
link. So, when a phone on the remote site tries to call a phone at the central site, the call
is rejected because of insufficient bandwidth. This means that until bandwidth is available, the only way to reach the central site is by sending the call over a different path
such as the public switched telephone network (PSTN). This requires the caller to hang up
and dial a number that will use the PSTN instead of the WAN link. The solution for this
issue is to configure AAR groups.
The following sections examine how SRST and AAR work and how each is configured.
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SRST Overview
SRST is a service that can run on a number of Cisco devices such as the 1861, 2821, and
3925, just to name a few. The SRST device is located at the remote site and is typically
the same device that is used for the remote site PSTN gateway.
When phones cannot communicate with any of the Communications Managers at the
central site, they register with the SRST device. The device then accepts and responds to
SCCP requests. All calls that are not destined for devices at the remote site are then routed out to the PSTN.
Because the SRST device is responsible for routing calls, it must have some type of route
plan programmed in it, just like a normal H.323 gateway.
The majority of the configuration is done on the SRST device itself. This part of the configuration requires some IOS telephony programming experience.
Configuring SRST
The configuration of an SRST solution occurs in two places. The first is on the SRST
device, and the second is in the Communications Manager. First take a look at a sample
of the configuration on the SRST device itself:
access-code fxo 9
default-destination pattern 2002
dialplan-pattern 1 547....
ip source-address 10.1.1.2 port 2000
keepalive 30
max-ephones 24
max-dn 48
transfer-pattern 5472...
voicemail 5479100
Table 6-4 provides nine commands and definitions for the configuration process.
In addition to these commands, further dial plan and port configuration are required. It is
recommended that only those familiar with IOS devices and telephony configuration of
such devices perform this portion of the configuration.
After the SRST device is configured, you must configure Communications Manager so
that phones can use this service. If the SRST device is the phone’s default gateway at the
remote site, fewer configurations are required. However, if the SRST device is not the
default gateway for the phone, an SRST reference must be created.
An SRST reference is used to instruct the phone to register with the appropriate device if
it cannot communicate with a Communications Manager. The following steps show how
to create an SRST reference. If the SRST device is the phone’s default gateway, skip to the
“Assign an SRST Reference to a Device Pool” section, later in this chapter.
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Table 6-4 Configuring SRST
Command
Description
access-code fxo 9
Signifies that when a 9 is the first digit, the call is routed out an
FXO port.
default-destination
pattern 2002
Causes all inbound calls that do not match a registered directory
number to be routed to extension 2002.
dialplan-pattern 1 547 . . . For inbound calls (calls to an IP phone in a Cisco SRST system)
where the calling party number matches the dial-plan pattern, the
call is considered a local call and has a distinctive ring that identifies the call as internal. Any calling party number that does not
match the dial-plan pattern is considered an external call and has a
distinctive ring that is different from the internal ringing. For outbound calls, the dialplan-pattern command converts the calling
party’s extension number to an E.164 calling party number.
Outbound calls that do not use an E.164 number and go through a
Primary Rate Interface (PRI) connection to the PSTN, can be
rejected by the PRI link as the calling party identifier.
ip source-address 10.1.1.2 Specifies the IP address of the interface that services all SCCP
port 2000
requests.
keepalive 30
Defines the keepalive interval in seconds.
max-ephones 24
Determines the maximum number of phones that are allowed to
register to this device.
max-dn 48
Determines the maximum number of directory numbers that are
allowed to register to this device.
transfer-pattern 5472 . . . Determines to what numbers calls can be transferred. In this case,
calls can only be transferred to seven-digit numbers that begin
with 5472. IP phone directory numbers that have registered to
SRST can also be transferred without additional configuration.
voice mail 5479100
Defines the number that is dialed when users press the Voicemail
button on their phones.
Creating an SRST Reference to a Device Pool
Follow these steps to create an SRST reference to a device pool:
Step 1.
From within CCMAdmin, select System > SRST.
Step 2.
Click the Add New link.
Step 3.
A screen similar to that shown in Figure 6-26 displays. In the Name field,
enter a name for this SRST reference.
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Chapter 6: Configuring CUCM Features and Services
Figure 6-26 SRST Reference Configuration
Step 4.
Enter the port number that is used for SCCP in the Port field. In most cases,
this should be left at 2000.
Step 5.
In the IP Address field, enter the IP address in the interface of the SRST
device that is going to service SCCP requests.
Step 6.
In the SIP Network/IP Address field, enter the server that SIP phones will
register to when SRST is active.
Step 7.
In the SIP Port field, enter the SIP port of the SRST gateway. This is typically 5060.
Step 8.
If the SRST device contains a self-signed or certificate-authority-issued certificate, select the Is SRST Secure? check box.
Step 9.
Click Save to add this SRST reference.
After you configure all the necessary SRST references, you must define which references
will be used by which devices. This is done by assigning an SRST reference to the device
pools. The following steps show how this is accomplished.
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Assign an SRST Reference to a Device Pool
Follow these steps to assign an SRST reference to a device pool:
Step 1.
From within CCMAdmin, select System > Device Pool.
Step 2.
To limit the results, enter search criteria in the search field and click the
Find button.
Step 3.
From the list that is generated, select the device pool on which you want to
define an SRST reference.
Step 4.
The device pool configuration screen displays. Select an SRST reference from
the SRST Reference drop-down list. You will notice that, in addition to the
SRST reference you created, the Use Default Gateway option is provided. If
the SRST device is the phone’s default gateway, select this option. To prohibit
phones from using SRST, leave the SRST reference set to Disable.
Step 5.
Click the Save button.
After all device pools have been configured, the SRST configuration process is complete.
Configuring AAR
Automated Alternate Routing (AAR) is used to reroute a call over an alternate path when
the call fails because of lack of bandwidth. This is accomplished by creating AAR groups
that define a prefix that should be assigned to a number if the call fails because of inadequate bandwidth. Figure 6-27 shows an example.
Central Office
Outside Access = 9
Remote Office
On-Net Number
5001
On-Net Number
4201
PSTN
Seven Digit Number
544-5001
Seven Digit Number
548-4201
Figure 6-27 AAR Group Example
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Chapter 6: Configuring CUCM Features and Services
The user at the remote office only needs to dial 4201 to reach the phone in the central
office. If there is enough bandwidth, the call will be successful. However, if there is insufficient bandwidth, the call needs to be rerouted over the PSTN. Because 9 is used as the
outside access code, the number that needs to be dialed to reach extension 4201 through
the PSTN is 95484201. With AAR properly configured, the dialed digits of 4201 are
automatically changed to 95484201 and the call is sent out the PSTN.
Note If an external phone number mask is assigned (for example, 612548XXXX), the
prefix digits are added to the end result of the mask’s transformation. Hence, in this example, 916125484201 would be the end result.
To configure AAR, you simply need to create AAR groups and assign them to directory numbers.
Creating an AAR Group
The following steps show how to create an AAR group:
Step 1.
From within CCMAdmin, select Call Routing > AAR Group.
Step 2.
Click the Add New link.
Step 3.
Enter a name for the AAR group in the Name field and click Save.
Step 4.
A screen like that shown in Figure 6-28 displays. Because this is the first
group, there is only one field to populate. In the Dial Prefix field, enter the
digits that will be added to the beginning of the dialed number when a device
from within this group dials another device within the same group.
Step 5.
Click Save.
Step 6.
To see how the configuration changes when more than one group exists, create another group by clicking the Add New link.
Step 7.
Enter a name for the ARR group in the Name field and click Insert.
Step 8.
A screen similar to that shown in Figure 6-29 displays. In the Dial Prefix
field, enter the digits that will be added to the beginning of the dialed number
when a device from within this group dials another device within this group.
Step 9.
On this screen, there are additional fields that do not exist when only one
AAR group is created. In the Dial Prefix (From <AAR GROUP NAME>)
field, enter the digits that will be added to the beginning of the dialed number
when a device from within this group dials a device within the other group.
Step 10. In the Dial Prefix (To <AAR GROUP NAME>) field, enter the digits that will
be added to the beginning of the dialed number when a device in the other
group dials a device within this group.
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Figure 6-28 AAR Group Configuration
Step 11. Click Save.
Now the AAR groups must be assigned to lines.
Assign an AAR Group to a Line
The following steps show how to assign an AAR group to a line on a phone:
Step 1.
From within CCMAdmin, select Device > Phone.
Step 2.
Enter search criteria in the Search field to limit the results, and click the
Find button.
Step 3.
Select the desired phone from the list that displays.
Step 4.
Click the desired line on the left side of the screen.
Step 5.
On the Directory Number Configuration page, select the desired AAR group
from the AAR Group drop-down list, as shown in Figure 6-30.
Step 6.
Click the Save button.
Note When you make a change to the directory number configuration and click Save, the
phone might reset. If a caller is currently on a call, the phone will reset after the call is ended.
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Figure 6-29 AAR Group Configuration (Multiple Groups)
Figure 6-30 Assigning an AAR Group to a Line
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Summary
This chapter covered an abundance of information. It began by exploring the configuration required for features such as call park, meet-me conferences, and call pickup groups.
Then the configuration of forced authorization codes and client matter codes was discussed. Next, configuration tasks required for the integration of Unity were explored.
This included creating voicemail ports and assigning them to line groups. The line groups
were then assigned to a hunt list, and finally a hunt pilot was configured that pointed to
the hunt list.
Next, the configuration of IP phone services was discussed. This section included the
steps required to make an IP phone service available and discussed how an administrator
can subscribe a phone to a service. Following IP phone services, Extension Mobility was
explored, and the features and configuration tasks were explained.
Media resources were then covered. This section included the features and functions of
transcoders, CFBs, MTPs, MoH servers, and annunciators. After explaining the configuration tasks for each of these, the chapter discussed the function of media resource
groups and lists. The configuration tasks of adding media resources to groups, and then
the groups to lists, were also explored.
SRST and AAR were discussed in the next section. It explained how SRST can help maintain system functionality for a remote site in the case of a WAN outage. It also discussed
how AAR can allow a call to successfully be routed over an alternate path if the call is
rejected because of bandwidth limitations.
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Chapter 7
Unity Predeployment Tasks
One of the most widely deployed phone system add-ons is voicemail. For a number of
years, Unity was the voicemail solution of choice for Communications Manager.
However, over the last few years, a new Cisco voicemail solution has become quite popular. The new product is called Unity Connection. Because both platforms integrate well
with Communications Manager, both are covered in the following chapters. There are
many similarities between these products. They are so comparable that much of what one
learns about one product can be directly applied to the other. Whenever possible, the
chapters are written so that they are applicable regardless of which product you are using.
Although Unity and Unity Connection are similar, they are not identical. They offer different features. Because both have the word Unity in their name, it can sometimes cause
confusion when discussing either product. For clarification the following terms are used
throughout the messaging section of this book to help you clearly identify which product is being discussed:
■
Unity: Used when the topic applies only to Unity
■
Unity Connection: Used when the topic only applies to Unity Connection
■
Unity/Connection: Used when the topic applies to both products
For the majority of tasks that an administrator performs on a day-today basis, the main
difference is the look and feel of the administrative interface. Because the interfaces are
not identical, step-by-step instructions for both interfaces are offered.
If your solution is going to be effective, proper configuration is essential. The first step to
proper configuration is verifying that the integration is correct and that all predeployment tasks have been completed. This chapter includes step-by-step instructions for completing predeployment tasks, such as verifying integration, defining system parameters,
and creating templates, distribution lists, and class of service (CoS). First you look at the
administrative interface to see how these tasks are accomplished within Unity.
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Accessing and Navigating Unity Administrator
Most of the configurations discussed throughout the messaging portion of this book are
accomplished through a web-based administration interface. Because this is the main
administration interface, it is used for most administration tasks, so it is important that you
become familiar with it. This section introduces you to the Unity administrative interface
(Unity Administrator). The Unity Connection interface is reviewed in the next section.
To access Unity Administrator, open Internet Explorer (6.0 or higher) and enter the server’s name followed by /sa/web. If the server’s name is DTWUNITY, for example, you
enter DTWUNITY/sa/web in the address bar of Internet Explorer. Unlike when using
CM, you should not be prompted for a username and password. You are authenticated
based on your Window’s Active Directory (AD) credentials. If you are logged in as a user
who does not have an associated Unity subscriber with administrative rights, you will not
have access to this interface. You need to log in to the network as a user that has Unity
administrative rights.
Unity Administrator is often referred to as System Administrator (SA). The term SA has
been used for this interface for a number of years, whereas the UA term is relatively new.
For this reason, the term SA will be used throughout this book.
Note It is important to understand that administrative rights for Unity are not associated
with Windows rights. Having administrator rights for Windows does not automatically
give you administrative rights for Unity. To access SA, your Windows account must have
an associated Unity subscriber that has Unity administrative rights. During the installation,
an administrator was selected. Use this account for now, and how to add administrative
rights to subscribers is discussed later.
After you log in to SA, you see an interface similar to that shown in Figure 7-1. In Figure
7-1, each section of this interface is labeled. The section labeled 1 is referred to as the
navigation panel. This panel is used to access the various configuration screens. The section labeled 2 is referred to as the title strip, which contains icons that enable you to perform functions such as adding, deleting, and saving. The section labeled 3 is referred to
as the page body. This is where the majority of the actual work is performed. From the
page body, you enter the specific configuration information for the object on which you
are working.
The following example should help you to better understand how this interface works; it
is an example of changing the extension number that is associated with a subscriber.
Each step includes a screen shot of the SA.
Note Although a screen shot is provided here for almost every configuration task, it is
not the general practice in this book. Step-by-step screen shots are important in this chapter to introduce you to the interface. However, throughout the rest of the book, screen
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shots are provided to help reinforce the topic being discussed and do not illustrate each
step as they do in this example.
2
3
1
Figure 7-1
Step 1.
Unity Administrator Interface
Log on to SA by entering the server’s name followed by /Web/SA. Figure 7-2
shows the main SA screen. Note that there is no title strip in this figure.
Note If you are accessing SA using the Unity server’s keyboard and screen, there is an
icon on the desktop that is labeled System Administrator. Clicking this icon should open
the System Administrator interface.
Step 2.
From the navigation panel, click the Subscribers link, which is shown in
Figure 7-2.
Step 3.
Figure 7-3 shows that the title strip displays and the page body changes after
you select the of the object you want to configure. Also note that the navigation panel changes and presents you with all the configuration options for that
type of object. By looking in the navigation panel, you are in a Subscriber
configuration, and by looking at the heading in the page body, you are in the
profile portion of the Subscriber configuration. For simplicity’s sake, this
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screen is referred to as the Subscribers > Profile page. You see many references of this type of syntax (called navigation syntax) throughout this portion
of the book, so be sure that you are comfortable with this concept.
Figure 7-2
SA Main Screen
Note In future step-by-step tasks, navigation syntax will be used to help you quickly
understand at exactly which screen you should be. For example, if the navigation syntax
were Call Handlers > Call Transfer, you would click Call Handlers from the main navigation menu and then select Call Transfer.
Step 4.
In the page body, the extension is 4321. Figure 7-4 illustrates this. To change
the extension, simply change the number that displays in the Extension field
to 4123. Although colors do not show in the screen captures of this book,
you can see the colors on the screen if you follow these steps on a live system. The icon that looks like a floppy disk will change colors. Before you
change the extension it is grayed out; when the extension number is changed,
it is blue. The disk displaying in blue signifies that changes have been made
but not saved. The disk displaying in gray indicates that there is no new information that needs to be saved. In the title strip, that there is an asterisk next
to the subscriber’s name. This is another indication of unsaved data. Click the
Disk icon to save the change.
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Figure 7-3
Subscribers > Profile
This example was fairly simple. Other Unity tasks will be more detailed. The goal of this
example is to introduce you to the SA interface and some of the syntax that is used in the
step-by-step tasks.
Before moving on, look at the other icons that display in the title strip. In addition to the
disk, there is a blue magnifying glass, a blue plus sign, a red X, a book framed by a yellow button, and a question mark framed by another yellow button. In addition, an icon of
a running man displays when you run reports. A brief explanation of each follows:
■
Disk (Save icon): Displays in blue if there is unsaved data. Clicking this icon saves
any unsaved data.
■
Magnifying glass (Find icon): Used to find items. Clicking this icon opens a
search criteria window.
■
Plus sign (Add icon): Used to add a new object.
■
X (Delete icon): Used to delete the current object.
■
Book (Help icon): Opens an online documentation window.
■
Question mark (Field Help): Displays a question mark next to each field.
Clicking a question mark opens a help window that explains that field.
■
Running man (displays when reports are running)
causes the selected report to be queued.
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Figure 7-4
Unity Subscriber Screen
Note These icons are sometimes referred to as sweet tarts.
In the earlier example, you were introduced to navigation syntax. Now look at one more
example to make sure that you are comfortable with the concept.
The navigation syntax for this example is Subscribers > Class of Service > Transfer.
Figure 7-5 shows the main navigation menu. Class of Service is under the Subscribers
heading. After selecting Class of Service, the menu in the navigation panel changes to
display a list of the configuration screens that deal with class of service. From this list,
select Transfer, as shown in Figure 7-6. To return to the main navigation menu, click the
blue arrow located in the upper-left corner of the navigation panel.
Now that you are familiar with the SA interface and the syntax that will be used in this
portion of the book, first look at the tasks that need to be accomplished to ensure a
properly configured system.
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Figure 7-5
Unity Main Screen
Accessing and Navigating Unity Connection
Administrator
The Unity Connection administrative interface is accessed by entering the IP address of
the Unity Connection server followed by “/cuadmin” in the address bar of a browser. For
example, if the server’s IP address is 10.1.1.10, you would enter https://10.1.1.10/cuadmin.
You will be presented with a login screen. In this book, this interface is called CUAdmin.
Enter the username and password for the CUAdmin page and click Login. A screen similar to the one shown in Figure 7-7 appears. Unlike Unity, Unity Connection does not
require integration to Microsoft AD. For this reason, you need to log in to CUAdmin
even if you are already logged in to AD.
Note Unity Connection supports the following browsers:
* Microsoft Internet Explorer (IE) 7 when running in Microsoft Windows XP SP3
* Microsoft Internet Explorer (IE) 8 when running in Microsoft Windows XP SP3 or
Microsoft Vista SP2
* Firefox 3.x when running in Microsoft Windows XP SP3, Microsoft Vista SP2, or
Safari 4.x when running on Apple MAC OS X
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Figure 7-6
Unity Submenu Screen
The layout of this interface is fairly intuitive. The left side of the screen is the navigation
panel, and the right side is where the configuration information is entered. After you
select an item from the navigation panel, the configuration page appears, as seen in
Figure 7-8. To become familiar with this interface, use the following steps to change the
voicemail password for a user.
Step 1.
To change the password of a user, you need to navigate to the user search
screen. Figure 7-8 shows the navigation panel fully expanded. This enables
you to examine the subchoices under each main heading. For example, the
first main heading you see is Users. Clicking only the main heading expands
and collapses the subchoices. To navigate to a configuration page, click the
subchoice. In this case, you want to select the subchoice Users under the
main Users heading. To make this easier throughout the rest of the text, navigation syntax will be used. Using navigation syntax, this step is simplified to
“From the menu on the left, select Users > Users.”
Step 2.
The users are listed as seen in Figure 7-8. If the users you are looking for are
not displayed in the list, you can search by alias, extension, first name, last
name, or display name. Enter the search criteria and click Find.
Step 3.
Select the user from the list that appears by clicking the alias. A screen similar
to the one shown in Figure 7-9 appears.
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Figure 7-7
Unity Connection Administrator
Step 4.
There are drop-down menus at the top of the screen. Mouse over Edit and a
list similar to that seen in Figure 7-10 appears; select Change Password.
Step 5.
Enter the new password in the Password field, as seen in Figure 7-11.
Step 6.
Enter the password again in the Confirm Password field.
Step 7.
Click Save.
In this example, you were introduced to navigation syntax. Now look at one more example to make sure that you are comfortable with the concept.
The navigation syntax for this example is Call Management > Call Routing > Direct
Routing Rules. Figure 7-12 shows the main navigation menu.
Now that you have a good understanding of how to navigate the interfaces for both
Unity and Unity Connection, look at what needs to be configured before you start to add
users. The rest of this chapter is split in two parts: Unity and Unity Connection. This
enables you to jump directly to the portion that is of most interest to you.
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Figure 7-8
Search Users
Unity Integration Verification
Before Unity can be integrated with any system, you need to ensure that the all required
PBX configuration is completed. The configuration tasks required on the PBX are based
on the type of PBX with which you are integrating. This book only addresses the configuration tasks required when integrating with a Cisco solution. If you are integrating with
a traditional PBX, you need to refer to the installation guides and release notes available
at Cisco.com. A list of all integration guides can be found by searching “unity configuration guides” at Cisco.com.
At one time, more than 150 PBXs were supported and approved to work with Unity. That
number has been drastically reduced. The main reason that Cisco acquired Activoice, the
original developer of Unity, was so that Cisco could offer a feature-rich unified messaging solution that would integrate well with Communications Manager. This does not
mean that Cisco removed all support for traditional PBXs. There are times when
Communications Manager will be integrated with an existing PBX. By allowing Unity to
integrate with both Communications Manager and the existing PBX, Unity becomes a
much more attractive solution to some customers. Cisco has since added support for
other Voice over IP (VoIP) solutions, such as Session Initialization Protocol (SIP). The following sections discuss tasks that should be completed to ensure that the Cisco IP-based
solution with which Unity will integrate is configured properly.
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Figure 7-9
Edit User Basics
Communications Manager Integration
In this book, it is expected that most Unity installations will be configured to operate
with Communications Manager. Before Unity can communicate with Communications
Manager, certain tasks must be accomplished on the Communications Manager and the
Unity systems. In Chapter 6, “Configuring CUCM Features and Services,” the
Communications Manager side of the voicemail configuration was discussed. If necessary, refer back to that chapter to review the configuration of the required components.
Because the actual configuration steps have already been discussed, the following sections will focus on how to verify whether all required configuration has been completed.
Voicemail Port Configuration
Unity communicates with PBXs through a port. With traditional PBXs, the ports are
most often analog lines. To enable this communication, a voice board is installed in the
Unity server. The voice board usually has either four or 12 analog ports, which connect
back to analog ports on the PBX. As of Unity 8.0, voice boards are no longer supported,
and integration with a PBX is accomplished by using PMIG or TIMG, which is discussed
later. When Unity is integrated with Communications Manager, analog lines are not used.
Instead, virtual ports are configured on the Communications Manager. The only physical
connectivity between the Unity server and the Communications Manager is Ethernet.
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Figure 7-10
Edit Menu
When these ports are configured in Unity, a name and a Directory Number (DN) are
assigned to them. The DN is significant only to the Communications Manager. Unity
does not need to know what the DN is because Unity is concerned only with the name
assigned to the port. Typically, the name looks something like Cisco-UM-VI1. This naming convention is only a suggestion; any name can be used.
The Unity Telephony Integration Manager (UTIM) is used to configure the voicemail
port names in Unity. The UTIM verifies that Unity can communicate with the port after
you enter the name. If you are not sure of the name of the voicemail port, you can find it
by using the Communications Manager configuration interface, as outlined in the steps
that follow:
Step 1.
Log on to the CCMADMIN interface.
Step 2.
Select Advanced Feature > Voice-mail > Cisco Voice-mail Port.
Step 3.
Click the Find button. On the left side of the screen, all the voicemail ports
display. You are concerned with the prefix portion of the name. That is the
portion that displays before the number. For example, if the name were
CiscoUM-VI1, the prefix would be CiscoUM-VI.
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Figure 7-11
Change Password
Unity Telephony Integration Manager (Communications Manager)
The current integration settings can be found in the Unity’s SA interface but can only be
changed using UTIM. To view the integration settings, open SA and click Integration.
This can be found under the System heading on the left side. If there are multiple switch
types, you can view the desired integration by clicking the appropriate integration in the
list on the left side of the integration page.
The configuration for the integration is accomplished using UTIM. When configuring
UTIM, enter information about the Communications Manager with which you are integrating Unity. The following checklist provides information necessary before applying UTIM:
✓ Integration Name: This name is used to identify the integration. If the name is changed,
make sure that you use a name that makes it easy to identify the integration.
✓ Communications Manager Cluster Name: This name is used for reference only. Choose a
name that helps to easily identify the cluster.
✓ Communications Manager IP Addresses: Unity needs to know the IP addresses of both the
primary and secondary Communications Managers with which it will be communicating.
✓ TCP Port: This is the TCP port number that Unity will use for connecting to
Communications Manager. The default value is 2000. Leave it at the default.
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✓ Starting RTP Audio Port: This is the first IP port number that is used for Real-time
Transport Protocol (RTP) sessions. Leave this at the default.
✓ MWI On/Off DNs: Message Waiting Indicator (MWI) DNs are configured in
Communications Manager. When these digits are dialed on behalf of a phone, the MWI
lights, or goes on or off depending upon which DN is entered.
✓ Number of Ports: This is the number of ports for which you are licensed. If you are implementing a dual-switch solution, this is the number of ports that you want to have communicate with Communications Manager.
✓ Voice Mail Port Prefix Name: This is the name of the voicemail port, not including any
numbers at the end of the name. For example, if the voicemail ports were named CISCOUM1-VI1 through CISCOUM1-VI12, you would use CISCOUM1-VI as the prefix name.
✓ Port Settings: The voicemail ports can be set to perform multiple functions. Before you
installed Unity, you should have discussed what functions were to be handled by each
port. The functions are as listed in Table 7-1.
✓ Trunk Access Codes: Additional digits that Communications Manager must dial when
transferring a call to another PBX.
Note If licensing for the Audio Messaging Interchange Specification (AMIS) protocol
networking is purchased for this system, an additional parameter of AMIS Delivery can be
listed. This allows the port to send and receive AMIS messages. AMIS is discussed in more
detail in Chapter 10, “Implementing Unity Networking.”
Figure 7-12
Navigation Syntax Example
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Table 7-1
Voicemail Port Settings
Parameter
Description
Description
Enables the port to be used by Unity
Answer Calls
Enables the port to answer incoming calls
Message Notification
Enables the port to make outgoing calls for
message notification purposes
Dialout MWI
Enables the port to be used to send MWIs
Telephony Record and Playback (TRAP)
Connection
Enables the port to be used for recording and
playing messages when a subscriber is using
the web or an email client
UTIM is part of the installation process. If the Unity server is already loaded, this information should already be configured. You can check and change the integration information by running UTIM. To run UTIM, follow these steps:
Step 1.
On the Unity Server, choose Start > Programs > Unity > Manage
Integrations.
Step 2.
To show the Communications Manager clusters, from the UTIM window,
click the CUCM/CCM, which displays in the left column.
Step 3.
Select the name of the Communications Manager cluster that you want to
manage by clicking its name.
Note If more than one Communications Manager cluster displays, make certain that you
select the correct one. If configured properly, the name assigned to each cluster enables
you to determine which one you want to access. If the name does not help you identify the
cluster, you need to identify it by the IP address.
Step 4.
A screen similar to Figure 7-13 displays. There are four tabs along the upper
right portion of the screen labeled Servers, MWI, Ports, and Advanced.
Step 5.
Click the Servers tab to make certain that the server parameters are brought
to the front. From this screen, you can change the display name of the cluster,
the voicemail port device name prefix, and the cluster security mode. To
change any of these settings, simply enter the desired value. If you want to
change the IP address in the CM, TCP port, or TLS number, highlight the
address of the server that you want to change and click the Modify button.
Enter the new information and click OK.
Step 6.
Click the MWI tab to change the MWI values.
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Figure 7-13
UTIM Communications Manager Screen
Step 7.
To change the functions that each voicemail port is allowed to perform, click
the Ports tab and set each port. You can refer to Table 7-1 for an explanation
of each function.
Step 8.
If you need to change the starting RTP port number, select the Advanced tab
and set a starting number. Leave this value at the default.
Step 9.
After you have changed all the values you want, click Save. When you save,
you are prompted to restart Unity. Restarting Unity causes active calls to
drop, so it is best to do this after hours. A restart is required for the changes
to take effect.
Step 10. You can close the UTIM by clicking the X in the upper-right corner.
After you have verified that the information is correct, Communications Manager and
Unity should be able to communicate. The simplest way to check this is to dial the DN
assigned to the first voicemail port. If you are dialing from a subscriber’s phone, you
should be asked to enter a password; otherwise, you should hear the opening greeting.
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SIP Integration
Since the release of Unity 4.0, SIP integration has been supported. SIP is an open standard protocol that is used for call setup. SIP provides the same function in a SIP integration as Skinny Client Control Protocol (SCCP) does in a Communications Manager integration. The configuration required on the SIP proxy server is outside the scope of this
book. The next section describes the general configuration steps required on the SIP
proxy server. For more detailed information refer to “Cisco SIP Integration Guide for
Unity,” which can be found at Cisco.com. Currently Unity is supported with SIP when
using a Cisco SIP proxy server. The configuration steps that follow are based upon the
assumption that the Cisco SIP proxy server is being used.
SIP Configuration
If you are integrating with a SIP solution, it is assumed that you have a solid understanding of SIP components and how the protocol works. With this understanding in mind,
now look at an overview of the steps required on the SIP server side.
Of course, before you can integrate Unity into a SIP solution, you must have a fully
installed and functioning SIP solution. This includes the installation and setup of a Cisco
SIP proxy server.
After the server is installed, the phones need to be configured so that calls are forwarded
to the Unity system when the called party is on the phone or does not answer. You will
set up the phone to forward to the SIP address of the Unity system. The format of this
address is the same as all SIP addresses: host@server. The host portion of the address will
be the line number used for Unity.
On the SIP gateway, you need to configure an application session. This is accomplished
by configuring a VoIP dial-peer using the application session command.
On the SIP gateway, it is required that you disable the SIP media inactivity timer. This is
done by issuing the following command at the gateway configuration prompt:
no timer receive-rtcp
Finally, you need to configure a VoIP dial peer on the gateway that enables a dual-tone
multifrequency (DTMF) relay. The commands to employ this are issued at the start of the
configuration prompt and are as follows:
Dial-peer voice <dial peer number> voip
Session target ipv4:<ip_address)
Session protocol sipv2Dtmf-relay rtp-ntme
After you have made certain that all the SIP server configuration tasks are completed, you
need to configure the Unity portion of the integration.
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Table 7-2
Voicemail Port Settings
Parameter
Description
Enabled
Enables the port to be used by Unity
Answer Calls
Enables the port to answer incoming calls
Message Notification
Enables the port to make outgoing calls for message notification
purposes
TRAP Connection
Enables the port to be used for recording and playing messages
when a subscriber is using the web or an email client
Unity Telephony Integration Manager (SIP)
Just as with Communications Manager integration, UTIM is used to configure the integration parameters for a SIP integration. Before you run UTIM, you need to make sure
that you have the information that will be entered in UTIM. The following checklist provides the information necessary before applying UTIM:
✓ Integration Name: This name is used to identify the integration. If this name is changed,
make sure that you use a name that makes it easy to identify the integration.
✓ Cluster Name: This name is used for reference only. Choose a name that helps to easily
identify the server. This is also referred to as the Display Name.
✓ IP Addresses: Unity needs to know the IP addresses of both the primary and secondary
SIP proxy server with which it will be communicating.
✓ Number of Ports: These are the number of ports that will be used between the SIP server
and Unity. This can be the same number of ports for which the system is licensed, unless
you are using Unity in a dual-switch integration environment.
✓ Contact Line Number: This is the number that is associated with Unity. Unity registers
this name with the SIP proxy server.
✓ Unity Port: This is the TCP port number that Unity uses to connect to the SIP server.
Leave it at the default. Only change this number if you know that a different port is being
used for SIP in your environment.
✓ Preferred Codec: This is the codec that Unity first tries to use when placing outgoing calls.
✓ Authentication Name and Password: If SIP proxy authentication is being used, this is the
username and password that Unity uses.
✓ Port Settings: The voicemail ports can be set to perform multiple functions. Before you
installed Unity, you should have decided what functions were being handled by each port.
The functions are as listed in Table 7-2.
✓ Trunk Access Codes: Additional digits must be dialed when transferring a call to another PBX.
Note If licensing for AMIS networking is purchased for this system, an additional parameter of AMIS Delivery can be listed. This enables the port to send and receive AMIS messages. AMIS is discussed in more detail in Chapter 10.
When you have all the information available, you can configure the SIP parameters for an
existing integration or create a new integration using UTIM.
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To add a new SIP integration, follow these steps:
Step 1.
On the Unity Server, choose Start > Programs > Unity > Manage
Integrations.
Step 2.
From the UTIM interface, select Integration > New.
Step 3.
The Welcome to the Telephony Integration Setup Wizard window displays.
Select SIP as the phone system type. Click Next.
Step 4.
On the following screen, enter an integration name and a cluster name. The
integration name should be something that helps you easily identify which
phone system goes with this integration. The cluster name helps identify the
specific SIP server in this integration. The cluster name appears as the display
name when modifying this integration. After you have entered this information, click Next.
Step 5.
On the next screen, enter the IP addresses for the primary and secondary SIP
proxy server, and the port that will be used. Click Next.
Step 6.
The next screen requests the number of ports that you want to configure for
this integration. If you are integrating only with a single SIP system, you
should enter the total number of ports you have purchased. If you are integrating with multiple PBX systems, enter only the number of ports that you
have decided to give to the SIP portion of the integration. Click Next.
Step 7.
The following screen enables you to enter the Contact Line Name, Unity SIP
Port, Preferred Codec, and Preferred Transport Protocol. After entering this
information, click Next.
Step 8.
On the next screen, you must enter the username and password for SIP
authentication, if you are using it. If you are not applying authentication, deselect the check box and click Next.
Step 9.
If you have other integrations, the following screen requests the access code
to be used to send calls to the other systems. Click Next after entering this
information.
Step 10. The next screen allows you to reassign subscribers to this integration. If you
want to reassign subscribers to this integration, select the desired subscribers
and click Next.
Step 11. The following screen allows you to reassign all handlers to this integration. If
you want to reassign call handlers to this integration, select the desired call
handlers and click Next.
Step 12. The final screen shows a summary of what you have chosen. Click Finish to
complete the configuration.
Step 13. You are now prompted to allow Unity to restart. Restarting Unity causes
active calls to drop, so it is best done after hours. A restart is required for the
changes to take effect.
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To change an existing integration, follow these steps:
Step 1.
On the Unity Server, choose Start > Programs > Unity > Manage
Integrations.
Step 2.
From the UTIM window, click the SIP integration that will display in the left
column to show the SIP clusters.
Step 3.
Click the name of the SIP cluster that you want to manage.
Step 4.
A screen similar to that shown in Figure 7-14 displays. Notice that there are
four tabs along the upper portion of the screen: Servers, SIP Info, Ports, and
Advanced.
Figure 7-14
UTIM SIP Screen
Step 5.
Click the Servers tab to make certain that the server parameters are brought
to the front. From this screen, you can change the display name of the cluster.
To change the IP address or TCP port number, highlight the address of the
server that you want to change and click the Modify button. Enter the new
information and click OK.
Step 6.
You can change the Contact Line Name, SIP Port, Preferred Codec, and
Preferred Transport Protocol by clicking the SIP Info tabs and changing the
values. You can also change the SIP authentication settings from this screen.
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Step 7.
To change the functions that each voicemail port is allowed to perform, click
the Ports tab, and set each port as you desire. You can refer to Table 7-2 for
an explanation of each function.
Step 8.
The Advanced tab enables you to change RTP, DTMF timing, and disconnect
tone trim. It is recommended that these be left at the default.
Step 9.
After all changes are made, click the Save button. After changes are saved, you
are prompted to restart Unity, which will cause active calls to drop. It is best if
this is done after hours. A restart is required for the changes to take effect.
Step 10. You can close the UTIM by clicking the X in the upper-right corner.
After you have verified that the information is correct, Unity should be able to communicate within the SIP environments. The simplest way to check this is to dial the DN
assigned to the voicemail port. If you are dialing from a subscriber’s phone, you should
be asked to enter a password; otherwise, you should hear the opening greeting.
PIMG/TIMG Integration
In the past, voice boards were used to integrate with traditional PBXs. Since the release
of Unity 8.0, voice boards are no longer supported. To integrate with many traditional
PBXs, a PBX IP Media Gateway (PIMG) or T1 IP Media Gateway (TIMG) device must be
used. These devices have analog or digital ports (depending on the PBX they are connecting to) and an Ethernet port. The analog or digital port is used to connect to the PBX.
The device converts the signal from the PBX to SIP and sends the SIP signal out the
Ethernet port to the Unity server.
PIMG/TIMG Configuration
The PIMG/TIMG integration is similar to the Communications Manager and SIP integrations. In some ways, it is even a little simpler.
Unity Telephony Integration Manager (PIMG/TIMG)
Just as with a Communications Manager and SIP integration, UTIM is used to configure
the integration parameters for PIMG/TIMG integration. Before you run UTIM, you need
to make sure that you have the information that will be entered in UTIM. The following
checklist provides the information necessary before applying UTIM:
✓ Integration Name: This name is used to identify the integration. If this name is changed,
make sure that you use a name that makes it easy to identify the integration.
✓ PIMG type: You must select the model of PIMG that you are using from the drop-down list.
✓ SIP Port: This is the SIP TCP port number of the PIMG unit. It should always be 5060.
✓ Number of Phone Lines: These are the number of ports that will be used between the
PIMG and PBX. This can be the same number of ports for which the system is licensed,
unless you are using Unity in a dual-switch integration environment.
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✓ Contact Line Name: This is the number (name) that is used to contact Unity. It is also the
name that will be registered to the PIMG device.
✓ Display Name: This name is used for reference only. Choose a name that helps to easily
identify the server.
✓ IP Addresses: Unity needs to know the IP address of the PIMG device.
✓ Unity SIP Port: This is the TCP port number on the Unity server. Leave it at the default.
Only change this number if you know that a different port is being used for SIP in your
environment.
✓ Preferred Codec: This is the codec that Unity first tries to use when placing outgoing calls.
✓ Trunk Access Codes: Additional digits must be dialed when transferring a call to another
PBX.
When you have all the information available, you can configure the SIP parameters for an
existing integration or create a new integration using UTIM.
To add a new SIP integration, follow these steps:
Step 1.
On the Unity Server, choose Start > Programs > Unity > Manage
Integrations.
Step 2.
From the UTIM interface, select Integration > New.
Step 3.
The Welcome to the Telephony Integration Setup Wizard window displays.
Select Circuit-Switched via PIMG (including TIMG) as the phone system
type. Click Next.
Step 4.
On the following screen, enter an integration name. The integration name
should be something that helps you easily identify which phone system goes
with this integration. Click Next.
Step 5.
On the next screen, click Next and enter the display name, PIMG type, IP
address of the PIMG device, SIP port, and the number of phone lines connected. After this information is entered, click OK.
Step 6.
The following screen shows a summary of the information that you just
entered. Click Next.
Step 7.
If Simplified Message Desk Interface (SMDI) will be employed for this integration, select Yes. This PIMG unit is the serial master. If you are not using
SMDI, click Next.
Step 8.
The following screen allows you to enter the Contact Line Name, Unity SIP
Port, and Preferred Codec. After entering this information, click Next.
Step 9.
If you have other integrations, the following screen requests the necessary
access code to send calls to the other systems. Click Next after entering this
information.
Step 10. The next screen allows you to reassign subscribers to this integration. If you
want to do so, select the desired subscribers and click Next.
Step 11. The following screen allows you to reassign all handlers to this integration. If
you want to do so, select the desired call handlers and click Next.
Step 12. The final screen shows a summary of what you have chosen. Click Finish to
complete the configuration.
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Step 13. You are now prompted to allow Unity to restart. Restarting Unity causes
active calls to drop, so it is best done after hours. A restart is required for the
changes to take effect.
To change an existing integration, follow these steps:
Step 1.
On the Unity Server, choose Start > Programs > Unity > Manage
Integrations.
Step 2.
From the UTIM window, click the PIMG integration that will display in the
left column.
Step 3.
Click the name of the PIMG that you want to manage.
Step 4.
Notice that there are four tabs along the upper portion of the screen that
appear: PIMG, SIP Info, Ports, and Advanced.
Step 5.
Click the PIMG tab to make certain that the PIMG parameters are brought to
the front. From this screen, you can change the display name, the PIMG type,
the IP address, and the SIP port number used. To change any of these settings, simply enter the desired value.
Step 6.
You can change the Contact Line Name, Unity SIP Port, and Preferred Codec
by clicking the SIP Info tab and changing the values.
Step 7.
To change the functions that each voicemail port is allowed to perform, click
the Ports tab and set each port as you desire. You can refer to Table 7-2 for an
explanation of each function.
Step 8.
The Advanced tab allows you to change RTP, DTMF timing, and disconnect
tone trim. It is recommended that these be left at the default.
Step 9.
After all changes are made, click the Save button. After changes are saved, you
are prompted to restart Unity, which will cause active calls to drop. It is best if
this is done after hours. A restart is required for the changes to take effect.
Step 10. You can close the UTIM by clicking the X in the upper-right corner.
After you have verified that the information is correct, Unity should be able to communicate with the PIMG device. The simplest way to check this is to dial the DN assigned to
the voicemail port. If you are dialing from a subscriber’s phone, you should be asked to
enter a password; otherwise, you should hear the opening greeting.
Defining Unity System Configuration
Although it is possible to begin adding users now, you should first make sure that the
system settings are appropriate for your environment. Many technicians are often tempted
to accept system defaults and assume that they can go back and fix them later. In most
cases, you can go back and change any setting you need, but it is more efficient and less
time consuming if you had set things up properly at the beginning.
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Note I once had a student who had a configuration issue that could have been avoided if
the integrator had spent 2 minutes changing a few parameters before adding users. The
integrator was not familiar with the system and assumed that, if necessary, the parameters
could be changed later. To compound matters, the integrator then told the student how to
fix the problem but not the easy way. The problem could have been fixed in about 5 minutes but instead took 3 days. The integrator was not invited to bid on any more projects.
The old saying applies to such situations: An ounce of prevention is worth a pound of
cure. The issue in this case was that the template used was not configured to have the
user’s phone ring when someone dialed the extension from within the auto-attendant.
The following sections cover topics associated with system settings, such as schedules,
languages, recording, and port settings. Because these settings affect all subscribers on
the system, they can be thought of as global settings. If you log on to SA, you see the
settings that are listed under the System heading in the navigation panel on the left side
of the screen. The topics covered in these sections include configuration, schedules, holidays, authentication, and ports. You might notice that licensing and integration also display under the System heading in SA. However, these are informational only and not configurable from the SA. They are not covered in the section.
Note In some more recent versions of Unity, Short Message Peer-to-Peer (SMPP) support
was added. This feature enables you to send message notifications to Short Message
Service (SMS) devices, such as cell phones. More details on this feature can be found by
searching “Setting Up Client SMS Applications” at Cisco.com.
Creating Schedules and Holidays
Just because something is first in a list doesn’t mean it is the first thing you do. You have
probably noticed that we have started the configuration process on the last five major
headings in the navigation panel. The navigation menu was designed more for ease of use
than as an ordered list of tasks. So, just as we did not begin at the first major area, we are
not going to begin at the first area under System. Because you define a default schedule
in the Configuration area, it would only make sense to have the schedules defined first.
After defining schedules, adding holidays is discussed.
Schedules are assigned to subscribers, and call handlers and are used to define what
hours Unity should consider opened and closed. Unity has the capability to play up to
five different greetings. Two of these greetings depend on the schedule. The Standard
greeting plays during open hours, and the Closed greeting plays during closed hours.
A default schedule is defined and is assigned to all subscribers and call handlers when
they are created. After an object is created, you can assign a different schedule to it. A
total of 64 schedules can be defined.
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Schedules are configured at the System > Schedules level of SA. Step-by-step instructions on how to perform the following schedule-related tasks are as follows:
■
Viewing/changing an existing schedule
■
Adding a new schedule
■
Selecting a default schedule
■
Deleting a schedule
View and Change a Schedule
To view and change existing schedules, use the steps that follow:
Step 1.
From within SA, select System > Schedules.
Step 2.
From the title strip, select the Find icon (magnifying glass). A list of schedules
displays.
Step 3.
Highlight the desired schedule and click View.
Note The main part of the schedule screen is a grid that represents the seven days of the
week, divided into half-hour increments. Check marks indicate the open hours.
Step 4.
To modify the schedules, place check marks on the half hours that you want
to be considered open. Remove the check marks from times to be considered closed.
Note You can copy the open and closed hours of one day to another day, and to all
weekdays and weekends by using the Copy Day’s Schedule function at the bottom of the
screen. Select the source day from the first drop-down list and the destination day(s) from
the second drop-down list; then click Copy Day’s Schedule.
Step 5.
At the top of the screen, there is a check box labeled Observe Holidays. If
this box is selected, this schedule is treated as if all hours are closed on holidays that are created in Unity.
Step 6.
After all changes are made, click the Save icon in the title strip. The changes
do not take effect until you have saved them.
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Add a Schedule
To add a new schedule, similar steps are performed. The following are the required steps:
Step 1.
From within SA, select System > Schedules.
Step 2.
From the title strip, select the Add icon (blue plus sign). An Add a Schedule
window displays.
Step 3.
Enter the name of the new schedule. You can choose to base the new schedule on an existing one. Doing this causes the new schedule to have the same
open and closed hours as an existing one. Choose whether the new schedule
should be based on an existing one or not and click the Add button.
Step 4.
The new schedule displays.
Note The main part of the schedule screen is a grid that represents the seven days of the
week, divided into half-hour increments. Check marks indicate the open hours.
To modify the schedules, place check marks on the half hours that you want
to be considered open. Remove the check marks if you want that time to be
considered closed.
Note You can copy the open and closed hours of one day to another day, all weekdays,
or weekends by using the Copy Day’s Schedule function at the bottom of the screen.
Select the source day from the first drop-down list and the destination day(s) from the second drop-down list; then click Copy Day’s Schedule.
Step 5.
At the top of the screen, there is a check box labeled Observe Holidays. If
this box is selected, this schedule is treated as if all hours are closed on holidays that are created in Unity.
Step 6.
After all changes are made, click the Save icon in the title strip. The changes
do not take effect until you have saved them.
Define a Default Schedule
The default system schedule that is selected during installation is the Weekdays schedule.
The default schedule is changed at the System > Configuration > Settings level of SA. All
the settings at this level are discussed in the next section. This section deals only with the
default schedule portion of this level. To change the default schedule, follow these steps:
Step 1.
From within SA, select System > Configuration> Settings.
Step 2.
The first field on this screen is Default Schedule. Select the schedule from the
drop-down list.
Step 3.
Click the Save icon.
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To remove a schedule from Unity, follow these steps:
Step 1.
From within SA select, System > Schedules.
Step 2.
From the title strip, click the Find icon (magnifying glass). A list of schedules
displays.
Step 3.
Highlight the desired schedule, and click View.
Step 4.
The selected schedule displays on the screen. Click the delete icon (red X)
from the title strip.
Step 5.
You are prompted to determine to which schedule you want to reassign all
objects that are currently using this schedule. Select a schedule from the
drop-down list and click Delete.
Now that you have the schedules defined, the next logical thing to configure is the holiday schedule. Holidays are configured so that Unity knows which days the company will
be closed. Unity plays the closed greeting the entire day on holidays.
Add a Holiday
To add holidays, follow these steps:
Step 1.
From within SA, select System > Holidays.
Step 2.
The current list of holidays displays on the screen.
Step 3.
To add a new holiday, click the Add icon (the plus sign).
Step 4.
An Add a Holiday window displays. Select the date from the drop-down list
and click Add.
Step 5.
Repeat Steps 3 and 4 for each holiday that you want to add.
Note Keep in mind that you do not have to save these additions. Saving is required only
when an object is changed. When an object is added, it is immediately written.
You have the option of copying holidays that were defined for one year to another year.
This can be a great timesaver, but you need to make sure that the holidays you are copying are on the same date in the year to which you are copying them. To copy holidays
from one year to another, simply click the appropriately labeled button. This causes all
the holidays from the earlier year to be copied. Remember to remove or change any incorrect dates.
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Modify or Delete a Holiday
To change or delete a holiday, follow these steps:
Step 1.
From within SA, select System > Holidays.
Step 2.
The current list of holidays displays.
Step 3.
Click the holiday that you want to change or delete. The fields in the Edit
Holiday For field should now show the date you selected.
Step 4.
To change the holiday’s date, edit the fields in the Edit Holiday For area and
click the Save icon.
Step 5.
To delete this holiday, click the Delete icon. You are prompted to confirm the
deletion. Click OK.
Now that you have all the schedules and holidays configured, system settings should be
completed. The next section discusses a number of global system settings that should be
defined before the system is put into production.
Defining Configuration Settings
Configuration settings are general Unity settings that define schedules, security, languages, and a number of other things. All of these settings will be configured using the
SA interface. This section is divided into subsections so that you can easily follow the
settings shown here in the order in which they are presented in SA. If you go to the
System > Configuration level in SA, you see the eight selections that will be discussed in
this section. Table 7-3 lists these topics and provides a brief description of each.
All the settings discussed throughout this section are configured under the System >
Configuration level of the Unity Administrator pages.
Settings
The first configuration level under System > Configuration is Settings. From this level,
you configure items that deal with general settings within Unity. The following steps walk
you through the Settings screen. Figure 7-15 shows the screen that contains most of the
fields that will be discussed in these steps.
Step 1.
From within SA, select System > Configuration > Settings.
Step 2.
Select the schedule that you want to use as the default schedule from the
drop-down list.
Step 3.
If you want time to be expressed in 24-hour format (1900 hours instead of
7:00 p.m.), select the Use 24-Hour Time Format for Conversations and
Schedules check box.
Step 4.
Next there is a box labeled Enable Spelled Name Search. Typically you want
other subscribers to spell by name, so this box should be selected. If you
want to allow them to dial only by extension, deselect this box.
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Figure 7-15
Table 7-3
Settings Screen
SA Configuration Settings Selections
Topic
Description
Settings
General settings such as schedules, log files,
and RSA settings
Software Versions
List of the version number of each Unity
process
Recordings
Time limits for silence and clip length
Contacts
Name and number of those responsible for the
system
Phone Languages
Phones languages that are available and loaded
GUI Languages
Languages available and loaded for the Unity
Administrator pages
Message Security
Determines whether messages will be
encrypted
Message Subjects
Defines the formatting for message subject
heading
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Step 5.
In the Message Locator field, enter the number of times you want the “still
searching” prompt to play when the user is using message locator.
Step 6.
The Responding to Phone Menus fields define how long Unity waits for the
following actions:
■
Milliseconds that Unity waits for the user to enter the first digit
■
Milliseconds that Unity waits for the user to enter additional digits when
entering names or extensions
■
Milliseconds that Unity waits for the user to enter additional digits when
entering phone menu commands
■
The number of times that Unity replays the menu if the user does not
respond
Enter the desired values if each field.
Step 7.
If you have implemented an RSA server for use with Unity, select the RSA
Two Factor check box. Typically this check box is deselected.
Step 8.
The check box labeled Display Fields Required for Cisco Unity Bridge
Networking on Subscribers Profile Page is only used if you are using a bridge
server to integrate with an Octel system. Typically this box is left deselected.
Step 9.
If you want messages that are left by a subscriber to include the subscriber’s
name or ID, leave the Disable Identified Subscriber Messaging check box
selected. If this box is selected, the subscriber leaving the message will have
to log in to be identified.
Step 10. Enter the URL used to access the Cisco Personal Communications Assistant
(CPCA) pages in the Internet Address field. Unity includes this address in an
email message notification. If CPCA is not being used, leave this field blank.
Step 11. Set the number of days that you want to keep log files in the Cleanup
Interval Portion of the Files Cleanup field for each type of log.
The Settings screen contains four more areas. Three of these areas are not configurable
and are used to initiate replication or to view information. The last area enables you to
configure substitute objects. Figure 7-16 shows these areas.
The first area is titled Replicate Cisco Unity Directory Objects. You can force a replication between AD and Unity to occur from here. When you click the Changed Objects or
the All Objects button, Unity replicates with AD immediately. The button that you select
determines whether you want to force a complete replication or only look for changed
objects. This action should be performed only during off hours.
The next area is titled Cisco Unity Computer Settings. This area displays the computer
name of the Unity server and the domain to which it belongs.
The third area on this screen is titled Disk Usage. This area displays the number and size
of the logical hard drives in the Unity server.
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Figure 7-16
Settings Screen (Cont’d)
The final area is titled Substitute Objects and contains four configurable parameters:
Recipient, Owner, After Message Call Handler, and Exit Call Handler.
The Recipient parameter is used to define where messages should be delivered when they
are left to a call handler or interview handler whose original recipient has been deleted.
The Owner parameter is used to define which subscriber or public distribution list will
become the owner of an object if the object’s current owner is deleted.
The Recipient and Owner parameters are configured in the same way, so the following
steps show how to configure either:
Step 1.
Select either Subscriber or Public Distribution List from the Recipient dropdown list.
Step 2.
The button next to this field is labeled either Select Subscriber or Select
Distribution List based on whether Subscriber or Public Distribution List
was selected. Click this button.
Step 3.
If you are choosing a subscriber, a window displays that allows you to enter
search criteria. After they are entered, click the Find button and select the
desired subscriber from the results listed in the Matching Subscribers box. If
you are choosing a public distribution list, a window displays containing the
lists. Select the desired public distribution list.
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Step 4.
After you have selected the subscriber or distribution list, click the Select
button.
The last two parameters are similar in function and configuration. The first of the two
deals with where a call is sent if the subscriber selected for the After Message Action of
an object is deleted. The second defines where a call is sent when it exits a directory handler if the original destination object has been deleted. The following steps can be used
to configure either of these parameters:
Step 1.
Click the Select Call Handler button.
Step 2.
A window displays that allows you to enter search criteria. After they are
entered, click the Find button and select the desired call handler from the
results listed in the Matching Call Handlers box.
Step 3.
Click the Select button.
Software Versions
The second configuration level under System > Configuration is Software Settings. This
area is strictly informational. The page is used to find the version of several Unity components and Windows version information. Because configuration cannot be performed on
this screen, we move on to the next area.
Recordings
The third configuration level under System > Configuration is Recordings. The
Recordings page deals with time limits for silence and clip length settings. Figure 7-17
shows this screen and all its settings.
To set these configurations, follow these steps:
Step 1.
From within SA select, System > Configuration > Recordings.
Step 2.
Select the amount of time you want to have clipped off the end of a recording
when the octothorpe (#) is pressed in the field labeled Allowed Time for
Recording, in Milliseconds. This is performed so that the tone produced
when the octothorpe (#) is pressed is not included in the messages. This value
normally does not need to be changed unless the tone is being heard in messages or the ends of messages are being clipped off.
Note The octothorpe is referred to by different names in different countries. For example, it is called the pound sign in the United States and the hash in England.
Note The next value, labeled Allowed Time for Short Recording, in Seconds, cannot be
changed. This value determines the permitted length of a message before Unity considers it
a long message. This is statically set to 10. In Step 5, you see how the value is used.
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Figure 7-17
Recordings Screen
Step 3.
Determine how long Unity will wait for someone to start talking by entering
a value in the box labeled Allow How Much Silence Before Time Out, in
Seconds. If the caller leaving a message does not begin talking within the
amount of time set here, Unity assumes that the caller is not going to leave
a message.
Step 4.
The field labeled Discard Any Recording Less Than, in Seconds is used to
prevent Unity from recording hang-ups. Typically, a value of 1 is sufficient for
this field. Enter the desired value.
Step 5.
The last two fields on this page determine how many seconds of silence Unity
allows in a message. Two different values are set here. The first one, labeled
Short Recording, is the value used during the first ten seconds of the message. As discussed in the note following Step 2, a short recording is defined as
10. The second value, labeled Long Recording, is used when the message that
is being recorded is longer than 10 seconds. Typically, these values can be left
at the default.
Step 6.
Click the Save icon at the top of the page.
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Contacts
The fourth configuration level under System > Configuration is Contacts. In the Contacts
area, administration and support contact information is entered. This information helps
others determine who is responsible for the system. Figure 7-18 shows this screen.
Figure 7-18
Contacts Screen
To enter this information, go to System > Configuration > Contacts in SA and enter the
appropriate names and numbers. When you are finished, click the Save icon.
Phone Languages
The fifth configuration level under System > Configuration is Phone Languages. This is
where you determine the phone languages that will be active on the Unity system. The
phone language is the language that the Unity system employs when talking to a caller.
One of the languages loaded must match the Windows operating system locale that was
selected during the Windows install. Figure 7-19 shows this screen.
To configure the Phone Language settings, follow these steps:
Step 1.
From within SA, select System > Configuration > Phone Languages.
Step 2.
The number of languages for which the system is licensed, and the number
loaded, display at the top of the page under the heading License Counts. If
the system is licensed for more than one language, add any of the languages
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listed in the field labeled Available. Highlight the desired language and click
the -> arrow. The language should now display in the loaded field.
Figure 7-19
Phone Languages Screen
Note Only languages that were chosen during the installation display are available. To
add languages that do not display in the list, you must run the Cisco Unity Installation and
Configuration Assistant (CUICA). Because running CUCIA causes some downtime, it
should be run during a planned off-hours period.
Step 3.
From the drop-down list labeled Default Phone Language, select the language that you want Unity to employ by default for all objects.
Step 4.
From the drop-down list labeled Default Text-To-Speech Language, select
the language that you want Unity to use by default when reading emails to
subscribers over the phone.
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GUI Languages
The sixth configuration level under System > Configuration is GUI Languages. This is
the language in which the SA will be displayed. Figure 7-20 shows this screen.
Figure 7-20
GUI Languages Screen
To access this configuration screen, go to System > Configuration > GUI Languages
from within SA. You can move any language that displays in the Available list over to the
Loaded list by highlighting the desired language and clicking the -> arrow.
Message Security
The next setting is Message Security. Here you can define whether messages should be
encrypted. You have the choice to encrypt all messages, no messages, or only messages
marked private. You can also enable message aging. If enabled, encrypted messages will
automatically be unavailable after the specified time.
Message Subjects
The Message Subjects settings determine how the subject line of a message will appear. In
most cases, these can be left at the default. However, you can change them if you desire.
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Configuring Authentication Settings
Now examine the next area of the system, the authentication settings. This encompasses
the authentication settings used when a subscriber logs in to CPCA. If you have chosen
to employ anonymous authentication for SA, these settings apply to SA and Status
Monitor logins as well. The configuration of these settings, which is outlined in the steps
that follow, is fairly simple:
Step 1.
From within SA, select System > Authentication.
Step 2.
The screen shown in Figure 7-21 displays. The first configuration on this
screen determines whether the users’ logon names are stored on their local PC
and, if so, the length of time stored. To have the logon name stored, select the
Remember Logons For check box and enter the number of days that you
want it to be stored in the box to the right of the label. If you do not want the
logon name stored on the client, make sure that this box is deselected. For
security reasons, it is recommended that you do not have the login names
stored on the client.
Figure 7-21
Step 3.
Authentication Screen
If you chose to have the logon name stored, the next field will be available for
configuration. This field determines whether the users’ passwords are stored
on their local PC and, if so, the length of time stored. To have the password
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stored, select the Remember Passwords For check box, and enter the number
of days you want to have it stored in the box to the right of the label. If you
do not want to have the password stored on the client, make sure that this
box is deselected. For security reasons, it is recommended that you do not
have the passwords stored on the client.
Step 4.
The Session Duration field allows you to set how long a subscriber remains
logged in to the interface if no activity has taken place. In other words, if the
session is idle for the amount of minutes entered in this field, the subscriber is
automatically logged out. The default is 20 minutes, and in most cases, this
should be adequate.
Step 5.
The third configuration field on this page determines whether blank passwords are allowed. To allow blank passwords, deselect the Disallow Blank
Password check box. Select this box if you do not want to permit blank passwords. For security reasons, it is recommended that this field be checked so
that blank passwords cannot be used.
Step 6.
The next check box is used to enable lockout polices. If you want to limit the
number of times users can enter the incorrect password, this box should be
selected. If this box is left deselected, users can enter incorrect passwords an
unlimited number of times.
Step 7.
If lockout polices are enabled, limits must be set. Enter the number of minutes that a user will be locked out in the box labeled Accounts Are Locked
Out For. Enter the number of times that users can enter incorrect passwords
before they are locked out in the box labeled Accounts Will Lock Out After.
Enter the number of minutes in which Unity will reset the login attempts in
the box labeled Reset Account Lockout Counters After.
Step 8.
Click the Save icon.
Configuring Ports
Ports is the last area of system configuration in this section. As discussed previously,
voicemail ports are used for communication between Unity and Communications
Manager. These ports are created on Communications Manager. As shown earlier in this
chapter, the Unity portion of configuration of these ports can be done using UTIM.
However, these ports can also be modified from within SA. Regardless of how these
ports are configured, the results are the same. Figure 7-22 shows the System > Ports
screen. From this interface, you can properly configure the ports according to their
desired purpose. Table 7-4 lists the functions a port can be configured to perform.
Note If licensing for AMIS networking is purchased for this system, an additional parameter of AMIS Delivery can be listed. This allows the port to send and receive AMIS messages. AMIS is discussed in more detail in Chapter 10.
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Figure 7-22
Table 7-4
Ports Screen
Voicemail Port Settings
Parameter
Description
Extension
Should contain the extension number that is
configured in Communications Manager for
this port
Enabled
Enables the port to be used by Unity
Answer Calls
Enables the port to answer incoming calls
Message Notification
Enables the port to make outgoing calls for
message notification purposes
Dialout MWI
Enables the port to be used to send MWI
notifications
TRAP Connection
Enables the port to be used for recording and
playing messages when a subscriber is using
the web or an email client
Before you start to configure the port, decide what functions are needed and which ports
will provide these functions. While making these decisions, keep in mind that outbound
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calls are sent from the higher-numbered ports and incoming calls are answered on the
lower-numbered ports. With this in mind, it makes sense to configure the higher-numbered ports for functions such as MWI and message notification. This leaves the lowernumbered ports available to answer incoming calls. It is recommended that approximately
75 percent of the ports should be configured to answer calls, whereas 25 percent should
be configured for MWI and message notification. As you can see in Figure 7-22, other
than the extensions, all values are enabled or disabled by selecting and deselecting the
appropriate check box. To configure the ports, locate System > Ports in SA, enter the
extension number, and enable the desired function(s) for each port. Although it is not
required to configure the extension number in these fields, it is recommended.
After the system configuration settings are completed, there are a few more tasks to
attend to before loading subscribers. The next sections discuss how to configure system
access rights and account policies.
Configuring Unity System Access and Policies
Andrew Grove, the chairman of Intel, wrote a book called Only the Paranoid Survive.
Although this might seem like an extreme title, there is much truth to it, especially when it
comes to computers. Because Unity has the capability to access both voicemail and email,
it is important to secure your system. Part of the securing process is setting up account
polices and class of service (CoS). Account policies allow you to define parameters for
subscribers’ phone passwords. These parameters can prevent subscribers from using passwords that could easily be compromised. CoS defines the rights a subscriber has within
the system, including the accessibility to Unity and the features they can use. Because
these settings affect subscribers, it is best to configure them before adding subscribers.
Defining Account Polices
Account policies are basically about securing the subscribers’ access to Unity. The link to
Account Policy is located under the major heading of Subscribers. Two things are configured here: password restrictions and phone lockout policies.
We examine the configuration of password restrictions first. Password requirements are
set here, including when a password expires, the required length of a password, and
whether the password must be unique when changed. In Figure 7-23, you can see the settings for this screen.
To configure phone password restrictions, follow these steps:
Step 1.
From within SA, select Subscribers > Account Policy > Phone Password
Restrictions.
Step 2.
Set the length of time for which passwords are valid. Click the Days Until
Password Expires radio button and enter the number of days for which the
password will be valid. This determines how often subscribers have to change
their passwords. If you have decided to allow the users to never have to
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change their passwords, click the Password Never Expires radio button. It is
recommended that you set passwords to expire.
Figure 7-23
Phone Password Restrictions Screen
Step 3.
Click the labeled Minimum Number of Characters in Password radio button
and enter the desired number in the corresponding field. If you want to permit
blank passwords, click the Permit Blank Password radio button instead. It is
recommended that a password be required and that it be at least eight digits.
Step 4.
Click the Number of Passwords to Remember radio button. In the corresponding field, enter the number of previously used passwords that you want
Unity to keep in its history. The subscriber cannot use any previous password
if it is still in Unity’s history. If you do not want Unity to keep a history, click
the Do Not Keep Password History radio button. This allows a subscriber to
reuse the same password when the password expires. It is recommended that
Unity keep at least three passwords in history.
Step 5.
The last thing to configure is whether Unity should check against trivial passwords. Any of the following are considered trivial passwords:
■
Password with all the same digits
■
Consecutive digits
■
The subscriber’s extension
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Select the Check Against Trivial Passwords for Extra Security check box if
you want this feature to be active. It is recommended that this be selected.
Step 6.
Click the Save icon.
The second area of configuration at this level is Phone Lockout Polices. These
settings determine how many times a subscriber can enter the incorrect phone
password before being locked out. The length of time a subscriber will be
locked out of the account is also configured here. The steps that follow
describe how to configure the settings that are shown in Figure 7-24:
Figure 7-24
Phone Lockout Policies Screen
Step 1.
From within SA, select Subscribers > Account Policy > Phone Lockout
Polices.
Step 2.
Check the Account Lockout radio button.
Step 3.
Enter the number of times that a subscriber can enter the incorrect password
in the Lock Account After (Field) Invalid Attempts field. If the No Account
Lockout radio button labeled is checked, the account never locks regardless
of how many times invalid passwords are entered. For security reasons, it is
recommended that the Lock Account After (Field) Invalid Attempts radio
button be selected.
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Step 4.
Enter the number of minutes Unity waits before resetting the attempts counter in the Reset Count After (Field) Minutes field.
Step 5.
The last thing to configure on this screen is how long the account will be
locked out if a subscriber exceeds the maximum invalid attempts. If you want
the account to be locked until a system administrator unlocks it, click the
Forever radio button. You can also choose to have the account locked for a
specified amount of time by clicking the Minutes radio button and entering
the number of minutes in the corresponding field.
Step 6.
Click the Save Icon.
Now that account policies have been defined, CoS needs to be addressed.
Configuring Class of Service
Each subscriber in the system has an associated CoS that is basically a list of rights and
restrictions. The CoS defines how subscribers are allowed to access the system and how
they can use the system after they have accessed it. Because subscribers are assigned to a
CoS when it is created, it is best to have all CoSs composed before adding users.
The CoS in Unity components fit into eight categories. Navigate to Subscribers > Class of
Service in SA; the eight categories display as submenus under the Class of Service heading. Table 7-5 provides each submenu and a brief description of each of the categories.
Before you configure CoS, it is best to record how many CoSs you need and how each
will be configured. Typically, there are at least two CoSs: one for administrators and one
for basic users. Unity has two default CoSs, so if you need only two, you can simply edit
the defaults.
It is a good idea to leave the Default Administrator CoS alone. This is the CoS that allows
administrative access. If you need a limited-administrative-access CoS, it is best to create
a new one.
The following tasks must be performed to add and edit a CoS. Each task is followed by
step-by-step instructions.
Adding a CoS
Step 1.
From within SA, select Subscribers > Class of Service > Profile.
Step 2.
Click the Add icon (plus sign) in the title strip.
Step 3.
A window displays, as shown in Figure 7-25, prompting you for a CoS name.
From this screen, you can also choose to create the new CoS based on an
existing one. This is useful if you need a new CoS that is similar to an existing one. Enter a name and select the CoS that you want to use as a base, or
select New CoS.
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Figure 7-25
Step 4.
Adding a CoS
Click Add and the new CoS is added. You are not required to save it. The saving action is required only when an object is modified, not when it is created.
Modifying a CoS
Modifying a CoS might require the configuration of all eight CoS categories or perhaps
only one. For this reason, we will examine the configuration of each of these categories
separately. However, before you can modify a CoS, you must first find it.
Finding a CoS to Modify
To find the CoS that you want to modify, follow these steps:
Step 1.
From within SA, select Subscribers > Class of Service > Profile.
Step 2.
Click the Find icon. A list of CoSs displays.
Step 3.
Select the CoS that you want to modify and click View.
Modifying a CoS Profile
Now that the CoS has been found, the first configuration area is the CoS profile. The following steps are required to modify the profile of a CoS:
Step 1.
From within SA, select Subscribers > Class of Service > Profile.
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Table 7-5
CoS Components
Category
Description
Profile
Defines the name of the CoS and subscriber’s
name setting and directory listing.
Subscribers
Enables assignment of subscribers to CoS.
System Access
Defines the level of administrative access for a
subscriber.
Transfer
Specifies whether a subscriber can use Unity’s
holding and screening features.
Messages
Defines the message length and other messagerelated settings.
Greetings
Defines the greeting length.
Licensed Features
Defines which licensed features the sub
scriber has.
Restriction Tables
Defines locations to which Unity can make
calls on behalf of a subscriber. Restriction
tables would be used, for example, for message
notification.
Step 2.
As you can see in Figure 7-26, the first field is the name of the CoS. If you
want to change it, enter the new name.
The next two settings determine whether a subscriber is allowed to add or
modify alternate extensions. Alternate extensions are used to associate a
phone number other than the subscriber’s extension to a subscriber’s mailbox. This is typically used to associate a subscriber’s cell phone number to
the mailbox. When subscribers call from their cell phones, they are prompted
to log in.
Step 3.
To allow the subscriber to view alternate extensions that are defined by the
administrator, select the Subscribers Can View Administrator-Defined
Alternate Extensions radio button. To allow subscribers to add and manage
their own alternate extensions, select the Subscribers Can Add, Modify and
Delete Their Own Alternate Extensions radio button.
Step 4.
Click the Save icon.
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Figure 7-26
CoS Profile
Modifying CoS Subscribers
You can also modify which subscribers are assigned to a CoS. During the creation
process, a subscriber is assigned to a CoS. The CoS that subscribers are assigned to is
determined by the subscriber template that is used. If, for any reason, you need to move a
subscriber to a different CoS, follow these steps:
Step 1.
From within SA, select Subscribers > Class of Service > Subscribers. The
screen shown is Figure 7-27 displays.
Step 2.
As you can see, three actions can be performed. You can view the subscribers
in this CoS, move (reassign) subscribers to a different CoS, or add (assign)
subscribers to this CoS. Click the radio button associated with the action that
you want to perform.
Step 3.
To view subscribers in this CoS, click the View radio button and click the
Find button. You can enter search criteria in the field in front of Find to narrow your search. A list of subscribers that match the criteria is displayed.
Step 4.
To move a subscriber from the current CoS to another CoS, click the
Reassign radio button. Figure 7-28 shows a screen that is slightly different
than Figure 7-27. Enter search criteria in the field in front of the Find button
and click Find. From the list that displays, select the subscriber who you want
to move. Now, from the drop-down list next to the Reassign button, select
the CoS to which you want to move him. Click Reassign.
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Figure 7-27
CoS - Subscribers (View)
Figure 7-28
CoS - Subscribers (Reassign)
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Step 5.
To move a subscriber to the current CoS, click the Assign radio button.
Enter search criteria in the field in front of the Find button and click Find.
From the list that displays, select the subscriber who you want to add. Click
Assign.
Step 6.
It is not necessary to save these changes, as they are saved automatically.
Modifying CoS System Access Settings
The CoS of subscribers determines the administrative level they have on the system. A
number of rights can be granted. Table 7-6 shows these rights and describes what access
can be assigned to them.
Table 7-6
CoS Subscriber Rights
Enable
Read
Edit
Add
Delete
CoS access
X
X
X
X
Directory handlers access
X
X
X
X
Subscribers access
X
X
X
X
X
X
X
X
UA access
Unlock accounts
X
X
Public distribution lists
Schedules/holidays
X
Restriction tables access
X
Routing tables access
X
Call handlers access
X
Status monitor access
X
Reports access
X
Network access
X
Diagnostic access
X
Technician functions
access (configuration,
licensing, ports, and
switch pages)
X
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Figure 7-29 illustrates these fields as they display on the screen.
Figure 7-29
CoS - System Access
To assign system access rights to a CoS, follow these steps:
Step 1.
From within SA, select Subscribers > Class of Service > System Access.
Step 2.
Select the Cisco Unity Administrator Application Access check box.
Step 3.
Select the appropriate boxes to assign the rights that you want this CoS to have.
Step 4.
Click the Save icon.
Modifying CoS Transfer Settings
A CoS can define whether a subscriber can use the call-screening and -holding options.
To configure these options, follow these steps:
Step 1.
From within SA, select Subscribers > Class of Service > Transfer.
Step 2.
To allow subscribers to determine whether Unity announces who is calling
before they accept the call, select the Subscribers Can Change Call
Screening Options check box.
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Step 3.
To allow subscribers to configure Unity to place an incoming call on hold
when they are already on the phone, select the Subscribers Can Change Call
Holding Options check box.
Step 4.
Click the Save icon.
Modifying CoS Message Settings
In the CoS message setting area, four things are defined:
Step 1.
The length of a message that a subscriber with this CoS can leave for another
subscriber.
Step 2.
Whether a subscriber can send messages to public distribution lists.
Step 3.
Whether messages that a subscriber deletes from the phone are moved to the
Deleted Items folder.
Step 4.
Whether, after listening to a message, a subscriber can have Unity dial the
extension number of another subscriber or outside caller who left the message. This feature is called Live Reply.
Figure 7-30 shows the configuration screen for these settings.
Figure 7-30
CoS - Message Settings
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To configure these settings, follow these steps:
Step 1.
From within SA, select Subscribers > Class of Service > Messages.
Step 2.
In the Maximum Length of Message Subscribers Can Record, in Seconds
field, enter the maximum length in seconds of a message from a subscriber
with this class of service, to another subscriber.
Step 3.
Select the Subscribers Can Send Messages to Public Distribution Lists
check box if you want to allow subscribers with the CoS to send messages to
distribution lists.
Step 4.
Select the Deleted Messages Are Copied to the Deleted Items Folder check
box to allow messages that are deleted from the phone to be stored in the
Deleted Items folder. Keep in mind that enabling this can have a negative
impact on disk space.
Step 5.
To allow subscribers to use the Live Reply feature to reach other subscribers,
select the Subscribers Can Reply to Messages from Other Subscribers by
Calling Them check box.
Step 6.
To allow subscribers to use the Live Reply feature to reach unidentified
callers, such as an outside caller, select the Subscribers Can Reply to
Messages from Unidentified Callers by Calling Them check box.
Step 7.
Click the Save icon.
Modifying CoS Greetings Settings
The greetings configuration for CoS defines the maximum length of a greeting. To configure the settings, follow these steps:
Step 1.
From within SA, select Subscribers > Class of Service > Greetings.
Step 2.
Set the maximum number of seconds that you want for a subscriber’s greeting in the Maximum Greeting Length, in Seconds field.
Step 3.
You can configure a message that callers must listen to before they can leave a
message. This recording plays after the subscriber’s greeting. Select the Play
Recording Only for Unidentified Callers radio button to create a recording
that is heard only by outside callers, or select the Play Recording to All
Callers radio button to record a message that all callers hear.
Step 4.
Use the Click to Record button to record a message. Your phone should ring
and you can use it to record the message. If your phone doesn’t ring, make
sure that you have the phone selected as the recording device. You can confirm this by clicking the down arrow next to the word Recording and select
Recording Devices from the drop-down menu. More information on the
record functions can be found later in this chapter in the “Creating Public
Distribution Lists” section.
Step 5.
Click the Save icon.
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Modifying CoS Features Settings
The features to which a subscriber has access are, in part, determined by the features
configured in the CoS to which they are assigned.
The features that can be assigned to a subscriber are determined by the license that was
purchased. The following is a list of the features that you might be able to assign
through the CoS:
■
FaxMail: Fax management through the phone
■
Text-to-Speech: Emails to be heard over the phone
■
Unity Assistant: Subscriber’s settings managed through the web
■
Unity Inbox: Voicemail retrieved through the web
To configure these settings, follow these steps:
Step 1.
From within SA, select Subscribers > Class of Service > Features. The
screen shown in Figure 7-31 displays.
Figure 7-31
CoS - Licensed Features
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Step 2.
Select either Voice Messaging or Unified Messaging for the type of subscriber. A voice-messaging subscriber is someone who uses Unity exclusively
for voicemail. A unified-messaging subscriber is someone who can access a
single mailbox for voicemail, email, and faxes.
Step 3.
Select the check boxes of the features that you want subscribers of this
CoS to have.
Step 4.
Two settings determine the size and quantity of personal distribution lists
available to a subscriber. Enter the maximum number of lists that a subscriber
can have in the Lists Available to the Subscriber field. Enter the maximum
number of members that each list can contain in the Maximum Members per
List field.
Step 5.
Click the Save icon.
The last area to be configured in CoS is Restriction Tables. Because restriction tables deal
with call management, they are covered in Chapter 9, “Call Management.”
Now that the CoS has been configured, we are almost ready to begin the process of
adding subscribers, which is covered in Chapter 8, “User/Subscriber Reference.” Before
moving on, one last topic needs to be discussed—public distribution lists.
Creating and Managing Unity Public Distribution Lists
The final material discussed in this section is public distribution lists (PDL). PDLs allow a
single message to be sent to a group of people. Although subscribers can be added to
PDLs manually, it is more efficient to add them during subscriber creation. This can be
accomplished because the template that is used during subscriber creation can contain
the PDLs to which the subscriber is assigned. PDLs can contain both subscribers and
other PDLs. For example, you might have three groups, one for Detroit sales, one for
New York sales, and one for Chicago sales. Each of these groups will contain subscribers
from the associated location. You could then create a PDL called USSales and simply add
the three sales PDLs, instead of having to add each subscriber again.
Creating Public Distribution Lists
To create and configure PDLs, follow these steps:
Step 1.
From within SA, select Subscribers > Public Distribution Lists > Profile. The
screen shown in Figure 7-32 displays.
Step 2.
To add a new PDL, click the Add icon (plus sign). Enter the name of the PDL
in the window that displays. You can choose to create a completely new PDL,
base it on an existing list, or import an Exchange group. Choose the appropriate option for this group and click Add.
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Figure 7-32
Public Distribution Lists
Step 3.
On the Profile screen, as shown in Figure 7-26, notice that the name and
owner fields are populated. Each PDL must have an owner. By default, the
owner will be the subscriber that created the PDL.
Step 4.
You can record a name for this PDL by clicking the Record button on the
media master control panel. The panel that has VCR-like buttons is the media
master, and the Record button is the round (red) button. The recorded name is
played back when a subscriber dials the PDL by name.
Note It is a good idea to become familiar with the media master interface because you
will be using it within SA when recording messages for objects, such as call handlers and
subscribers. The first time that you click the Play or Record button on the media master
panel, the Phone Record and Playback Settings window displays. This window prompts
you for the extension number of the phone that you will use for recording and the name
of the Unity server. This displays only the first time that you use this interface from a particular PC.
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Figure 7-32 illustrates the four buttons on this control panel. The first is an arrow that
points down. When this button is clicked, you are presented with a menu that offers the
following eight options:
■
Paste: Enables you to paste audio that is on the clipboard into the current
recording
■
Paste from File: Enables you to paste an audio file into the current recording
■
Copy: Enables you to copy the current recording to the clipboard
■
Copy to File: Enables you to copy the current recording to a file
■
Playback Devices: Enables you to choose to use either a phone or the sound card
of the PC for playback
■
Record Devices: Enables you to choose to use either a phone or the sound card
of the PC for recording
■
Options: Enables you to change the name of the server and the extension that
you entered when the Phone Record and Playback Settings box appeared, the
first time you used the media master on the PC
■
About: Displays version details for the media master
Note The next button is the Play button (an arrow that points to the right). This arrow is
blue if there is a recording already present for the object. To listen to the recording, click
the Play button. The phone should ring, and the recording is played when the phone is
answered.
The third button is the Stop button (square). To stop the recording, click this button.
The final button is the Record button (red circle). To record a message, click the Record
button. The phone should ring. Answer the phone and record the message after the beep.
In some cases an informational window displays in SA asking whether you want to overwrite the current recording before you can record the message.
Take a few minutes to get use to this interface, because you will find yourself using it often
within SA.
Step 5.
You can assign an extension by entering the desired extension number in the
Extension field. An extension is not required but is recommended. If a new
extension is not assigned, a subscriber can only send a message by using the
phone to dial the PDL by name.
Step 6.
If you want the PDL to display in the email address book, select the Show
Distribution List in E-mail Server Address Book check box.
Step 7.
Click the Save icon.
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Managing PDL Members
As mentioned earlier, the easiest way to add a subscriber to a PDL is at the point of creation. You can also add or make changes afterward manually from within SA. To view,
add, or remove members from PDLs, follow these steps:
Step 1.
Navigate to Subscribers > Public Distribution Lists > Members of the PDL
that you want to manage. The screen shown in Figure 7-33 should display.
Figure 7-33
PDL Members
Step 2.
You have the option of viewing, adding, or removing members from the PDL.
Select an action by clicking the appropriate radio button at the top of the
screen. If you choose to add members, you must select either Selected
Subscriber or Public Distribution Lists from the drop-down list.
Step 3.
Enter the search criteria for the subscriber (or public distribution list) that you
want to manage in the field to the left of the Find button and click Find. If
you are simply viewing, the PDL list displays. If you are adding or removing, a
list displays with either an Add to List button or a Remove button. Select the
subscriber that you want to manage and click the Add or Remove button.
Step 4.
Click Save.
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The settings that are configured in this chapter affect the entire system. Now that the system is set up, you must conduct tests to ensure that the configurations you made are
going to deliver the results you are anticipating. Take the time to test that the various settings you configured, such as schedules, holidays, and port functions, to name a few, are
configured properly.
Unity Connection Integration Verification
In most environments, Unity Connection is integrated with CM. However, CM is not the
only system that can integrate with Unity Connection. When integrating with CM, it can
use SCCP or SIP. To integrate with other phone systems, SIP is used. When integrating
with a Communications Manager Express system, it can use SIP straight through because
CME understands SIP. However, to integrate with many traditional PBXs, a PIMG or
TIMG device must be used. These devices have analog or digital ports (depending on the
PBX it is connecting to) and an Ethernet port. The analog or digital port is used to connect to the PBX. The device converts the signal from the PBX to SIP and sends the SIP
signal out the Ethernet port to the Unity Connection server.
This book focuses on the Communications Manager integration. Because the focus of
this book is administration, it is assumed that Unity Connection is already integrated with
CM. But we take a few minutes to explore the Unity Connection integration settings.
Communications Manager Integration
In this book, it is expected that most Unity Connection installations will be configured
to operate with Communications Manager. Before Unity Connection can communicate with Communications Manager, certain tasks must be accomplished on the
Communications Manager and the Unity systems. In Chapter 6, the Communications
Manager side of the voicemail configuration was discussed. You might need to refer back
to Chapter 6 to see how to configure the required components. Because the actual configuration steps have already been discussed, this section focuses on the process that verifies that all required configuration has been completed.
The integration settings can be viewed from within the CUAdmin web page. Under the
Telephony Integration heading you find Phone System, Port Group, Port, and Trunk. This
is where you locate all the integration settings. The following steps take you through the
process of verifying the integration settings:
Step 1.
From within CUAdmin, navigate to Telephony Integrations > Phone System.
Step 2.
A list of the phones systems that are integrated with Unity Connection
appears, as seen in figure 7-34. The name of the default integration is
PhoneSystem. If you do not see that name, it means that someone changed
the name of the default integration or deleted it and created a new one. Click
the name of your phone system.
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Figure 7-34
Phone Systems
Step 3.
The phone system basic information page appears. From this page, you can
change the parameters listed in Table 7-7.
Step 4.
Each phone system must have a port group associated with it. Navigate to
Telephony Integrations > Port Group to see a list of the port groups configured in your system. You will notice that next to the name of the port group
is the name of the affiliated phone system.
Step 5.
Select a port group from the list (you can have only one port group in your list).
The parameters on this page vary based on the type of integration you have.
Table 7-8 lists the parameters found in an SCCP integration.
Step 6.
As the name implies, port groups consist of ports. Navigate to Telephony
Integrations > Port to see a list of the ports configured on your system. You
will notice that next to the name of the port is the name of the affiliated
phone system.
Step 7.
Select a port from the list. Depending on the type of integration, the parameters on this page can vary slightly. Table 7-9 lists those found with an SCCP
integration.
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Table 7-7
Phone System Parameters
Parameter
Description
Phone System
Phone System Name
Descriptive name for the phone system.
Default TRAP Phone System
This check box determines whether this will be the default
Telephone Record and Playback (TRAP) system. It is always
selected if it is the only phone system.
Message Waiting Indicators
Send Message Counts
If selected, Unity Connection sends a message count when a
new message arrives, even if the MWI is already lit. When
deselected, Unity Connection will not send the message count
when a new message arrives if the MWI is already lit. Default:
Not selected
Use Same Port for Enabling
and Disabling MWIs
If selected, the MWI off request is sent using the same port
that the MWI on request used. Default: Not selected
Force All MWIs Off for This
Phone System
When selected, all MWIs for this system will be turned off
when the Synchronize All MWIs on This Phone System Run
button is clicked. Default: Not selected
Synchronize All MWIs on
This Phone System
When the Run button is clicked, all the MWIs for the system
are synchronized.
Call Loop Detection by
Using DTMF
Enable for Supervised
Transfers
When this box is selected, Unity Connection uses DTMF to
detect and reject calls transferred using supervised transfer.
This helps prevent the message prompt from being recorded,
which can occur during a call loop. Default: Not selected
Enable for Forwarded
Message Notification
Calls (by Using DTMF)
When this box is selected, Unity Connection uses DTMF to
detect and reject message notifications that are forwarded
from a remote device such as a cell phone. This helps prevent
false new message call notifications, which can occur during a
call loop. Default: Not selected
DTMF Tone to Use
Determines the DTMF tone to be used for loop detection.
Default: A
Guard Time
Determines the length of time that DTMF is played when
DTMF is used for loop detection. Default: 2500 milliseconds
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Table 7-7
Phone System Parameters
Parameter
Description
Call Loop Detection
by Using Extension
Enable for Forwarded
Message Notification
Calls (by Using Extension)
When selected, Unity Connection checks the extension to
determine whether a message notification is being forward to
it from a remote device such as a cell phone. It helps prevent
false new message call notifications, which can occur during
a call loop.
Phone View Settings
Enable Phone View
If selected, Phone View is enabled on this phone system.
Default: Not selected
CTI Phone Access User
Name
The name of the application user that is associated to user
phones must be entered for Phone View to function.
CTI Phone Access Password
The password of the CTI Phone Access user must be entered
for Phone View to function.
Enable Outgoing Calls
Allows Unity Connection to use this phone system for outgoing calls such as message notifications. Default: Selected
Disable All Outgoing Calls
Immediately
When selected, Unity Connection cannot use this phone
system to place outgoing calls. Default: Not selected
Disable All Outgoing Calls
Between
When selected, outgoing calls are disabled based on the times
entered. Default: Not selected
Defining Unity Connection System Configuration
After the system is installed and the integration is complete, you need to start adding
users. Unity Connection includes a number of default objects that can be used so that
you can begin adding users immediately. It is recommended that you take some time to
evaluate how your system is going to be utilized before you jump in and start adding
users. When you have a good understanding of the types of users you have and the manner in which the system will be used, you might determine that some of the default settings aren’t appropriate for your environment.
The following sections discuss systemwide settings. These are settings that affect the system and its users as a whole.
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Table 7-8
SCCP Port Group Parameters
Parameter
Description
Display Name
A descriptive name for the port group.
Integration Method
Displays the type of integration (SCCP, SIP,
and PIMG).
Device Name Prefix
The prefix used for the voicemail ports created
in CM.
Reset Status
Displays whether the port group needs to be
reset.
Enable Message Waiting Indicators
When selected, the ports of this port group
can be used to send MWI requests. Default:
Selected
MWI On Extension
The extension number configured in
Communications Manager for MWI On.
MWI Off Extension
The extension number configured in
Communications Manager for MWI Off.
Delay Between Requests
The number of milliseconds between MWI
requests. Default: 0
Maximum Concurrent Requests
The maximum number of MWI requests that
can be sent at the same time. Default: 0
Retries After Successful Attempt
The number of times that the MWI request
should be resent after a success is reported to
ensure success. Default: 0
Retry Interval After Successful Attempt
The number of milliseconds between MWI
retries after a success is reported. Default: 5
Defining General Configuration
The first area to configure is the General Configuration settings. These are settings that
define things such as time zone, language, recording settings, and default search parameters. The following steps show how to configure the general configuration settings:
Step 1.
From within CUAdmin, navigate to System Settings > General
Configuration. The screen shown in Figure 7-35 appears.
Step 2.
The first field is the Time Zone field. You cannot change this from CUAdmin.
It can only be changed from the CLI. Because the CLI is not used by most
administrators, it is not covered in this book.
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Figure 7-35
General Configuration
Step 3.
From the System Default Language drop-down list, make sure that the
desired language is selected. This is the language that system prompts are
played in.
Step 4.
From the System Default TTS Language drop-down list, be sure that the
desired TTS language is selected. This determines what language Text-toSpeech (TTS) uses if there is no TTS for the language configured for the calling phone.
Step 5.
From the Recording Format drop-down list, select the codec that you want
to use for recorded messages. The default is G.711 Mu-law. The other available choices are PCM-linear, G.711 a-law, G.729a, G.726, and GSM 06.10.
Step 6.
In the Maximum Greeting Length field, enter the maximum number of seconds that a record greeting should exist for a call handler. The default is 90
and the maximum is 1200.
Step 7.
The Target Decibel Level for Recordings and Messages field determines the
volume established when audio normalization is used. The default is –26. If
the volume of messages and recordings is too low, set this value higher. If messages and recordings are too loud, adjust the setting lower. Keep in mind that
you are working with negative numbers, so –40 is actually lower than –29.
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Table 7-9
SCCP Port Parameters
Parameter
Description
Enable
Enables the port to be used by Unity. Default: Selected
Port Name
Displays the name of the voicemail port.
Restart Button
Restarts the port when clicked.
Phone System
Displays the name of the phone system that uses this port.
Port Group
Displays the name of the port group that the port belongs to.
Server
The name of the Unity Connection server that controls this
port.
Answer Calls
Enables the port to answer incoming calls. Default: Selected
Perform Message Notification
Enables the port to make outgoing calls for message notification purposes. Default: Selected
Send MWI Requests
When selected, allows this port to be used to send MWI
requests. Default: Selected
Allow TRAP Connections
When selected, allows the port to be used for recording and
playing messages when a subscriber is using the web or an
email client. Default: Selected
Outgoing Hunt Order
Determines the place in the hunt order for this port.
Security Mode
Used to determine the security mode. Default: Nonsecure
SCCP Device Name
Displays the name of the device as configured in CM.
Certificate
When clicked, the certificate for this port is displayed.
Step 8.
From the Default Partition drop-down list, select the partition that is to be
assigned to objects when they are first created.
Step 9.
From the Default Search Scope drop-down list, select the search scope that is
to be assigned to objects when they are first created.
Note Partitions and search scopes are similar to partitions and calling search spaces in
CM. They are covered in more detail in the “Configuring Unity Connection System Access
and Policies” section, later in this chapter.
Step 10. The When a Recipient Cannot Be Found field determines what Unity
Connection executes with a Simple Mail Transfer Protocol (SMTP) message
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that is addressed to an invalid address. By default, Unity Connection sends a
Non-Deliverable Receipt (NDR). You can also choose to have the message
sent to a smart host so that it can try to deliver it.
Note A smart host is a mail relay server that aides in the delivery of email. It can be used
as a central mail server, which helps reduce the administration in larger environments.
Step 11. If you changed any settings on this page, click Save.
Now that general configuration settings are complete, we move on to defining mailbox
quotas. Unlike Unity, Unity Connection uses an on-box message store so that quotas are
configured within Unity Connection. Unity uses Exchange as its message store, so quotas
are configured on the Exchange store if you are using Unity. The next section discusses
how to set system-wide mailbox quotas.
Defining Mailbox Quotas
How many times have you heard, “I’m sorry, but the mailbox of the party you are calling
is full . . . click”? Don’t you just love that? This section examines how to configure mailbox quotas so that others can hear that message. Well, actually it’s to keep your system’s
drive from filling up.
Unity Connection enables you to set three quota levels. When users connects to the first
level, they are warned that the mailbox is almost full. When they tie into the second
level, they are prevented from sending voicemails. When they hit the third level, the mailbox is truly full and messages cannot be sent or received. Unity Connection enables you
to configure systemwide quotas, but you can override these quotas at the user configuration level. The following steps illustrate the configuration process for systemwide mailbox quotas.
Step 1.
From within CUAdmin, navigate to Message Storage > Mailbox Quotas. The
screen shown in Figure 7-36 appears.
Step 2.
Check the Custom radio button that is to the right of Warning Quota and
enter the number of megabytes allowed for message storage before a warning
message is played. The default is 12.
Note G.729 requires 60 KB of disk space for every minute of speech. This means that the
default of 12 MB will allow about 200 minutes of messages. G.711 requires approximately
a half a megabyte per minute of recorded speech, meaning that the default setting of 12
MB will allow about 25 minutes of messages.
Step 3.
Click the Custom radio button located to the right of Send Quota and enter
the number of megabytes allowed for message storage before a user can no
longer send messages. The default is 13.
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Figure 7-36
Mailbox Quotas
Step 4.
Click the Custom radio button positioned to the right of Send/Receive
Quota and enter the number of megabytes allowed for message storage
before a user can no longer send or receive messages. The default is 14.
Step 5.
By default, the Full Mailbox Check for Outside Caller Messages check box
is not selected. When deselected, Unity Connection does not verify whether
the mailbox is full. This means that the caller can leave a message even if the
mailbox is full. To make sure that this doesn’t occur, select the box. This setting only affects outside callers.
Step 6.
Click Save to save the settings.
In addition to setting the mailbox size, you can set messaging aging policies that determine how long messages can stay on the system. The next section discusses how to create and configure message policies.
Configuring Message Aging Policy
Now that mailbox quotas are set, we define message-aging polices. Message-aging policies determine how Unity Connection will acknowledge messages based on their age.
This can help keep mailboxes clean, but if misconfigured, this can also cause messages to
be deleted without ever being heard.
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Two message policies are preconfigured on Unity Connection; the Default System Policy
and the Do Not Age Messages Policy. Default System Policy is a fairly conservative rule.
It permanently deletes deleted messages after 15 days. The Do Not Age Messages Policy
is even more conservative because it doesn’t accomplish anything and it is not even
enabled by default.
You can edit these policies or create new ones. To configure a new policy, follow these
steps:
Step 1.
From within CUAdmin, navigate to Message Storage > Message Aging Policy.
Step 2.
Click the Add New button.
Step 3.
Enter a descriptive name for the new policy, and click Save.
Step 4.
A screen similar to figure 7-37 appears. By default, a message-aging policy is
not enabled when it is fist created. It cannot be edited until it is enabled. To
edit this policy, select the Enable check box.
Figure 7-37
Step 5.
Message-Aging Policy
The three options in the Message Aging Rules Based on When the Message
Was Last Modified section are similar. They define how many days Unity
Connection waits before taking the specified action. For example, Move New
Messages to the Saved Messages Folder in __ Days determines how many
days after a new message arrives that it is automatically moved to the Saved
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Messages folder. By default, the only one of these options that is enabled is
the last one, which permanently deletes deleted messages. To enable an
option, select the check box and enter the desired number of days. To disable
an option, deselect the check box.
Warning Take care when enabling these options because it could result in messages
being deleted without ever being heard. For example, if this were configured to move new
messages to the Saved folder after 5 days and then move saved messages to the Deleted
folder after 5 days, a user that did not have phone access for 2 weeks would have unheard
messages in the Deleted folder. It only gets worse if the option to permanently delete deleted messages is enabled.
Step 6.
The next section, Secure Message Aging Rules Based on When the Message
Was Created, determines how secure messages are handled. The only difference between the two options is that the first one, Permanently Delete
Secure Touched Messages That Are Older Than __ Days, only applies to
messages that the user has “touched” by an action such as saving, opening, or
deleting. The second option, Permanently Delete All Secure Messages That
Are Older Than __ Days, takes action regardless of whether the message had
been “touched.” To enable an option, select the check box and enter the
desired number of days. To disable an option, deselect the check box.
Step 7.
Click Save to save the settings.
Now that you have message-aging policies configured, it’s time to move on to the settings that determine the hours of operation. The next sections discuss how to create
schedules and holidays.
Creating Schedules and Holidays
You want to confirm that when people calls your company, they hear a message that is
accurate and informative. For example, if you call on a Saturday, you don’t want to hear
the standard greeting that tells you to please wait for your call to be answered if the company is actually closed. You want to be informed that the company is closed and to notify you when it will reopen. To have Unity Connection play a different greeting based on
hours of operation, you must configure schedules and holidays.
View and Change a Schedule
To view and change existing schedules, use the following steps:
Step 1.
From within CUAdmin, navigate to System Settings > Schedules.
Step 2.
Select the schedule that you want to edit from the list that appears.
Step 3.
A screen similar to that shown in Figure 7-38 appears. The grid offers a graphical representation of the open hours. The solid boxes with a plus sign in
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Configuring Cisco Unified Communications Manager and Unity Connection: A Step-by-Step Guide
them signify open hours. Above the grid in the Schedule Details section, you
see a summary of the open hours. In this example, the open hours are 8:00
a.m. to 5:00 p.m.
Figure 7-38
Schedules
Step 4.
You can either add additional hours to be open or edit the current hours. To edit
the current hours, click the Edit button. A screen similar to Figure 7-39 appears.
Step 5.
Enter a descriptive name in the Name field.
Step 6.
Using the drop-down lists, select the desired start and end times. If you want
the end time to be midnight, you can simply select the End of Day check box.
Step 7.
Select the check box next to the days of the weeks that you want these open
hours to be assigned to.
Step 8.
Click Save to save the settings.
Step 9.
To verify that the new hours are in the schedule, navigate to System Settings
> Schedules.
Step 10. If you want to add additional open hours to this schedule, click the Add New
button in the Schedule Details section, as shown previously in Figure 7-38.
Now repeat Steps 5–8.
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Figure 7-39
Edit Schedule Details
Now that system settings are configured, it’s time to look at system access and policies.
The next section discusses system access and policies.
Configuring Unity Connection System Access and
Policies
Before you put your system into production, make sure that it is configured as you
intended and used only by those you intended. Just as with any computer system today,
make sure that it is secure. Because different types of people will be accessing it for different purposes, you need to configure access for a few types of permissions. For example, you need to define what users are allowed to access the administrative interface, but
you also need to determine how many times a user can enter an invalid password before
she is locked out. The following sections discuss the various types of permissions that
you should be concerned with and describe how to configure them.
Configuring Authentication Rules
Authentication rules deal with passwords and login attempts. They define the characteristics of a password and describe how Unity Connection deals with failed login attempts.
Unity Connection comes with two default authentication rules. They can be edits, or you
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can create additional rules. The following steps show how to create a new rule as well as
explain the aspects of each setting:
Step 1.
From within CUAdmin, navigate to System Settings > Authentication Rules.
Step 2.
A screen appears that lists the existing authentication rule. Click the Add
New button below that list.
Step 3.
A screen similar to the one shown in Figure 7-40 appears. Enter a descriptive
name in the Display Name field.
Figure 7-40
New Authentication Rule
Step 4.
In the Failed Sign-In field, enter the maximum number of times that a user is
permitted to enter an incorrect password. If you do not want to set a limit,
select the No Limit for Failed Sign-Ins check box.
Step 5.
The Reset Failed Sign-In Attempts Every field determine how many minutes
must pass before Unity Connection resets the failed sign-in counter. For
example, if the failed sign-in field is set to 5 and the user enters the wrong
password four times and then waits the time specified in this field, Unity
Connection will act as if there were no failed sign-ins. Enter the desired number of minutes in the field.
Step 6.
The Lockout Duration field determines the length of time that an account
stays locked when the failed sign-in attempts have been exceeded. Enter the
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desired number of minutes in this field. If you want the account to be locked
until an administrator unlocks it, select the Administrator Must Unlock
check box.
Step 7.
The Minimum Duration Between Credential Changes field determines how
many minutes expire between password changes. The default setting is 1440
minutes, which means that a user must wait one day since the last time he
changed his password before he can change it again. Enter the desired number
of minutes.
Step 8.
The Credential Expires After field determines when a password expires. The
default is 180, which means that users must change their passwords approximately twice a year. Enter the desired number of days. If you do not want
passwords to expire, select the Never Expires check box.
Step 9.
The Expiration Warning Days field determines when a user will be warned
that her password is about to expire. Enter the number of days prior to the
expiration date that you want the warning played. If you enter 0, the user will
not be warned.
Step 10. The Minimum Credential Length field determines the minimum length for a
password. Enter a value between 1 and 64. The default is 6.
Step 11. The Stored Number of Previous Credentials field determines how many previous passwords Unity Connection will store and prevents you from using the
password again. Enter the desired number. The default is 5.
Step 12. If the Check for Trivial Passwords box is selected, Unity Connection will not
allow a user to create passwords considered to be trivial. Unity Connection
considers the following trivial:
■
Digits all the same
■
Consecutive digits
■
Primary extension
Step 13. Click Save to save the settings.
Keep in mind that users can log in from phones and from a web page. For this reason, you
want to configure authentication rules for each type of log in.
Just as you restrict access to the system by defining password characteristics, you need to
restrict user capability after users are logged in. The next section discusses how to configure restriction tables to help prevent misuse of certain features.
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Configuring Restriction Tables
Certain features, such as faxing and call transfers, require UC to be able to make outbound calls. Because users configure what numbers UC should dial for these various
functions, you need to be able to restrict what numbers can be configured. If restrictions
are not in place, it could result in unexpected toll charges or toll fraud.
To set these restrictions, you must create restriction tables and assign them to a class of
service (CoS). Each user is assigned to a CoS; hence the restriction table assigned to the
CoS affects the user.
UC is preconfigured with five restriction tables, which in most cases are sufficient in
North America. But, there might be a time that you need to create or change one. The
following steps show how to change the Default Transfer restriction table.
Step 1.
From within CUAdmin, navigate to System Settings > Restriction Tables.
Step 2.
Select Default Transfer (or the restriction table that you want to modify) for
the list. A screen similar to the one shown in Figure 7-41 appears.
Figure 7-41
Restriction Table
Step 3.
The Display Name field should contain a descriptive name.
Step 4.
In the Maximum Length of Dial String field, enter the maximum number of
digits allowed in a single dial string. If a user enters a string larger than this
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number, it is rejected before it is even checked against the patterns defined in
the table.
Step 5.
In the Minimum Length of Dial String field, enter the minimum number of
digits permitted in a single dial string.
Step 6.
When you create a new pattern, you can set it to be blocked or allowed. To
set all new patterns to be blocked, select the New Restriction Patterns Are
Blocked by Default check box.
Step 7.
The Restriction Patterns section of this page is where you configure the patterns that will be blocked and allowed. The wildcards used for the restriction
table are different than those used in CM. A ? matches a single digit, and a *
matches any digit and any number of digits. For example, the first pattern in
Figure 7-41 is 91???????*. This pattern matches a pattern that starts with a 91
and is followed by at least seven digits. To modify an existing pattern, simply
edit the Pattern field.
Note New patterns are positioned at the top of the list and, by default, are allowed.
Because the pattern of * is entered automatically, when you add a new pattern, you need to
be careful. If you accidentally save this pattern at the top position, you have, in effect,
allowed nearly any pattern to be entered by a user.
Step 8.
To add a new pattern, click the Add New button below the pattern. A new
pattern appears at the top of the list. Edit this pattern and select the Block
check box. If you are creating a pattern that you want to allow, do not select
the Block check box.
Step 9.
You can also delete patterns by selecting the check box to the far left of the
pattern and then clicking the Delete Selected button.
Step 10. To change the order of the patterns, click the Change Order button. All the
patterns appear in a box in their current order. Highlight the pattern that you
want to move and use the up and down arrow below the box to move it. After
you are done rearranging the patterns, click the Save button.
Step 11. If you did not change the order of the patterns, click the Save button to save
the changes. If you did change the order, your changes were saved when you
clicked Save in this step.
Because restriction tables are assigned to a CoS, they should be configured before the
CoS. After they are configured, you can move on to configuring classes of service. The
next section covers how to create various CoSs.
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Configuring CoS
UC offers many features, and you might not want to allow all of your users full entry, or
you might want to control what level of access they have to them. To do this, you must
create various classes of service. This section will walk you through the steps to create a
new class of service and offer explanations for each setting. The following steps illustrate
this process:
Step 1.
From within CUAdmin, navigate to Class of Service > Class of Service.
Step 2.
A list of currently configured classes of service appears. Click the Add New
button located under the list.
Step 3.
A screen similar to the one shown in Figure 7-42 appears. Enter a descriptive
name in the Display Name field.
Figure 7-42
New CoS
Step 4.
The Allow User to Record Name check box determines whether a user is
allowed to record his own name. If this box is selected, the user will be
prompted to record a name during the first-time enrollment process. To permit users to record their name, ensure that the box is selected. If you do not
want users to record a name, make sure that the check box is not selected.
Step 5.
In the Maximum Length field below the Allow User to Record Name field,
enter the maximum number of seconds desired for recording a name.
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Step 6.
The Allow Users to Choose to Be Listed in the Directory check box determines whether users are permitted to decide whether they want to be listed in
the directory. Typically, you want to make sure that this box is not selected.
Step 7.
In the Maximum Length field under the Greetings heading, enter the maximum number of seconds desired for a user’s greeting. The default is 90.
Step 8.
UC allows a user to access voicemail from his email client. One option is to
configure it as an IMAP client. To permit user access, select the Allow Users
to Access Voicemail Using an IMAP Client check box. Under this setting,
you can define the type and portions of the message that you allow them to
retrieve. If you enable this feature, you must select one of the access settings.
The choices are
Step 9.
■
Allow Users to Access Message Bodies: Enables full access to all
voicemails.
■
Allow Users to Access Message Bodies Except on Private Messages:
Enables full access to all emails except those marked private.
■
Allow Users to Access Message Headers Only: Enables the user to see
that there is a new voicemail, but not download the message.
Users can check voicemail from a web page called UC Messaging Inbox. To
enable this feature, you must select the Allow Users to Use the Messaging
Inbox and RSS Feeds check box.
Step 10. The Allow Access to Advanced Features check box determines whether the
user will be able to access email stored on Exchange and have entry to voice
recognition services. To enable these features, select this check box and the
check boxes below for the features that you desire.
Step 11. UC has the ability to send messages to a third-party transcription service. To
enable this feature for users, select the Provide Transcriptions of Voice
Messages (SpeechView) check box. You must also define whether secure
messages will be allowed to be transcribed. If this feature is enabled, you
must also select one of the following options:
■
Do Not Transcribe Secure Messages: Secure message will not be sent
to the transcription service.
■
Allow Transcriptions of Secure Messages: Secure message will be sent
to the transcription service, but users will not receive these transcriptions on their notification devices.
■
Allow Transcriptions of Secure Messages to Be Sent to Notification
Devices: Secure message will be sent to the transcription service, and
users will receive these transcriptions on their notification devices.
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Step 12. The next three check boxes determine the features that are accessible to the
user. Select the box for each feature that you want users to have in this class
of service The available features are
■
Allow Users to Use Personal Call Transfer Rules: Personal call transfer
rules allow a user to define how a call is handled based on the call information and other factors such as time of day.
■
Allow Users to Use the Messaging Assistant: The Messaging Assistant
is a web-based interface that allows users to configure the settings for
their voicemail.
■
Allow Users to Use Unified Client to Access Voicemail: This permits
users to access voicemail using the Cisco Unified Personal
Communicator.
Step 13. Multiple extensions can be assigned to a single user. This is accomplished by
defining alternate extensions. You can choose to allow users to view and/or
assign their own alternate extensions. Select the Allow Users to View
Administrator-Defined Alternate Extensions check box to permit access to
the alternate extensions that an administrator has assigned to their accounts.
To allow users to personally assign extensions, select the Allow Users to
Manage Their User-Defined Alternate Extensions check box.
Step 14. The Maximum Length field located under the Message Length heading
determines the maximum length of a message left by the user to another user
in the system. Enter the desired number of seconds in the field.
Step 15. The next four check boxes determine message options for the user. Select the
desired options. The available options are
■
Allow Users to Send Messages to System Distribution Lists: Enables
users to send messages to distributions lists.
■
Delete Messages Without Saving to Deleted Items Folder: When
selected, messages are deleted immediately and not saved to the Deleted
Items folder.
■
Users Can Reply to Messages from Other Users by Calling Them:
Enables a user to press 4-4 while listening to a message and respond to
the sender of the message if the sender is another UC user. This is often
referred to as Live Reply.
■
Users Can Reply to Messages from Unidentified Callers by Calling
Them: Enables a user to press 4-4 while listening to a message and call
the sender of the message if the sender is not a UC user. This is often
referred to as Live Reply.
Step 16. The Require Secure Messaging field determines what messages are marked as
secure. Select the desired setting from the drop-down list. The available settings are
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Step 17.
■
Always: All messages are marked secure.
■
Never: No messages are marked secure.
■
Ask: Only marked secure when user selects the secure option from the
special delivery options.
■
Private: Mark private messages secure.
The two fields located under Private Distribution List determine the maximum number of private lists per user and the maximum number of members
per list. Enter the maximum number of lists a user can have in the Maximum
List per User field, and enter the maximum number of members per list in the
Maximum Members per List field.
Step 18. The Allow Users to Change Call Screening Options check box determines
whether the user can enable call screening. It is recommended that this rarely
be used because it can affect call-processing performance.
Step 19. The Allow Users to Change Call Holding Options check box determines
whether callers can change call-holding options. If enabled, use Message
Assistant to configure this.
Step 20. Restriction tables determine what type of calls UC is allowed to place. The
configuration details were covered in the previous section. Select a restrictions table for each of the three types of calls that UC can place. They are
■
Outcalling: Restricts the number that a user can enter for UC to use for
record and playback functions
■
Transfer: Restricts the number that a user can enter for UC to use for
transfer functions
■
Fax: Restricts the number that a user can enter for UC to use for fax
functions.
Step 21. Click Save to save the settings.
The CoS defines what types of functions and features a user has access to, but it does
not define any administrative access. Administrative access is determined by the roles
that are assigned to a user.
Understanding Roles
Roles determine the administrative level that’s accessible to a user. There are eight predefined roles. They cannot be edited, and additional roles cannot be added. Each role permits access to functions that common types of administrators need. Table 7-10 lists the
roles and their functions.
For a closer look at the exact privileges a role offers, navigate to System Settings >
Roles. Select the role that you are interested from the list that appears. A screen similar to
that shown in Figure 7-43 appears. The allowable applications are listed in detail under
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Table 7-10
Predefined Roles
Role
Function
Audio Text Administrator
Administers call handlers, directory handlers,
and interview handlers
Greeting Administrator
Manages call handler recorded greetings
through the phone
Help Desk Administrator
Resets user passwords and unlocks user
accounts, views user settings
Mailbox Access Delegate Account
Access to all messages through messaging
APIs
Remote Administrator
Administers the database using remote management tools
System Administrator
Access to all connection administrative functions, reports, and tools for server and users
Technician
Access to functions that enable management
of system and phone system integration settings, viewing of all system and user settings,
running all reports and diagnostic tools
User Administrator
Access to all user administration functions,
user reports, and user administration tools
the Role Privileges heading. You can also see which users are assigned to this role by
clicking the Role Assignments button.
Now that you are familiar with the use and function of roles, there is only one more
access-related component. Now move on to learn about partitions and search scopes.
Defining the Dial Plan
This section examines the configuration of partitions and search spaces. They are similar
to the partitions and calling search spaces that are employed in CM. Within UC, they
can be used to restrict a user’s accessibility to objects. This is beneficial in a multitenant
environment.
Because these are nearly identical to partitions and call search spaces within CM, not
much time is spent covering the details of how they work. If you need to refresh your
memory, refer to Chapter 5, “Configuring Class of Service and Call Admission Control.”
The main difference is that calling search spaces are simply referred to search spaces in UC.
As you recall, partitions are assigned to prevent or permit accessibility to objects. A
search space is a list of partitions. The search space assigned to a user determines his or
her level of access.
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Figure 7-43
Role Assignments
Partitions are assigned to the following:
■
Users with mailboxes (primary extension)
■
User alternate extensions
■
Contacts (including VPIM contacts)
■
System call handlers
■
Directory handlers
■
Interview handlers
■
System distribution lists
■
VPIM locations
Search spaces are assigned to the following:
■
Users with mailboxes
■
Routing rules (both direct and forwarded)
■
Phone directory handlers
■
System call handlers
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■
Voice-enabled directory handlers
■
VPIM locations
Search spaces contain partitions, so partitions must be created first. To create a partition,
navigate to Dial Plan > Partitions, and click the Add New button. On the next screen,
enter a name for the partition and click the Save button. On the following screen, enter a
description for the partition and click the Save button.
After you have created all the partitions, create search spaces. Use the following steps to
create a search space:
Step 1.
From within CUAdmin, navigate to Dial Plan > Search Space.
Step 2.
Click the Add New button.
Step 3.
On the next page, enter a name for the search space and click the Save button.
Step 4.
A screen similar to that shown in Figure 7-44 appears. Enter a description for
the search space in the Description field.
Figure 7-44
Step 5.
Search Spaces Configuration
In the Unassigned Partition box, highlight the partition that you want to
assign to this search space and click the Up Arrow directly above the
Unassigned Partition box.
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Step 6.
The partition selected should now appear in the Assigned Partition box.
Repeat Step 5 until all the desired partitions are listed in the Assigned
Partition box.
Step 7.
After all the partitions have been added, you can change the order in which
they display. Remember, the order in which they display determines which
partition is searched first. To change the order, highlight the partition that you
want to move and click the Up or Down Arrow to the right of the box.
Note You can remove a partition from a search space by highlighting it and clicking the
Down aArrow below the Assigned Partition box. The selected partition should now
appear in the Unassigned Partition box.
Step 8.
Click Save to save the settings.
After the partitions and search spaces are created, they can be assigned to various
objects. You will see how they are assigned Chapters 8 and 9 as we explore the objects
that might require them.
Summary
That’s all there is to it. Well, not really. There is much more, but these steps complete
the chapter. All the hard work you have done up to now is about to pay off. In this
chapter, you first learned how to verify a proper integration with Communications
Manager for Unity and Unity Connection. Then system configuration settings such as
schedules, holidays, and languages were discussed. In addition, authentication and port
settings were examined. This chapter also discussed the configuration of account policies and class of service (CoS) settings. The next chapter examines the process of
adding subscribers and users.
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Chapter 8
User/Subscriber Reference
After a proper integration between Unity and Communications Manager is achieved and
the predeployment tasks discussed in the previous chapter are completed, users can be
added. Unity refers to users as subscribers, whereas Unity Connection refers to them simply as users. To keep things simple, this chapter uses the term users as a general term that
refers to both Unity subscribers and Unity Connection users. In this chapter, the different
types of users are examined, and the proper usage for each is discussed. Next you
explore the process for adding, importing, and managing users. Within the Managing
Users section, various administrative tasks are discussed, ranging from learning how to
reset a subscriber password to granting a user administrative access. Each task includes
step-by-step instructions.
Note Throughout this chapter, the term subscriber is used when referring to a Unity user.
The term user is used when discussing Unity Connection users. When the subject matter is
referring to both Unity subscribers and Unity Connection users, the term user is used.
In Chapter 1, “CUCM and Unity Connection Overview,” you learned about the various
types of users. This chapter discusses how to create and manage users.
Defining Various Types of Subscribers
Even though you are already familiar with the different types of subscribers, we take a
minute to briefly review the different types of subscribers.
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Exchange
The Exchange subscriber is the most common subscriber found in a Unity deployment.
As the name implies, this type of subscriber uses Exchange for its message store.
Exchange subscribers can be either voicemail-only or unified messaging subscribers.
Voicemail-only subscribers use the Exchange server as a place to store only their voicemail messages. They are not able to retrieve their email through Unity or their voicemail
through a PC. The unified messaging subscribers have both their email and voicemail
stored on the same Exchange server and can access both types of messages from a
phone or a PC.
These types of users can have many different ways to access Unity, such as through the
Internet or over a phone. The user’s CoS determines exactly what entry points he can use.
Networked Subscribers
In addition to the Exchange subscriber, there is also a Network subscriber that is created
only if some type of Unity networking is deployed. Because these users are directly
related to Unity networking, they are discussed in Chapter 10: “Implementing Unity
Networking.”
Unity Connection Users
Unity Connection has two types of users, one with a mailbox and one without. You
might wonder why you would ever create a user on a voicemail system that doesn’t have a
mailbox. It is recommended for administrative purposes. In other words, do not use your
normal voicemail user login information for administrative tasks. There are also objects
that are called contacts that are similar to a network user in Unity. A contact is simply a
pointer to another system. These are created so that people that have a voicemail account
on a different system can appear in the directory and have messages forwarded to them.
Creating Users
The actual process of creating a single user is, at the surface, a simple task. However,
when examined closely, you see that each user has well over 50 individual settings. It is
logical to assume that this process is so simple because it requires only a few pieces of
information and a few keystrokes. To ensure that the user is properly configured, it
requires much more work. Each time a user is added, every setting must be given a value.
So, creating a user requires the configuration of more than 50 settings, but at the same
time, a user can be created with just a few keystrokes. How is this possible? The creation
of a user is based on what is known as a template. The values configured in the template
are applied to the newly created user. The next section discusses the use and creation of
user templates.
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Exploring Templates
When you create or import users, a large number of them have similar attributes.
Templates automatically enable you to assign the same settings to each user during the
creation process. The use of a template is not optional. Each time you add a user, a template must be chosen.
Before creating templates, plan how each user will use the system. Templates enable you
to configure everything from the user’s time zone to what happens to a call if a caller
presses 4 while listening to the greeting. Because of their configuration, it is possible to
find yourself creating many templates. Be careful not to get carried away. Remember, the
templates are used to help assign settings to a group of users that are going to have similar, but not necessarily identical, attributes. In most cases, the template you create should
apply to at least five users. If you find that you are creating a large number of templates
and applying each of them to a small number of users, you might want to rethink your
deployment strategy. Although each user has unique settings, there will be a large number of settings that are the same for the majority of users.
When determining how many templates you need, it is a good idea to try to organize
users into groups. These groups should begin at a very general level. The members in
these groups require the same type of features and access to Unity/Connection. Assume
that you are implementing unified messaging, but only some of the users need to retrieve
their email from the phone. From this, we can assume that there are at least three types of
users. One group, which is needed in all environments, is the administrator group. The
other two types of users are those that need access to text-to-speech and those that
don’t. There might be a requirement for additional groups, but this lets you determine the
different templates you might need.
Refer to the following list of questions that should be considered when determining the
necessary number of templates. They are directly related to the various settings that
should be configured in the templates. In addition to answering these questions, it is recommended that you walk through the process of creating a test template to become
familiar with the settings that are configured.
Step 1.
How many CoSs are there?
Typically you will need at least one template for each CoS you have created.
Step 2.
How many schedules are there?
Step 3.
Are all users in the same time zone?
Step 4.
Are all users on the same phone switch?
Step 5.
Are multiple mailstores (Exchange servers) being used?
Step 6.
Will there be call accounting for outbound calls?
Step 7.
How many billable departments are there?
The system can create a report of each outbound call that it places and associate it with a billing ID.
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Step 8.
Will all users have the same password restrictions?
Step 9.
Will users be allowed to control their own conversation flow? If not, how
many base flows are needed?
This deals with things such as menu styles, time format, and message playback order.
Step 10. Will all phones ring when dialed from within the system?
This refers to what happens when a user’s extension is dialed from within the
system. Typically, you want the phone to ring, but in some cases, you might
want the call to go directly to voicemail or a different extension.
Step 11. Will all phones ring the same number of times when transferred?
Step 12. What greetings are enabled by default?
A user can have multiple greetings. You need to determine which greeting is
enabled by default and whether the same greetings are enabled for all users.
Figure 8-1
Subscriber and Subscriber Template Submenus
Step 13. While listening to a greeting, what options will a caller have? Will this be the
same for all phones?
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While listening to a greeting, the caller can be given a list of options such as
“To skip this message, press <option number>” or “To reach an operator, press
<option number>.”
Step 14. How long can messages be? Will the length be the same for all phones?
Step 15. How many distribution lists are there?
Step 16. Will message notification be used? How many groups will require this?
This is not necessarily a complete list of the questions you need to ask, but it gives you a
good idea of the types of information you need to gather. After you have assembled this
type of information, you are ready to start creating templates. Because the creation of
Unity templates is somewhat different than that of Unity Connection templates, the
details are discussed separately in the following two sections. The first section discusses
how to create Unity templates and describe how to use them to create Unity subscribers.
The second section focuses on these same tasks for Unity Connection.
Creating Unity Subscriber Templates
Before moving on to the actual process of creating subscriber templates, it is important
that you understand that these templates are significant only at the point of creation. This
means that the setting of a template affects the subscriber only when the subscriber is
first created. If you create 50 users using a template, changes that you make to that template after the subscribers are created do not affect the subscriber’s settings. If you need
to change certain settings on a large number of subscribers, use the Bulk edit utility. This
utility is discussed in Chapter 11, “Exploring Unity/Connection Tools.”
Figure 8-1 displays a split screen that shows the submenu for a subscriber and the submenu for a subscriber template. This figure shows that the submenus are similar.
There are more than 50 configurable settings for each subscriber, so there must be at
least that many settings for each subscriber template. It is important to understand the
effect of each setting. Table 8-1 provides an overview of the types of settings that each
submenu contains.
Table 8-1 Subscriber Template Submenus
Submenu
Description
Profile
General settings, such as schedules and time zones, are set in this area.
Information relating to the subscribers’ names is also set here, such as how
their Windows domain usernames are generated.
Account
This area enables you to set a billing ID for subscribers and determine
whether the subscriber’s account will be available for use as soon as it is created or whether an administrator will have to unlock it first.
Passwords
This area enables you to configure default passwords and expiration settings.
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Table 8-1
Subscriber Template Submenus
Submenu
Description
Conversation
The Conversation area enables you to configure how the Unity system interacts with a subscriber. This includes things such as volume, menu style, time
format, message playback order, and many other settings. If you allow the
subscribers access to Cisco Personal Communication Assistant (CPCA),
subscribers can change most of these settings themselves.
Call Transfer
This area determines what action is taken when a subscriber’s extension is
dialed from within Unity. Most of the time, this occurs when an outside
caller dials the extension from the auto-attendant.
Greetings
The Greetings area configures which greetings are enabled, what action is
allowed during the greeting, and what action is taken after the greeting.
Caller Input
During a greeting, a caller has a number of choices, such as to skip the
greeting by pressing # or to reach an operator by pressing 0. The caller
input settings determine what happens if a digit is pressed while the greeting is being played.
Messages
The settings in this area determine how long messages can be, whether
callers can edit messages, and the actions taken after taking a message. The
extension to use for the Message Waiting Indicator (MWI) is also configured here.
Distribution
Lists
The distribution lists to which the subscriber belongs are configured here.
Message
Notification
The devices used to notify a subscriber of a new message are configured in
this area.
Features
The features such as broadcast messages, message locator, and message
security are configured in this area.
Note The conversation settings are important to the success of the system because they
allow control over the settings to make the new system more acceptable to the end user. I
was once asked to deliver some end-user training for a company whose employees were
frustrated with the system. As it turned out, 90 percent of their concerns related to the
conversation settings. After they learned to configure these settings themselves, the
majority of the issues were resolved.
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Two user templates are created during the installation process. These are the default administrator and the default subscriber templates. As their names imply, they each have settings
that are commonly found in administrative-level subscribers and standard subscribers.
Note Although you can use the default templates to create subscribers, it is best to confirm the context of the default template’s settings first. Do not assume that the default settings are the best settings for your deployment.
At this point, you should have an idea of the different templates that you need. I’ll go
through the process of creating and managing a template so that you fully understand all
the settings that must be configured. I begin by adding a new template. The following are
the steps required to add a new subscriber template:
Step 1.
From within SA, select Subscribers > Subscriber Template.
Step 2.
Click the Add icon in the title strip.
Step 3.
A window such as the one in Figure 8-2 displays. Enter the name for the new
subscriber template.
Figure 8-2 Subscribers > Subscriber Template (Adding)
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Step 4.
If there is an existing template similar to the one you are creating, click the
Based on Existing Template radio button and choose the existing template
from the drop-down list. Otherwise, click the New Template radio button and
click Add.
The creation of a template is the easy part. Now you need to configure the settings within each submenu. As Figure 8-1 shows, there are 11 submenus under Subscriber
Templates. The configuration of settings in each submenu follows.
Configuring Subscriber Template Profile Settings
We start by configuring the template’s Profile settings. Figure 8-3 illustrates the Subscriber
Template Profile page, and the following steps describe the configuration process:
Figure 8-3 Subscribers > Subscriber Template > Profile
Step 1.
From the previous steps, click Profile from the list on the left side of the
screen and move on to the next step. Otherwise, from within SA, select
Subscribers > Subscriber Template. If the name of the template you want to
manage does not display in the title strip, click the Find icon, and select the
desired subscriber template. After the template is active, click Profile from
the list on the left side of the screen.
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Step 2.
The name of the template displays in the Name field on this screen. This name
is chosen when the template is created. You can change it here if you want.
Step 3.
Select the desired Class of Service, Active Schedule, and Time Zones from
the appropriate drop-down lists.
Step 4.
In a dual-switch integration, choose the phone system that the subscriber’s
phone is on in the Switch field. This field displays only with a dual-switch
integration.
Step 5.
The Message Playback Speed setting enables you to speed up or slow down
the speed at which voicemail is played back. Select the desired setting from
the drop-down list.
Step 6.
The Conversation Speed setting permits you to speed up or slow down the
speed at which Unity’s prompts are played to the subscribers. Select the
desired setting from the drop-down list.
Step 7.
By default, subscribers have access to all directory handlers. This enables subscribers to address messages and add users to private distribution lists through
the phone. To restrict subscribers to access only certain directory handlers,
select the Limit Access To radio button. Now click Select. A window appears
from which you can select the directory handlers to which the subscribers will
have access. In many cases, this can be left set to Not Limited.
Step 8.
Under the Display Name Generation heading, choose either First Name
Then Last Name or Last Name Then First Name. This setting should be similar to the existing names in Exchange.
Step 9.
To force subscribers to record a name and greeting and change their passwords on the next login, select the Set Subscriber for Self-Enrollment at
Next Login check box. Subscribers can choose to be listed in the directory
during the self-enrollment process, if the subscribers’ CoS allows this option.
Step 10. To list subscribers in the auto-attendant directory, select the List in Phone
Directory check box.
Step 11. Select the Show Subscriber in E-mail Server Address Book check box if you
want the subscriber’s address to display in the Outlook address book. If this
box is deselected, other subscribers cannot use the Outlook address book to
address messages to the subscriber. When an Exchange server has two
accounts for each user, one for email and one for voicemail, it is a good idea
to deselect this box. This prevents an email user from mistakenly sending
email to a voicemail inbox.
Step 12. Finally, you need to choose how to generate the Exchange alias. Use this
when creating subscribers who do not already have AD accounts. An
Exchange and AD account is created when the subscriber is added, and the
alias generated is based on this selection. If there are existing Exchange
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accounts, make sure that the format you choose matches the format of the
existing Exchange aliases.
Step 13. Click the Save icon in the title strip.
Configuring Subscriber Template Account Settings
Next, configure account information for the template:
Step 1.
If you are continuing from the previous steps, click Account from the list on
the left side of the screen and continue to the next step. Otherwise, from
within SA, select Subscribers > Subscriber Template. If the name of the template you want to manage does not appear in the title strip, click the Find
icon and select the desired subscriber template. After the template is active,
click Account from the list on the left side of the screen.
Step 2.
The Cisco Unity TUI Access Status check box enables you to lock the
account at the point of creation. You might want the account to be locked at
the point of creation so that the account cannot be used before the deployment is completed. Decide whether you want the accounts you create, using
this template, to be locked upon creation and select the check box accordingly.
Step 3.
The other setting for account information is a billing ID. A billing ID determines which subscriber or department is billed for outbound calls placed by
Unity. A billing ID is assigned to an individual mailbox or a group of mailboxes. If you want to assign the same billing ID for all subscribers using this template, enter it here. If you plan to have different billing IDs for each user or
not use them at all, leave this field blank.
Step 4.
Click the Save icon in the title strip.
Configuring Subscriber Template Passwords Settings
The Subscriber Template Passwords page enables you to define password-specific settings. From this page, you can set default passwords and phone password settings. Figure
8-4 illustrates how the settings are configured.
To configure these settings, follow these steps:
Step 1.
From the previous steps, click Passwords from the list on the left side of the
screen and move on to the next step. Otherwise, from within SA, select
Subscribers > Subscriber Template. If the name of the template you want to
manage does not display in the title strip, click the Find icon, and select the
desired subscriber template. After the template is active, click Passwords
from the list on the left side of the screen.
Step 2.
Under the Phone Password Settings heading, there are three settings. These
settings determine whether the subscribers are able to change their passwords,
whether subscribers must change their passwords on the next login, and
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whether the password will expire. If you do not want the users to be able to
change their passwords, select the User Cannot Change Password check box.
Figure 8-4 Subscribers > Subscriber Template > Passwords
Step 3.
You want to force subscribers to change their passwords on their first login.
select the User Must Change Password at Next Login check box to do so.
Step 4.
To enhance security, it is a good idea to force subscribers to change their
passwords on a regular basis. However, a password can be set to never expire.
If you choose to configure the phone’s password so that it never expires,
select the Password Never Expires check box. If this box is left deselected,
the password expires based on the setting you make in Account Policy.
Although you can set a password never to expire, this is not recommended.
Step 5.
The subscribers’ default passwords are determined by the passwords set in the
template. In the Phone Password for New Subscribers field, enter the desired
default phone password.
Step 6.
The next field is not configurable. The Last Phone Password Change field
displays the date on which the phone password was last changed.
Step 7.
When new Exchange subscribers who do not have existing Windows domain
accounts are added, Windows domain accounts are created for them. In the
Password for New Windows Accounts field, enter the Exchange password
you want to have assigned this account.
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Step 8.
The final field on this screen in not configurable. The Last Windows
Password Change field displays the date the last time the Windows password
was changed.
Step 9.
Click the Save icon in the title strip.
Configuring Subscriber Template Conversation
The Subscriber Template Conversation settings enables you to configure how Unity interacts with a subscriber. This page has more configurable settings than any other page in
Unity, so there is more configuration. It is important to understand how these affect a
subscriber’s settings because the settings have a significant impact on the subscriber’s
satisfaction with the system. The following steps explain what effect the settings have
and how to configure them. Figure 8-5 shows the fields covered in Steps 1–8.
Figure 8-5 Subscribers > Subscriber Template > Conversation
Step 1.
From the previous steps, click Conversation from the list on the left side of
the screen and proceed to the next step. Otherwise, from within SA, select
Subscribers > Subscriber Template. If the name of the template you want to
manage does not display in the title strip, click the Find icon and select the
desired subscriber template. After the template is active, click Conversation
from the list on the left side of the screen.
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Step 2.
The first setting determines whether the subscriber hears full or brief menus.
It is a good idea to set the menus to full for users who are new to the system
because full menus explain more than brief menus. After users are familiar
with the menus, you might want to change to a brief menu. Choose the
desired menu style from the Menu Style drop-down list.
Step 3.
If multiple languages are installed, you can select the language from the
Language drop-down list. This determines the language for the subscriber
conversation.
Step 4.
Select a time format different from the system default by selecting 12-hour
or 24-hour format from the Time Format drop-down list. If you leave this
field set to System Default, the format selected under Setting >
Configuration is used.
Step 5.
The Conversation Styles fields determine which touch tones you use to do
things such as save and delete messages. If you are migrating from another
voicemail system, it is a good idea to become familiar with these conversations and pick the one closest to previous voicemail system conversations.
There are a number of options to choose from. The Standard Conversation is
a good choice for locations where users are unfamiliar with a certain voicemail menu format. The Optional Conversation 1 is a good choice if users are
familiar with an Octel-like menu system. The Hospitality Conversation is
only used when Unity’s Hospitality and Property Management System is
deployed.
Step 6.
From the Send Message Style drop-down list, you can select Standard or
Streamline. The Streamline option enables the subscriber to send messages
and select sending options using fewer keystrokes than the Standard style.
Select the desired style from the Send Message Style drop-down list.
Step 7.
To allow subscribers to use Unity’s voice recognition, select the Allow Access
to Voice-Recognition Features check box. If this is enabled, you can also
select the Use Press-or-Say Phone Input Style check box to permit a user to
say a number in response to a prompt instead of having to press the number
on the keypad.
Step 8.
Unity offers a number of additional conversation styles that sound similar to
other voicemail systems. This enables users to apply the same keys to perform
actions such as delete, save, and so on, as they did on their previous system.
Users can select the conversation style from within Unity Assistant. The conversations they can select from are determined by which are selected in the
Conversation Styles Offered in Cisco Unity Assistant section. Select the
check box of each conversation you would like to make available.
The next three sets of settings deal with phone menu responses, exit actions,
and message addressing and sending. Figure 8-6 shows the fields covered in
Steps 9–16.
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Figure 8-6 Subscribers>Subscriber Template>Conversation (Continues)
Step 9.
There are four values that require configuration in the Responding to Phone
Menus section. The first three values deal with timing:
■
Milliseconds to wait for subscriber to press the first key
■
Milliseconds to wait for additional key presses when entering names,
extensions, and passwords
■
Milliseconds to wait for additional key presses when entering phone
menu commands
The fourth value determines the number of times the menu will repeat if the
subscriber does nothing. These values can be left blank if you want to use the
ones you configured under the System Configuration. If you want values that
are different than those configured in System Configuration, enter the desired
values in each field.
Step 10. Under the When Exiting the Conversation heading, you can configure where
subscribers are sent when they end the subscriber conversation. A number of
choices are available from the drop-down list; normally this is configured to
send the subscriber to a call handler or hang up. Select the destination from
the Send Subscriber To drop-down list. If Call Handler is selected, click the
Select Call Handler button to choose the desired call handler.
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Step 11. To have subscribers prompted to address a message before being allowed to
record it, select the Address Message and Then Record It check box. To
allow subscribers to record a message and then address it, deselect this box.
Step 12. To configure how a subscriber is prompted to address messages to other subscribers, select one of the available choices from the Identify Addressee By
drop-down list: Spelling the Last Name Then First Name, Spelling the First
Name Then Last Name, or Enter the Extension.
Step 13. To have Unity confirm the name of each addressee as it is added, select the
Confirm Addressee by Name check box. If deselected, this feature is disabled.
Step 14. If the Perform Name Suggest on Spelled Names check box is selected, Unity
will search for name matches after each key is pressed (starts searching after
the second key). If it finds a single match, it plays the name of the match.
When it finds between two and six matches, it states how many matches have
been found. If the box is deselected, this feature is disabled.
Step 15. To allow subscribers to enter addresses one after the other, select the
Continue Adding Names After Each Addressee check box. After the user is
done entering the addresses, she presses 2 to send the message. If this box is
not selected, the user must press 1 after each address to enter another one.
Step 16. When the Send Message When Subscriber Hangs Up or Call Is
Disconnected check box is selected, Unity automatically sends the message
when the call disconnects. If this box is not selected, the subscriber must
press # for the message to be sent.
The next group of settings to configure in the conversation area deal with the
messages playback option. Figures 8-7, 8-8, and 8-9 show the fields that are
covered in these steps.
Step 17.
When a subscriber is listening to messages, he can fast-forward and rewind
the message. The amount of time that is applied to these functions is defined
by the value entered in the Fast-Forward Messages By and Rewind
Messages By fields. The default is 5 seconds and is adequate in most cases.
To change this value, simply enter the new value in the appropriate field.
Step 18. If the Confirm Deletions of New and Saved Messages check box is selected,
a subscriber will be prompted to confirm each time that she attempts to
delete a message. If you have a user that frequently deletes messages by mistake, this might be a feature you should turn on.
Step 19. The Mark a Message as Saved Upon Hangup or Disconnection field determines how Unity will mark messages that are being listened to when the call
is disconnected. If the box is selected, the message is marked read. If left deselected, the message is marked as new.
Step 20. To have Unity play the extension number of the subscriber that sent a message along with the subscriber’s recorded name, select the Announce
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Sender’s Extension for Messages from Subscribers check box. In most
cases, this box is left deselected.
Figure 8-7
Subscribers > Subscriber Template > Conversation (Continues)
Step 21. To have Unity play the Automatic Number Identification (ANI) of an unidentified caller that sent a message, select the Announce ANI for Messages
from Unidentified Callers check box.
Step 22. To have the recorded name of the subscriber played when he logs in to Unity,
select the Subscriber’s Recorded Name check box.
Step 23. An alternate greeting is used when a subscriber is out of the office for a long
period of time, such as vacation. It is important that the subscriber remember
to turn this greeting off when he or she returns. Configure Unity to inform
the subscriber whether the alternate greeting is enabled when he logs in by
selecting the Alternate Greeting Notification check box. Starting with Unity
4.04, when a subscriber enables the alternate greeting, he can automatically
schedule a time for the alternate greeting to be disabled.
Step 24. Unity offers a follow-me feature. This feature is used to have calls sent to a
number other than users’ desk phones. It is similar to a call-forward feature.
When enabled, Unity will not send the call to the SB’s desk phone. To have
the SB notified when this feature is on, select the Follow Me Transfer
Notification check box. If an SB leaves this feature on, she is notified the next
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time she checks her voicemail. She is then offered the option of turning it off
or leaving it on.
Step 25. To have Unity start playing new messages as soon as the SB logs in, select the
New Messages Automatically check box. This prevents the SB from having to
press a key before new messages are played.
Step 26. To configure Unity to inform the subscriber of the total number of new messages and of the message count, select the box next to each type of message
count you want the subscriber to hear. The choices are Message Count
Totals, Voice Message Counts, Email Message Counts, and Fax Counts. It is
best to offer only the message counts for the messages the subscriber is likely
to retrieve using the phone. For example, if the subscriber retrieves only
voicemail using the phone, only Voice Message Counts is chosen.
Step 27.
To announce the number of saved messages to the subscriber, select the
Saved Message Count check box.
Step 28. Figure 8-8 shows the next set of fields. To allow subscribers to determine the
types of messages they want to hear, select the Message Type Menu check
box. When this option is enabled, the subscriber hears the following message:
“Press 1 to hear voice messages, press 2 to hear emails, press 3 to hear faxes,
press 4 to hear receipts.”
Figure 8-8 Subscribers > Subscriber Template > Conversation (Continues)
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Step 29. The order in which new messages are played can be modified. Under the New
Message Play Order heading is a list of the various types of messages. The
messages are played as they appear from top to bottom. To change the order,
highlight the type of message you want to move, and then click the Move Up
or Move Down button. Repeat this process with all the message types until
they are in the desired order.
Step 30. When playing new messages, the default behavior of Unity is to play the oldest message first. To play the newest message first, choose Newest First from
the Then By drop-down list. In most cases, it is best to leave this setting at
Oldest First.
Step 31. The order in which saved messages are played can be modified. Under the
Saved Message Play Order heading is a list of the various types of messages.
The messages are played as they appear from top to bottom. To change the
order, highlight the type of message you want to move, and then click the
Move Up or Move Down button. Repeat this process with all the message
types until they are in the desired order.
Step 32. When playing saved messages, the default behavior of Unity is to play the
newest message first. To play the oldest message first, choose Oldest First
from the Then By drop-down list. In most cases, it is best to leave this setting
at Newest First.
Step 33. Figure 8-9 shows the next set of fields. Under the Before Playing Each
Message, Play heading you can choose to play the sender’s information, message number, time the message was sent, and message duration. To configure
what is played, select the box next to each type of information that you want
to have played.
Step 34. To play the time a message was sent after a message is played, select the Time
the Message Was Sent check box.
Step 35. The Drop Call Recovery feature enables an SB to call Unity back when the
call has been disconnected and pick up where he was when the call dropped.
To enable this, select the Enable DCR for Calls Dropped During Message
Playback and Enable DCR for Calls Dropped While Addressing or
Recording Messages check boxes. You can also modify the time in which a
subscriber must call back in for this feature to be invoked by entering the
desire number of minutes in the After Disconnect, Recover Calls For fields.
Step 36. Click the Save icon in the title strip.
Configuring Subscriber Template Call Transfer
The subscribers’ template call transfer settings determine what happens to a call when an
outside caller attempts to reach a subscriber’s phone from the auto-attendant. An outside
caller can attempt to reach a subscriber by entering the subscriber’s extension number
or by using Unity’s dial-by-name feature. When Unity finds a match for either the
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extension number or the name, it looks at the subscriber’s call transfer settings to determine where to send the call. One would think that Unity would send the call to the
PBX so that the call could be delivered to the phone. Although this is the most common
action, it is not always desired. A number of circumstances can dictate that the call not
be sent to the phone.
Figure 8-9 Subscribers > Subscriber Template > Conversation (Continued).
If the call is sent to the phone, it can be transferred to the PBX in one of two ways,
Released to Switch and Supervised Transfer.
A Released to Switch transfer enables Unity to send the call to the PBX and is then finished with that call. If the called party does not answer or the phone is busy, it is the
responsibility of the PBX to forward the call back to Unity. You might think that if the
PBX sent the call back to Unity, Unity would just loop the call back to the PBX because
the call transfer settings dictate it. The reason it does not loop is because the call transfer
settings do not affect calls that are forwarded to Unity from the PBX. If the transfer settings were applied to every call, unanswered and busy calls would loop forever.
The second type of transfer is called Supervised Transfer. On a supervised transfer,
Unity holds open the port used to transfer the call and tries to determine whether the call
is answered. When Unity determines that the call is answered, Unity releases the call and
the port is closed. If Unity determines that the call is busy or is not answered within the
number of rings defined for this transfer, Unity pulls the call back and plays the subscriber’s
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greeting. Because Unity maintains control of the call during a supervised transfer, additional features can be implemented, such as holding options and call screening.
Figure 8-10 shows the Call Transfer Setting page. The following steps describe how to
configure the various settings and discuss the effect they have on transferred calls.
Figure 8-10
Subscribers > Subscriber Template > Call Transfer.
Step 1.
From the previous steps, click Call Transfer from the list on the left side of
the screen and proceed to the next step. Otherwise, from within SA, select
Subscribers > Subscriber Template. If the name of the template you want to
manage does not appear in the title strip, click the Find icon and select the
desired subscriber template. After the template is active, click Call Transfer
from the list on the left side of the screen.
Step 2.
Calls can be handled differently during open hours and closed hours. A call
can also be handled differently if the follow-me feature is enabled. Because of
this, it might be necessary to configure transfer settings for both situations.
To accomplish this, use the Transfer Rule Applies To drop-down list to select
the schedule or follow feature that you want to configure. Table 8-2 illustrates
the six types of transfer rules that can be configured.
Step 3.
To enable the type of call transfer that you select in Step 2, select the Enable
radio button.
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Table 8-2
Transfer Rules
Type
Description
Standard
This transfer rule applies during open hours. It
also applies at all hours if no other transfer
rule is enabled.
Closed
If enabled, the Closed transfer rule applies
during closed hours.
Alternate
If the enabled, it supersedes the other transfer
rules and is applied all hours on all days.
Follow-Me Here
It is used with the follow-me feature.
Follow-Me Home
It is used with the follow-me feature.
Follow-Me Mobile
It is used with the follow-me feature.
Step 4.
To configure whether, and to which phone, a call will be sent, select the Yes,
Ring Extension For radio button. If you do not want the phone to ring and
for the call to be sent straight to the subscriber’s greeting, select the No (Send
Directly to Subscriber’s Greeting) radio button. The call can also be sent to
an extension other than the one associated with the subscriber by selecting
the Yes, Ring This Extension radio button. Set the transfer setting to send the
call to the subscriber’s extension. If No (Send Directly to Subscriber’s
Greeting) is selected, the remaining steps in this section do not apply.
Step 5.
By default, Unity plays a prompt to the caller informing her that the call is
being transferred. To disable this prompt, select the Do Not Play the “Wait
While I Transfer Your Call” Prompt check box.
Step 6.
If the call is transferred to an extension, the type of transfer must be selected
under the Transfer Type heading. If holding and screening options are
desired, select the Supervise Transfer radio button. If these features are not
needed, select the Release to Switch radio button. If Release to Switch is
selected, the remaining steps in this section do not apply.
Step 7.
Unity has the capability to queue calls for subscribers if they are on the phone
when a call is transferred from Unity. You must choose one of three options
for Unity to know whether the subscriber’s extension is busy. The choices are
■
Always hold: Informs the caller that the extension is busy and places the
caller on hold
■
No holding: Prompts the caller to leave a message or dial another
extension
■
Ask caller: Informs the caller that the extension is busy and offers the
choice of holding, leaving a message, or dialing another extension
Select the desired action by clicking the associated radio button.
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Note When implementing the option to allow the caller to hold, remember that this will
hold a port open while the caller is on hold. If this option is offered to a number of subscribers, it could adversely affect Unity’s ability to handle incoming calls.
Step 8.
Unity has the capability to gather caller information for screening purposes
on a supervised transfer. There are four options under the Gather Caller
Information heading. These options are as follows:
■
Announce: Unity announces “transferring call” when the subscriber
answers the call.
■
Introduce: Unity announces whom the call is for when the subscriber
answers the phone. This feature is used when more than one person
receives calls on the same phone.
■
Confirm: Unity asks the subscriber to press 1 to accept the call or press
2 to send the call to voicemail.
■
Ask caller’s name: Unity asks the callers to record their names and plays
the recorded names to the subscriber when the phone is answered.
Select the options you want to enable by selecting the radio button next to
each option.
Step 9.
Click the Save icon in the title strip.
Configuring Subscriber Template Greetings
The Greetings setting of the template enables you to define what greetings are enabled
and how they are configured. A subscriber can use up to five greetings, each played
under different circumstances. Table 8-3 lists these greetings and the function of each.
Table 8-3
Greetings
Greeting
Function
Standard
Plays during open hours unless overridden by another greeting. If the Closed
greeting is not enabled, Standard plays after hours.
Closed
Plays during closed hours as defined in the schedule associated with the subscriber.
Busy
Plays when the call is transferred to Unity because the subscriber’s extension
was busy.
Internal
Plays when the caller is another subscriber on the system.
Alternate
When enabled, overrides all other greetings. This greeting plays all hours, all
days until disabled.
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Figure 8-11 shows the Subscriber Template Greetings configuration screen. The following
steps discuss the effects of these settings and describe how to configure them.
Figure 8-11 Subscribers>Subscriber Template>Greetings
Step 1.
After applying the previous steps, click Greetings from the list on the left side
of the screen and proceed to the next step. Otherwise, from within SA, select
Subscribers > Subscriber Template. If the name of the template you want to
manage does not appear in the title strip, click the Find icon, and select the
desired subscriber template. After the template is active, click Greetings from
the list on the left side of the screen.
Step 2.
From this screen, you can enable and configure each of the five greetings.
From the Greeting drop-down list, select the greeting you want to configure.
Start with the Standard greeting.
Step 3.
Under the Status heading, the greeting can be enabled or disabled. If the
greeting is disabled, it will not be used. The Standard greeting is enabled by
default and cannot be disabled. If you are configuring a greeting other than
the standard greeting, select the Enabled radio button to enable this type of
greeting.
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Step 4.
Under the Source heading, the source of the message is chosen. The three
choices for the source are as follows:
■
System: Plays the prerecorded greeting “Sorry, <subscriber’s name> is
not available.” If the subscriber has not yet recorded a name, the greeting “Sorry, the subscriber at extension <subscriber’s extension number>
is not available” is played.
■
Recording: Plays the greeting that the subscriber recorded.
■
Blank: Plays no greeting and goes directly to the after-greeting action.
Typically, Recording is chosen for this setting so that the subscribers can create their own greetings. Select the desired source by clicking the appropriate
radio button.
Step 5.
If Recording is chosen, you can use the media master control panel (the VCRlike panel next to the record label) to record a greeting. This interface also
enables you to copy, cut, and paste the recording.
Step 6.
To allow callers to try to reach another extension or select options during a
greeting, select the Allow Caller Input check box. Notice that caller input is
underlined; this is a link to the Caller Input page. The Caller Input page is covered later in this section.
Step 7.
After the greeting is played, the caller has the opportunity to leave a message.
To enable this, click the Take Message radio button. If you want a different
action to take place, click the Send Caller To radio button and select the
desired destination. Table 8-4 lists the destination options and actions from
the drop-down list.
When choosing to send the call to a call handler, directory handler, interview
handler, or another subscriber, you have to specify the specific handler or
subscriber. A Select Type_of_Object_You_Selected button becomes available
to the right of the type of object you select.
For example, if you select the option to send the call to a subscriber, choose
the Select Subscriber button. When you click this button, a search criteria
window displays. Enter the appropriate criteria and click Find. Select the
desired object from the list. If a subscriber or call handler is selected, you
must also select whether the call should be sent to the greeting or the phone
extension. To send the call to the extension, select Attempt Transfer For
from the Conversation drop-down list. To send the call to the subscriber’s
greeting, select Send to Greeting For from the Conversation drop-down list.
Note Because you have to choose to send the call to the greeting, or attempt to transfer
it, has caused many administrators hours of frustration and troubleshooting. Make sure
that if you want the phone of the subscriber to whom the call is transferred to ring, you
must select Attempt Transfer For.
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Table 8-4
Send Caller To Destinations
Destination
Action
Broadcast Message Administrator
Sends the call to a conversation for sending
broadcast messages.
CVM Mailbox Reset
Sends the call to a conversation that allows the
caller to reset the mailbox. This is only available when the Community Voice-mail (CVM)
package is being used.
Call Handler
Sends the call to the selected call handler.
Caller System Transfer
Sends the callers to a prompt that allows them
to enter another extension to which they
would like to be transferred.
Directory Handler
Sends the call to the directory handler you
select.
Greetings Administrator
Sends the call to a conversation that allows
them to manage the greetings of call handlers
that they own.
Hang Up
Disconnects the call.
Hotel Checked Out
This option works in concert with Cisco
Unity’s Hospitality and Property Management
Integration. It allows guests to archive their
messages when checking out.
Interview Handler
Sends the call to the interview handler you
select.
Sign-in
Sends the call to subscriber sign-in.
In Archived Mailbox
Enables subscribers to access messages on the
Cisco Unity server by entering their old Cisco
Unity primary extension.
Subscriber
Sends the call to another subscriber’s greeting
or extension depending upon how it is configured.
Subscriber System Transfer
Enables callers to transfer to another extension
after they log in with subscriber credentials.
They can then transfer to any number that
their restriction table allows.
Step 8.
By default, Unity plays a prompt after the subscriber greeting that instructs
the caller to leave a message after the tone. Because many subscribers include
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these instructions in their greeting, you might want to disable this prompt.
You can do so by selecting the Do Not Play the “Record Your Message at
the Tone” Prompt check box.
Step 9.
If callers remain silent after being offered the opportunity to leave a message,
they can be reprompted. To have a caller reprompted, select the Reprompt
the User After This Many Seconds of Silence check box and enter the number of seconds to wait for the caller to begin speaking. Then enter the number
of times you want the caller to be reprompted; the maximum is 100.
Step 10. Click the Save icon in the title strip.
Configuring Subscriber Template Caller Input
When callers reach a greeting, a list of options is made available to them. Often the caller
is given the option to dial another extension or reach an operator by pressing a certain
digit. Adding options to a spoken greeting isn’t enough; these options must also be configured under the Caller Input section of the template.
The following steps show how to configure the Caller Input settings, which are shown in
Figure 8-12.
Figure 8-12
Subscribers> Subscriber Template > Caller Input.
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Step 1.
After applying the previous steps, click Caller Input from the list on the left
side of the screen and move on to the next step. Otherwise, from within SA,
select Subscribers > Subscriber Template. If the name of the template you
want to manage does not appear in the title strip, click the Find icon and
select the desired subscriber template. After the template is active, click
Caller Input from the list on the left side of the screen.
Step 2.
The first setting that must be configured determines whether callers are able
to reach a different extension when listening to a subscriber’s greeting. This is
commonly allowed because callers might rather try to reach someone else
instead of leave a message. To allow a caller to enter another extension while
listening to a greeting, select the Allow Callers to Dial an Extension During
Greeting box check.
Step 3.
If the option to allow callers to dial another extension during the greeting is
enabled, an interdigit timeout must also be configured. The Milliseconds to
Wait for Additional Digits setting is the amount of time Unity waits before
deciding that the caller has finished pressing digits. The default value is 1500
milliseconds, which is a second and a half. Typically, this value is adequate. If,
during a greeting, callers are transferred or receive error messages from Unity
before they are finished dialing an extension, increase this value.
Step 4.
From the dial pad on the screen, you can configure the action that is taken
when a digit is pressed. A button can be configured for five types of actions.
Table 8-5 provides descriptions and actions for caller input.
Step 5.
■
Ignore key: No action is taken when digit is pressed.
■
Skip greeting: The greeting is skipped and Unity proceeds to the aftergreeting action.
■
Take message: The caller can press this key to cause Unity to take a
message.
■
Say goodbye: Unity plays a good-bye message and disconnects the call.
■
Send caller to: This option allows various destinations for the caller.
Table 8-5 provides descriptions and actions for caller input.
Click the digit you want to configure, and then select the action by clicking
the appropriate radio button.
As with the Greetings settings, when choosing to send the call to a call handler, directory handler, interview handler, or another subscriber, you have to
designate the specific handler or subscriber.
Step 6.
If the Lock This Key to the Action (Don’t Wait for an Additional Keypress)
check box is selected, an extension that begins with this digit cannot be
entered. Unity then transfers the call to the destination assigned to that key
without waiting to see whether the caller is going to enter other digits. Only
enable this field on digits that are not leading digits for any extensions.
Step 7.
After you have all the digits configured, click the Save icon in the title strip.
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Table 8-5
Caller Input Actions
Action
Description
Alternate Contact Number
Sends the call to the number that you specify
in the Number to Dial field.
Broadcast Message Administrator
Sends the call to a conversation for sending
broadcast messages.
CVM Mailbox Reset
Enables the caller to reset the mailbox (available with Community Voice-mail package).
Call Handler
Sends the call to the selected call handler.
Caller System Transfer
Enables the caller to transfer to another extension after he logs in with subscriber credentials. He can then transfer to any number that
his restriction table allows.
Directory Handler
Sends the call to the directory handler you
select.
Easy Sign-In
Sends the call to a login process that asks for
the password for this mailbox.
Greetings Administrator
Sends the call to a conversation that allows
users to manage the greetings of call handlers
that they own.
Hang Up
Disconnects the call.
Hotel Checked Out
This option works in concert with Cisco
Unity’s Hospitality and Property Management
Integration. It enables guests to archive their
messages when checking out.
Interview Handler
Sends the call to the interview handler you
select.
Sign-In
Sends the call to the subscriber sign-in.
Sign-In Archived Mailbox
Enables subscribers to access messages on the
Cisco Unity server by entering their old Cisco
Unity primary extension.
Subscriber
Sends the call to another subscriber’s greeting
or extension, depending upon how it is configured.
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Table 8-5
Caller Input Actions
Action
Description
Subscriber System Transfer
Enables callers to transfer to another extension
after they log in with subscriber credentials.
They can then transfer to any number that
their restriction table allows.
Configuring Subscriber Template Messages Settings
The Subscriber Template Messages section enables you to configure caller message-specific attributes, such as message length, post-message options, caller options, and MWI
settings.
Figure 8-13 shows the various settings that are configured from this screen. The following
steps illustrate how to configure these settings and describe the effect on each setting.
Figure 8-13 Subscribers > Subscriber Template > Messages
Step 1.
From the previous steps, click Messages from the list on the left side of the
screen and proceed to the next step. Otherwise, from within SA, select
Subscribers > Subscriber Template. If the name of the template you want to
manage does not appear in the title strip, click the Find icon, and select the
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desired subscriber template. After the template is active, click Messages from
the list on the left side of the screen.
Step 2.
To set the maximum length of a message that an outside caller can leave, enter
the value in seconds in the Maximum Message Length, in Seconds box. This
setting applies only to messages left from outside callers. The limit for subscribers is set by the subscriber’s CoS.
Step 3.
After the message is recorded, select either the Say Goodbye or the Send
Caller To radio button. When choosing Send Caller To, the destination must
be chosen. The destinations are the same as those under the Send Caller To
drop-down list in the caller input section.
Step 4.
If you want to let callers listen to, change, or delete the message they leave,
the select the Callers Can Edit Message check box.
Step 5.
Allow the caller to label a message urgent by selecting the Ask Caller for
Their Preference radio button under the Mark Messages as Urgent heading.
To mark a message always urgent without asking the caller, select the
Always radio button, or to never mark the message urgent, select the Never
radio button.
Step 6.
Callers can be given the option to mark a message private. When messages
are marked private, they can be encrypted based on the message security settings. To always have a message marked private, select the Always radio button in the Mark Messages as Private section. To never have a message
marked private, select the Never radio button. To allow the caller to decide
whether he wants to have the message marked private, select the Ask Caller
for Their Preference radio button.
Step 7.
Select the language in which Unity plays its prompt from the Language That
Callers Hear drop-down list. When this field is set to Inherited, it uses the
language used by the parent call handler.
Step 8.
Under the Message Waiting Indicators (MWIs) heading, enable MWI by
selecting the Use MWI for Message Notification check box. Entering an X
in the Extension box causes the extension associated with the subscriber to
be activated. If you place an extension number in this field, that extension’s
MWI is activated when users created from the template receive a voicemail. If
the Send Count check box is selected, the message count appears on the
screen of phones that support this feature.
Step 9.
If you do not want to use MWI, simply deselect the Use MWI for Message
Notification check box.
Step 10. Click the Save icon in the title strip.
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Configuring Subscriber Template Distribution Lists Settings
The most efficient way to add subscribers to a public distribution list is by adding them
through the subscriber template. The following steps show how this is done:
Step 1.
From the previous steps, click Distribution Lists from the list on the left side
of the screen and proceed to the next step. Otherwise, from within SA, select
Subscribers > Subscriber Template. If the name of the template you want to
manage does not appear in the title strip, click the Find icon and select the
desired subscriber template. After the template is active, click Distribution
Lists from the list on the left side of the screen.
Step 2.
Figure 8-14 shows that the available Public Distribution Lists are displayed
on the left side of the screen and the New Subscribers Added To lists are on
the right side. To move a list from one column to the other, highlight the
desired list and select either the >> button or the << button.
Figure 8-14 Subscribers > Subscriber Template > Distribution Lists
Step 3.
Click the Save icon in the title strip.
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Configuring Subscriber Template Message Notification Settings
Message notification allows a subscriber to be informed through a pager, phone, or
other external device when a new message arrives. This type of notification is very useful for subscribers who are seldom in the office, such as outside salespeople or service
technicians.
It might seem odd that you can set up message notification in the subscriber template.
After all, you will not want to send a message to the same pager when any subscriber,
who was created using this template, receives a new message. However, the subscriber
template is an excellent place to create the same type of flow and generic notification
settings for a large number of subscribers. After the subscribers are added, you can have
them fill in the details such as the specific phone numbers using Cisco Personal
Communications Assistant (CPCA).
More than a dozen devices can be configured for message notification. This includes
eight phones, four pagers, and a number of text-based devices. Message notification can
be configured to inform the subscriber of any new voicemails, faxes, and emails. The
settings on this screen determine what type of message triggers a notification and what
device(s) are notified.
The following steps show how these settings are configured and discuss the effect they
produce:
Step 1.
From the previous steps, click Message Notification from the list on the left
side of the screen and move on to the next step. Otherwise, from within SA,
select Subscribers > Subscriber Template. If the name of the template you
want to manage does not appear in the title strip, click the Find icon, and
select the desired subscriber template. After the template is active, click
Message Notification from the list on the left side of the screen.
Step 2.
Figure 8-15 shows the first few settings that must be configured. First you
must select which device you want to configure. From the Device drop-down
list, choose a device.
Step 3.
The next few configuration fields vary slightly depending on whether you
choose a phone or an email as the notification device. If you choose a phone
or a pager, enter the phone number of that device in the Phone Number
field. After dialing the phone number, enter any digits that still need to be
dialed in the Extra Digits field. Extra digits are typically needed for pagers
that use PINs.
If you choose a device that will use an email address as the delivery mechanism, such as a text pager or Text for Visual Messaging Interface (VMI), enter
the email address in the To field. Then enter the pilot number for Unity in the
From field. Some cell phones will act as pagers and can call back the number
that is in the From field. Figure 8-16 shows these fields.
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Figure 8-15 Subscribers > Subscriber Template > Message Notification (Phone
or Pager).
Note VMI is the old name for CPCA Inbox, which enables a subscriber to check messages from a web page.
Step 4.
When a phone is chosen as the notification device, you can set it so that the
extra digits are not sent until a connection is detected, as seen in Figure 8-15.
To enable this option, select the Try to Detect Connection check box. If you
want it to wait a specified number of seconds, instead of trying to detect a
connection, select the Seconds to Wait Before Dialing Extra Digits check
box, and enter the number of seconds to wait in the box to the right of that
field. You can use the Seconds to Wait option if Unity is having trouble
detecting a connection because of various factors, such as poor line quality.
In some cases you might need to use both the Try to Detect a Connection
and Seconds to Wait options. It might be necessary to set the parameters and
then test these settings a few times before getting them to work properly with
different paging companies.
Step 5.
When a device that uses an email address is selected, enter the text that you
want to appear in the notification in the Text field, as seen in Figure 8-16. A
message count can be included in this notification by selecting the Include
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Voice-mail, E-mail, and Fax Counts check box. Caller information can also
be included by selecting the Include Caller Information check box.
Figure 8-16
Device)
Subscribers > Subscriber Template > Message Notification (E-Mail
Step 6.
To enable this device to be used for message notification, you must select the
Enabled radio button. It is not enough to simply configure the device—you
must ensure that the Enable button is selected.
Step 7.
Under the Notify Subscriber Of heading, select the types of messages of
which the subscriber should be notified. As shown in Figure 8-16, you can
select all messages or only one type, such as voicemail. You can further narrow it down by selecting only urgent messages.
Step 8.
The next area of the screen, shown in Figure 8-17, is a grid that represents the
days of the week, with each day broken into half-hour segments. This grid is
used to define at what hours of the day notifications are sent out. The boxes
that have a check mark represent the time of day that notifications will be
sent. Edit the schedule as needed.
Note You can copy the open/closed hours of one day to another day, and to all weekdays or weekends, by using the Copy Day’s Schedule field under the grid. Select the
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Source Day from the first drop-down list and the Destination Day(s) from the second
drop-down list. Then click Copy Day’s Schedule, as shown in Figure 8-17.
Figure 8-17 Subscribers > Subscriber Template > Message Notification
Step 9.
The notification can be configured to be sent immediately or to wait a specified number of minutes. If cascading notification is being deployed, some
devices are configured to wait for a certain number of minutes before being
notified. The term cascading notification means that notification is sent, and
if the message has not been retrieved after a specified number of minutes, a
different device is notified. If, after an additional amount of time, the message
is still not retrieved, another device is notified. An example of cascading flow
might look something like this:
1.
First notification is sent to your pager.
2.
You don’t retrieve the message.
3.
Ten minutes later, a notification is sent to your cell phone.
4.
You don’t retrieve the message.
5.
Ten minutes later a notification is sent to your boss’s pager.
6.
You’re busted!
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Step 10. Enter the number of minutes that Unity should wait before notifying this
device in the Send Initial Notification After How Many Minutes field. Unity
only sends the notification if the message has not been retrieved after this
amount of time.
Step 11. To have Unity send another notification to this device only when a new message arrives, select the Restart Notification Each Time a New Message
Arrives radio button. To invoke what some refer to as the “nag factor,” you
can have the device notified every couple of minutes until the message is
retrieved by selecting the Repeat Notification if There Are Still New
Messages After This Many Minutes radio button. Enter the number of minutes in the field to the right of this label.
Note If the device you are configuring is a text pager or Text for VMI, none of the following settings display. Simply click the Save icon in the title strip to complete this task.
Step 12. Under the If Device Does Not Answer heading, enter the number of rings
Unity should wait before hanging up, how many times Unity should try again,
and how many minutes Unity should wait between attempts in the appropriate fields.
Step 13. Under the If Device Is Busy heading, enter the number of times Unity should
try again and how many minutes Unity should wait between attempts in the
appropriate fields.
Step 14. If the notification fails, you can specify another device to be informed by
selecting the device from the If Notification Fails, Send Notification To
drop-down list.
Step 15. Click the Save icon in the title strip.
Configuring Subscriber Feature Settings
The Features configuration page allows you to enable and configure certain features for
SBs. This area deals with broadcast messages, message locator, and message encryption.
You might notice that this area does not include all features that Unity offers. I refer to
the features in this section as the enhanced messaging features. The following steps offer
more details on each feature and show how to configure them:
Step 1.
From the previous steps, click Features from the list on the left side of the
screen and proceed to the next step. Otherwise, from within SA, select
Subscribers > Subscriber Template. If the name of the template you want to
manage does not appear in the title strip, click the Find icon and select the
desired subscriber template. After the template is active, click Features from
the list on the left side of the screen. A screen similar to Figure 8-18 appears.
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Figure 8-18 Subscribers > Subscriber Template > Features
Step 2.
Broadcast messages are messages that are sent to all users on the Unity server.
These messages do not cause the MWI to light and are played before other
messages. The user cannot skip them. They can be configured to expire automatically. Because these messages affect everyone on the system, you want to
limit their accessibility. To grant rights to send broadcast messages only to
users on this Unity server, select the Subscriber Can Send Broadcast
Messages to Subscribers on This Server check box. To allow subscribers to
send them to users on all Unity servers that are networked together, select the
Subscriber Can Send Broadcast Messages to Subscribers on Multiple
Servers check box. If you want to allow SBs to modify broadcast messages,
select the Subscriber Can Update Broadcast Messages Stored on This
Server check box.
Step 3.
Message locator allows SBs to search their voicemails when calling in to
Unity. When this feature is enabled, the SB can search by extension, subscriber name, or outside caller phone number. Because this feature can be
processor intensive, take care when determining how many SBs have access to
it. To enable this feature, select the Subscribers Can Use Message Locator
check box. To allow the results of a search to appear on the screen of a Cisco
IP Phone, select the Subscriber Can Use Phone View Visual Message
Locator check box. By default, message locator does not search non-delivery
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receipts (NDR) or return (read and delivery) receipts. To include these in the
search, select the Include Receipts in Searches check box.
Step 4.
Unity has the ability to encrypt messages so that only the intended recipient
can listen to it. By default this is disabled. To leave it disabled, ensure that the
Do Not Encrypt Messages radio button is selected. To encrypt only messages marked as private, select the Encrypt Only Private Messages radio
button. To enable encryption for all messages, select the Encrypt All
Messages radio button.
Step 5.
Click the Save icon in the title strip.
At this point, you have completed creating a subscriber template. Make sure to take some
time to figure out the number of templates you will need, based on the information discussed at the beginning of this chapter. Add the templates to Unity using the preceding
steps as a guide.
Now that the subscriber templates are created, you can start to add subscribers. The next
section explains how to add Exchange-based subscribers. A lot of configuration is necessary to get to this point. However, the success of your deployment rests heavily upon an
accurately prepared foundation. Okay, enough of this foundation stuff; now start adding
subscribers!
Creating New Unity Subscribers
Now that you have the subscriber templates constructed, the process of creating a user is
fairly simple. To create a subscriber, enter a few pieces of information and click Add, and
the user is created. Because subscriber creation is based on one of the templates you created, all the additional settings will be populated. There are times when you want to
change the settings of a subscriber; the process for making the most common changes
comes later in this chapter in the “Managing Users” section.
This section focuses on the creation of Exchange-based subscribers. The next section
discusses the import process for existing Exchange subscribers.
The following steps illustrate how to add a new Exchange-based subscriber to Unity:
Step 1.
From within SA, select Subscribers > Subscribers.
Step 2.
Click the Add icon (plus sign) in the title strip.
Step 3.
A window like that shown in Figure 8-19 displays. Select the New
Subscriber radio button and select Exchange from the drop-down list to the
right of the label.
Note If you have more than one Exchange server, click the Select button, which appears
next to the name of the default Exchange server near the bottom of the page. A search
criteria window opens. Click the Find button and select the Exchange server from the list
provided. The reason you perform this step first is that if you fill in the other fields and
then select a different Exchange server, all the information you entered disappears and
you need to reenter it.
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Figure 8-19 Add Subscriber Window
Step 4.
Enter the first and last name of the new subscriber in the fields labeled as
such. The display name is automatically generated. This is the name that is displayed in Exchange and the SA. If needed, you can edit this.
Step 5.
An extension number must be assigned to every subscriber, and this should
be the extension number that is assigned to the subscriber’s phone. Enter the
subscriber’s extension number in the Extension field. In rare cases, the extension number in Unity is different than the actual extension assigned to the
subscriber’s phone, but this should be avoided. All extensions in Unity must
be unique—that is, each extension can only be assigned to one subscriber.
Step 6.
In the Fax ID field, enter the number that a caller dials to send a fax to this
subscriber.
Step 7.
Select the subscriber template that is used during the creation of this subscriber from the Template drop-down list.
Step 8.
The Exchange alias is automatically populated based on the settings you
select on the Subscriber Template Profile screen, but this can be edited if
needed. One of the options available was not to populate the Exchange Alias
field automatically. If the aforementioned was selected, this field must be
manually configured.
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Step 9.
As mentioned in the note at the beginning of these steps, you should have
already chosen the desired Exchange server. Be aware that selecting a different Exchange server at this point will cause all the information entered on this
page to disappear and require reentering.
Step 10. If more than one mailstore is available, select the one you want from the
Mailstore drop-down list.
Step 11. Click the Add button.
Step 12. The information is processing, and it can take up to 15 seconds. During this
time, an Exchange mailbox and Windows domain account is being created.
The Unity-specific information is being added to the SQL database. After the
user is created, you return to the subscriber profile screen of the newly created subscriber.
Note You do not have to click the Save icon when adding a user. That is because you
only have to save when you are changing something. Whenever something new is added, it
is written to the database automatically.
So, that is all there is to it. The subscriber you added is now ready for use. As you can
see, adding a subscriber is a much simpler process than creating subscriber templates.
Because you use a template during the creation process, the majority of the subscriber
settings are automatically configured, which shows the value of the subscriber template.
However, there are three types of subscriber configurations that are not configured using
the subscriber template. These are private lists, alternate extensions, and alternate names.
These settings are not part of the subscriber template because they are unique for each
subscriber.
Private lists are lists of other subscribers with whom a subscriber can communicate when
sending voicemail. Instead of having to add each subscriber as a recipient, the subscriber
can select a private list as the recipient, and the voicemail will be sent to everyone in the
list. Subscribers are encouraged to create their own private lists, but these lists can be
created and managed using SA.
Alternate extensions are used for multiple purposes. When an alternate extension is
added to a subscriber, Unity treats all calls to or from that extension as if they are to or
from the subscriber’s primary extension. Think of it as a way to associate multiple extensions to one mailbox. Imagine that a subscriber has multiple extensions on the system.
When someone calls the subscriber and the call is not answered, the caller is forwarded
to the same mailbox regardless of which of the subscriber’s extensions the caller dialed.
This feature is also used to allow subscribers easy message access when calling from an
outside phone. When calling from the outside, a subscriber must press the asterisk (*) key
and enter an extension number and password. By adding the phone number of phones
that are commonly used to access voicemail, such as a cell phone, this process is simplified. Because Unity recognizes the incoming caller ID as an alternate extension, the subscriber is simply being prompted for a password, just as if calling from an office phone.
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Alternate names enable you to enter additional names that a subscriber can go by such as
nicknames, shortened names (Bob instead of Robert), or a maiden name. These names are
included in a search when voice recognition is used.
Configuring private lists, alternate extensions, and alternate names are covered in the
“Managing Users” section of this chapter.
Importing Unity Subscribers
When integrating Unity with an existing Exchange server, you can import the users from
the existing mailstore. This can be done one at a time or in bulk by using the Unity Bulk
Import Wizard. When importing a large number of users, use the Bulk Import Wizard.
When importing only one or two users, SA can be used.
The following steps illustrate how to import a single Exchange user. The steps for importing a Domino user are very similar:
Step 1.
From within SA, select Subscribers > Subscribers.
Step 2.
Click the Add icon.
Step 3.
Select the Import Existing Exchange User radio button.
Step 4.
Click the Select button. A search criteria window displays like that shown in
Figure 8-20.
Step 5.
Select the Exchange radio button.
Step 6.
Ensure that the user’s domain is in the Domain field.
Step 7.
You can search for the user by alias, first name, or last name. Select one of
these choices from the Find By drop-down list and enter the search criteria in
the next field. Click Find.
Step 8.
A list of users that match the search displays. Click the first name of the user
you want to import. An Add Subscriber window like that shown in Figure 821 displays.
Step 9.
Only the Extension, Fax ID, and Template fields are edited on this screen. All
other fields are populated using the information imported from the mailstore
account. Enter the extension number in the Extension field. If the subscriber
has a fax extension, enter it and select the subscriber template you want to
use from the Template drop-down list.
Step 10. Click Add.
As mentioned previously, subscribers can be added using the Bulk Import Wizard. This
utility can import users from an existing mailstore or from a CSV file. When importing
from a CSV file, Unity adds Exchange mailboxes and Windows domain accounts for the
imported users if these accounts do not already exist. Because the accounts already exist
when importing from an Exchange server, there is no need for Unity to create these.
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When importing from a Domino server, only Unity subscribers are created; no AD
accounts are automatically created for these subscribers.
Figure 8-20 Exchange User Search Screen
The Import utility functions in much the same way, regardless from which source you are
importing. The following steps are required when importing from a CSV file:
Step 1.
Create a CSV file. There are more than 40 values that can be entered in the
CSV file for each subscriber. Of these fields, only three are required:
FIRST_NAME, LAST_NAME, and DTMF_ACCESS_ID. The remaining
fields are populated with the settings defined in the subscriber template you
choose during the import.
Step 2.
Using the start menu on the Unity server, choose Start > Programs > Unity >
Cisco Unity Bulk Import.
Step 3.
The Import Wizard starts. Click Next.
Step 4.
You are asked whether you want to import the users from a CSV file or AD.
Choose CSV and click Next.
Step 5.
You can accept the default for the log files location and click Next.
Step 6.
The next page asks you to select what type of subscriber you are importing.
In most cases, you will choose either Unified Messaging or Voice-mail Only.
Choose the desired type of subscriber and click Next.
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Figure 8-21 Importing an Exchange User
Step 7.
On the following screen, accept the default to create new mailboxes and
Windows account. Click Next.
Step 8.
Choose a template to apply for the users you will import. Select the appropriate subscriber template and click Next.
Step 9.
Choose a domain and container in which to create the new users. Click
Browse, highlight the appropriate container, and click OK. You will return to
the previous window; click Next.
Step 10. Select your server in the next window. Make sure that you have highlighted
your server before continuing. Click Next.
Step 11. Select the CSV file you want to use. Click Browse. Locate the CSV file and
click Next.
Step 12. The data is examined and you are presented with a summary of findings.
Click Next.
Step 13. Select the box of each subscriber you want to import. You can also click
Select All. Click Next.
Step 14. Confirm that you want to import the data that is shown in the summary
screen. Click Next.
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Step 15. When the import is complete, you see a final summary screen. You can view
the output and error log from this screen. Click Finish.
Now that the subscribers are added, your system is ready for basic voicemail use. Users
can now configure their personal settings, such as passwords and greetings, by dialing the
voicemail pilot number and going through subscriber self-enrollment.
Next you examine how these same tasks are accomplished within Unity Connection.
Creating Unity Connection User Templates
Now that you understand how to create Unity user templates, you need to explore how
to configure user templates for Unity Connection. Many of the tasks are similar, so those
of you that have read the Unity section might feel that you have read some of this
already. Each section is purposefully not dependent on the previous one. This is done so
that you can go directly to the section you are interested in and retrieve the information
without having to bounce back and forth across multiple sections or chapters.
Before moving on to the actual process of creating user templates, it is important that
you understand that these templates are significant only at the point of creation. This
means that the setting of a template affects the user only when the user is first created. If
you create 50 users using a template, changes you make to that template after the users
are created do not affect the user’s settings. If you need to change certain settings on a
large number of users, use the Bulk edit mode. This utility is discussed in Chapter 11.
There are more than 80 configurable settings for each user, so there must be at least that
many settings for each user template. It is important to understand the effect of each setting. Figure 8-22 shows the Edit drop-down list that displays each configurable category
of a user template.
Table 8-6 lists the types of settings that each section contains.
Note The Phone Menu, Playback Message and Send Message settings are important to
the success of the system, because allowing control over the settings makes the new system more acceptable to the end user. I was once asked to deliver some end-user training
for a company whose employees were very frustrated with the system. As it turned out,
90 percent of their concerns related to the Phone Menu, Playback Message and Send
Message settings. After they learned to configure these settings themselves, the majority
of the issues were resolved.
Two user templates are created during the installation process. These are the administrator template and the voicemail user template. As their names imply, they each have settings that are commonly found in administrative-level users and voicemail users.
Note Although you can use the default voicemail user template to create users, it is best
to confirm the template’s settings first. Do not assume that the default settings are the best
settings for your deployment.
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Figure 8-22 User Template
At this point, you should have an idea of the different templates you need. We go
through the process of creating and managing a template so that you fully understand all
the settings that must be configured. We begin by adding a new template. The following
steps are required to add a new user template when creating users with mailboxes:
Step 1.
From within Unity Connection Administrator (UCA), select Templates > User
Templates.
Step 2.
Click the Add New button.
Step 3.
A window similar to Figure 8-23 displays. To create a template for a standard
type of user, select User Template With Mailbox from the User Template
Type drop-down list.
Note You would select User Template Without Mailbox from the User Template Type
drop-down list when creating a template that will be used to create administrative users.
Step 4.
From the Based on Template drop-down list, select the existing template that
you want to use as a starting point for creating the new template.
Step 5.
Enter a unique name for the template in the Alias field.
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Figure 8-23 Templates > User Templates (Adding)
Step 6.
Enter a description name in the Display Name field.
Note Often in a single Unity Connection server environment, you only need to configure
the first four fields.
Step 7.
From the Mailbox Store drop-down list, select the mailbox store that the
mailboxes are created on for users created using this template.
Step 8.
If you are implementing the fax feature of Unity Connection, you can select
the fax server from the Outgoing Fax Server drop-down list.
Step 9.
From the Phone System drop-down list, select the phone system to be associated with users.
Step 10. Click Save.
The creation of a template is the easy part. Now you need to configure the settings within each submenu. As Figure 8-22 shows, there are 13 submenus under User Templates.
The configuration of settings in each submenu follows. We start by configuring the template’s basic settings. Figure 8-24 shows the User Templates Basic page, and the following
steps describe how to configure these settings:
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Table 8-6
User Template Categories
Category
Description
User Template Basics
General settings, such as schedules, location,
class of service, and time zones are set in this
area.
Password Settings
This area enables you to determine whether the
user’s account will be available for use as soon
as it is created or whether an administrator will
have to unlock it first. It also defines whether
users have the ability to change their own
password and whether it expires automatically.
Change Password
This area enables you to configure default
passwords.
Roles
This area enables you to define what roles are
assigned to users created using this template.
Transfer Rules
This area determines what action is taken when
a user’s extension is dialed from within Unity.
Most of the time, this occurs when an outside
caller dials the extension from the auto-attendant.
Message Settings
The settings in this area determine the length
permitted for messages, whether callers can
edit messages, and post-message actions.
Message Actions
Determines what Unity Connection does with
each type of message. There are four types
that can be configured: Voicemail, Email, Fax,
and Delivery Receipt.
Caller Input
Provides a number of choices, such as to skip
the greeting by pressing # or to reach an operator by pressing 0. The Caller Input settings
determine what happens if a digit is pressed
while the greeting is being played.
Mailbox
Defines the size of a user’s mailbox.
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Table 8-6
User Template Categories
Category
Description
Phone Menu, Playback Message Settings,
Send Message Settings
These settings enable you to configure how
the system interacts with a user. This includes
things such as volume, menu style, time format, message playback order, and many other
settings. If you allow the users access to the
web-based Cisco Unity Connection Assistant,
users can change most of these settings themselves.
Greetings
The Greetings area configures which greetings
are enabled, what action is allowed during the
greeting, and what action is taken after the
greeting.
Post Greeting Recording
This area enables you to determine whether an
additional greeting is played after the user’s
greeting and, if so, which greeting to play.
Notification Devices
The devices used to notify a user of a new
message are configured in this area.
Configuring User Template Basics Settings
The following steps describe how to configure the settings on the User Templates
Basic page:
Step 1.
From the previous steps, select User Template Basic from the drop-down Edit
menu. Otherwise, from within UCA, select Templates > User Templates.
Select the template you want to manage from the list that appears. After the
template is active, select User Template Basic from the drop-down Edit menu.
Step 2.
The Alias and Display Name fields were defined when the template was created. You can change them now, but in most cases, they are left alone.
Step 3.
The Display Name Generation selection determines how the display name is
formatted. Select either First Name, Then Last Name or Last Name, Then
First Name.
Step 4.
The Outgoing Fax Server field was defined when the template was created.
You can change it now, but in most cases, it is left alone.
Step 5.
From the Partition drop-down list, select the partition that the primary
extension of the user is associated with.
Step 6.
From the Search Space drop-down list, select the search space that is associated to objects created using this template.
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Figure 8-24 Templates>User Templates>Edit>User Templates Basics.
Step 7.
The Phone System field is defined when the template is created. You can
change it now, but in most cases, it is left alone.
Step 8.
Select the desired Class of Service and Active Schedule from the appropriate
drop-down lists.
Step 9.
To assure that users record their name and greeting and change their passwords on the next login, select the Set User for Self-Enrollment at Next
Login check box. Users can choose to be listed in the directory during the
self-enrollment process if the users’ CoS allows this option.
Step 10. To list users in the auto-attendant directory, select the List in Phone
Directory check box.
Step 11. If a message is not successfully delivered, Unity Connection can send a notice
to the sender. To enable this select the Send Non-Delivery Receipts on Failed
Message Delivery checkbox. To disable this feature, deselect the check box.
Step 12. To allow a user to automatically log in to voicemail when calling from the
extension that is associated to his mailbox, select the Skip PIN When Calling
from a Known Extension check box. When this is enabled, the user will not
need to enter a password.
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Step 13. The Use Short Calendar Caching Poll Interval field is only used if calendar
integration is enabled. When this box is selected, Unity Connection will use
the short calendar caching poll interval value set in the external services configuration page to determine how often to update calendar information.
Step 14. The Location section of this page allows you to configure physical address,
time zone, language, and billing ID information. Configure these parameters
as needed.
Step 15. Click Save.
Configuring Password Settings
Next, configure account information for the template:
Step 1.
From the previous steps, select Password Settings from the drop-down Edit
menu. Otherwise, from within UCA, select Templates > User Templates.
Select the template you want to manage from the list that appears. After the
template is active, select Password Settings from the drop-down Edit menu.
Step 2.
You can configure settings for either the voicemail password or the web application (Cisco Personal Communication Assistant). Select which password you
want to configure settings for from the Choose Password drop-down list.
Step 3.
The Locked by Administrator check box allows you to lock the account at
the point of creation. You might want the account to be locked so that the
account cannot be used before the deployment is complete. Decide whether
you want the accounts you create using this template to be locked upon creation and select the check box accordingly.
Step 4.
To prevent the user from being able to change her password, select the User
Cannot Change check box. In most cases, you want users to be able to
change their passwords, so this is left deselected.
Step 5.
Typically, you want newly created users to change their password on their
first login, so the User Must Change at Next Sign-In box should be selected.
Step 6.
To override the password expiration settings, you can select the Does Not
Expire check box. This will prevent the password from expiring. It is recommended that users be required to change passwords, so in most cases, this
box is left deselected.
Step 7.
From the Authentication Rule drop-down list, select the authentication rule
that you want to assign to this template.
Step 8.
Click Save.
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Configuring Template Passwords
The User Template Change Password page enables you to define the default password for
this template.
To configure these settings, follow these steps:
Step 1.
From the previous steps, select Change Password from the drop-down Edit
menu. Otherwise, from within UCA, select Templates > User Templates.
Select the template you want to manage from the list that appears. After the
template is active, select Change Password from the drop-down Edit menu.
Step 2.
You can configure the voicemail or the web application (Cisco Personal
Communication Assistant) password. Select which password you want to configure settings for from the Choose Password drop-down list.
Step 3.
Enter the password you want to have assigned to users that are created using
this template in both the Password and Confirm Password fields.
Step 4.
Click Save.
Configuring Roles
Roles define the administrator rights of a user. This was discussed in more detail in the previous chapter. The following steps illustrate to the process of assigning roles to a template:
Step 1.
From the previous steps, select Roles from the drop-down Edit menu.
Otherwise, from within UCA, select Templates > User Templates. Select the
template you want to manage from the list that appears. After the template is
active, select Roles from the drop-down Edit menu.
Step 2.
Select the desired roles from the Available Roles box, and click the Up Arrow
above that box to move them to the Assigned Roles box.
Step 3.
Click Save.
Configuring User Template Transfer Rules
The template transfer rules determine the outcome of a call when an outside caller
attempts to reach a user’s phone from the auto-attendant. An outside caller can attempt
to reach a user by entering the user’s extension number, using Unity Connection’s dialby-name feature or by speaking the name if voice recognition is used. When Unity
Connection finds a match for either the extension number or the name, it looks at the
user’s transfer rules settings to determine where to send the call. One would think that
Unity Connection would send the call to the CM so that the call could be delivered to
the phone. Although this is the most common action, it is not always desired. A number
of circumstances can dictate that the call not be sent to the phone.
If the call is sent to the phone, it can be transferred to the Communications Manager in
one of two ways. The first is called Release to Switch. In this type of transfer, Unity
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Connection sends the call to the Communications Manager and is finished with that call.
If the called party does not answer or the phone is busy, it is the responsibility of the
Communications Manager to forward the call back to Unity Connection. You might
think that if the Communications Manager sent the call back to Unity Connection, Unity
Connection would just loop the call back to the Communications Manager because the
call transfer settings dictate it. The reason it does not loop is because the transfer rules
do not affect calls that are forwarded to Unity Connection from the Communications
Manager. If the transfer rules were applied to every call, unanswered and busy calls would
loop forever.
The second type of transfer is called Supervised Transfer. On a supervised transfer,
Unity Connection holds open the port used to transfer the call and tries to determine
whether the call is answered. When Unity Connection establishes that the call is
answered, Unity Connection releases the call and the port is closed. If Unity Connection
decides that the call is busy or unanswered within the number of rings defined for this
transfer, Unity Connection pulls the call back and plays the user’s greeting. Because
Unity Connection maintains control of the call during a supervised transfer, additional
features can be implemented, such as holding options and call screening.
Figure 8-25 shows the standard Transfer Rule page. The steps that follow describe how to
configure the various settings and discuss the effect they have on transferred calls:
Figure 8-25 Templates > User Templates > Edit > Transfer Rules (Standard).
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Step 1.
From the previous steps, select Transfer Rules from the drop-down Edit
menu. Otherwise, from within UCA, select Templates > User Templates.
Select the template you want to manage from the list that appears. After the
template is active, select Transfer Rules from the drop-down Edit menu.
Step 2.
There are three rules that can be configured:
■
Standard rule: Determines how the transfer is handled during open
hours.
■
Closed rule: If enabled, determines how the transfer is handled during
closed hours.
■
Alternate rule: If enabled, will override the Standard and Closed rule
and the transfer will be handled according to its settings.
Select the transfer rule you want to configure.
Step 3.
The Rule Name field is static and cannot be changed.
Step 4.
The Status section determines whether the rule is enabled. The Standard rule
cannot be disabled. If you are configuring one of the other rules, you must
enable it for it to take effect. You can select Enable with No End Date and
Time to have it always enabled. If you want to have it enabled until a specific
time and date, select Enable Until and end the desired date and time.
Step 5.
In the Transfer Action section, select Extension to have the call transferred
to the user’s extension. If you do not want the call to be sent to the user’s
phone, select Greeting and the call will be transferred directly to the greeting.
Step 6.
If the call is transferred to an extension, the type of transfer must be selected
under the Transfer Type heading. If holding and screening options are
desired, select the Supervise Transfer radio button. If these features are not
needed, select the Release to Switch radio button. If Release to Switch is
selected, the remaining steps in this section do not apply.
Step 7.
In the Rings to Wait field, enter the number of rings that Unity Connection
should wait before sending the call to the user’s voicemail greeting.
Step 8.
Unity Connection can let the caller know that it is going to transfer the call.
To enable this feature, select the Play the “Wait While I Transfer Your Call”
Prompt check box.
Step 9.
Unity Connection has the capability to queue calls for users if they are on the
phone when a call is transferred from Unity Connection. You must choose
one of three options for Unity Connection to use if the user’s extension is
busy. The choices are
■
Send the Caller to Voicemail: If selected, the call is not placed on hold
but sent directly to voicemail.
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■
Put Callers on Hold Without Asking: If selected, the caller is placed on
hold. The third choice,
■
Ask Callers to Hold: If selected, informs the caller that the extension is
busy and offers the choice of holding.
Select the desired action by clicking the associated radio button.
Note When implementing the option to allow the caller to hold, remember that this will
hold a port open while the caller is on hold. If this option is offered to a number of subscribers, it could adversely affect Unity Connection’s capability to handle incoming calls.
Step 10. Unity Connection has the capability to gather caller information for screening
purposes on a supervised transfer. There are four options under the Call
Screen heading. These options are as follows:
■
Tell Me When the Call Is Connected : Unity Connection announces
“transferring call” to the user when the user answers the call.
■
Tell Me Who the Call Is For: Unity Connection announces whom the
call is for when the user answers the phone. This feature is used when
multiple users receive calls on the same phone.
■
Ask Me If I Want to Take the Call: Unity Connection asks the user to
press 1 to accept the call or press 2 to send the call to voicemail.
■
Ask for Caller’s Name: Unity Connection asks the caller to record his
name and plays the recorded name to the user when the phone is answered.
Select the options you want to enable by selecting the radio button next to
each option.
Step 11. Click Save.
Configuring User Template Messages Settings
The User Template Messages section enables you to configure caller message-specific
attributes, such as message length, post-message actions, and caller options.
Figure 8-26 shows the various settings that are configured from this screen. The following steps illustrate the configuration process of these settings and describe the effect of
each setting:
Step 1.
From the previous steps, select Message Settings from the drop-down Edit
menu. Otherwise, from within UCA, select Templates > User Templates.
Select the template you want to manage from the list that appears. After the
template is active, select Message Settings from the drop-down Edit menu.
Step 2.
To set the maximum length of a message that an outside caller can leave, enter
the value in seconds in the Maximum Message Length, in Seconds box. This
setting applies only to messages left from outside callers. The limit for users is
set by the user’s CoS.
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Figure 8-26 Templates > User Templates > Edit > Message Settings
Step 3.
If you want to let callers listen to and change or delete the message they leave,
select the Callers Can Edit Message check box.
Step 4.
The Language setting defines what language is used to play the system.
When Inherit Language from Caller is selected, the language that was set by
the handler or rule that transferred the call is used. If it is set to Use System
Default Language, the system default language is used. To specify a particular language, choose the radio button next to the drop-down list of languages.
If only one language is installed on this system, the drop-down box is not
available.
Step 5.
Allow the caller to label a message urgent by selecting the Ask Caller radio
button under the Message Urgency heading. To mark a message always urgent
without confirmation from the caller, select the Mark Urgent radio button or,
to never mark the message urgent, select the Mark Normal radio button.
Step 6.
To have messages that are left by outside or unknown callers marked secure,
select the Mark Secure check box.
Step 7.
After the message is recorded, the call can be sent to a number of destinations. Typically, the call is sent to the goodbye call handler, but you can pick
from any of the following:
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■
A call action such as hang-up: A specific action must be selected from
the drop-down list.
■
A call handler: A specific call handler must be selected from the dropdown list. When sending to a call handler, you must select either
Attempt Transfer or Go Directly to Greeting. Typically, Greeting is
selected.
■
An interview handler: A specific interview handler must be selected
from the drop-down list.
■
A directory handler: a specific directory handler must be selected from
the drop-down list.
■
A conversation such as sign-in or greetings administrator: A specific
conversation must be selected from the drop-down list.
■
Another user’s mailbox: A specific user must be selected from the
drop-down list. When sending to another user, you must select either
Attempt Transfer or Go Directly to Greeting. To have the user’s phone
ring, select Attempt Transfer.
Step 8.
Select the language in which Unity plays its prompt from the Language That
Callers Hear drop-down list. When this field is set to Inherited, it uses the
language used by the parent call handler.
Step 9.
Click Save.
Configuring User Template Message Actions
The Message Actions section determines what Unity Connection does with a message
when it receives it. You would assume that it accepts the message and then delivers it to
the proper mailbox. However, there are times when you might not want this to occur.
Unity Connection can be configured to handle four different types of messages: voicemail, email, fax, and delivery receipts. Different message actions can be applied to each
type. The actions that can be applied are
■
Accept the Message: Accepts and delivers the message to the user’s mailbox
■
Reject the Message: Rejects the message and attempts to send a non-delivery receipt
to the sender
■
Relay the Message: Forwards the message to the address specified in the Relay
Address field
■
Accept and Relay the Message: Accepts and delivers the message to the user’s mailbox and forwards a copy of the message to the address specified in the Relay
Address field
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Figure 8-27 Templates > User Templates > Edit > Message Settings.
Figure 8-27 shows the various settings that are configured from this screen. The following steps illustrate the configuration process of these settings and describe the effect on
each setting:
Step 1.
From the previous steps, select Message Actions from the drop-down Edit
menu. Otherwise, from within UCA, select Templates > User Templates.
Select the template you want to manage from the list that appears. After the
template is active, select Message Actions from the drop-down Edit menu.
Step 2.
In the Message Actions section, select the desired action for each type of
message.
Step 3.
Replaceable tokens are used when the message is configured to be relayed. To
configure this, select the desired token and press the arrow that points to the
relay address. For example, if the domain for the relay address is cisco.com
and the email name is the same as the alias defined in Unity Connection, the
relay address would be %Alias%@cisco.com. The result for a user with the
alias of djones would be djones@cisco.com.
Step 4.
Click Save.
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Configuring User Template Caller Input
When callers reach a greeting, a list of options is made available to them. Often the caller
is given the option to dial another extension or reach an operator by pressing a certain
digit. Adding options to a spoken greeting isn’t enough; these options must also be configured under the Caller Input section of the template.
The following steps illustrate how to configure the Caller Input settings:
Step 1.
From the previous steps, select Caller Input from the drop-down Edit menu.
Otherwise, from within UCA, select Templates > User Templates. Select the
template you want to manage from the list that appears. After the template is
active, select Caller Input from the drop-down Edit menu.
Step 2.
On the first screen that appears, you can select the key you want to configure. But before you do that, configure the Wait for Additional Digits,
Prepend Digits to Dialed Extensions setting. The Wait for Additional Digits
setting is the amount of time Unity Connection waits before deciding that the
caller has finished pressing digits. The default value is 1500 milliseconds,
which is a second and a half. Typically, this value is adequate.
Figure 8-28 Templates > User Templates > Edit > Caller Input.
Step 3.
The Prepend Digits setting determines whether any digits are prepended to
the extension a caller enters. This is useful if your callers are used to entering
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four digits but the actual extension number is seven digits. For example, the
caller can enter 3322, but the full extension that the call needs to be transferred to is 5423322. By entering 542 in the Digits to Prepend field and
selecting the Enable check box, the call would be transferred to 5423322.
Step 4.
Now you can start configuring the Caller Input section and keys. Select the key
you want to configure. A screen similar to that shown in Figure 8-28 appears.
Step 5.
The Key field is static and simply displays the number of the key you are
configuring.
Step 6.
The next field determines whether the key is locked. If the Ignore Additional
Input (Locked) check box is selected, an extension that begins with this digit
cannot be entered. Unity Connection transfers the call to the destination
assigned to that key without waiting to see whether the caller is going to enter
other digits. Only enable this field on digits that are not leading digits for any
extensions.
Step 7.
The Action section determines where the call is sent when the key is pressed.
Choose from the following:
Step 8.
■
A call action such as a hang-up: A specific action must be selected from
the drop-down list.
■
A call handler: A specific call handler must be selected from the dropdown list. When sending to a call handler, you must select either
Attempt Transfer or Go Directly to Greeting. Typically, Greeting is
selected.
■
An interview handler: A specific interview handler must be selected
from the drop-down list.
■
A directory handler: A specific directory handler must be selected from
the drop-down list.
■
A conversation such as sign-in or greetings administrator : A specific
conversation must be selected from the drop-down list.
■
Another user’s mailbox: A specific user must be selected from the
drop-down list. When sending to another user, you must select either
Attempt Transfer or Go Directly to Greeting. To have the user’s phone
ring, select Attempt Transfer.
Click Save. Return to the Caller Input page to select other keys to configure.
Configuring User Template Mailbox Settings
The Mailbox Settings section enables you to configure aging and mailbox size limits. As
you might recall, mailbox size limits can be configured as a system setting. If you choose
to define limits here, the system’s limits will not be available for users created using this
template.
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Figure 8-29 Templates > User Templates > Edit > Mailbox.
Figure 8-29 shows the settings that are configured from this screen. The following
steps illustrate the configuration process for these settings and describe the effect of
each setting:
Step 1.
From the previous steps, select Mailbox from the drop-down Edit menu.
Otherwise, from within UCA, select Templates > User Templates. Select the
template you want to manage from the list that appears. After the template is
active, select Mailbox from the drop-down Edit menu.
Step 2.
User can request read receipts so that they are notified when a message has been
retrieved. To have the receipts sent, select the Respond to Requests for Read
Receipts check box. To prevent these receipts from being sent, deselect this box.
Step 3.
From the Message Aging Policy drop-down list, select the desired policy.
Step 4.
When Use System Settings under the Mailbox Quotas heading is selected,
the mailbox limits are set by the system settings. To set customs limits, select
Custom and set the limits as you desire.
Note When calculating the size for the mailbox quota, it is helpful to understand the
number of minutes of voice that can be stored per megabyte. When using G.711, approximately 2 minutes of recorded voice can be stored per megabyte of space. When using
G.729, approximately 17 minutes of recorded voice can be stored per megabyte of space.
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Step 5.
From the Mailbox Store drop-down list, select the mailbox store the user’s
mailbox should be created on.
Step 6.
Click Save.
Configuring User Template Phone Menu
The User Template Phone Menu settings enable you to configure conversation parameters
such as volume, speed, and whether brief or full menus are played. From this page, you
can also configure menu timeout settings and set up what the user hears when he logs in
and out. Figure 8-30 illustrates most of the settings on this page. The following steps
explain the effect of the settings and describe the configuration process:
Figure 8-30 Templates > User Templates > Edit> Phone Menu.
Step 1.
From the previous steps, select Phone Menu from the drop-down Edit menu.
Otherwise, from within UCA, select Templates > User Templates. Select the
template you want to manage from the list that appears. After the template is
active, select Phone Menu from the drop-down Edit menu.
Step 2.
The first setting determines whether the user hears full or brief menus. It is
recommended that you set the menus to full for users who are unaccustomed
to the system because full menus explain more than brief menus. After users
are familiar with the menus, you might want to change to a brief menu.
Choose the desired menu style from the Menu Style drop-down list.
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Step 3.
Select the volume at which the Unity Connection conversation is played from
the Conversation Volume drop-down list. The choices are Quieter, Medium,
and Louder. Medium should be adequate for most users.
Step 4.
The Conversation Speed setting enables you to speed up or slow down the
speed at which voicemail is played back. Select the desired setting from the
drop-down list.
Step 5.
Select a time format by selecting 12-hour or 24-hour format below the Time
Format heading.
Step 6.
The Touchtone Conversation field determines which touch tones you would
use to save and delete messages. If you are migrating from another voicemail
system, it is a good idea to become familiar with these conversations and
choose the one most similar to previous voicemail system conversations. Use
the classic conversation if you are uncertain which conversation to use.
Step 7.
The Message Locator field allows users to search their voicemails when calling
in to Unity Connection. When this feature is enabled, the users can search by
extension, username, or outside caller phone number. Because this feature can
be processor intensive, take care when determining how many users have access
to it. To enable this feature, select the Enable check box under the Finding
Messages with Message Locator heading. If enabled, select the order in which
the message should be checked from the Message Locator Sort Order list.
Step 8.
Enter the number of times the menu should repeat if the user does not
respond to the menu in the Times to Repeat Menu When User Does Not
Respond field.
Step 9.
Enter the number of milliseconds Unity Connection waits before determining
the user is not going to respond by entering the desire value in the Wait for
First Touchtone or Voice Command field.
Step 10. Enter the number of milliseconds Unity Connection waits before it determines whether the user is finished pressing touch tones when entering names,
extensions, and PINs by entering the desired value in the Wait for Additional
Key Presses When Entering Names, Extensions, and PINs field.
Step 11. Enter the number of milliseconds Unity Connection waits before it determines whether the user has completed pressing touch tones when entering
multiple digit menu options by entering the desired value in the Wait for
Additional Key Presses When Entering Multiple Digit Menu Options field.
Step 12. When users log in to the system, they can be greeted by hearing their recorded names so that they know they are logged in to the correct account. To
enable this greeting, select the User’s Recorded Name check box.
Step 13. An alternate greeting is used when a user is out of the office for a long period
of time, such as vacation. It is important that the user remember to turn this
greeting off when he returns. Configure Unity Connection to inform the user
whether the alternate greeting is enabled when he logs in by selecting the
Alternate Greeting Notification check box.
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Step 14. If the User’s New Messages Automatically check box is selected, users are
taken directly to new messages after login.
Step 15. Under the When Exiting the Conversation heading, you can configure
where users are sent when they end the user conversation. A number of
choices are available. Typically, you set this to send the caller to the opening
greeting call handler or the hang-up call action. However, you can choose
from the following:
■
A call action such as a hang-up: A specific action must be selected from
the drop-down list.
■
A call handler: A specific call handler must be selected from the dropdown list. When sending to a call handler, you must select either
Attempt Transfer or Go Directly to Greeting. Typically, Greeting is
selected.
■
An interview handler: A specific interview handler must be selected
from the drop-down list.
■
A directory handler: A specific directory handler must be selected from
the drop-down list.
■
A conversation such as sign-in or greetings administrator: A specific
conversation must be selected from the drop-down list.
■
Another user’s mailbox: A specific user must be selected from the
drop-down list. When sending to another user, you must select either
Attempt Transfer or Go Directly to Greeting. To have the user’s phone
ring, select Attempt Transfer.
Step 16. Click Save.
Configuring User Template Playback Message Settings
The playback message settings determine the order in which messages will be played and
the information that is included during playback. The setting in this area has a large
impact on creating a positive user experience. It is important that you take time to consider the impact these settings will have on the users before you configure them. If you
grant access to Unity Connection Messaging Assistant, users can change most of these
fields from that interface.
Figures 8-31 and 8-32 show the settings that are configured in this area. The following
steps illustrate the configuration process of these settings:
Step 1.
From the previous steps, select Playback Message Settings from the dropdown Edit menu. Otherwise, from within UCA, select Templates > User
Templates. Select the template you want to manage from the list that appears.
After the template is active, select Playback Message Settings from the dropdown Edit menu.
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Figure 8-31 Templates > User Templates > Edit > Playback Message Settings.
Figure 8-32 Templates > User Templates > Edit > Playback Message Settings
(Continued)
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Step 2.
From the Message Volume drop-down list, select the volume that messages
should be played back at.
Step 3.
From the Message Speed drop-down list, select the speed that messages
should be played back at.
Step 4.
To configure Unity Connection to inform the user of the total number of new
messages and of the message count, select the check box next to each type
of message count you want the user to hear. The choices are
■
Total of All Message Count
■
Voice Message Count
■
Email Message Count
■
Fax Message Count
■
Receipt Message Count
It is best to offer only the message counts for the messages the user is likely
to retrieve using the phone. For example, if the user retrieves only voicemail
using the phone, only Voice Message Counts is chosen.
Step 5.
To announce the number of saved messages to the user, select the Saved
Message Count check box.
Step 6.
To announce the number of draft messages to the user, select the Draft
Message Count check box.
Step 7.
To allow users to determine the types of messages they want to hear, select
the Message Type Menu check box. When this option is enabled, the user
hears the following message: “Press 1 to hear voice messages, press 2 to hear
emails, press 3 to hear faxes, press 4 to hear receipts.”
Step 8.
Users can configure the order in which new messages are played. Under the
New Message Play Order heading is a list of the various types of messages.
The messages are played as they appear from top to bottom. To change the
order, highlight the type of message you want to move, and then click the Up
or Down Arrows below the box. Repeat this process with all the message
types until they are in the desired order.
Step 9.
When playing new messages, the default behavior of Unity Connection is to
play the oldest message first. To play the newest message first, choose
Newest First from the Then By drop-down list. In most cases, it is best to
leave this setting at Oldest First.
Step 10. Users can configure the order in which saved messages are played. Under the
Saved Message Play Order heading is a list of the various types of messages.
The messages are played as they appear from top to bottom. To change the
order, highlight the type of message you want to move, and then click the up
or down arrows below the box. Do this with all the message types until they
are in the desired order.
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Step 11. When playing saved messages, the default behavior of Unity Connection is to
play the newest message first. To play the oldest message first, choose Oldest
First from the Then By drop-down list. In most cases, it is best to leave this
setting at Newest First.
Step 12. When playing deleted messages, the default behavior of Unity Connection is
to play the oldest message first. To play the newest message first, choose
Newest First from the drop-down list below the Deleted Message Play
Order heading.
Step 13. Under the Before Playing Each Message, Play heading, you can choose to
play the sender’s information, message number, time the message was sent,
Automatic Number Identification (ANI), and message duration. To configure
what is played, select the check box next to each piece of information that
you want to have played.
Step 14. When a subscriber is listening to messages, he can fast-forward and rewind
the message. The amount of time that is applied to these functions is defined
by the value entered in the Fast-Forward Messages By and Rewind
Messages By fields. The default is 5000 milliseconds (5 seconds) and is adequate in most cases. To change these values, simply enter the new value in the
appropriate field.
Step 15. If the Create a Message Bookmark check box is selected, Unity Connection
will set a bookmark at the point of the message the user was at if the caller
gets disconnected. If the caller dials back in shortly after the disconnection,
she can pick up where she left off.
Step 16. By default, when a call is disconnected in the middle of a user listening to a
message, the message is left marked as new. To change this so that the message is marked as saved, select the Saved radio button near the Mark a New
Message heading.
Step 17.
To play the time a message was sent after a message is played, select the Time
the Message Was Sent check box.
Step 18. To require that the user confirm the deletion of messages, select the Confirm
Deletions of New and Saved Messages check box. This is useful if you have
users that often delete messages by mistake.
Step 19. Click Save.
Configuring User Template Send Message Settings
The send message settings define how the user interacts with Unity Connection when
sending messages. It also determines whether the user can send broadcast messages. This
is another area that can impact whether the user has a positive experience and therefore a
positive attitude toward Unity Connection.
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Figure 8-33 shows the settings that are configured in this area. The following steps illustrate how to configure these settings:
Figure 8-33 Templates > User Templates > Edit > Send Message Settings.
Step 1.
From the previous steps, select Send Message Settings from the drop-down
Edit menu. Otherwise, from within UCA, select Templates > User
Templates. Select the template you want to manage from the list that appears.
After the template is active, select Send Message Settings from the dropdown Edit menu.
Step 2.
To allow a user to send broadcast messages, select the User Can Send
Broadcast Messages to Users on This Server check box.
Step 3.
To allow a user to update broadcast messages, select the User Can Update
Broadcast Messages Stored on This Server check box.
Step 4.
The Enter a Recipient By drop-down list determines the user’s search options
for sending messages. You can select to search by first then last name, last
then first name, or extension. In most cases, you should select Entering the
Extension from this list.
Step 5.
To have Unity Connection play the name of the addressee, select the Confirm
Recipient by Name check box. Typically, this is left deselected.
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Step 6.
To allow a user to enter multiple addresses, select the Continue Adding
Names After Each Recipient check box. If this box is not selected, the user
must press 1 after each address to enter another one.
Step 7.
If the Automatically Add Recipients to Addressing Priority List check box
is selected, Unity Connection will acknowledge the addresses that the user
chose. This allows Unity Connection to initially offer those addresses as a
possible match.
Step 8.
To permit a user to save a message as a draft, select the Allow Users to Save
Draft Messages check box.
Step 9.
If the Retain Urgency Flag When Forwarding or Replying to Messages
check box is selected, the urgency marking will stay with the message when it
is replied to or forwarded.
Step 10. The When a Call Is Disconnected or the User Hangs Up selections determine whether a message is sent, discarded, or saved as a draft if the call is disconnected before the user sends the message. Select the desired action.
Step 11. Click Save.
Configuring User Template Greetings
The greetings setting of the template allows you to define what greetings are enabled and
how they are configured. A user can play up to five greetings, each played under different circumstances. Table 8-7 lists these greetings and the function of each.
Table 8-7
Greetings
Greeting
Function
Standard
Plays during open hours unless overridden by another greeting. If the
Closed greeting is not enabled, Standard plays after hours.
Closed
Plays during closed hours as defined in the schedule associated with the
user.
Busy
Plays when the call is transferred to Unity because the user’s extension was
busy.
Internal
Plays when the caller is another user on the system.
Alternate
When enabled, overrides all other greetings. This greeting plays all hours, all
days until disabled.
Error
Plays when a caller presses an invalid key.
Holiday
Plays on dates defined as holidays.
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Figure 8-34 Templates > User Templates > Edit > Greetings.
Figure 8-34 shows the User Templates Greeting configuration screen. The following steps
discuss the effects of these settings and describe how to configure them:
Step 1.
From the previous steps, select Greetings from the drop-down Edit menu.
Otherwise, from within UCA, select Templates > User Templates. Select the
template you want to manage from the list that appears. After the template is
active, select Greetings from the drop-down Edit menu.
Step 2.
A list of greetings appears. Select the greeting you want to configure.
Step 3.
The Status section determines whether the greeting is enabled. The Standard
greeting is always enabled and cannot be disabled. If you are configuring one
of the other greetings, you must enable it for it to take effect. You can select
Enable with No End Date and Time to have it always enabled. If you want to
have it enabled until a specific time and date, select Enable Until and enter
the desired date and time.
Step 4.
Under the Callers Hear heading, the source of the message is chosen. The
three choices for the source are as follows:
■
System Default Greeting: Plays the prerecorded greeting “Sorry, <user’s
name> is not available.” If the user has not yet recorded a name, the
greeting “Sorry, the user at extension <user’s extension number> is not
available” is played.
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■
My Personal Recording: Plays the greeting that the user recorded.
■
Nothing: Does not play a greeting and defaults to the after-greeting
action.
Typically, My Personal Recording is chosen for this setting to allow the user
to create his own greetings. If this is chosen, you can disable the “Record
Your Message at the Tone” prompt by deselecting the Play the “Record Your
Message at the Tone” Prompt check box. Select the desired source by clicking the appropriate radio button.
Step 5.
To prevent the caller from transferring to another extension during the greeting or attempting to skip the greeting, select the Ignore Caller Input check
box. Typically this is not selected, but it can be useful in some situations.
Step 6.
By default, Unity Connection will only transfer calls to numbers that are
assigned to users or call handlers. To allow Unity Connection to transfer calls
to other numbers, select the Allow Transfers to Numbers Not Associated
with Users or Call Handlers check box.
Step 7.
If callers remain silent after being offered the opportunity to leave a message,
they can be reprompted. To have a caller reprompted, enter the number of
times she should be reprompted in the Times to Re-prompt Caller field.
Step 8.
Enter the number of seconds Unity Connection should wait between playing
the prompt again in the Delay Between Re-prompts field.
Step 9.
After the greeting is played, the caller has the opportunity to leave a message.
To enable, click the Call Action radio button and select Take Massage from
the drop-down list. Table 8-8 provides additional options.
Step 10. If more than one language is installed, greetings can be recorded in multiple
languages. To record a greeting, select the language from the Select Language
drop-down list and click the Play/Record button. A recording panel opens.
Click the red button on the panel. Enter your extension when prompted.
When the phone rings, answer it and record the greeting. Typically, you
would not record a greeting when creating a template.
Step 11. Click Save.
Configuring User Template Post-Greeting Recording
Unity Connection enables you to configure a greeting to be played after the user’s greeting. You can also select whether all callers hear the recording or just unidentified callers.
To configure the post-greeting settings, follow these steps:
Step 1.
From the previous steps, select Post Greeting Recording from the drop-down
Edit menu. Otherwise, from within UCA, select Templates > User
Templates. Select the template you want to manage from the list that appears.
After the template is active, choose Post Greeting Recording from the dropdown Edit menu.
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Table 8-8
Call Action
Action
Description
Call action such as hang-up
A specific action must be selected from the
drop-down list.
Call handler
A specific call handler must be selected from
the drop-down list. When sending to a call
handler, you must select either Attempt
Transfer or Go Directly to Greeting.
Typically, Greeting is selected.
Interview handler
A specific interview handler must be selected
from the drop-down list.
Directory handler
A specific directory handler must be selected
from the drop-down list.
Conversation such as sign-in or greetings
administrator
A specific conversation must be selected from
the drop-down list.
Another user’s mailbox
A specific user must be selected from the
drop-down list. When sending to another user,
you must select either Attempt Transfer or Go
Directly to Greeting. If you prefer to have the
user’s phone ring, select Attempt Transfer.
Step 2.
If you do not want Unity Connection to play a post-greeting recording, simply select the Do Not Play Recording radio button and click Save. If you do
want a post-greeting recording to play, proceed to Step 3.
Step 3.
To have the post-greeting recording played for all callers, select the Play
Recording to All Callers radio button. If you want to play the recording
specifically for unidentified callers, click the Play Recording Only to
Unidentified Callers radio button.
Step 4.
Select the desired greeting from the Post Greeting Recording Selection
drop-down list. Post-greeting recordings must be created before they can be
selected here.
Step 5.
Click Save.
Configuring User Template Message Notification Settings
Message notification allows a user to be advised through a pager, phone, or other external device when a new message arrives. This type of notification is useful for users who
are seldom in the office, such as outside salespeople or service technicians.
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It might seem odd that you can set up message notification in the user template. After
all, you will not want to send a message to the same pager when any user who was created using this template receives a new message. However, the user template is an excellent
place to create the same type of flow and generic notification settings for a large number
of users. After the users are added, you can have them fill in the details such as the specific phone numbers using CPCA.
There are five devices that can be configured for message notification. This includes three
phones, one pager, and an STMP (email) address. Message notification can be configured
to notify the user of any new voicemails or faxes. The settings on this screen determine
what type of message triggers a notification and what devices are notified.
The following steps illustrate how these settings are configured and describe the effect
they produce:
Step 1.
From the previous steps, select Notification Devices from the drop-down
Edit menu. Otherwise, from within UCA, select Templates > User
Templates. Select the template you want to manage from the list that appears.
After the template is active, select Notification Devices from the drop-down
Edit menu.
Step 2.
A page appears that lists the five devices that can be configured. Select the
device you want to configure.
Step 3.
Figure 8-35 illustrates the required settings that must be configured. The
Enabled check box must be selected for this device to receive notifications.
Step 4.
The Display Name field is used to identify the device. In most cases, the
existing display names are acceptable, but you can change them if you desire.
Step 5.
The notification can be configured to be sent immediately or to wait a specified number of minutes. If cascading notification is being deployed, some
devices are configured to wait for a certain number of minutes before being
notified. The term cascading notification means that notification is sent, and
if the message has not been retrieved after a specified number of minutes, a
different device is notified. If, after an additional amount of time, the message
is still not retrieved, another device is notified. An example of cascading flow
might look something like this:
Step 6.
1.
First notification is sent to your pager.
2.
You don’t retrieve the message.
3.
Ten minutes later, a notification is sent to your cell phone.
4.
You don’t retrieve the message.
5.
Ten minutes later, a notification is sent to your boss’s phone.
6.
You’re busted!
Enter the number of minutes that Unity Connection should wait before notifying this device in the Delay Before First Notification Attempt field. Unity
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Connection only sends the notification if the message has not been retrieved
after appropriated time.
Figure 8-35 Templates > User Templates > Edit > Notification Devices (Phones
and Pager).
Step 7.
To invoke what some refer to as the “nag factor,” you can have the device notified at specified intervals until the message is retrieved. Select the Repeat
Notification if There Are Still New Messages check box, and then enter the
number of minutes in the Notification Repeat Interval field.
Step 8.
The On Notification Fail setting enables you to specify another device to be
notified if the notification fails. To do this, select the Send To radio button
and select the device from the drop-down list. If you do not want to have
another device notified, select the Do Nothing radio button.
Step 9.
Under the Notify Me Of heading, select the types of messages of which the
user should be notified. As shown in Figure 8-35, you can select all messages
or only one type, such as voicemail. You can further narrow it down by
selecting only urgent messages.
Step 10. The rest of the fields vary depending on whether you choose a phone or email
as the notification device. Figure 8-36 shows the fields that are available if
you choose SMTP as the delivery mechanism. You have the option to send
transcriptions. You can choose to send them for all voicemails or only those
marked urgent. Because you are configuring a template, you would not enter
information in the To, From, Message Header, Message Text, or Message
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Footer fields. In most cases, the values for these fields vary based on the specific user. To have information like caller name, caller ID, urgency marking,
and time message left in the email, select the Include Message Information in
Message Text check box. To include the number of new messages the user
has, select the Include Message Count in Message Text check box. To
include a link to CPCA in the email, select the Include a Link to Cisco PCA
in Message Text check box. If you are configuring an SMTP device, you can
now click Save.
Figure 8-36 Templates> User Templates > Edit > Notification Devices (Email).
Step 11. If you choose a phone or a pager, you have a number of options to choose
from. Table 8-9 lists the available fields.
Step 12. Click Save.
At this point, you have completed creating a user template. Be sure to analyze the number of templates required based on the information discussed at the beginning of this
chapter. Add the templates to Unity Connection using the preceding steps as a guide.
Now that the user templates are created, you can start to add users. The next section
explains the process of adding users to the template. A lot of configuration is required to
accomplish this task. However, the success of your deployment rests heavily upon a properly prepared foundation. Okay, enough of this foundation stuff; let’s start adding users!
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Table 8-9
Phone or Pager Notification Fields
Field
Description
Phone Number
The number to dial when sending notifications. This field is normally left blank when configuring templates.
Extra Digits
Digits that are sent after the call is connected. This can be used
when sending a notification to a pager. This field is normally left
blank when configuring templates.
Duration to Wait Before
Dialing Extra Digits
The number of seconds to wait before sending the extra digits.
Ring to Wait
The number of rings to wait before determining whether the call
is going to be answered. This should be set lower than the number
of rings that an answering machine or service (voicemail) waits
before answering a call. This prohibits the notification from being
answered by the service.
Busy Retry Limit
The number of times that the notification will be attempted when
the device is busy.
Busy Retry Interval
The length of time between the notification retry when the device
is busy.
RNA Retry Limit
The number of times that the notification will be tried when the
device does not answer.
RNA Retry Interval
The length of time between the notification retries when the
device does not answer.
Phone System
The phone system that is used for this notification device.
Prompt for User ID on
Notification
If selected, the user will have to enter his ID (extension number)
and PIN to retrieve the messages. Even if this is not selected, the
user will only be required to enter his PIN.
Creating New Unity Connection Users
Now that you have the user templates created, the process of creating a user is fairly simple. To create a user, enter a few pieces of information and click Save, and the process is
complete. Because user creation is based on one of the templates you created, all the
additional settings will be populated. On occasion you might opt to change the settings
of a user. The process for implementing the most common changes comes later in this
chapter in the “Managing Users” section.
This section focuses on the creation of users with mailboxes. The next section discusses
how to import existing users.
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The following steps show how to add a new Exchange-based user to Unity:
Step 1.
From within UCA, select Users > Users.
Step 2.
Click the Add New button.
Step 3.
A window such as that shown in Figure 8-37 displays. Select User with
Mailbox from the User Type drop-down list.
Figure 8-37 Add User Window
Step 4.
Select the template this user should be based on from the Based on Template
drop-down list.
Step 5.
Enter an alias for the user in the Alias field. The alias is the ID a user will use
to log in to the system from a web page. The alias must be unique.
Step 6.
Enter the first and last name of the new user in the fields labeled as such.
Step 7.
The display name is used by the voice recognition server when callers speak
the name of the person they are trying to reach. It is also used to play the
name of the user if the user has not recorded a name. The display name is
automatically generated based on the settings you define in the template. You
can change it if you desire by entering the specified name in the Displayed
Name field.
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Step 8.
The SMTP Address field is optional. This is used to identify a user that is
using an SMTP client. If you do not enter anything, the display name is used
to create the address.
Step 9.
If more than one mail store exists in your environment, select the mail store
from the Mailbox Store drop-down list to be used for this user.
Step 10. An extension number must be assigned to every user, and this should be the
extension number that is assigned to the user’s phone. Enter the user’s extension number in the Extension field. In rare cases, the extension number in
Unity Connection is different than the actual extension assigned to the user’s
phone, but this should be avoided. All extensions in Unity Connection must
be unique—that is, each extension can only be assigned to one user.
Step 11. A cross-server transfer occurs when a call is transferred to a user on a different Unity Connection server. If the cross-server transfer fails, the call is sent
to the extension number entered in the Cross-Server Transfer Extension
field. Enter the desire extension in this field.
Step 12. In the Outgoing Fax Number field, enter the number that provides a user
with access to send faxes for printing purposes.
Step 13. Click the Save button.
Step 14. After the user is added to the database, you will be returned to the Edit User
Basics screen.
So, that is all there is to it. The user you added is now ready for use. As you can see,
adding a user is a much simpler process than creating user templates. Because you use a
template during the creation process, the majority of the user settings are automatically
configured, which shows the value of the user template. However, there are five types of
user configurations that are not configured using the user template:
■
Private lists
■
Alternate extensions
■
Alternate names
■
External service account
■
SMTP proxy addresses
These settings are not part of the user template because they are unique for each user.
Private lists are lists of multiple users with whom a user can communicate when sending
voicemail. Instead of having to add each user as a recipient, the user can select a private
list as the recipient, and the voicemail will be sent to everyone in the list. Users manage
their own private lists.
Alternate extensions are used for multiple purposes. When an alternate extension is
added to a user, Unity Connection regards all calls to or from that extension as if they are
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to or from the user’s primary extension. Consider it as a way to associate multiple extensions to one mailbox. Imagine that a user has multiple extensions on the system. When
someone calls the user and the call is not answered, the caller is forwarded to the same
mailbox regardless of which of the user’s extensions the caller dialed. This feature is also
used to allow users easy message access when calling from an outside phone. When calling from the outside, a user must press the asterisk (*) key and enter an extension number
and password. By adding the phone number of phones that are commonly used to access
voicemail, such as a cell phone, this process is simplified. Because Unity recognizes the
incoming caller ID as an alternate extension, the user is simply being prompted for a password just as if calling from an office phone.
Alternate names allow you to enter additional names that a subscriber can be referred to,
such as nicknames, shortened names (Bob instead of Robert), or a maiden name. These
names are included in a search when voice recognition is used.
External service accounts are used to allow integration with email, calendar, and conference services. The external service accounts must be configured by the system engineer
before they can be assigned to users.
SMTP proxy addresses are required when a user is going to use an IMAP client to send
and receive messages. The SMTP proxy address should be the same as the user’s corporate email address.
Configuring private lists, alternate extensions, alternate names, external service
accounts and SMTP proxy addresses are covered in the “Managing Users” section, later
in this chapter.
Importing Unity Connection Users
When integrating Unity Connection with CM, you can import users from CM. You can
also channel users from a Lightweight Directory Access Protocol (LDAP) server. Some
configuration is required to properly integrate to a Communications Manager or LDAP
server. While the integration steps are beyond the scope of this book, the steps for the
actual importing process are covered.
The following steps show how to import a CM user. The process of importing an LDAP
user are similar:
Step 1.
From within UCA, select Users > Import Users.
Step 2.
Select the Communications Manager server from the Find End Users In dropdown list.
Step 3.
From the drop-down list located next to the Where heading, select the field
you want to search. You can search by extension, first name, last name, or
alias. If alias is selected, the search is based on the user ID in CM.
Step 4.
In the next drop-down list, select how you want the search to match the criteria you enter. The choices are Begins With, Contains, and Is Exactly.
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Step 5.
Enter the search string in the list field and click Find.
Step 6.
Select the check box next to each user you want to import.
Step 7.
Click the Import Selected button. Optionally, you can click the Import All
button if you want to select all the users that appear on the page.
Step 8.
After the users are imported, a summary of the import process will appear
below the Status heading at the top of the page.
As mentioned previously, users can be added using the Bulk Administration Tool. This
utility can import users from a CSV file.
The following steps are required when importing users from a CSV file:
Step 1.
Create a CSV file. There are more than 80 values that can be entered in the
CSV file for each user. Of these fields, only three are required: Alias,
Template Alias, and Extension. Many of the remaining fields are populated
with the settings defined in the user template you choose during the import.
Step 2.
From within UCA navigate to Tools > Bulk Administration Tool. A screen
similar to that shown in Figure 8-38 should appear.
Figure 8-38 Bulk Administration Tool
Step 3.
Select the Create radio button under the Select Operation heading.
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Step 4.
Select the User with Mailbox radio button under the Select Object Type
heading.
Step 5.
When the CSV file contains information already defined in the template,
Unity Connection must be aware of the information to be used. This process
is accomplished by selecting No next to the User Template heading. If you
choose to have the information in the template used, select Yes.
Step 6.
Click the Browse button. Navigate to the CSV file and click Open.
Step 7.
In the Failed Objects Filename field, enter the name that will be implemented for the error log in the event of problems that can occur with the creation
of objects.
Note If errors occur, you can view the errors by clicking the Download the Failed
Objects File link. The link appears just below the Status heading. If there are no errors, the
link does not appear on the page.
Step 8.
Click the Submit button.
Step 9.
After the import is complete, a summary of the results appears at the top of
the page below the Status heading.
Now that the users are added, your system is ready for basic voicemail use. Users can
now configure their personal settings, such as passwords and greetings, by dialing the
voicemail pilot number and going through the process of user self-enrollment.
Unity Connection Contacts
Before you move on to managing users, take a look at another type of Unity Connection
object called a contact. Contacts are typically used for people who do not have a voicemail account on your system, but you would like users on the system to be able to send
them voicemail and look them up on the directory.
Contacts are created in much the same way that users are, but the configuration process
is much easier because they do not have a mailbox in the system. As with users, a contact
template must be specified when creating a contact.
Creating Unity Connection Contact Templates
The creation of a contact template is fairly simple. To create the template, follow these steps:
Step 1.
From within UCA, select Templates > Contact Templates.
Step 2.
Click the Add New button.
Step 3.
Enter a unique name for the template in the Alias field.
Step 4.
Enter a descriptive name in the Description field.
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Step 5.
If this user is a VPIM contact, select a location from the Delivery Location
drop-down list.
Step 6.
Click Save.
Step 7.
A new screen appears. From the Partition drop-down list, select the partition
you want to assign to contacts that are created using this template.
Step 8.
If desired, you can also enter the contact’s city and department information in
the appropriate fields.
Step 9.
Click Save.
Creating Unity Connection Contacts
The following steps illustrate how to create a Unity Connection contact:
Step 1.
From within UCA, select Contacts > Contacts.
Step 2.
Click the Add New button.
Step 3.
A window like that shown in Figure 8-39 displays.
Figure 8-39 New Contact
Step 4.
Enter an alias for the user in the Alias field. The alias must be unique.
Step 5.
Enter the first and last name of the new user in the appropriate fields.
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Step 6.
The display name is used by the voice recognition server when a caller speaks
the name of the person he is trying to reach. It is also used to play the name
of the contact if there is no recorded name. The display name is automatically
generated based on the settings you define in the template. You can change it
if you desire by entering the desired name in the Displayed Name field.
Step 7.
Select the template this contact should be based on from the Contact
Template drop-down list.
Step 8.
Click Save.
Step 9.
A new screen similar to Figure 8-40 appears. The information you entered in
the previous screen is used to populate the first four fields. To record the contact’s name, click the Play/Record button. A recording panel opens. Click the
red button on the panel. Enter your extension when prompted. When the
phone rings, answer it and record the greeting.
Figure 8-40 Configuring a New Contact
Step 10. To have the contact listed in the directory, you must select the List in
Directory check box.
Step 11. The partition that you assigned to the template is assigned to the contact. If
you want to change it, select the desired partition from the Partition dropdown list.
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Step 12. To allow a call to be transferred to the contact, select the Transfer Enable
check box and enter the number that the call should be transferred to in the
Transfer Extension field.
Step 13. The City and Department fields are populated by the information in the contact template. You can change these if you want.
Step 14. When configuring a VPIM contact, select the VPIM delivery location from
the Delivery Location drop-down list.
Step 15. When configuring a VPIM contact, enter the remote mailbox number in the
VPIM Remote Mailbox Number field.
Step 16. When configuring a VPIM contact, a local extension can be assigned. Enter
the desired extension the Local Extension field.
Step 17.
Users on the system can reach contacts by using the voice recognition “Call”
command. Enter the contact’s work, home, and cell phone number in the
Dialed Work Phone, Dialed Home Phone, and Dialed Mobile Phone fields.
Step 18. Users can set up personal call-routing rules in Unity Connection. These rules
can be based on incoming phone numbers. To allow Unity to recognize the
contact so that rules can be set up against their numbers, you must enter their
work, home, mobile, and up to two other numbers in the Phone Numbers to
Identify Contact for Personal Call Transfer Rules section.
Step 19. Click the Save button.
Alternate names can be assigned to contacts that might be known by a name other than
their actual name, such as a maiden name. This allows them to be found in the directory if
someone searches using a variation of their name. To configure alternate names for a
Unity Connection contact, use the following steps:
Step 1.
From the previous steps, select Alternate Names from the drop-down Edit
menu. Otherwise, from within UCA, select Contacts > Contacts. Select the
contact you want to manage from the list that appears. After the contact is
active, select Alternate Names from the drop-down Edit menu.
Step 2.
Enter the first and last name in the appropriate fields and click Add New.
Step 3.
Click Save.
STMP proxy addresses are used to map a contact’s email address to his Unity Connection
account. To configure an SMTP proxy address for a Unity Connection contact, simply
add his email address to the STMP proxy field. The following steps illustrate the process:
Step 1.
From the previous steps, select SMTP Proxy Addresses from the drop-down
Edit menu. Otherwise, from within UCA, select Contacts > Contacts. Select
the contact you want to manage from the list that appears. After the contact is
active, select SMTP Proxy Addresses from the drop-down Edit menu.
Step 2.
Click Add New.
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Step 3.
Enter the user’s email address in the blank field.
Step 4.
Click Save.
Managing Users
After users are created, there might be some basic day-to-day administration required.
For the most part, as long as things don’t change, the user administration tasks should be
minimal. Then again, things don’t tend to go unchanged for long. The following sections
are a collection of common user administrative tasks. No task discussed in these sections
is difficult and, if performed often, can be done without reference to the sections. The
goal is to provide a quick reference for you when you are required to perform an unfamiliar administrative task. Use this reference the first few times you perform any of these
tasks. Although they are not difficult, some tasks can cause hours of troubleshooting if
incorrectly configured. The tasks are divided into four categories:
■
User access
■
Call transfer and greetings
■
Message access and notification
■
Conversation management settings
Note The steps shown for each task are designed so that they can stand on their own.
That is, each individual set of steps is not dependent upon other sets of steps. This is
designed intentionally so that the reader does not have to jump back and forth between
multiple sections to complete one task. This format does, however, require the repetition of
the same steps in multiple tasks. The text might seem repetitive, but to offer a task-independent reference, this format is required.
Managing User Access
A subscriber can access Unity in a number of ways. The following sections cover administrative tasks that are associated with subscriber access, such as passwords, CoS, and
licensed features. Because the exact steps differ between Unity and Unity Connection,
the steps for each are offered.
Unlocking an Account
Users often lock themselves out by entering the incorrect password too many times.
These thresholds are configured on the Subscriber > Account Policy > Phone Password
Restrictions page in Unity and on the System Settings > Authentication Rules page on
Unity Connection.
There are two types of accounts that can become locked. The first is the voice mailbox.
When this is locked, the user cannot access her messages from the phone. She still has
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access to messages through CPCA and viewmail, if she has these features. The second
type of account, web access, when locked, prohibits all access. This also prevents any
user from accessing the administrate interfaces.
To unlock either of these in Unity, follow these steps:
Step 1.
From within SA, select Subscribers > Subscribers. Click the Find icon and
search for the user you want to manage.
Step 2.
When the name of the desired subscriber appears in the title strip, select
Account from the menu that appears on the left side of the screen.
Step 3.
To allow the subscriber access to messages from a phone, deselect the Cisco
Unity TUI Account Status check box.
Step 4.
To allow the subscriber access to CPCA and viewmail, deselect the Cisco
Unity GUI Access Status check box. Remember, only subscribers who have a
CoS that allows them access to these interfaces are able to access them
regardless of what this setting is.
Step 5.
Click the Save icon in the title strip.
To unlock either of these in Unity Connection, follow these steps:
Step 1.
From within UCA, select Users > Users. Enter the appropriate search criteria
and click the Find button to search for the user you want to manage.
Step 2.
Click the name of the user you want to manage.
Step 3.
Navigate to Edit > Password Settings.
Step 4.
Select the type of account you want to unlock from the Choose Password
drop-down list.
Step 5.
If the account is locked because of invalid sign-in, the time that the account
was locked is shown in the Time Locked Due to Failed Sign-In Attempts field.
If the account is locked because of an administrative locking, the time that the
account was locked is shown in the Time Locked by Administrator field.
Step 6.
Click the Unlock Password button to unlock the account.
Step 7.
Click Save.
Resetting Passwords
Another common issue is when the user forgets his/her password. This phenomenon has
baffled system administrators for years. However, with the average person having to keep
track of numerous passwords, it is a common occurrence.
To change a subscriber’s password in Unity, follow these steps:
Step 1.
From within SA, select Subscribers > Subscribers. Click the Find icon and
search for the user you want to manage.
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Step 2.
When the name of the desired subscriber appears in the title strip, select
Phone Password from the menu that appears on the left side of the screen.
Step 3.
In the Password field, enter the new password.
Step 4.
In the Confirm Password field, enter the new password again.
Step 5.
Click the Save icon.
To change a subscriber’s password in Unity Connection, follow these steps:
Step 1.
From within UCA, select Users > Users. Enter the appropriate search criteria
and click the Find button to search for the user you want to manage.
Step 2.
Click the name of the user you want to manage.
Step 3.
Navigate to Edit > Change Password.
Step 4.
Select Voice Mail or Web Application from the Choose Password dropdown list.
Step 5.
In the Password field, enter the new password.
Step 6.
In the Confirm password field, enter the new password again.
Step 7.
Click Save.
Changing a Subscriber’s Extension
A user’s extension might need to be changed if they move to a different office or
department.
To change a subscriber’s extension in Unity, follow these steps:
Step 1.
From within SA, select Subscribers > Subscribers. Click the Find icon and
search for the user you want to manage.
Step 2.
When the name of the desired subscriber appears in the title strip, select
Profile from the menu that appears on the left side of the screen.
Step 3.
In the Extension field, enter the new extension number.
Step 4.
Click the Save icon.
To change a subscriber’s extension in Unity Connection, follow these steps:
Step 1.
From within UCA, select Users > Users. Enter the appropriate search criteria
and click the Find button to search for the user you want to manage.
Step 2.
Click the name of the user you want to manage.
Step 3.
Navigate to Edit > User Basics.
Step 4.
In the Extension field, enter the new extension number.
Step 5.
Click Save.
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Changing a Subscriber’s CoS
Many of the features that you want to allow a user access to are configured in the CoS. If
you want to give the feature to all users in the CoS, you can simply edit their current
CoS. However, if you want to grant certain features to only a subset of users that share
the same CoS, you need to create a new CoS based on the current CoS, edit the new
CoS, and assign it to those users. This section includes the steps required to add various
features to CoS.
The following steps show how to change a Unity subscriber’s CoS:
Step 1.
From within SA, select Subscribers > Subscribers. Click the Find icon and
search for the user you want to manage.
Step 2.
When the name of the desired subscriber appears in the title strip, select
Profile from the menu that appears on the left side of the screen.
Step 3.
In the Class of Service field, select the desired class of service from the dropdown list.
Step 4.
Click the Save icon.
The following steps show how to change a Unity Connection user’s CoS:
Step 1.
From within UCA, select Users > Users. Enter the appropriate search criteria
and click the Find button to search for the user you want to manage.
Step 2.
Click the name of the user you want to manage.
Step 3.
Navigate to Edit > User Basics.
Step 4.
In the Class of Service field, select the desired class of service from the dropdown list.
Step 5.
Click Save.
Granting Access to Licensed Features (FaxMail, Text-to-Speech, CPCA)
To allow subscribers to have access to various licensed features, you must grant them
rights. These rights are authorized based on the subscriber’s CoS. If you want to permit
all the subscribers of a particular CoS these rights, you can simply edit their current CoS.
However, if you do not want to authorize these rights to all subscribers in a CoS, you
need to create a new CoS, grant the rights to that CoS, and assign the new CoS to the
selected subscribers.
To add access to the features of an exiting CoS in Unity, follow these steps:
Step 1.
From within SA, select Subscribers > Class of Service.
Step 2.
Click the Find icon, and a list of CoSs displays.
Step 3.
Highlight the CoS you want to edit and click View.
Step 4.
Select Features from the left side of the screen.
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Step 5.
To allow subscribers to manage faxes using the phone, select the FaxMail
check box. This feature requires that your organization has a supported thirdparty fax solution.
Step 6.
To allow subscribers to listen to their email over the phone, select the Text-toSpeech for E-mail Messages check box.
Step 7.
To allow subscribers to configure their personal setting through a web browser, select the Cisco Unity Assistant check box.
Step 8.
To allow subscribers to check their voicemail through a web browser, select
the Cisco Unity Inbox (Visual Messaging Interface) check box.
Note All of these features require a license for each subscriber who is assigned to the
feature.
To add access to the features of an existing CoS in Unity Connection, follow these steps:
Step 1.
From within UCA, select Class of Service > Class of Service.
Step 2.
Click the name of the CoS you want to edit.
Step 3.
To allow users to access voicemails for an email client, select the Allow Users
to Access Voice Mail Using an IMAP Client check box. If this feature is
enabled, you must also select which part of the message they can receive
through email. To let them receive all messages, select the Allow Users to
Access Message Bodies radio button. If you do not want them to be able to
receive messages marked private through email, select the Allow Users to
Access Message Bodies Except on Private Messages radio button. To permit them to only see the header information such as time stamp and sender,
select the Allow Users to Access Message Headers Only radio button.
Step 4.
To allow users to use the Unity Connection Messaging Inbox (CPCA), select
the Allow Users to Use the Messaging Inbox and RSS Feeds check box.
Step 5.
To provide users with the ability to access email or use voice recognition, you
must first select the Allow Access to Advanced Features check box. To allow
access to email, select the Allow Access to Email in Third-Party Message
Stores check box. To allow a user to use voice recognition, select the Allow
Users to Use Voice Recognition check box.
Step 6.
If a transcription service is configured in your environment, you must grant
users access to that service for them to be able to use it. To enable this feature, select the Provide Transcriptions of Voice Messages check box. By
default, secure messages will not be transcribed. To enable transcription of
secure messages, select the Allow Transcriptions of Secure Messages radio
button. While this will enable transcription of secure messages, they will not
be sent to notification devices unless you select the Allow Transcriptions of
Secure Messages to Be Sent to Notification Devices radio button.
Step 7.
Click Save.
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Granting Additional System Access Rights
Unity enables you to assign subscribers limited administrative rights that allow them to
deal with certain administrative tasks. These rights are strictly Unity rights and do not
correspond to any AD rights. By assigning limited rights to some subscribers, you can
enable them to assist in various tasks such as unlocking accounts, changing passwords,
and running reports. These rights are based on the subscribers’ CoS.
Unity connection also enables you to grant users administrative rights, but this is done by
assigning roles to user. It is not a function of the CoS. Roles have already been discussed
in detail. This section simply shows how to change the role assignment.
The following steps describe the process of adding administrative rights to an existing CoS:
Step 1.
From within SA, select Subscribers > Class of Service.
Step 2.
Click the Find icon, and a list of CoSs displays.
Step 3.
Highlight the CoS you want to edit and click View.
Step 4.
Select System Access from the left side of the screen.
Step 5.
Figure 8-41 shows the various rights that can be assigned to a CoS. Based on
what tasks you want the subscriber to perform, select the appropriate check
boxes. For the most part, these rights are self-explanatory. Keep in mind that
when you grant one right, another right can be inherited. For example, when
you select the Can Unlock Subscriber Accounts and Change Passwords
check box, the Subscriber Access > Read right is granted. Inherited rights are
not automatically removed when you remove the parent right. You need to
manually remove any inherited rights.
Step 6.
Click the Save icon in the title strip.
The following steps describe how to add administrative rights to Unity Connection users
by assigning roles:
Step 1.
From within UCA, select Users > Users. Enter the appropriate search criteria
and click the Find button to search for the user you want to manage.
Step 2.
Click the name of the user you want to manage.
Step 3.
Navigate to Edit > Roles.
Step 4.
To add roles to this user, select the desired roles from the Available Roles
box and click the up arrow above that box to move them to the Assigned
Roles box.
Step 5.
To remove roles from this user, select the roles you want to remove from the
Assigned Roles box and click the Up Arrow above that box to move them to
the Available Roles box.
Step 6.
Click Save.
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Figure 8-41 Subscriber >CoS >System Access.
Managing Call Transfer and Greetings
The following sections explore how to adjust a user’s settings that affect the outside
caller’s experience when they are transferred to a user. This includes things such as callscreening options and one-key transfer settings.
Allowing Screening and Hold Options
Using screening options, a user can have the system request the caller’s name and play it
to the user before transferring the call. The user can then choose to take the call or send
it to voicemail. The system can also be configured so that the user can choose to have the
system ask callers to hold if they are on other calls. When either of these features is used,
the voicemail port is held open until the call is transferred to a phone. It is best to use
these options sparingly to enable call-screening and holding options.
The following steps demonstrate how to configure these features in Unity:
Step 1.
From within SA, select Subscribers > Class of Service.
Step 2.
Click the Find icon, and a list of CoSs displays.
Step 3.
Highlight the CoS you want to edit and click View.
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Chapter 8: User/Subscriber Reference
Step 4.
Select Transfer from the left side of the screen.
Step 5.
To allow subscribers to enable call-screening options, select the Subscribers
Can Change Call Screening Options check box.
Step 6.
To allow subscribers to enable call-holding options, select the Subscribers
Can Change Call Holding Options check box.
Step 7.
Click the Save icon in the title strip.
The following steps demonstrate how to configure these features in Unity Connection:
Step 1.
From within UCA, select Class of Service > Class of Service.
Step 2.
Click the name of the CoS you want to manage.
Step 3.
Scroll down to the Call Transfer section.
Step 4.
To allow subscribers to enable call-screening options, select the Allow Users
to Change Call Screening Options check box.
Step 5.
To allow subscribers to enable call-holding options, select the Allow Users to
Change Call Holding Options check box.
Step 6.
Click Save.
Changing Maximum Greeting Length
The maximum length of a user’s greeting is determined by the CoS. Follow these steps to
change this value in Unity:
Step 1.
From within SA, select Subscribers > Class of Service.
Step 2.
Click the Find icon, and a list of CoSs displays.
Step 3.
Highlight the CoS you want to edit and click View.
Step 4.
Select Greetings from the left side of the screen.
Step 5.
Enter the desired number of seconds in the Maximum Greeting Length, in
Seconds field.
Step 6.
Click the Save icon in the title strip.
Follow these steps to change the value in Unity Connection:
Step 1.
From within UCA, select Class of Service > Class of Service.
Step 2.
Click the name of the CoS you want to manage.
Step 3.
Enter the desired number of seconds in the Maximum Length Seconds field
located under the Greetings heading.
Step 4.
Click Save.
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Enabling and Disabling Greetings
A user can have up to five greetings configured. Both the subscriber and the administrator have the ability to enable or disable these greetings. The steps for doing this from
within SA in Unity are as follows:
Step 1.
From with SA, select Subscribers > Subscribers.
Step 2.
Click the Find icon and enter search criteria that will match the subscriber
account you plan to edit, and click Find.
Step 3.
Choose the subscriber from the list that displays.
Step 4.
Select Greetings from the left side of the screen.
Step 5.
Choose the greeting you want to enable/disable from the Greeting dropdown list, as shown in Figure 8-42.
Figure 8-42 Subscribers > Subscribers > Greetings
Step 6.
Select either the Enabled - No End Date and Time or the Disabled radio button. You can also enable it with an automatic end date and time by selecting
Enable Until and entering the time and date it should automatically become
disabled.
Step 7.
Click the Save icon in the title strip.
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The steps for configuring the process with UCA in Unity Connection are as follows:
Step 1.
From within UCA, select Users > Users. Enter the appropriate search criteria
and click the Find button to search for the user you want to manage.
Step 2.
Click the name of the user you want to manage.
Step 3.
Navigate to Edit > Greetings.
Step 4.
Choose the desired greeting from the list that displays.
Step 5.
Select either the Greeting Enabled with No End Date and Time or the
Disabled radio button. You can also enable it with an automatic end date and
time by selecting Enable Until and entering the time and date it should automatically become disabled.
Step 6.
Click Save.
Modifying Caller Input Options
Users’ greetings can include options that allow an outside caller to be transferred to
another number by pressing a key on the dial pad. These are referred to as one-key transfers. For these to work, the administrator must configure the caller input options, which
is the same as configuring caller input for a template. The steps that follow show how to
configure these options in Unity:
Step 1.
From within SA, select Subscribers > Subscribers.
Step 2.
Click the Find icon and enter search criteria that will match the subscriber
account you plan to edit, and click Find.
Step 3.
Choose the subscriber from the list that displays.
Step 4.
Select Caller Input from the left side of the screen.
Step 5.
To allow callers to enter another extension while listening to a greeting, select
the Allow Callers to Dial an Extension During Greeting check box.
Step 6.
If the option to allow callers to dial another extension during the greeting is
enabled, an interdigit timeout must also be configured. This is the amount of
time Unity waits before deciding that the caller has finished pressing digits.
This setting is Milliseconds to Wait for Additional Digits. The default value
is 1500 milliseconds, which is a second and a half. Typically, this value is adequate. If, during a greeting, callers are transferred or receive error messages
from Unity before they are finished dialing an extension, increase this value.
Step 7.
From the dial pad on the screen, configure the action that will be taken
when a digit is pressed. A button can be configured for the following five
types of actions:
■
Ignore Key: No action is taken when digit is pressed.
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Step 8.
■
Skip Greeting: The greeting is skipped and Unity proceeds to the aftergreeting action.
■
Take Message: The caller can press this key to cause Unity to take a
message.
■
Say Goodbye: Unity plays a good-bye message and disconnects the call.
■
Send Caller To: You can select various destinations. Table 8-10 lists the
available options.
Click the digit you want to configure, and then select the action by clicking
the appropriate radio button.
Note When you choose to send the call to a call handler, directory handler, interview
handler, or another subscriber, you have to designate the specific handler or subscriber.
Step 9.
If the Lock This Key to the Action (Don’t Wait for an Additional Keypress)
check box is selected, an extension that begins with this digit cannot be
entered. Unity then transfers the call to the destination assigned to the key.
This occurs without checking whether the caller will input additional digits.
Only enable this field on digits that are not leading digits for any extensions.
Step 10. After you have all the digits configured as desired, click the Save icon in the
title strip.
The steps that follow show how to configure these options in Unity Connection:
Step 1.
From within UCA, select Users > Users. Enter the appropriate search criteria
and click the Find button to search for the user you want to manage.
Step 2.
Click the name of the user you want to manage.
Step 3.
Navigate to Edit > Caller Input.
Step 4.
On the first screen that appears, select the key you want to configure as well
as the Wait for Additional Digits and Prepend Digits to Dialed Extensions
settings. The Wait or Additional Digits setting is the amount of time Unity
Connection waits before deciding whether the caller has finished pressing digits. The default value is 1500 milliseconds, which is a second and a half.
Typically, this value is adequate.
Step 5.
The Prepend Digits setting determines whether any digits are prepended to
the extension that a caller enters. This is useful if your callers are accustomed
to entering four digits but the actual extension number is seven digits. For
example, the caller can enter 3322 but the full extension that the call needs to
be transferred to is 5423322. By entering 542 in the Digits to Prepend field
and selecting the Enable check box, the call would be transferred to 5423322.
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Table 8-10
Send Caller to Options
Action
Description
Alternate Contact Number
Sends the call to the number that you specify in the Number
to Dial field.
Broadcast Message
Administrator
Sends the call to a conversation for sending broadcast messages.
CVM Mailbox Reset
Enables the caller to reset the mailbox (available with
Community Voice-mail package).
Call Handler
Sends the call to the selected call handler.
Caller System Transfer
Enables the caller to transfer to another extension after he logs
in with subscriber credentials. He can then transfer to any
number that his restriction table allows.
Directory Handler
Sends the call to the directory handler you select.
Easy Sign-In
Sends the call to a login process that asks for the password for
this mailbox.
Greetings Administrator
Sends the call to a conversation that allows users to manage
the greetings of call handlers that they own.
Hang Up
Disconnects the call.
Hotel Checked Out
This option works in concert with Cisco Unity’s Hospitality
and Property Management Integration. It allows guests to
archive their messages when checking out.
Interview Handler
Sends the call to the interview handler you select.
Sign-In
Sends the call to the subscriber sign-in.
Sign-In Archived Mailbox
Enables subscribers to access messages on the Cisco Unity
server by entering their old Cisco Unity primary extension.
Subscriber
Sends the call to another subscriber’s greeting or extension,
depending upon how it is configured.
Subscriber System Transfer
Enables callers to transfer to another extension after they log
in with subscriber credentials. They can then transfer to any
number that their restriction table allows.
Step 6.
Now you can start configuring the caller input keys. Select the key you want
to configure. A screen similar to that shown in Figure 8-43 appears.
Step 7.
The Key field is static and simply displays the number of the key you are
configuring.
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Figure 8-43 Edit Caller Input
Step 8.
The next field determines whether the key is locked. If the Ignore Additional
Input (Locked) check box is selected, an extension that begins with this digit
cannot be entered. Unity Connection transfers the call to the destination
assigned to that key without waiting to see whether the caller is going to enter
other digits. Only enable this field on digits that are not leading digits for any
extensions.
Step 9.
The Action section determines where the call is sent when the key is pressed.
Table 8-11 lists the available options.
Step 10. Click Save. Return to the Caller Input page to select other keys to configure.
Managing Message Access, Notification, and Indication
Unity/Connection enables subscribers to manage their messages from both the phone
and an email client. To do this efficiently, the system offers many features that impact
how messages can be sent, received, and managed. The following sections discuss various
settings that affect these features.
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Table 8-11
Caller Input Actions
Action
Function
Call action such as hang-up
A specific action must be selected from the
drop-down list.
Call handler
A specific call handler must be selected from
the drop-down list. When sending to a call
handler, you must select either Attempt
Transfer or Go Directly to Greeting.
Typically, Greeting is selected.
Interview handler
A specific interview handler must be selected
from the drop-down list.
Directory handler
A specific directory handler must be selected
from the drop-down list.
Conversation such as sign-in or greetings
administrator
A specific conversation must be selected from
the drop-down list.
Another user’s mailbox
A specific user must be selected from the
drop-down list. When sending to another user,
you must select either Attempt Transfer or Go
Directly to Greeting. To have the user’s phone
ring, select Attempt Transfer.
Allowing Subscribers to Send to Distribution Lists
You can choose to allow or prohibit subscribers to send messages to public distribution
lists. This setting is based on a subscriber’s CoS. The following steps explain how to configure this in Unity:
Step 1.
From within SA, select Subscribers > Class of Service.
Step 2.
Click the Find icon, and a list of CoSs displays.
Step 3.
Highlight the CoS you want to edit and click View.
Step 4.
Select Messages from the left side of the screen.
Step 5.
To allow subscribers with this CoS to send messages to distribution lists,
select the Subscribers Can Send Messages to Public Distribution Lists
check box. Deselect this box if you do not want subscribers with this CoS to
send messages to distribution lists.
Step 6.
Click the Save icon in the title bar.
The following steps explain the configuration process for Unity Connection:
Step 1.
From within UCA, select Class of Service > Class of Service.
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Step 2.
Click the name of the CoS you want to manage.
Step 4.
Scroll down to the Messages Options section.
Step 5.
To allow subscribers with this CoS to send messages to distribution lists,
select the Allow Users to Send Messages to System Distribution Lists check
box. Deselect this box if you do not want subscribers with this CoS to send
messages to distribution lists.
Step 6.
Click Save.
Allowing Messages Deleted from the Phone to Be Saved in the Deleted Items
Folder
By default, when a Unity subscriber deletes a message from the phone, the message is
deleted from the system after the phone is hung up. However, in Unity Connection, the
opposite is true. You can configure Unity to store the message in a subscriber’s Deleted
Items folder. Subscribers can then retrieve messages they have deleted from the phone
through Outlook or Unity Inbox (version 4.03 or later). The following steps explain how
to enable this feature:
Step 1.
From within SA, select Subscribers > Class of Service.
Step 2.
Click the Find icon, and a list of CoSs displays.
Step 3.
Highlight the CoS you want to edit and click View.
Step 4.
Select Messages from the left side of the screen.
Step 5.
Select the Deleted Messages Are Copied to the Deleted Items Folder
check box.
Step 6.
Click the Save icon in the title bar.
Enabling Live Reply for a Subscriber
A feature called Live Reply allows subscribers to have Unity connect them with the
extension of a subscriber that left a message by pressing 44 at the end of listening to the
message. This feature is granted based on the subscriber’s CoS. To enable these features
in Unity, follow these steps:
Step 1.
From within SA, select Subscribers > Class of Service.
Step 2.
Click the Find icon, and a list of CoSs displays.
Step 3.
Highlight the CoS you want to edit and click View.
Step 4.
Select Messages from the left side of the screen.
Step 5.
To allow subscribers access to Live Reply to reach other subscribers, select
the Subscribers Can Reply to Messages from Other Subscribers by Calling
Them check box.
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Step 6.
To allow subscribers access to Live Reply to reach unidentified callers, select
the Subscribers Can Reply to Messages from Unidentified Callers by
Calling Them check box.
Step 7.
Click the Save icon in the title bar.
To enable these features in Unity Connection, follow these steps:
Step 1.
From within UCA, select Class of Service > Class of Service.
Step 2.
Click the name of the CoS you want to manage.
Step 3.
Scroll down to the Messages Options section.
Step 4.
To allow user to access Live Reply to reach other users, select the Users Can
Reply to Messages from Other Users by Calling Them check box.
Step 5.
To allow users to access Live Reply to reach unidentified callers, select the
Users Can Reply to Messages from Unidentified Callers by Calling Them
check box.
Step 6.
Click Save.
Creating Private Lists
Private lists are similar to public distribution lists but are private to the users for whom
they are created. Either the administrator or the user can create these lists. The following
steps illustrate the process of list creation from SA in Unity:
Step 1.
From within SA, select Subscribers > Subscribers.
Step 2.
Click the Find icon and enter search criteria that will match the subscriber
account you plan to edit, and click Find.
Step 3.
Choose the subscriber from the list that displays.
Step 4.
Select Private Lists from the left side of the screen.
Step 5.
Select an unused list number from the Private Lists drop-down list.
Step 6.
Enter a name for the list in the Name of List field.
Step 7.
Click the Change Members button. A search criteria window displays.
Enter search criteria that match the members you want to add to this list
and click Find.
Step 8.
A list of matching objects displays on the left side of the window. Highlight
the objects you want to add to the private list and click the >> button.
Note You can choose multiple objects by holding the Ctrl key down as you click on the
object’s name.
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Step 9.
After all desired members have been added, click the Save button.
Step 10. Record a name for the list by clicking the red dot on the media master control
panel. This should cause the phone to ring. Answer the phone and speak the
name of the private list.
Step 11. Click Save.
The following steps explain how to create these lists from UCA in Unity Connection:
Step 1.
From within UCA, select Users > Users. Enter the appropriate search criteria,
and click the Find button to search for the user you want to manage.
Step 2.
Click the name of the user you want to manage.
Step 3.
Navigate to Edit > Private Distribution Lists.
Step 4.
A new window opens. Essentially, you are now logged in to the CPAC interface as the user you are configuring.
Step 5.
Click the New Private List button.
Step 6.
In the Name field, enter a name for the list.
Step 7.
You can record a name for the list by clicking the red button on the panel.
Enter your extension if prompted. When the phone rings, answer it and
record the name.
Step 8.
If you are using voice recognition in your environment, you can enter alternate names if the name assigned to the list is not pronounced like it is spelled.
If needed, enter a phonetically spelled name as an alternate name.
Step 9.
To add users to the list, click the Add Members button.
Step 10. A new window opens. The types of objects you can add appear near the top
of the window. You can add users, distribution lists, private lists, and remote
contacts to private lists. Select the type of object you want to add to the list.
Step 11. Enter the appropriate search criteria and click Find.
Step 12. Select the check box next to the name of object (user, list, and so on) that you
want to add to the list.
Step 13. Click Add Members.
Step 14. The window closes and you return to the CPCA Private List page. The objects
you added to the list are listed on the Private List Members heading. Click
Save and close the window.
Configuring Message Notification
The user or the system administrator can configure message notification. If possible,
encourage your users to control and configure their own message notification using
CPCA. If it is necessary to configure message notification through the administrative
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interface, the process is very similar to configuration through a template. The following
steps describe how to configure message notification for a Unity subscriber:
Step 1.
From within SA, select Subscribers > Subscribers. Click the Find icon, and
search for the user you want to manage.
Step 2.
When the name of the desired subscriber appears in the title strip, select
Message Notification from the menu that appears on the left side of the screen.
Step 3.
First you must select which device you want to configure. From the Device
drop-down list, choose a device.
Step 4.
Depending on whether you choose a phone or an email device as the notification device, the next few configuration fields vary slightly. If you choose a
phone or a pager, enter the phone number of that device in the Phone
Number field. After dialing the phone number, enter any digits that still need
to be dialed in the Extra Digits field. Extra digits are typically needed for
pagers that use PINs.
Step 5.
If you choose a device that will use an email address as the delivery mechanism, such as a text pager or Text for VMI, enter the email address in the To
field. Then enter the pilot number for Unity in the From field. Some cell
phones act as pagers and can call back the number that is in this field.
Step 6.
When the notification device is a phone or pager, you can set it so that the
extra digits are not sent until a connection is detected. To enable this option,
select the Try to Detect Connection check box. If you want it to wait a specified number of seconds instead of trying to detect a connection, select the
Seconds to Wait Before Dialing Extra Digits check box and enter the number of seconds to wait in the box to the right of that field. You can use the
Seconds to Wait option if Unity is having trouble detecting a connection
because of poor line quality. In some cases, you might need to use both the
Try to Detect a Connection and the Seconds to Wait options. It might be
necessary to set the parameters and then test these settings a few times before
getting them to work properly with different paging companies.
When a device that uses an email address is selected, enter the text that you
want to display in the notification in the Text field.