®
ReadiVoice
Administration &
Maintenance Guide
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Contents
Purpose . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . xv
Document Conventions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . xvi
Support . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . xvii
What is the ReadiVoice System? . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1
How ReadiVoice Conferencing Works . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1
Starting and Joining a Conference . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2
Controlling a Conference . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2
Managing Your ReadiVoice System . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3
Operator Interface . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3
Provisioning Interface . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4
Administration Interface . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4
Components of the ReadiVoice System . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5
ReadiVoice-PSTN Systems . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5
ReadiVoice-IP Systems . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 9
ReadiVoice-IP Call Processing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 10
“Dialing” in ReadiVoice-IP Systems . . . . . . . . . . . . . . . . . . . . . . . . . . . . 11
ReadiVoice Call Flow . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 12
How Quick-Start and Conference Continuation Affect Call Flow . . . 12
How One-Click Conference Affects Call Flow . . . . . . . . . . . . . . . . . . . . 12
How A Two-Password Configuration Affects Call Flow . . . . . . . . . . . 14
How IP Tributaries Affect ReadiVoice-IP Call Flow . . . . . . . . . . . . . . . 14
Maintaining Bridge Information . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 18
Adding a New Bridge . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 18
Deleting a Bridge . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 21
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Managing Routing Lists . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 27
Working with Number Groups . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 30
Working with Access Classes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 33
Deleting an Access Class . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 34
Adding a Provider . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 43
Deleting a Provider . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 46
Working with Subscriber Groups . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 46
Deleting a Subscriber Group . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 59
Managing System Access . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 63
Adding an Internal User . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 63
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Deleting an Internal User . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 66
Accessing the Monitoring Tools . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 69
Checking System Status . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 71
CDR Data by Subscriber . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 73
Viewing Operator Information . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 77
Using the Channel State Monitor . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 86
Quick Maintenance Checklist . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 92
Understanding Informix Backup . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 93
Archives . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 93
Backing Up Logical Logs . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 96
Daily Maintenance Tasks . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 97
Weekly Maintenance Tasks . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 103
Database Maintenance Tasks . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 105
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Rebooting the Sun Server . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 112
Rebooting a Bridge . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 113
Configuring Dial-Out Billing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 126
Pager Notification Setup . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 131
Email Notification Setup . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 132
Enabling Conference Recording . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 137
Enabling SNMP Logging . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 141
Changing the Web Server Port . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 142
Viewing Critical Logs . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 145
Database Troubleshooting . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 147
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Contents
Dial-Out Channel Testing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 149
Lock/Unlock . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 158
Adding a Prompt Set . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 167
Renaming a Prompt Set . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 168
Deleting a Prompt Set . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 168
About Initial Greetings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 181
Voice Prompt File Reference . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 182
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CDR Processing Database . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 251
Informix CDR Tables . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 252
SNMP Reference . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 261
Using the SNMP Log Files . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 266
What are IP Tributaries? . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 271
Implementing IP Tributaries . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 274
Figures
Figure 1-1
ReadiVoice-PSTN system diagram . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 6
ReadiVoice-PSTN INCR signaling flow . . . . . . . . . . . . . . . . . . . . . . . . . . 8
ReadiVoice-IP system diagram . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 9
ReadiVoice call flow . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 13
Figure 1-2
Figure 1-3
Figure 1-4
System Administration home page . . . . . . . . . . . . . . . . . . . . . . . . . . . . 17
Login dialog box . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 17
Bridges page . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 20
Translation Numbers page . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 23
Bridge Groups page (routed system only) . . . . . . . . . . . . . . . . . . . . . . . 26
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Routing Lists page (routed system only) . . . . . . . . . . . . . . . . . . . . . . . . 27
Edit Routing List page (routed system only) . . . . . . . . . . . . . . . . . . . . 28
Number Groups page (routed system) . . . . . . . . . . . . . . . . . . . . . . . . . . 31
Access Classes page . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 33
Access Phone Numbers page . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 36
Access Phone Numbers page after search . . . . . . . . . . . . . . . . . . . . . . . 38
Modify Existing Access Phone Number page . . . . . . . . . . . . . . . . . . . 39
Application Providers page . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 43
Add Provider page . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 45
Subscriber Group page . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 47
Edit Group page (top portion) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 48
Company and Billing sections of Edit Group page . . . . . . . . . . . . . . . 50
Call Flow and Voice Prompt section of Edit Group page . . . . . . . . . . 51
Conference Options section of Edit Group page . . . . . . . . . . . . . . . . . 53
Quick Provisioning Setup page (top) . . . . . . . . . . . . . . . . . . . . . . . . . . 61
Quick Provisioning Setup page (bottom) . . . . . . . . . . . . . . . . . . . . . . . 62
Passwords page . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 64
Operator Registered Subscriber Groups page . . . . . . . . . . . . . . . . . . . 65
Invalid Subscriber Password Table page . . . . . . . . . . . . . . . . . . . . . . . 67
System Status page . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 71
Conference Status page . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 72
Perfmeter tab of SNMP Monitor window . . . . . . . . . . . . . . . . . . . . . . . 82
SNMP Monitor Properties dialog box . . . . . . . . . . . . . . . . . . . . . . . . . . 83
Statistics tab of SNMP Monitor window . . . . . . . . . . . . . . . . . . . . . . . 84
Histograms window of SNMP Monitor . . . . . . . . . . . . . . . . . . . . . . . . . 85
Channel State Monitor page, listing bridges in the system . . . . . . . . 86
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Checking Informix users . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 100
System Configuration page . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 118
Critical Logs page . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 145
Bridge critical log page . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 146
[DTMF_CMDS]section of default ive.inifile . . . . . . . . . . . . . . . . . . . . 157
Prompt Sets page . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 168
Recording dial-out process . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 213
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Figures
Name record process . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 224
Operator wait process . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 226
Account options process . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 227
Quick start setting process . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 234
Roll call setting process . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 235
Exit announce process . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 238
Rejoin conference process . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 243
Other in-conference events . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 250
SNMP overview . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 262
Sample SNMP log file . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 267
Sample histogram log file . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 270
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Tables
Group Information fields . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 49
Billing Information fields . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 51
Conference Options . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 54
purgeCDRcommand options . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 106
purgeCDRexamples . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 107
Alarm code format . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 136
Error code descriptions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 136
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Tables
SNMP log file fields . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 267
Histogram log file fields . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 270
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About This Manual
This introduction provides a brief overview of the ReadiVoice Administration &
Maintenance Guide, describes the conventions used in this manual, and
explains how to get additional information or support.
Purpose
This manual is for administrators of a ReadiVoice system. It covers:
•
•
•
•
Administering the ReadiVoice application. Describes using the Web-based
System Administration interface to add, modify, and view the system and
user data.
Monitoring the ReadiVoice system. Describes using the monitoring tools
in the System Administration interface to monitor the status of the system,
usage levels, conference data, and operator data.
Maintaining the ReadiVoice system. Includes routine maintenance tasks,
backup and database maintenance procedures, and other maintenance
tasks you may need to perform from time to time.
Configuring the ReadiVoice system. Includes procedures for changing
how the system works and what capabilities, features and options are
enabled. Many of these tasks involve editing configuration files and
working at the UNIX command prompt.
•
•
Customizing and branding the ReadiVoice system. Describes customizing
the system for your company and customizing how specific sets of users
experience the system.
Diagnosing and troubleshooting the ReadiVoice system. Describes tools
and procedures for identifying problems with the system.
In addition, the appendices of this manual provide reference information
about the system, including voice prompts and call flows, CDR (call detail
record) data, and SNMP data.
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Document Conventions
This document uses the following typographical conventions.
Typeface
bold
Usage
Names of fields, screens, windows, dialog boxes,
and other user interface elements; for example:
1
Type the number into the Phone Number
field and click Dial.
2
Click Cancel to close the dialog box.
italics
New terms, book titles, or emphasis; for example:
According to the VERITAS Cluster Server User
Guide, crash tolerant applications are sometimes
referred to as cluster friendly applications.
code
Computer output, command references within text,
and filenames; for example:
Performs the initial configuration and reads the
.vcsrcfile
code, bold
Command line entries, for example:
>> Type cp ../default_group.ini
group.ini
.
code, bold & italics
Command line variables, for example:
>> Type cp ../default_group.ini
groupnn.inireplacing nn with the
subscriber group number.
SMALL CAPS
Specific keys on the keyboard, for example:
>> Move the cursor by pressing TAB or
SHIFT+TAB.
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About This Manual
Support
Recognizing that technology alone cannot solve today’s complex challenges,
Polycom Global Services provides the industry’s best technical support staff
and programs to let you concentrate on the task at hand. ReadiVoice users can
select from a variety of support solutions to obtain the level of support that
best meets their needs.
Before contacting your Polycom Global Services representative for technical
assistance, gather as much information as possible about your situation. Any
information you can provide helps us assess the problem and develop an
appropriate solution.
Polycom Global Services Telephone and Email
If you have comments or questions about ReadiVoice or if you need technical
assistance, contact:
800-827-7782
303-223-5223
•
•
•
Polycom Global Services in U.S.A
Denver metro or outside U.S.A.
Email: techsupport@polycom.com
Polycom Global Services on the Web
For more information about Polycom Global Services, go to
http://www.polycom.com and select the Support Solutions link under the
Global Services category.
Polycom Technical Publications Department
If you have comments or questions about this or any other ReadiVoice
documentation, contact:
Polycom Technical Publications Department
1765 West 121st Avenue
Westminster, CO 80234-2301 U.S.A.
Email: TechPubs@polycom.com
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1
Introducing the ReadiVoice® System
This chapter offers a general overview of the ReadiVoice conferencing system
and its features, functionality, and components.
What is the ReadiVoice System?
The ReadiVoice is the industry-leading, on-demand conferencing system. It
provides an easy-to-use, reliable, scalable, and full-featured conferencing
solution.
A ReadiVoice conference may contain up to 300 participants, including the
conference leader(s).
How ReadiVoice Conferencing Works
The ReadiVoice system is subscriber-based. A subscriber is someone, such as a
chairperson or conference leader, for whom you’ve set up conferencing access.
Each subscriber has access to a specified number of conferencing ports and can
use them at any time.
Through the provisioning process, you add subscribers to the ReadiVoice
system’s database. Each subscriber is given an access phone number for dialing
into the system and a subscriber password that identifies the subscriber. An
access number can be shared by many subscribers or private (assigned to only
one subscriber). If the access number is shared, the subscriber is also given
either:
•
•
An access code that everyone uses to access the subscriber’s conferences.
A participant password that participants use to access the subscriber’s
conferences (the subscriber uses the subscriber password).
Each conferencing subscriber is a member of a subscriber group. Subscriber
groups are logical groupings of related subscribers who share the same default
settings and options (for instance, a specific prompt set). You can create groups
for specific purposes or you can use one subscriber group for all subscribers.
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Starting and Joining a Conference
Once provisioned, a subscriber can hold a conference at any time. No advance
reservation is necessary or possible. Subscribers can start conferences from
any touchtone telephone by dialing the access phone number and providing
their access code (if any) and subscriber password when prompted.
Participants join a conference by dialing the access phone number and
providing their access code or participant password. If the account permits,
the subscriber can also specify a conference security code that participants
must know to join the conference.
Controlling a Conference
A subscriber can have only one conference active at a time. During a
conference, subscribers can control the conference and dial out to additional
participants using:
Telephone touchtone (DTMF) commands — The ReadiVoice system
supports a complete set of touchtone commands. Subscribers can use these to
control the conference, enable or disable features, and dial out. Participants
can use these to control their own lines and use features to which they’ve been
given access.
The touchtone command set is completely configurable at the system level. In
addition, you can enable or disable specific features at the subscriber group or
individual subscriber level.
The Web-based Moderator interface — The Moderator lets your conferencing
subscribers or their designated moderators set up, start, monitor, and control
their conferences over the Internet. To log into the Moderator, users enter the
subscriber’s access phone number, access code (if any), and password.
The Moderator provides the conference control features that are enabled in the
subscriber profile for the conference. Most commands available to the
subscriber as touchtone commands are also available in the Moderator.
By default, conferences start when the subscriber arrives and end when the
subscriber leaves. This can be modified as follows:
•
•
•
If the Quick Start option is enabled, participants don’t have to wait on hold
for the subscriber. The conference starts when the first participant arrives.
If Continuation is enabled, the conference doesn’t end when the subscriber
leaves. It continues until the last participant leaves.
The system can be configured to end a conference (or call an operator)
containing only one to three lines (configurable) after a specified delay.
You can use this feature to handle problems such as someone forgetting to
hang up a speaker phone in a conference room.
For more detailed information about how subscribers and participants interact
with the system to set up, start, control, and join conferences, see the ReadiVoice
Subscriber Guide.
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Introducing the ReadiVoice® System
Managing Your ReadiVoice System
Day-to-day management and operation of your ReadiVoice system is
accomplished using operator/maintenance stations, which are typically standard
PCs connected to your company’s LAN (local area network) or WAN (wide
area network). From these PCs, your company’s staff can access the
Web-based ReadiVoice applications for:
Operators — Respond to operator requests from subscribers (and, if
authorized, participants) to resolve problems, answer questions, or provide
other assistance.
Provisioners — Add, modify, and delete subscriber accounts.
System Administrators — Configure and maintain the ReadiVoice system,
create new subscriber groups, and set up other internal users and their
passwords. They can also perform the provisioning functions.
To use a Web-based ReadiVoice application, users launch a supported
browser (Microsoft Internet Explorer version 5.5 or later), point it to the correct
Internet address (URL) for the application, and log in when prompted.
The Operator application and some parts of the Administration interface also
require a Java virtual machine (plug-in), which the user is prompted to
download if it isn’t already installed. Voyant has certified these Java applets
with the Sun Microsystems Java Runtime Engine (JRE) 1.3.1 and 1.4.2, on
Microsoft Windows 2000 and Microsoft Windows XP.
These applets should work on other platforms and operating systems and with
newer versions of the JRE, but that depends on proper implementation by the
operating system and Java virtual machine. Polycom hasn’t tested other possible
combinations and can’t be responsible for implementation or compatibility issues
beyond our control.
Note:
Operator Interface
The ReadiVoice Operator interface lets operators monitor the conferences on
the system, answer operator requests from their subscribers and participants,
and perform various tasks to assist users.
The ReadiVoice system doesn’t require operators. Providing operator services is
entirely optional.
Note:
Operators may have access to all conferences or they may be limited to the
conferences of a specific subscriber group or groups, depending on their login
name and the system configuration.
When someone requests an operator, the system sends an audible and visual
alert to the available operators. An operator can answer the oldest request in
the queue or select a specific request from a list.
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When operators answer a request or select a conference to monitor, the system
registers them for the conference and displays all the available information
about the conference and its participants.
If an operator’s duties include provisioning as well as answering operator
requests, the operator can switch to the Provisioning interface by clicking the
Provisioning button on the Operator page.
For more information about the Operator interface and operator tasks, see the
ReadiVoice Operator Guide.
Provisioning Interface
The ReadiVoice Provisioning interface lets provisioners create and modify
ReadiVoice subscriber accounts. A subscriber account authorizes someone to
use your ReadiVoice system.
We recommend automating your initial subscriber provisioning. Contact Polycom
for help with this. The ReadiVoice system’s Provisioning interface is available for
manual provisioning and for viewing or modifying subscriber account information.
Note:
To set up an account for a customer, a provisioner enters subscriber and billing
information and defines the subscriber’s conference access and features.
All subscribers belong to a subscriber group. You can use one subscriber group
for everyone, or you can have multiple subscriber groups, each used for a
specific company, access method, or other affinity group. A subscriber group
can have its own default voice prompt set, account settings, and number group
(controlling access numbers and routing).
Like operators, provisioners may have access to all subscriber groups or they
may be limited to a specific subscriber group, depending on their login name
and the system configuration.
The Provisioning interface lets provisioners create new subscriber accounts or
search for an existing account and then edit or disable it.
For more information about the Provisioning interface and provisioning tasks,
see the ReadiVoice Provisioning Guide.
Administration Interface
The ReadiVoice Administration interface lets administrators monitor the
system, create new subscriber groups, customize the interfaces, create and
maintain user logins, set default parameters, and configure bridges.
This manual describes the Administration interface and administrative tasks.
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Introducing the ReadiVoice® System
Components of the ReadiVoice System
A ReadiVoice system includes:
•
•
A Conference Allocation and Control System (CACS) server.
Up to twelve InnoVox media servers (conferencing bridges), providing up
to 5,760 ports in a single system, or one InnoVox media server, providing
4032 ports in a single 13U bridge cabinet.
•
An optional boot server from which the bridges boot.
The specific configuration of a ReadiVoice system depends on its capacity
(number of ports), type of media server(s), and options.
The CACS and boot server may be the same server. All system components
mount into a standard 19-inch rack.
Two types of ReadiVoice systems are available. There are both hardware and
software differences between the two types, although the core ReadiVoice
software is the same (merely configured differently). The two types are for
different network environments:
ReadiVoice-PSTN uses the Public Switched Telephone Network (PSTN). The
bridge or bridges make circuit-switched connections over standard digital
telephony spans.
ReadiVoice-IP uses Voice Over IP (VOIP). The bridge makes packet-switched
Internet Protocol (IP) connections over Ethernet.
ReadiVoice-PSTN Systems
The PSTN version of ReadiVoice has two call routing configurations:
Fixed Access — Each access telephone number terminates on a specific bridge
in the system. This configuration is called a non-routed system.
Intelligent Network Call Routing (INCR) — Access telephone numbers
aren’t assigned to specific bridges. Instead, the ReadiVoice system uses a
carrier common channel signaling network (such as SS7 or Tollfree Gateway)
to dynamically route calls among available bridges. This configuration is
called a routed system. A routed system makes more efficient use of the bridges
because you can distribute the conferencing load and specify bridge
availability and priority.
Figure 1-1 provides a high-level view of the major components in a routed
ReadiVoice-PSTN system attached to a carrier common channel signaling
network. A fixed access system is identical, but without the Service Control
Point (SCP) connecting it to the carrier common channel signaling network.
The SCP may or may not be required for a routed system, depending on your
carrier.
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The SCP (if required), subscriber database, and Web server can physically
reside on the CACS server. In large systems, however, these components may
be distributed among multiple servers in order to distribute the processing
load.
Figure 1-1 ReadiVoice-PSTN system diagram
InnoVox
Bridges
Carrier
common
Subscriber access via
channel
Web-based Moderator
signaling
network
Internet
SCP (if required)
PSTN
(See note)
CACS
Operator/Maintenance
Station
Database
Note: The InnoVox bridges are connected to
the PSTN via a switch certified by the local
country regulatory authority.
When a caller dials a ReadiVoice-PSTN subscriber’s access phone number on
a typical routed ReadiVoice system:
1
The originating central office switch recognizes the number as requiring
carrier common channel signaling network handling.
2
The carrier switch sends a query through the carrier common channel
signaling network to the ReadiVoice CACS, which allocates bridge
resources to that subscriber according to his or her subscriber account
settings.
3
The CACS returns a routing solution through the carrier common
channel signaling network to the originating carrier switch. It also
notifies the bridge of the upcoming call and the associated account profile
and features.
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Introducing the ReadiVoice® System
4
5
The carrier switch uses the routing solution to route the call to the
InnoVox bridge selected by the CACS call router.
Figure 1-2 summarizes this INCR signaling flow.
When the call reaches the bridge, the ReadiVoice system places it into an
interactive call flow script to guide the caller into the conference. In a
fixed access system, the script collects the access code (if needed). In all
systems, the script handles subscriber identification and collection of the
optional conference security code. Non-subscribers are put on hold until
the subscriber arrives, enters the subscriber password, and starts the
conference.
6
7
If the account permits, the call flow script gives the subscriber an
opportunity to review and change account options prior to starting the
conference.
Once the subscriber starts the conference, participants are prompted to
enter the conference security code (if applicable) and then placed into the
conference.
Figure 1-4 on page 13 summarizes the ReadiVoice basic call flow after the call
reaches the bridge or on a non-routed system.
If the Quick Start feature is enabled, participants don’t have to wait for the
subscriber. They’re placed directly into conference instead of on hold. If the
Roll Call feature is enabled, callers are prompted to record their name before
entering the conference.
Callers who enter the subscriber or participant password or the conference
security code incorrectly three times are played the appropriate message and
then either sent to an operator (if available) or disconnected, depending on
how your system is configured.
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Figure 1-2 ReadiVoice-PSTN INCR signaling flow
Start
Originating switch
recognizes that the
number requires carrier
common channel
signaling network
handling
User dials
ReadiVoice
access phone number
Originating switch
initiates carrier common
channel signaling
If applicable, collects
access code.
network query to CACS
Return associated
PSTN translation
number
Is the conference
underway?
Are there enough
ports?
Yes
Yes
No
Originating switch
completes call using
translation number
supplied by carrier
common channel
signaling network
ReadiVoice is not
available. Try again.
(Or special network
handling.)
Are there enough
ports?
No
Call placed in
ReadiVoice call flow
script
Select bridge,
allocate resources
Yes
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Introducing the ReadiVoice® System
ReadiVoice-IP Systems
All ReadiVoice-IP systems are routed. That is, the CACS uses SIP (Session
Initiation Protocol) signaling over IP to route calls dynamically among
available bridges.
Figure 1-3 provides a high level view of the components in a ReadiVoice-IP
system.
Figure 1-3 ReadiVoice-IP system diagram
Subscriber
access via
Web-based
Moderator
1
4
7
2
5
8
8
3
6
9
#
PSTN
*
SIP Phone
Gateway
1
4
7
2
5
8
8
3
6
9
#
*
Internet
SIP Phone
I P
Windows XP
SIP Phone
InnoVox
Bridge(s)
Ethernet
Switch
CACS
Operator/
Maintenance
Station
Gigabit
Conference
Database
100Base-T
Control
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ReadiVoice-IP Call Processing
When a caller dials a ReadiVoice-IP subscriber’s access phone number on a
typical ReadiVoice system:
1
The SIP Invite message from the caller‘s SIP user agent (perhaps a SIP
telephone) reaches the call-control portion of the CACS. If the SIP invite
contains no access code:
a
The CACS directs the call to a bridge with a free port and runs a bridge
script that prompts the caller for the access code.
b
The bridge returns that access code to the CACS.
2
The CACS uses the access code to identify the subscriber account. Its call
router allocates bridge resources according to the service description of
the subscriber just identified.
3
4
The CACS connects the call to the bridge just chosen.
When the call reaches the bridge, an interactive call flow script guides the
caller into the conference. The script handles subscriber identification and
collection of the optional conference security code. Non-subscribers are
put on hold until the subscriber arrives, enters the subscriber password,
and starts the conference.
5
6
If the account permits, the call flow script gives the subscriber an
opportunity to review and change account options prior to starting the
conference.
Once the subscriber starts the conference, participants are prompted to
enter the conference security code (if applicable) and then placed into the
conference.
Figure 1-4 on page 13 summarizes the ReadiVoice basic call flow after the call
reaches the bridge.
If the Quick Start feature is enabled, participants don’t have to wait for the
subscriber. They’re placed directly into conference instead of on hold. If the
Roll Call feature is enabled, callers are prompted to record their name before
entering the conference.
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“Dialing” in ReadiVoice-IP Systems
For the sake of simplicity, we refer to “dialed phone numbers” throughout this
document. Actually, in IP telephony, end points are identified by a Universal
Resource Identifier (URI).
This URI resembles an email address: phonecontact@domain. Domain is
pretty much what one would expect: something.com or something.org, for
example. Phonecontact can be anything that identifies a device known to the
domain, such as a name, a 10-digit phone number, or perhaps an extension
number.
The ReadiVoice-IP system uses the Session Initiation Protocol (SIP) to
communicate with other IP telephony equipment. It sends out formal SIP
URIs, but users can “dial” any of the following:
Formal SIP URI
The URI described above preceded by the name of the protocol and a
colon. For instance:
SIP:3032235000@voyanttech.com
SIP:brent@voyanttech.com
SIP:5000@voyanttech.com
Simplified URI
The URI with the protocol name omitted (similar to omitting the “http:” in
a Web address). For instance:
3032235000@voyanttech.com
brent@voyanttech.com
5000@voyanttech.com
URI Without Domain
If your system is configured with a default domain and port, it routes
addresses that don’t specify a domain to the default domain and port. For
instance:
3032235000
brent
5000
Since most of the world is still using the PSTN and ordinary phone numbers,
your ReadiVoice-IP system undoubtedly sits behind one or more gateways to
the circuit-switched network. When your system is given an ordinary phone
number with no domain specified (such as the example 3032235000 above), it
routes the call to the gateway (or to a router that sends it to the appropriate
gateway).
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ReadiVoice Administration & Maintenance Guide
ReadiVoice Call Flow
The basic call flow is the same for ReadiVoice-PSTN and ReadiVoice-IP (see
Figure 1-4). Various system-level and subscriber-level configurations and
settings affect the basic call flow, however. The sections that follow describe
some of these call flow differences.
How Quick-Start and Conference Continuation Affect Call Flow
By default, conferences start when the subscriber arrives and end when the
subscriber leaves. Two options can change this:
Quick Start
The conference starts as soon as the first participant arrives. Participants
can speak to one another prior to the arrival of the subscriber or hold a
conference without the subscriber. Quick Start is useful for certain types of
users, such as disaster management groups and others who need to meet
without relying on any one individual to start the meeting.
Subscribers should be aware that Quick Start conferences are less secure
than conferences requiring a subscriber password.
Conference Continuation
The conference can continue after the subscriber disconnects. In this case,
the conference ends when the last participant disconnects.
If the subscriber’s account permits, the subscriber can turn Conference
Continuation on or off for each conference. If the subscriber’s account has
Auto Continuation enabled, all the subscriber’s conferences start with
Conference Continuation turned on.
How One-Click Conference Affects Call Flow
In a One-Click Conference, participants click on a link or icon to enter the
conference. The ReadiVoice system identifies the conference from the link’s
URL and returns a Web page asking for the participant’s phone number. When
the system gets this number, it calls and puts the connection in conference (or
on music hold, if the subscriber isn’t present and Quick Start is off). The
subscriber can use the same link, since the Web page provides a checkbox to
indicate subscriber and a field for entering the subscriber password.
The Provisioning application generates the One-Click Conference link for a
new subscriber account. If the provisioners have email capability, they can
paste the link into an email message and send it to the subscriber for
distribution to participants.
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Figure 1-4 ReadiVoice call flow
(1) In an Intelligent Network Call Routing (INCR) system, the carrier's network takes care of this.
(2) If the Access Phone Number is shared, an Access Code is required to identify the specific subscriber.
(3) Depending on your system configuration, additional Access Code prompts may be possible.
(4) If account has Roll Call feature enabled.
(5) After three incorrect attempts, caller is routed to an operator (if available) or played a message and disconnected.
(6) For Quick Start conferences, participants bypass hold and are placed directly into conference.
(7) The caller remains on hold until the subscriber arrives and starts the conference.
(8) Subscription setting determines whether Conference Security Passcode is mandatory, optional, or unavailable. If optional,
the subscriber chooses whether to use a Conference Security Code for each conference.
(9) ACM (Application Control Mode), if enabled, removes callers from the call flow for an interactive session with an external application.
(10) Subscription setting determines whether subscriber has access to Account Options Menu.
(11) There is no request to press * if a Participant Password is enabled and specified.
(12) Skip if Prompt For Subscriber is set to Off; subscriber is not asked to press *, and, instead, a generic password request is played.
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How A Two-Password Configuration Affects Call Flow
In the traditional ReadiVoice call flows, the system:
•
Prompts all callers for an access code (unless the subscriber has a private
access number) to identify the conference.
•
•
Prompts callers to identify themselves as subscribers by pressing *.
Prompts the subscribers for their subscriber password.
In two-password call flows, no separate access code is used; both subscribers
and participants are given passwords. The system:
•
•
Prompts all callers for their password.
Uses it both to identify the conference and to distinguish between
subscribers and ordinary participants.
With two passwords, subscriber identification occurs without any additional
prompting by the system. This provides a shorter call flow.
How IP Tributaries Affect ReadiVoice-IP Call Flow
ReadiVoice-IP systems can be configured to enable the IP Tributaries feature.
To use this feature, callers must be authenticated by some means outside of the
ReadiVoice system, and the SIP INVITEmessage that they send to access the
system must include the identifying information.
Each piece of information contained in the SIP INVITEmessage takes the place
of information that the caller would otherwise be prompted to enter. For
instance:
•
If the SIP INVITEmessage includes an access code (traditional call flow),
then the system can identify the conference from the invite and only needs
to prompt for subscriber identification.
•
If the INVITEmessage includes both the access code and subscriber
password (traditional call flow) or either a participant password or a
subscriber password (two-password call flow), then the system can
identify both the conference and the caller’s role and doesn’t need to
prompt the caller for any information.
For security, the authentication mechanism can encrypt the passwords.
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Administering the ReadiVoice System
This chapter describes how to use the ReadiVoice system’s Web-based System
Administration interface to add, modify, and view the system and user data.
It covers:
•
•
•
•
•
•
•
•
•
•
•
Some of the functions available in the System Administration interface are
described elsewhere:
•
•
•
The changes you can make on the System Configuration page are
The various monitoring functions available in the System Administration
Installing multiple prompt sets and setting them up in the System
•
Using the Critical Logs page to help diagnose problems is described in
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Opening ReadiVoice System Administration
You can access the ReadiVoice System Administration interface from any
computer that can connect to the ReadiVoice system’s Web server and has a
compatible Web browser (Microsoft Internet Explorer, version 5.5 or later, is
compatible with all ReadiVoice interfaces).
Access is restricted to authorized users. You must know the correct user name
and password.
To access the System Administration pages:
1
Do one of the following:
— Point your browser to the internal user home page (for example,
http://www.rvoice.com/index2.html). Then click the System
Administration link in the navigation bar on the right.
— Point your browser directly to the System Administration page (for
example,
https://www.rvoice.com/provisioning/Adminframe.html).
navigation bar on the right provides links to pages for performing various
administrative functions.
2
In the navigation bar, under the System Administration heading, click
the link for the function you want.
All the System Administration links are restricted to authorized users. If
you aren’t already logged in as an authorized user, a login dialog box
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Maintaining Bridge Information
When Polycom installs your ReadiVoice system, we configure the application
with the correct bridge information. If any of this information changes, you
must update the system’s bridge table.
Contact your Polycom Global Services representative before making any changes
to your system’s bridge configuration.
Caution!
Adding a New Bridge
Installing a bridge (media server) involves:
•
Physically installing the hardware and making all cable connections.
Running the creator scripts to create the configuration files for the bridge.
•
•
Using the vbootptoolshell script to configure VBootP to bootstrap the
bridge.
•
Adding information about the bridge to the ReadiVoice database so that
the ReadiVoice application is aware of its presence and able to use it.
This section describes only the last of these tasks. The prerequisite installation
and configuration tasks are covered in the appropriate installation manual for
the system.
This manual includes information about adding bridges for reference
purposes; you shouldn’t need it. When you buy a new bridge from Polycom,
a field engineer installs the bridge and configures your system.
To add a new bridge to the ReadiVoice system:
1
In the System Administration navigation bar, click Bridges.
2
In the Add Bridge section at the bottom, if the new bridge’s ID number
isn’t in the Bridges list, stop here.
The installation of the bridge and VBootP configuration with the
vbootptoolshell script are incomplete.
— Make sure that the vbootptoolshell script was run.
— Verify the VBootP configuration.
— Make sure the bridge is powered on and properly bootstrapped
fromVBootP.
3
In the Add Bridge section at the bottom, if the ID number for the new
bridge is in the Bridges list, select it.
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4
In the Add Bridge section at the bottom, complete the following fields:
Reserve Ports — Enter the number of ports you want to reserve for
operator voice paths. These ports are not available for conferencing.
Trans DNIS Length — For IP systems, this is the number of digits in each
translation number. For PSTN systems, enter the number of DNIS (dialed
number identification service) digits the network delivers to the bridge. If
the network delivers non-delimited or single-delimited ANI/DNIS, the
ive.inifile also specifies a DNIS length, and the setting here must match
that one.
Depending on the system’s translation number type (see “Changing
also apply:
— If Translation Number Type is set to Random Translation Number,
the number of digits must exactly equal the length of the translation
numbers (not including the prefix, if any) assigned to the bridge (see
DNIS Length to a different length, remove the translation numbers
first.
— If Translation Number Type is set to Fixed Translation Number, the
number of digits must exactly equal the number of digits in each
subscriber’sExternalIdfield. If you change Trans DNIS Length to a
different length, a warning message tells you that you must update
your subscribers’ external IDs.
— If Translation Number Type is set to Fixed Translation Number with
3-digit code, the number of digits must be exactly three more than the
number of digits in each subscriber’sExternalIdfield. If you change
Trans DNIS Length to a different length, a warning message tells you
that you must update your subscribers’ external IDs.
Hrt Beat Interval — Enter the interval (in seconds) between heartbeat
messages from the bridge to the Conference Allocation and Control
System (CACS) server. The heartbeat message confirms that the bridge is
available and can communicate with the server. One second is generally
considered a reasonable interval.
Max Missed Hrt Beats — Enter the number of heartbeats that the CACS
can fail to receive before triggering an alarm. Eight is generally considered
a reasonable number.
Bridge Status — Select Busyout to hold the bridge back from use until
you’ve completed the bridge configuration.
5
6
Click Add Bridge.
The system provides a link to the Translation Numbers page.
Click the link, and then add translation numbers to the bridge as
The system confirms that the translation numbers have been added.
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7
8
In the navigation bar, click Bridges.
To make the bridge available for service, select In Service in its Bridge
Status field and click the Commit button (blue “c”).
The system confirms that the bridge’s status has been changed and
provides a link back to the Bridges page.
Figure 2-3 Bridges page
Modifying an Existing Bridge
1
In the System Administration navigation bar, click Bridges.
2
3
To change a bridge’s status from In Service to Busyout or vice versa,
select the status you want in the Bridge Status field.
To change a bridge’s number of reserved ports, DNIS length, or heartbeat
settings, edit the appropriate field.
If this is a routed system, you can’t change the DNIS length to something
different than the length of the bridge’s translation numbers. If you must
make this change, remove the translation numbers first (see “Changing
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Administering the ReadiVoice System
The value in the Bridge ID field must be unique and must correspond to the bridge’s
appBridgeID value in the vbootp.db file. If the bridge was properly configured in
VBootP, its Bridge ID was set correctly when it was added to the ReadiVoice
database, and it shouldn’t be changed.
Caution!
4
To save your changes, click the Commit button (blue “c”).
The system confirms that the bridge has been modified and provides a link
back to the Bridges page.
To change a bridge’s translation numbers, follow the procedure in “Changing
Deleting a Bridge
Before deleting a bridge, remove its translation number assignments. Then
remove the bridge from any bridge groups to which it belongs.
To remove a bridge from the ReadiVoice application:
1
In the navigation bar, click Bridges.
Find the bridge you want to delete and click its Delete button (red “–”).
2
If the bridge has IP addresses or translation numbers associated with it or
belongs to a bridge group, the system warns you of this and doesn’t delete
the bridge. Otherwise, it confirms deleting the bridge.
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Working with Translation Numbers
A translation number is the telephone network’s translation of the dialed
telephone number into a number used to route the call:
•
For a routed PSTN system, your carrier provides a block or blocks of
translation numbers. Depending on the translation number type (see
may randomly choose a number from this block, or it may use a
subscriber-specific fixed translation number. In either case:
a
When someone dials a ReadiVoice access number, the originating
switch queries the CACS, which determines who the subscriber is and
returns a translation number (including prefix) to tell the telephone
network how to route the call.
b
c
When the switch routes the call to the bridge, it sends the translation
number, with or without prefix, as DNIS (dialed number
identification system) digits.
The bridge sends the DNIS to the CACS, which uses the last
n
digits
(where is the Trans DNIS Length) to identify the conference.
n
•
For a non-routed system or an IP system, you must enter a block of
dummy translation numbers. The number of digits must be the number
In either type of system, each conference requires one translation number for
each different access number used to reach the conference (see “Working with
The maximum number of conferences is always the number of ports divided
by three. Therefore, the maximum number of translation numbers a bridge
might need is the smaller of:
•
•
The number of access classes times one-third the number of ports.
The number of ports.
Table 2-1 shows the results of this calculation for 480-port and 4032-port
bridges.
Table 2-1
Maximum number of translation numbers needed per bridge
InnoVox 480 with 480 ports
(max. 160 conferences)
InnoVox 4000 with 4032 ports
(max. 1344 conferences)
Access classes
Translation numbers
Access classes
Translation numbers
1
160
320
480
1
1344
2688
4032
2
2
3+
3+
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These numbers are the worst-case scenario: each conference contains three
lines, and each line used a different access nmber.
You may be able to reduce these numbers further, especially if most of your
subscribers use a single access number from your primary access class, and the
other access classes are used rarely. But, do so with caution. If a bridge runs
out of translation numbers, no more calls can connect to it, even though ports
are available.
Don’t add or change translation number assignments while conferences are under
way! Perform the following procedures after operational hours.
Caution!
Adding Translation Numbers to a Bridge
1
In the navigation bar, click Bridges.
2
Find the bridge to which you want to assign translation numbers and
click its Translations link.
translation numbers assigned to this bridge, the page contains only a blank
row with an Add button (green “+”) at the right.
Figure 2-4 Translation Numbers page
3
In the Prefix Number field, enter the prefix digits (the portion that the
switch strips off before sending DNIS digits) of the number block you’re
assigning. Enter only the digits (no punctuation).
In a non-routed or IP system, leave the prefix field blank.
4
5
In the Range Start field, enter the number that begins the block of
translation numbers. For instance, if your system receives only four DNIS
digits from the switch, this might be 1000
.
In the Range End field, enter the number that ends the block of
translation numbers. For instance, if your system receives only four DNIS
digits from the switch, this might be 1479
.
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Click the Add button.
The system assigns the translation numbers to the bridge. For a large
block, this may take several minutes. When finished, it confirms that the
translation numbers have been added and provides a link back to the
Translation Numbers page.
7
Click the link to return to the Translation Numbers page.
The new row appears in the list. A blank row below lets you add
additional blocks of numbers.
Repeat the above procedure to add another block of numbers.
Changing the Translation Numbers Assigned to a Bridge
1
In the navigation bar, click Bridges.
2
Find the bridge whose translation numbers you want to change and click
its Translations link.
contains a row for each block of contiguous translation numbers assigned
to the bridge.
3
4
To change a block of translation numbers, edit the fields that need to be
changed and click the row’s Commit button (blue “c”).
The system implements the changes you made. Depending on the extent
of the change, this may take several minutes. When finished, it confirms
the change.
To remove a block of translation numbers, click the row’s Delete button
(red “–”).
The system confirms the change.
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Working with Bridge Groups and Routing Lists
Bridge groups and routing lists apply only to routed systems (PSTN or IP). If
you have a non-routed system, the access phone number determines which
bridge receives a call, and this section doesn’t apply to you.
In a routed system, bridge groups and routing lists control the routing of calls
among bridges. Each subscriber is assigned to a routing list (by default, the one
assigned to the number group, but provisioners can change this). The routing
list specifies which bridge or bridges the subscriber’s conferences can use and
the priority order of those bridges.
Setting up bridge routing involves:
•
•
Setting up bridge groups, each containing one or more bridges.
Setting up routing lists, each specifying the bridge groups to which calls
should be routed in order of preference.
Managing Bridge Groups
This section applies only to routed systems. In a routed system, if you add
bridges or change bridge assignments, you must make changes to your bridge
groups. By default, your system contains the bridge group All Bridges, which
includes all bridges. You can’t modify or delete this group. You can add other
groups and modify or delete them.
Adding a Bridge Group
1
In the navigation bar, click Bridge Groups.
groups, starting with the default entry, All Bridges. A blank input field
and Add button (green “+”) let you add bridge groups. Check boxes below
each entry let you select which bridges are in each group.
2
3
Enter a name for the new group in the blank Bridge Group Name field.
Select the check box of each bridge you want to include in the group and
click the Add button.
The system confirms that the bridge group has been added and provides
a link back to the Bridge Groups page.
4
Click the link to reload the Bridge Groups page.
The new bridge group appears in the list.
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Figure 2-5 Bridge Groups page (routed system only)
Repeat the above procedure to add another bridge group. When you’re
finished, make the necessary changes to the routing lists. See “Managing
Modifying a Bridge Group
1
2
3
To change the bridges in the group, select or clear their check boxes.
To change the name of the group, edit its Bridge Group Name entry.
Click the entry’s Commit button (blue “c”).
The system confirms that the bridge group has been modified and
provides a link back to the Bridge Groups page.
4
Click the link to reload the Bridge Groups page.
Repeat the above procedure to modify another bridge group. When you’re
finished, make the necessary changes to the routing lists.
Deleting a Bridge Group
1
Click the bridge group’s Delete button.
The system confirms that the bridge group has been deleted and provides
a link back to the Bridge Groups page.
2
Click the link to reload the Bridge Groups page.
Repeat the above procedure to delete another bridge group. When you’re
finished, make the necessary changes to the routing lists.
When you delete a bridge group, the system doesn’t make its ID available for
reuse or renumber remaining groups. If you have four bridge groups and delete
the one whose ID is 3, the remaining bridge groups are 1, 2, and 4. The next
bridge group you add is assigned ID number 5. ID number 3 remains unused.
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Managing Routing Lists
This section applies only to routed systems (PSTN or IP). In a routed system,
routing lists control how calls are routed to the bridges in your system. Each
routing list contains one or more bridge groups, listed in order of preference,
to which calls can be routed.
By default, your system contains the routing list Any Bridge, which contains
the default bridge group All Bridges. You can’t modify or delete this routing
list (but need not use it). If you have a routed system, you can add other
routing lists and modify or delete them.
Adding a Routing List
1
In the System Administration navigation bar, click Routing Lists.
starting with the default entry, Any Bridge. A blank input field and Add
button (green “+”) let you add routing lists.
2
Enter a name for the new routing list in the blank Routing List Name
field and click the Add button.
The system confirms that the routing list has been added and provides a
link back to the Routing Lists page.
3
4
Click the link to reload the Routing Lists page.
The new routing list appears in the list. Initially, all new routing lists
contain a single bridge group, the default All Bridges group.
Modify the routing list to include the bridge groups you want in the
order you want them searched (see next procedure).
Repeat the above procedure to add another routing list.
Figure 2-6 Routing Lists page (routed system only)
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Modifying a Routing List
1
On the Routing Lists page, find the entry you want to modify and click
its Edit Routing List button.
included in this routing list in order of priority. If you just created this
routing list, it contains only one bridge group entry, All Bridges, with its
Search Order field set to 1.
2
To change an existing entry, select a different bridge group from the
entry’s drop-down list. Then click its Commit button (blue “c”).
The system confirms that the routing list has been modified and provides
a link back to the Edit Routing List page.
No two entries can have the same priority (Search Order value). To
change the priority order of the bridge groups, change the Bridge Group
associated with each Search Order value, not vice versa. To avoid
confusion, it’s best to keep the Search Order values of the entries in the list
in sequential order.
3
4
Click the link to reload the Edit Routing List page.
To add another entry to the list, in the blank row at the bottom, select the
bridge group you want, enter the next priority number in the Search
Order field, and click the Add button (green “+”).
The system confirms that the bridge group has been added to the routing
list and provides a link back to the Edit Routing List page.
5
6
Click the link to reload the Edit Routing List page.
The new bridge group appears in the list.
To delete an entry from the list, click its Delete (red “–”) button.
The system confirms that the bridge group has been removed from the
routing list and provides a link back to the Edit Routing List page.
Figure 2-7 Edit Routing List page (routed system only)
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7
Click the link to reload the Edit Routing List page.
The bridge group entry you deleted is gone from the list.
The system doesn’t update the Search Order numbers of remaining
entries. If there were three bridge groups with priority numbers 1, 2, and
3, and you delete the second, the two remaining entries are numbered 1
and 3. Change the entry numbered 3 to 2 in order to maintain sequential
numbering and avoid possible confusion in the future.
Repeat the above procedure to modify another routing list.
Renaming a Routing List
1
On the Routing Lists page, find the entry you want to rename, edit its
Routing List Name field, and click its Commit button (blue “c”).
The system confirms that the routing list has been modified and provides
a link back to the Routing Lists page.
2
Click the link to reload the Routing Lists page.
The renamed routing list entry appears in the list.
Repeat the above procedure to rename another routing list.
Deleting a Routing List
1
On the Routing Lists page, find the entry you want to delete and click its
Delete button (red “–”).
The system confirms that the routing list has been deleted and provides a
link back to the Routing Lists page.
2
Click the link to reload the Routing Lists page.
The routing list entry you deleted is gone from the list.
Repeat the above procedure to delete another routing list.
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Working with Number Groups
Number groups are logical groupings of telephone numbers used to control the
assignment of access phone numbers to subscribers. In a routed system, each
number group is associated with a routing list that controls the routing of calls
Each number group can have one or more subscriber groups associated with it.
Subscriber groups are logical groupings of related subscribers who share the
same customized user interface (for instance, a specific prompt set). When
someone signs up as a new subscriber, the ReadiVoice system automatically
assigns the subscriber an access phone number (or numbers) from the number
group associated with the subscriber group to which the new subscriber
belongs.
You can use a number group to reserve a specific access phone number (or
numbers) for a specific subscriber group (or groups). For instance, you may
have a subscriber group specifically for Acme Widgets Co. users. If you want
all Acme Widgets conferences to use the same access phone number,
888-555-1234, you can:
1
2
Create a number group called Acme Widgets.
Define the access phone number 888-555-1234 as a shared number
assigned to the Acme Widgets number group.
3
Assign the Acme Widgets number group to the Acme Widgets subscriber
group.
All access phone numbers must be assigned to a number group. One number
group, Default, exists automatically. If you don’t want to control which access
numbers are used by whom, you can assign all numbers to the Default number
group and use it for all subscriber groups.
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Adding a Number Group
1
In the System Administration navigation bar, click Number Groups.
number groups. Below that, a blank input field and Add button (green
“+”) let you add number groups to the list.
2
In the blank Name field, enter a name for the number group you want to
create. The name may be up to 30 characters long and may contain spaces
and punctuation.
Figure 2-8 Number Groups page (routed system)
3
4
If this is a routed system, choose the routing list to be used for this
number group.
Click the Add button.
The system confirms that the number group has been added and provides
a link back to the Number Groups page.
5
6
Click the link to reload the Number Groups page.
The new group appears in the list.
Add one or more access phone numbers to the number group. See
Repeat the above procedure to create additional number groups. To use a new
number group (and the access phone numbers in it), you must assign the
number group to one or more subscriber groups. See “Working with
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Modifying a Number Group
1
In the navigation bar, click Number Groups.
The Number Groups page appears, listing the available number groups.
2
3
To rename a number group, edit its name.
To change its routing list assignment, select the one you want in the
Routing List field.
4
Click the Commit button (blue “c”).
The system confirms that the number group has been modified and
provides a link back to the Number Groups page.
5
Click the link to reload the Number Groups page.
Repeat the above procedure to modify another number group.
Deleting a Number Group
You can delete a number group only if it has no associated subscriber group(s)
and no associated access phone numbers.
To delete a number group:
1
In the navigation bar, click Number Groups.
The Number Groups page appears, listing the available number groups.
2
Click the Delete button (red “–”) by the number group you want to
remove.
The system confirms that the number group has been deleted.
When you delete a number group, the system doesn’t make its ID
available for reuse or renumber remaining groups. If you have four
number groups and delete the one whose ID is 3, the remaining groups are
1, 2, and 4. The next number group you add is assigned ID number 5. ID
number 3 remains unused.
Repeat the above procedure to delete another number group.
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Working with Access Classes
Access classes are logical groupings of access phone numbers you can use to
group numbers by carrier, kind of number (such as toll-free), or other criteria.
You can create as many classes as you want. A subscriber can have one access
phone number from each access class, so multiple classes let you offer
subscribers multiple access numbers.
All access phone numbers belong to an access class. One class, LOCAL, exists
automatically. You can rename it if you want. If you don’t want to categorize
access numbers by carrier or other criteria and you don’t want to offer multiple
access numbers to subscribers, you can assign all numbers to one class.
Adding an Access Class
1
In the System Administration navigation bar, click Access Classes.
access classes. Below that, a blank input field and Add button (green “+”)
let you add access classes to the list.
2
3
In the blank Name field, enter a name for the access class. It may be up to
30 characters long and may contain spaces and punctuation.
Click the Add button.
The system confirms that the access class has been added and provides a
link back to the Access Classes page.
4
Click the link to reload the Access Classes page.
The new access class appears in the list.
Repeat the above procedure to create more access classes. Then, assign
access phone numbers to the classes. See “Working with Access Phone
Figure 2-9 Access Classes page
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Renaming an Access Class
1
In the navigation bar, click Access Classes.
The Access Classes page appears, listing the available access classes.
2
3
Find the class you want to rename and type the new name into its New
Name field.
Click its Commit button (blue “c”).
The system confirms that the access class has been modified and provides
a link back to the Access Classes page.
4
Click the link to reload the Access Classes page.
The new class appears in the list.
Repeat the above procedure to rename another access class.
Deleting an Access Class
You can’t delete an access class if it contains access phone numbers. First,
reassign those numbers to another class. You can do this only if the access
numbers have no subscribers associated with them. See “Changing an Access
To delete an access class:
1
In the navigation bar, click Access Classes.
The Access Classes page appears, listing the available access classes.
Click the Delete button (red “–”) by the access class you want to remove.
2
If the class has access numbers in it, the system displays an error message.
Otherwise, it confirms that the access class has been deleted and provides
a link back to the Access Classes page.
3
Click the link to reload the Access Classes page.
When you delete an access class, the system doesn’t make its ID available
for reuse or renumber the remaining classes. If you have four access
classes and delete the one whose ID is 3, the remaining classes are 1, 2, and
4. The next access class you add is assigned ID number 5. ID number 3
remains unused.
Repeat the above procedure to delete another access class.
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Administering the ReadiVoice System
Working with Access Phone Numbers
Access phone numbers are the telephone numbers (or, for IP systems, SIP
addresses) that customers dial to reach your ReadiVoice system. They’re often
toll-free numbers, and your carrier usually tells you which numbers to use. In
the ReadiVoice system, access numbers have several characteristics:
Access Type. A routed system can have two access types:
Private — A private number is assigned to only one subscriber. Since the
access phone number uniquely identifies a subscriber’s conference, no
access code is needed.
Shared — A shared number is assigned to multiple subscribers. Each
subscriber receives a unique access code that identifies the subscriber’s
conferences. Callers must enter the correct access code to join a conference.
For a non-routed PSTN system, all access phone numbers must be either
private or shared; you can’t mix the two. For an IP system, access numbers
are shared.
Number Group. Each access phone number belongs to one number group,
which can be associated with one or more subscriber groups. The access phone
number is available in subscriber groups that use its number group.
Access Class. Each access phone number belongs to an access class. Multiple
classes let you offer subscribers multiple access numbers, one from each class.
You can create classes for specific kinds of numbers, such as local or toll-free.
on page 118). If the network maps the numbers that callers dial to different
numbers for routing, the system must link the routing number (hidden
number) with the dialed number.
You can:
•
•
•
•
Add access phone numbers.
In a non-routed system, move access numbers from one bridge to another.
Globally reassign subscribers from one access number to another.
Change the access class of an access phone number (if no subscribers are
using it).
•
Change the hidden number associated with an access phone number (if
hidden number mapping is enabled).
•
Delete an access phone number (if no subscribers are using it).
You can’t change the access type or number group of an existing access
number. To change a number from private to shared or switch it to a different
number group, delete the number and then add it back.
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Adding Access Phone Numbers
To put the new access phone numbers into a new number group or access
class, create the number group or access class first. See “Adding a Number
To add an access phone number:
1
In the System Administration navigation bar, click Access Numbers.
is at the top. Below it is the Add panel.
2
3
In the Add panel, enter the access number you want to add. Enter only
the digits, with no dashes or spaces.
If your system uses hidden numbers, enter the new number’s hidden
number. Enter only the digits, with no dashes or spaces.
4
5
Set the new access number’s Number Group and Access Class fields.
Click the Add button (green “+”).
The system confirms that the access phone number has been added and
provides a link back to the Access Phone Numbers page.
6
Click the link to reload the Access Phone Numbers page.
Repeat the above procedure to add another access phone number.
Figure 2-10 Access Phone Numbers page
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Moving Fixed Access Numbers to Another Bridge
In non-routed systems, access numbers are assigned to specific bridges. If
necessary, you can move access numbers from one bridge to another. For
example, you may want to move some access numbers from an overloaded
bridge to another bridge that’s less loaded.
To move access numbers to a different bridge:
1
In the navigation bar, click Access Numbers.
2
In the Search panel, enter criteria to retrieve the access numbers you want
to move. Then click the Search button.
You can use the asterisk (*) as a wild card. For instance, enter 80033311*
to retrieve all numbers from 800-333-1100 to 800-333-1199.
The access numbers that match your search criteria appear at the bottom
3
4
Carefully check the list of access numbers retrieved. If the list contains the
numbers you want to move and only those numbers, continue. Otherwise,
set of numbers.
In the Move panel, select the bridge to which you want to move the
access numbers. Then click the Move button.
The system prompts you to confirm the move and reminds you to make
the telephone network changes needed to implement the move.
5
6
Click OK.
The system confirms that the access numbers have been moved and
provides a link back to the Access Phone Numbers page.
Click the link to reload the Access Phone Numbers page.
Repeat the above procedure to move additional blocks of access numbers.
When you’re finished changing bridge assignments, make the necessary
changes to switches or other network elements in order to correctly route calls
dialed to these access numbers.
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Figure 2-11 Access Phone Numbers page after search
Reassigning an Access Phone Number’s Subscribers
You can globally reassign subscribers from one access number to another. For
instance, if several subscribers are assigned to the shared access number
888-555-1000, and you need to reassign them to 888-555-2000, you can make
this change all at once without having to re-provision each individual
subscriber.
You can:
•
Reassign subscribers to a different existing access phone number. The
number must be in the same number group and access class.
•
Reassign subscribers to a new access phone number, adding it to the
ReadiVoice system in the process.
You must, of course, notify the reassigned subscribers of the change. If you
add a new access phone number to the system, you must also ensure that the
telephone network routes that number to the system properly.
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To reassign subscribers to a different access phone number:
1
In the navigation bar, click Access Numbers.
2
In the Search panel, enter criteria to retrieve the access number whose
subscribers you want to reassign. Then click the Search button.
The access numbers that match your search criteria appear at the bottom
3
4
In the Numbers panel, find the access number whose subscribers you
want to reassign and click its Change button (blue “c”).
In the New Access Phone Number field, enter the number you want to
assign to the subscribers. Enter only the digits, with no dashes or spaces.
If this is an existing access number, it must be in the same number group
and access class as the access number it’s replacing.
5
6
If you’re creating a new access phone number for these subscribers and
your system uses hidden numbers, enter the access number’s hidden
number in the New Hidden Phone Number field.
Click Commit New Access Number.
If you entered a new number, the ReadiVoice system adds it to the
database, copying the access type, number group, and access class from
the existing number. It confirms that all the existing number’s subscribers
have been reassigned to the access phone number you entered and
provides a link back to the Access Phone Numbers page.
7
Click the link to reload the Access Phone Numbers page.
Repeat the above procedure to reassign another access phone number’s
subscribers.
Figure 2-12 Modify Existing Access Phone Number page
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Changing an Access Phone Number’s Access Class
You can’t change an access phone number’s access class if subscribers are
assigned to it. First, reassign the subscribers to another access number. See
To change an access number’s access class:
1
In the navigation bar, click Access Numbers.
2
In the Search panel, enter criteria to retrieve the access number whose
access class you want to change. Then click the Search button.
The access numbers that match your search criteria appear at the bottom
3
Confirm that the access number you want to change is listed and has no
subscribers assigned to it (Used is set to No).
If the Used field is set to Yes, click that word to see a list of the number’s
page 40 or, if there are only one or two, open and edit them manually.
4
5
Click the Change button (blue “c”) of the number you want to change.
In the Existing Access Class list, select the class to which you want to
assign this access number. Then click Commit New Access Class.
The system confirms that the access class has been changed and provides
a link back to the Access Phone Numbers page.
6
Click the link to reload the Access Phone Numbers page.
Repeat the above procedure to change another access phone number.
Figure 2-13 Checking an Access Phone Number’s subscribers
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Administering the ReadiVoice System
Changing Hidden Numbers
This section doesn’t apply to IP systems, which don’t use hidden numbers.
If your system uses hidden numbers, you can use the Modify Existing Access
Phone Number page to change the hidden number associated with an access
number. Do this only if the routing number associated with the access number
is changed in the network.
To change an access phone number’s hidden number:
1
In the navigation bar, click Access Numbers.
2
In the Search panel, enter criteria to retrieve the access number whose
hidden number you want to change. Then click the Search button.
The access numbers that match your search criteria appear at the bottom
3
4
Find the access number whose hidden number you want to change and
click its Change button (blue “c”).
In the New Hidden Phone Number field, enter the new hidden number
to associate with this access phone number. Enter only the digits, with no
dashes or spaces. The hidden number must be unique (that is, not already
assigned to another access phone number).
5
6
Click Commit New Hidden Number.
The system confirms that the hidden number has been changed and
provides a link back to the Access Phone Numbers page.
Click the link to reload the Access Phone Numbers page.
Repeat the above procedure to change another access phone number’s hidden
number.
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Deleting an Access Phone Number
Before you delete an access phone number, keep the following points in mind:
•
•
Access phone number changes require telephone network changes to
ensure that calls are routed properly.
You can’t delete an access phone number that has subscribers assigned to
it. First, reassign the subscribers to another access number. See
To delete an access phone number:
1
In the navigation bar, click Access Numbers.
2
In the Search panel, enter criteria to retrieve the access number that you
want to delete. Then click the Search button.
The access numbers that match your search criteria appear at the bottom
3
Confirm that the access number you want to delete is listed and has no
subscribers assigned to it (Used is set to No).
If the Used field is set to Yes, click that word to see a list of the number’s
page 40 or, if there are only one or two, open and edit them manually.
4
5
Click the Delete button (red “–”) of the number you want to delete.
The system confirms that the number has been deleted and provides a link
back to the Access Phone Numbers page.
Click the link to reload the Access Phone Numbers page.
Repeat the above procedure to delete another access phone number.
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Working with Multiple Providers
You can configure ReadiVoice for access by multiple application service
providers (ASPs). For instance, your company can enter into a relationship with
Acme Conferencing, allowing Acme to sell services on your ReadiVoice
system, using their own custom interfaces (such as the Moderator interface
and voice prompts). To enable such a relationship, you configure Acme as a
provider in your ReadiVoice system, permitting their users to connect to it.
Adding a Provider
1
In the navigation bar, click Application Providers.
providers already defined in the system.
Figure 2-14 Application Providers page
2
Click Add Provider.
Click Create Provider.
3
4
The system confirms that the provider has been added and provides a link
back to the Application Providers page.
5
Click the link to reload the Application Providers page.
Repeat the above procedure to add another provider.
Table 2-2
Provider Information fields
Field
Description
Provider Name
Enter a login name for the provider. The maximum length is 30
characters. Required.
Password
Enter a password for the provider. The maximum length is 30
characters. Required.
Expiration Date
Select the expiration month and date and enter a four-digit year.
Required.
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Table 2-2
Provider Information fields (continued)
Field
Description
Allowed IP
Enter a specific IP or an IP mask that defines valid IP addresses from
which this provider can connect. For instance, if you enter
192.168.56.255, all IP addresses in the 192.168.56 subnet are valid.
API Access
Define which APIs the provider can access by entering the sum of the
authorized APIs’ values. Required.
The API Values are:
1
Moderator API
Operator API
Call Router API
Provisioning API
CDR API
2
4
8
16
For instance, to authorize access to the Moderator and Operator APIs
only, enter 3. To authorize access to the Moderator, Operator, and
Provisioning APIs, enter 11.
Currently, only the Moderator API can be used.
Company Name
Contact Person
Phone
Enter the provider’s company name. The maximum length is 30
characters.
Enter the name of the person responsible for this account. The
maximum length is 30 characters.
Enter a phone number for the contact person. The maximum length is
30 characters.
E-Mail
Enter an email address for the contact person. The maximum length is
50 characters.
Note
Enter any additional information about the provider that you need. The
maximum length is 256 characters.
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Figure 2-15 Add Provider page
Editing a Provider Account
1
In the navigation bar, click Application Providers.
2
Find the provider that you want to edit and click its Edit button (purple
“e”).
Click Create Provider.
3
4
The system confirms that the provider has been modified and provides a
link back to the Application Providers page.
5
Click the link to reload the Application Providers page.
Repeat the above procedure to modify another provider.
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Deleting a Provider
You can’t delete a provider account to which subscribers are assigned.
To delete a provider account from the system:
1
2
3
In the navigation bar, click Application Providers.
Find the provider you want to delete and click its Delete button (red “–”).
A dialog box asks you to confirm.
Click OK.
The system confirms that the provider has been deleted and provides a
link back to the Application Providers page.
4
Click the link to reload the Application Providers page.
Repeat the above procedure to delete another provider.
Working with Subscriber Groups
Subscriber groups are logical groupings of related subscribers who share the
same user interface (for instance, a specific prompt set). All subscribers are in
a group, so you must have at least one. Typically, you create more subscriber
groups as needed. For instance, you may create a subscriber group specifically
for Acme Widgets Co. users.
During ReadiVoice installation, a default subscriber group is automatically
created. Initial configuration of the system includes defining the initial value
settings for the default subscriber group.
A subscriber group’s settings are used as initial values for its new subscribers.
Therefore, the settings you make for a group are the defaults that provisioners
see when they add subscribers to the group.
After working with the subscriber group settings, you may realize that your
company doesn’t need or want to use all the available fields when
provisioning subscribers or modifying their accounts. See “Setting Up the
When you create a new group, the system copies the settings from the existing
group of your choice to save you time.
Each subscriber group can have a group administrator, who is the internal user
authorized to modify the group and view its CDR and conference information.
You can’t directly create or delete group administrators as you do other
internal users. You can do so only as part of creating or editing the group. You
can, however, change the group administrator’s password directly on the
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The following sections describe how to create, edit, and delete a group. For
information about adding subscribers to a group, see the ReadiVoice
Provisioning Guide.
Adding a New Subscriber Group
Every subscriber group must be associated with a number group, which
controls the access numbers available to the subscriber group. If you want to
link the new subscriber group with its own number group, create the number
To add a new subscriber group:
1
In the System Administration navigation bar, click Subscriber Groups.
the subscriber groups that exist.
2
3
In the Add panel, select an existing group from which to copy the initial
values and enter a name for the new group. The maximum length is 50
characters.
Click the Add New Group button.
The system creates the new group, copying its initial values from the
4
starting at 3.
Repeat the above procedure to add another subscriber group.
Figure 2-16 Subscriber Group page
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Editing a Subscriber Group
1
In the navigation bar, click Subscriber Groups.
The Subscriber Group page appears.
2
Find the group that you want to edit and click its Edit Group button
(purple “e”).
3
descriptions.
Figure 2-17 Edit Group page (top portion)
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Table 2-3
Group Information fields
Fielda
Description
Group Name
Leave the current name or change it if you wish. The maximum length is
50 characters.
Group Type
Choose Personal or Corporate from the list. Personal is for billing each
participant individually. Corporate is for billing a company for the entire
conference. If you choose Corporate, you must complete the Company
Information and Billing Information sections.
Provider Name
If your system has multiple providers, select the provider for this
subscriber group from the list. Otherwise, leave the default value of
NONE.
Number Group
Select the number group from which access phone numbers for this
group are assigned.
Administrator Name
Change the login name of the group administrator (the initial value is the
group name). This is the person authorized to change group information
and view CDR (call detail record) and conference information for this
group. The maximum length is 30 characters.
Administrator
Password
Change the password for this group administrator (the initial value is the
group name). The maximum length is 30 characters.
Access Class
Place a check next to the access class or classes that should be
selected by default for new subscribers in this group. A class typically
defines a type of access number, such as toll or toll-free.
Operator Notes
Enter any message or operator script that you want displayed to
operators when they view this group’s conferences. The maximum
length is 484 characters.
Subscriber User
Field A
Optional field for additional subscriber group information, such as a key
to another database. Whether and how you use this field is entirely up to
your company. The ReadiVoice system doesn’t use it.
Subscriber User
Field B
Second optional field for additional subscriber group information.
a. These are the default labels. All field labels that also appear on the New Subscriber page can be
customized.
4
If Group Type is Corporate:
a
Enter or modify the data in the Company Information fields. See
Table 2-4 for descriptions. All fields are required.
b
Enter or modify the data in the Billing Information section. See
Table 2-5 for descriptions.
Figure 2-18 shows these two sections.
The ReadiVoice system doesn’t use any of the data in the Billing
Information section. These fields are provided for your company to use as
it wishes, and the descriptions here are just general guidelines.
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Figure 2-18 Company and Billing sections of Edit Group page
Table 2-4
Company Information fields
Field
Description
Company Name
Phone
Enter the name of the company. The maximum length is 50 characters.
Enter the company’s telephone number. The maximum length is 30
characters.
Fax
Enter the company’s fax number. The maximum length is 30 characters.
Contact Name
Enter the name of the person to contact at this company. The maximum
length is 50 characters.
Contact Phone
Enter the contact person’s telephone number. The maximum length is
30 characters.
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Table 2-5
Billing Information fields
Fielda
Description
Billing Type
Select Credit Card or Telephone Number to choose the billing method
for this group’s conferences.
Card Type
For credit card billing, select the card type from the list.
For credit card billing, enter the 16-digit card number.
Credit Card Number
Expiration Date
For credit card billing, select the expiration month and enter the
four-digit year.
Billing Phone
Enter the phone number to be billed or the phone number from the
Company Information section.
This field is required, even if Billing Type is not Telephone Number.
Billing Address
Enter the billing address.
Billing User Field A
Optional field for additional billing information, such as a key to another
database. Whether and how you use this field is entirely up to your
company.
Billing User Field B
Second optional field for additional billing information.
a. These are the default labels. All field labels that also appear on the New Subscriber page can be
customized.
5
Configure the prompt settings in the Call Flow and Voice Prompt section
Figure 2-19 Call Flow and Voice Prompt section of Edit Group page
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Table 2-6
Call Flow and Voice Prompt fields and settings
Field/Settinga
Description
Prompt for Menu
Specifies whether subscribers can access the account options menu
when they dial into the system and identify themselves.
On — Subscribers in this group are given the opportunity to change the
account options. The account options menu enables subscribers to
make changes that affect all future conferences (including the one that a
subscriber is in the process of starting).
Off — Subscribers are not given the opportunity to access the account
options menu (but can still modify account options through the
Moderator interface).
Prompt for Subscriber
Indicates whether the call flow includes a prompt for the subscriber to
enter his or her subscriber password.
On — Callers hear “If you are the subscriber, press star” (this is the
default wording). Use this setting unless all subscribers in this group
must have a two-password (shared or private) call flow.
Off — Callers hear “At any time during this message, please enter your
passcode followed by the pound key.” Use this setting if only
two-password call flows are possible.
If this is set to Off for a subscriber account with a traditional (one
password) call flow, the subscriber has no opportunity to identify him or
herself when dialing in and must use the Moderator to start conferences.
Prompt for Security
Specifies the setting for the conference security code prompt.
No Prompt — The conference security feature isn’t available.
Optional Prompt — Subscribers are given the option of setting up a
conference security code, which participants must know in order to join
the conference.
Mandatory Prompt — Subscribers are required to set up a conference
security passcode.
Prompt Set
Specifies the voice prompt set for this subscriber group’s conferences.
a. These are the default labels. All field labels that also appear on the New Subscriber page can be customized.
6
Options and Roll Call Options. Each of the features has two settings:
Setting — Determines the feature’s default setting.
Subscriber Configurable — Determines whether subscribers can change
their account’s default setting later, using either the account options call
flow menu (if enabled) or the Moderator.
settings are the initial settings for subscribers in the group, but
provisioners can modify them for specific subscribers.
7
Continue with the Conference Options section by setting the
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8
If your system has any Application Control Mode (ACM) features
installed, place a check mark next to those that you wish to enable for this
subscriber group.
ACM applications must be both installed and activated to appear here.
Figure 2-20 Conference Options section of Edit Group page
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Table 2-7
Conference Options
Fielda
Description
Setting:
Auto Continuation
On — By default, conferences begin with continuation turned on (which
means that the conference continues until the last participant
disconnects).
Off — By default, conferences begin with continuation turned off (which
means that the conference ends when the subscriber disconnects).
Subscriber Configurable:
Yes — Subscribers can change the On/Off setting before a conference
begins through the account options menu (if available) or the Moderator
interface.
No — Subscribers can’t change the On/Off setting.
can override the default continuation behavior for a specific conference.
governs all of a subscriber’s conferences. See “How Continuation
Quick Start
Setting:
On — By default, conferences start as soon as the first participant dials
in. Participants don’t wait on hold for the subscriber to arrive.
Off — By default, conferences start when the subscriber arrives.
Participants who dial in before the subscriber wait on music hold until
the subscriber arrives.
Subscriber Configurable:
Yes — Subscribers can change the On/Off setting before a conference
begins through the account options menu (if available) or the Moderator
interface.
No — Subscribers can’t change the On/Off setting.
Quick start conferences are considerably less secure than conferences
that don’t start until the subscriber enters the subscriber password.
Listen Only Entry
Setting:
On — Conferences start in listen only mode. The subscriber is the only
conference participant who can speak to the conference; all other
participant lines are muted, and participants can’t unmute themselves.
Off — Conferences don’t start in listen only mode. All participants
(except those who are muted) can be heard.
Subscriber Configurable:
Yes — Subscribers can change the On/Off setting for the account,
affecting all future conferences.
No — Subscribers can’t change the On/Off setting. But, they can still
use the Moderator interface or touchtone commands to turn listen only
mode on or off in the current conference.
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Table 2-7
Conference Options (continued)
Fielda
Description
Waiting Room
Setting:
Off — The waiting room isn’t available. If a subscriber locks a
conference, callers after that are told that the conference is locked and
are disconnected.
On — The waiting room is available. If a subscriber locks a conference,
callers after that must wait on music hold to be admitted. The subscriber
is notified when someone wants to enter the conference and can use
touchtone commands or the Moderator interface to speak with, admit, or
disconnect the waiting caller(s).
On w/Waiting Room on Entry — Conferences start locked with the
waiting room available. All callers must wait to be admitted by the
subscriber.
Subscriber Configurable:
Yes — Subscribers can change the default setting for their account,
affecting all future conferences.
No — Subscribers can’t change the default. If a subscriber locks the
current conference, the waiting room either is or isn’t available,
depending on the setting for this feature.
Password Enable
Name Record
Determines whether subscribers can change their subscriber password
using the touchtone interface.
Setting:
On — The ReadiVoice system prompts participants to record their
names before placing them into conference. Conferees can request a
“roll call” of participants’ names during the conference. This feature must
be turned on if Entry/Exit Announcement is set to Name.
Off — The ReadiVoice system doesn’t record participants’ names. The
Entry/Exit Announcement setting can’t be Name.
Subscriber Configurable:
Yes — Subscribers can change the On/Off setting for the account,
affecting all future conferences.
No — Subscribers can’t change the On/Off setting.
Entry/Exit
Setting:
Announcement
Specifies whether the system signals the entry and exit of a participant
by playing a tone, the participant’s recorded name, or silence. If you
select Name, you must also turn on the Name Record feature.
Subscriber Configurable:
Yes — Subscribers can change the default setting for the account,
affecting all future conferences. If the announcement setting is Name,
and if the subscriber can turn off Name Record, then the subscriber
should be able to change the announcement setting, too.
No — Subscribers can’t change the default.
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Table 2-7
Conference Options (continued)
Fielda
Description
Conference
Continuation
Determines whether subscribers can override their accounts’ default
continuation behavior (specified by the Auto Continuation settings) for
a specific conference during that conference (using either a touchtone
command or the Moderator interface).
Yes — Subscribers can turn continuation on or off during their
setting for the account; only the current conference is affected.
No — Subscribers can’t override the default continuation behavior
during their conferences. Once a conference has started, the account’s
Auto Continuation setting controls the continuation behavior.
subscriber configurable, then the subscriber can neither change the
Dial Out Permission
Dial Out Prefix
Determines whether a subscriber can dial out from his or her
conferences.
Appears only if your system is configured for dial-out billing (see
calls a subscriber makes. This field and Dial Out Postfix together can’t
exceed 24 characters.
Dial Out Postfix
Appears only if your system is configured for dial-out billing. Enables
billing for dial-out calls the subscriber makes. This field and Dial Out
Prefix together can’t exceed 24 characters.
Recorder Dial Out
Determines whether a subscriber can choose to record a conference.
Conference recording is available only if your system is configured
Services representative about conference recording.
VIP Conference
On — Subscribers in this group have VIP Conference turned on by
default. VIP Conference subscribers are guaranteed access to the
number of ports specified in their subscriptions.
Polycom recommends selecting Off for this option. It’s best to grant VIP
status only to individual subscribers and only in limited numbers. VIP
subscribers have exclusive use of their ports at all times, permanently
removing those ports from the pool of available ports for non-VIP
conferences. Overuse of the VIP Conference option has a negative
impact on your system’s port utilization efficiency.
Off — Subscribers in this group have VIP Conference turned off by
default. This is the normal setting. Non-VIP subscribers’ ports come
from the general pool, and the full number specified in their
subscriptions is not guaranteed to be available at all times.
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Table 2-7
Conference Options (continued)
Fielda
Description
Conference Entry with
Count
On — Upon entering the conference, participants hear a private
message telling them how many people, including themselves, are now
in the conference.
Off — Participants hear a message stating only that they’re being
placed into conference.
Operator Request
Available
Specifies who may make operator requests from a subscriber’s
conferences: the subscriber only, the subscriber and participants, or
neither.
This setting doesn’t affect automatic operator requests resulting from a
caller’s failure to enter the correct password or conference security
code.
Conference
Termination Option
Specifies how the ReadiVoice system handles conferences that have
the minimum number of participants:
Off — The ReadiVoice system never terminates conferences.
Time, below), the conference is referred to an operator for termination.
If no operator is available, the system plays a message to the
conference asking the participants to press a touchtone key to continue.
If there’s no response, the system terminates the conference.
On – w/ Oper Override — The system plays a message to the
conference asking the participants to press a touchtone key to continue.
If there’s no response, the system terminates the conference.
You can set the minimum number of participants for your system to one,
Conference
Termination Time
Specifies the time interval (in minutes) that governs the automatic
termination of conferences that have the minimum number of
Off.
ACM Options
Application Control Mode (ACM) enables an external application to
interact with callers or conference participants. The options or settings
that appear here, if any, depend on which ACM features and external
applications are installed in your system.
ACM and some sample ACM applications to demonstrate its use are
included in the ReadiVoice Software Development Kit (SDK).
a. These are the default labels. All field labels that also appear on the New Subscriber page can be customized
via the Quick Provisioning Setup page described in “Setting Up the Provisioning Interface” on page 60.
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9
Click Commit Group.
The system either confirms that the group has been updated or tells you
which information is missing.
10 If the update failed, note what’s missing. Then click the link to return to
the Edit Group page, make the needed corrections, and click Commit
Group again.
Repeat the above procedure to edit another subscriber group.
How Continuation Settings Interact
interact to determine an account’s continuation behavior (whether a
conference continues after the subscriber leaves) and what control the
subscriber has over that behavior.
Table 2-8
Continuation settings at a glance
Auto Continuation
Setting
Subscriber
Configurable
Conference
Continuation
Effect
Yes
Continuation on. Subscriber can turn it off for the
current conference or for all future conferences.
Yes
Continuation on. Subscriber can turn it off for all future
conferences, but not for the conference currently under
way.
No
On
Continuation on. Subscriber can turn it off for the
conference currently under way, but not for all future
conferences.
Yes
No
No
Yes
No
Continuation on. Subscriber can’t turn it off at all. All of
the subscriber’s conferences are set to continue until
the last participant disconnects.
Continuation off. Subscriber can turn it on for the
conference currently under way or for all future
conferences.
Yes
No
Continuation off. Subscriber can turn it on for all future
conferences, but not for the conference currently under
way.
Off
Continuation off. Subscriber can turn it on for the
conference currently under way, but not for all future
conferences.
Yes
No
Continuation off. Subscriber can’t turn it on at all. All of
the subscriber’s conferences are set to end when the
subscriber disconnects.
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If they may do so, subscribers can change the continuation behavior of all
future conferences in two ways:
•
By accessing the account options menu when dialing into the system
(available if Prompt for Menu is turned on).
•
By logging into the Moderator and accessing its account options settings.
If they may do so, subscribers can change the continuation behavior of the
conference currently under way by entering the correct touchtone command
(the default is *8).
Deleting a Subscriber Group
You can’t delete a group that has active subscribers. First, you must change the
subscribers to a different group. For instructions, see the ReadiVoice
Provisioning Guide. You can’t delete the last remaining subscriber group.
Be sure you want to permanently remove all information about a group before you
delete it.
Caution!
To delete a subscriber group:
1
2
3
In the navigation bar, click Subscriber Groups.
Find the group you want to delete and click its Delete button (red “–”).
A dialog box asks you to confirm deleting the group.
Click OK.
The system confirms that the group has been deleted.
Repeat the above procedure to delete another subscriber group.
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Setting Up the Provisioning Interface
After working with the subscriber group settings, you may realize that your
company doesn’t need or want to use all the available fields when
provisioning subscribers or modifying their accounts.
You can customize the Provisioning interface so that provisioners see only the
fields you want them to use. You can also change the field names to match
your company’s usage.
decide what appears on the Add Subscriber and Edit Subscriber pages. You
can:
•
•
Hide fields and settings that your company doesn’t need.
A few fields are required by the system. They don’t have check boxes by
them, so you can’t turn them off (but you can rename them). You probably
want to require at least some subscriber and billing information.
Other fields may be necessary because of your system configuration (such
as Subscriber External ID) or to implement certain features (such as Dial
Out Prefix and Dial Out Postfix).
Change the names of fields and settings.
When renaming fields and settings, keep in mind that space is limited.
Check your changes to make sure they display properly.
If you rename fields and settings that apply to subscriber groups, those
changes also appear on the Edit Group page.
To customize what appears in the Provisioning interface:
1
In the System Administration navigation bar, click Quick Provisioning
Setup.
The Quick Provisioning Setup page appears.
2
To disable a field or setting (removing it from the Provisioning interface),
click its check box to clear it. To re-enable a field or setting, click its check
box again to select it.
Some ReadiVoice system configurations may require that certain fields and settings
be available in Provisioning even though the Quick Provisioning Setup page allows
you to leave them out. Be sure you know which fields are needed for your system
configuration.
Caution!
3
To change how a field or setting is labeled in the Provisioning interface,
edit the name shown in its text box.
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4
5
When you’re finished making changes, click the Update Quick
Provisioning Settings button.
The system confirms the provisioning setup changes and provides a link
back to the System Administration page.
Test the changes you’ve made:
— Go to the Provisioning interface and add a test subscriber. Verify that
the Add Subscriber page looks the way it should.
— Start a conference as the test subscriber and verify that all the features
you want to support are available and operational.
Repeat this procedure if you need to make additional changes.
Figure 2-21 Quick Provisioning Setup page (top)
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Managing System Access
The ReadiVoice system supports several kinds of users with different roles
and responsibilities. To give internal users (not subscribers) access to the
system, you create entries for them on the Passwords page, assigning each a
user name and password.
The user type (operator, provisioner, or administrator) controls which
functions a user may access. A group provisioner is assigned to a single
subscriber group and can add, change, or delete only subscribers within that
group. A group operator is assigned to one or more subscriber groups and can
monitor conferences and answer requests only from conferences within those
subscriber groups.
Similarly, a group administrator can edit only a specific subscriber group and
view only the CDRs from that group.
You can’t add or delete group administrators directly. Instead, you do so when
creating or editing the subscriber group. See “Working with Subscriber
password using the procedure described here.
The sections that follow describe how to add and delete internal users and
change their passwords.
Adding an Internal User
1
In the System Administration navigation bar, click Passwords.
the ReadiVoice system, showing the password, user type, and, if
applicable, group for each. A blank row at the end of the list allows you to
add a new user.
This page shows actual passwords, not asterisks, for all ReadiVoice users. Don’t
display it unless the monitor is secure from unauthorized personnel. Don’t walk
away from the monitor with this page displayed.
Caution!
2
3
In the User Name field of the blank row, enter the login name for the new
user. The maximum length is 30 characters.
In the Password field, enter the new user’s password. The maximum
length is 20 characters.
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Figure 2-23 Passwords page
4
5
In the User Type list, select the user type.
The user types are operator, provisioner, group operator, group
provisioner, and system administrator.
If the new user is a group provisioner or group operator, select from the
Group list the subscriber group to which this user has access.
For a group operator, you can add or change subscriber groups later (see
which the provisioner has access and it can’t be changed.
6
Click the Add button.
The system confirms that the user has been added and provides a link for
returning to the Passwords page.
7
8
Click the link to reload the Passwords page.
The new entry appears at the end of the list, above the blank row.
If the new user is a group operator and you want to add more subscriber
groups, click the new entry’s Register Groups link.
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Figure 2-24 Operator Registered Subscriber Groups page
9
Use the Add and Remove buttons to move selected groups into or out of
this operator’s Registered Subscriber Groups list.
10 When you’re finished setting up this group operator, click Passwords in
the navigation bar to return to the Passwords page.
Repeat the above procedure to add another user.
Changing an Internal User’s Password
1
In the navigation bar, click Passwords.
the system, showing the password, user type, and, if applicable, group for
each. Only the password can be modified.
To change a user’s login name, user type, or group, delete the existing
entry and add a new one. The Passwords page also lists the Telnet login
for the system. You can change the password for this login, but can’t delete
it.
2
3
Locate the user whose password you want to change.
Replace or edit the existing password and click the Commit button (blue
“c”).
The system confirms that the password has been modified and provides a
link for returning to the Passwords page.
4
Click the link to reload the Passwords page.
The user you modified has a new password.
Repeat the above procedure to change another password.
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Deleting an Internal User
1
In the navigation bar, click Passwords.
2
Locate the entry for the user you want to remove and click its Delete
button (red “–”).
The system confirms that the user has been deleted and provides a link for
returning to the Passwords page.
3
Click the link to reload the Passwords page.
The entry you deleted is gone from the list.
Repeat the above procedure to delete another user.
Defining Invalid Subscriber Passwords
As a security feature, the ReadiVoice system lets you specify certain number
combinations as not valid subscriber passwords. For instance, you can
disallow combinations such as 1111 and 1234.
The ReadiVoice system stores the invalid combinations in the invalid
subscriber password table. You can add entries to and delete entries from this
table. Adding entries has no effect on subscribers currently using the entries.
It only prevents these entries from being chosen in the future.
Adding Entries to the List of Invalid Subscriber Passwords
1
In the System Administration navigation bar, click Invalid Passwords.
2
3
In the Enter a Password to Add to the List field, type the number you
want to invalidate as a subscriber password.
Click the Add button (green “+”).
The system confirms that the entry has been added and provides a link for
returning to the Invalid Subscriber Password Table page.
4
Click the link to reload the Invalid Subscriber Password Table page.
The entry you added appears in the list.
Repeat the above procedure to add another invalid password entry.
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Deleting Entries from the List of Invalid Subscriber Passwords
1
In the navigation bar, click Invalid Subscriber Passwords.
2
3
Scroll through the list of invalid subscriber passwords and select the
number you want to delete.
Click the Delete button (red “–”).
The system confirms that the entry has been deleted and provides a link
for returning to the Invalid Subscriber Password Table page.
4
Click the link to reload the Invalid Subscriber Password Table page.
The entry you deleted is gone from the list.
Repeat the above procedure to delete another entry.
Figure 2-25 Invalid Subscriber Password Table page
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3
Monitoring the ReadiVoice System
This chapter describes the ReadiVoice system monitoring functions that you
can access with your browser. You can view the overall status of your system,
usage levels, bridge and span status, operator data, and conference data.
Accessing the Monitoring Tools
You can access the ReadiVoice system monitoring tools from any computer
that can connect to the ReadiVoice system’s Web server and has a compatible
Web browser (Microsoft Internet Explorer, version 5.5 or later).
Some functions also require a Java plug-in (Java Runtime Engine, or JRE). If
you don’t have the necessary plug-in, your browser should ask you if you
want to download and install it. You must do so to use those functions.
Polycom has tested these Java applets with the Sun Microsystems Java
Runtime Engine (JRE) 1.3 on Microsoft Windows 2000.
These applets should work on other platforms and operating systems and with
newer versions of the JRE, but that depends on proper implementation by the
operating system and Java virtual machine. Polycom hasn’t tested other
possible combinations and can’t be responsible for implementation or
compatibility issues beyond our control.
Access to the monitoring tools is restricted to authorized users. You must
know the correct user name and password.
To access ReadiVoice system monitoring tools:
1
Point your browser to the ReadiVoice home page for your system (for
example, http://rvoice.com/index2.html).
bar is on the right.
2
Select the link for the information you want to see:
— Click Show System Status to see information about disk usage and
running processes on your CACS server.
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— Click Show Conference Status to see information about current
conferences.
— Click Show Conference Report to view call detail record (CDR) data
for completed conferences sorted by subscriber.
— Click Show CDR Data to view CDR data sorted by conference.
— Click Operator Statistics to view data taken from requests for
operator assistance, or to view statistics calculated from the data.
— Click SNMP Monitor to view system statistics in real time.
— Click Channel State Monitor to see real-time display of the lines in
use on each bridge (not available in an IP system).
— Click Show Critical Logs to check for critical events in the bridge log
page 145).
If you aren’t already logged in as a system administrator, the login dialog
box appears.
3
Enter a user name and password authorized for administrative access.
Then click OK.
Figure 3-1 ReadiVoice internal user home page
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Checking System Status
Status link in the navigation bar.
The System Disk Usage and Informix Disk Usage panels show disk usage
statistics for your CACS server’s Solaris partitions and Informix chunks. For
each partition or chunk, they show:
KB Size — The total size of the partition or chunk in kilobytes.
KB Left — The amount of free space in kilobytes.
Capacity — The percentage of the total size that is used.
The Process Status panel shows the status (running or not running) and CPU
time of the ReadiVoice server processes. The processes listed depend on
whether this is a PSTN or IP system and how it’s configured. See “Stopping
ReadiVoice behavior is unpredictable unless all server processes are running.
Caution!
Figure 3-2 System Status page
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Viewing Conferencing Information
The Conference Information links in the navigation bar let you look at:
•
•
Information about the conferences currently running.
Summary CDR data, with one entry per subscriber, for a time period you
choose (up to one month).
•
•
Summary CDR data, with one entry per conference.
Detailed CDR data for a specific conference.
Call detail records (CDRs) are used for billing and provide detailed
information about a conference, its participants, and all features used and
events associated with it. If you’re a system administrator, you can look at any
CDR data in the database; if you’re a group administrator or group
provisioner, you can look only at the CDR data for your subscriber group.
Current Conferences Information
The Conference Status page appears when you click the Show Conference
Status link in the navigation bar. It provides an overview of the current
conferencing activity on the system.
The Conference Status page shows:
•
•
At the top, summary information about all currently running conferences.
Below the summary table, status information for each running conference,
showing details about each line in the conference.
Figure 3-3 Conference Status page
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CDR Data by Subscriber
When you click the Show Conference Report link in the navigation bar, the
CDR Data page displays CDR data by subscriber. Each entry in the
Conference Report panel shows the aggregate conferencing data for one
subscriber.
When the page first appears, it shows the subscriber data for the current date.
To see subscriber data for a different date or range of dates, change Begin
Date, End Date, or both, and click Search.
For each subscriber listed, the report shows:
•
•
•
•
Subscriber ID and name.
Total number of port minutes.
Number of dial-outs and dial-out minutes.
Number of times each billable feature was used.
To see more detail for a subscriber, click that subscriber’s ID in the list. The
system displays summary CDR data for the subscriber’s conferences, with one
entry per conference. This is the same display that you get by clicking Show
CDR Data and then searching for that subscriber ID and date range. See the
next section.
Figure 3-4 Viewing CDR data by subscriber
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CDR Data by Conference
When you click the Show CDR Data link in the navigation bar, the CDR Data
various criteria, such as subscriber ID or name, and set a date range.
When you click Search, the system retrieves data for the conferences within
your date range that match your search criteria. Each entry in the Conference
of a search for a specific subscriber ID.
If the date range includes today, the list may include conferences that are
currently running or whose CDRs haven’t been fully processed yet; these
records will be incomplete.
For each listed conference, the Conference Report panel shows:
•
•
•
•
•
Conference and subscriber IDs.
Bridge ID and card number on which conference ran.
Subscriber’s name.
Start time and duration (or Incomplete if the conference is still running).
Number of participants (or N/A if the conference is still running).
To see details for a specific conference, click its conference ID. The CDR Data
page displays detailed CDR data for the conference. See the next section.
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Detailed CDR Data for One Conference
While viewing CDR data by conference, click a conference ID for more
information about that conference. The CDR Data page displays detailed CDR
contains information about the conference and its participants. The
Participant Features panel contains records of billable feature usage. For
If the conference is still running or its CDRs haven’t been fully processed yet,
these records will be incomplete.
If you have a non-routed system using private (not shared) access numbers,
and you’re using hidden numbers, then the Conference Report panel shows
the hidden numbers instead of the DNIS numbers.
Figure 3-6 Viewing detailed CDR data for a conference
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Viewing Operator Information
The Operator Statistics page appears when you click the Operator Statistics
link in the navigation bar of the ReadiVoice internal user home page. From
here, you can:
•
•
Click View Requests in the navigation bar to see records of operator
requests (for all operators, a specific operator, or a specific subscriber).
Click View Statistics in the navigation bar to see operator request
statistics (for all operators, a specific operator, or a specific subscriber).
Operator request data is available only if operator data logging is enabled on
file each night at midnight containing all the operator request records for the
preceding day.
The Operator Statistics pages let you retrieve and view data from these log
files The operator logs are deleted after 31 days, so you can see operator
request data for up to a month.
Viewing Operator Request Records
1
2
In the Operator Statistics page navigation bar, click View Requests.
options in the navigation bar (For All Operators, For Specific Operator,
and For Specific Subscriber).
In the navigation bar, click the link that corresponds with the scope of the
records you want to retrieve.
The Requests Search page displays search fields that vary slightly
appear when you choose For Specific Subscriber.
3
4
If you chose For Specific Operator or For Specific Subscriber, select the
operator from the list or enter the subscriber’s ID (if you don’t know the
subscriber’s ID, click the Search link to its right to search for the
subscriber).
If you chose For All Operators, select the number of records you want to
see per page.
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Figure 3-7 The Operator Statistics Requests page
Figure 3-8 Searching for operator requests by subscriber
5
6
Set the Begin Date and End Date fields by doing one of the following:
— Click in the field and edit the date.
— Click the field’s calendar icon and click a date in the calendar that
appears.
Regardless of which dates you choose, records are only available for the
past 31 days and aren’t available for the current day (the system collects
the day’s data at midnight).
Click the Search button.
The ReadiVoice system displays the answered requests matching your
search criteria. For each request, it shows:
— Type of operator request (private or conference).
— Date and time the request was made.
— Operator ID of the operator who answered the request.
— How long it took an operator to answer the request.
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— How long the operator spent on the request (talking with the
subscriber or conference).
— Subscriber ID for the conference in question. Click the ID to retrieve a
conference report for that subscriber.
— Conference ID of the conference. Click the ID to retrieve a detailed
report for that conference.
Figure 3-9 Results of operator request search
Viewing Operator Request Statistics
1
In the Operator Statistics page navigation bar, click View Statistics.
The Statistics page appears. It looks similar to the Requests page
Operators, For Specific Operator, and For Specific Subscriber).
2
In the navigation bar, click the link that corresponds with the statistics
you want to generate.
The Statistics Search page displays search fields that vary slightly
that appear when you choose For All Operators.
3
If you chose For Specific Operator or For Specific Subscriber, select the
operator from the list or enter the subscriber’s ID (if you don’t know the
subscriber’s ID, click the Search link to its right to search for the
subscriber).
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Figure 3-10 Generating statistics for all operators
4
5
Set the Begin Date and End Date fields by doing one of the following:
— Click in the field and edit the date.
— Click the field’s calendar icon and click a date in the calendar that
appears.
Regardless of which dates you choose, records are only available for the
past 31 days and aren’t available for the current day (the system collects
the day’s data at midnight).
Click the Search button.
The ReadiVoice system displays operator request statistics for the
operator, subscriber, and/or time period matching your search criteria
— Minimum, maximum, and average wait for request to be answered.
— Minimum, maximum, and average time spent on a request.
— Sample sizes (number of requests included in statistics).
If you requested statistics for all operators, it also shows:
— Minimum, maximum, and average wait for canceled requests.
— Maximum number of requests in the queue at any one time.
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Figure 3-11 Results of generating statistics for all operators
Using the SNMP Monitor
The ReadiVoice system supports the Simple Network Management Protocol
(SNMP). SNMP lets you monitor the ReadiVoice system with standard
network administration tools, such as HP Openview from Hewlett Packard or
similar tools from companies such as SUN or IBM.
documents the system and bridge SNMP events. “Enabling SNMP Logging”
on page 141 describes SNMP logging, which writes SNMP data to log files on
an ongoing basis.
But, if you just want to check system statistics in real time, you can use SNMP
Monitor. SNMP Monitor is a Java applet that you run from your browser. It
provides a quick and convenient way to monitor the contents of the SNMP
Management Information Base (MIB).
To run SNMP Monitor:
1
On the ReadiVoice home page, click the SNMP Monitor link.
The Java applet loads, which may take a few moments. Then, the SNMP
Monitor window appears. The Perfmeter tab is selected, and it starts
graphing selected MIB data over time, both system-level and for each
Although the SNMP Monitor applet opens in its own window, your
browser must remain active and pointed to the Monitor host page. If you
close your browser or navigate away from the host page, SNMP Monitor
closes.
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Figure 3-12 Perfmeter tab of SNMP Monitor window
2
3
Point your mouse pointer at a specific data point on a graph.
A small window displays the MIB data value and time for that point on
the graph.
To customize which MIB elements are monitored, select Setup from the
Configuration menu.
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Figure 3-13 SNMP Monitor Properties dialog box
4
Change the polling interval and time horizon, if you want, and choose the
graphs you want displayed. Then click Apply.
The SNMP Monitor window reflects the changes you made.
5
6
Resize the SNMP Monitor window, if necessary, to display the graphs
you selected at a convenient size.
Make additional changes, if you want, in the SNMP Monitor Properties
dialog box. When you’re finished, click OK.
The SNMP Monitor Properties dialog box closes.
7
To see all MIB statistics for your system, click the Statistics tab.
The Statistics tab displays, in text form, all MIB elements, both for the
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Figure 3-14 Statistics tab of SNMP Monitor window
8
To see a record of the system’s conference and port usage levels over a
longer time period, select Histograms from the Histograms menu.
port usage levels over time.
The time period shown on the horizontal axis is always 24 hours, from
midnight to midnight. Therefore, the data to the left of the current hour is
from today and the data to the right of the current hour is from yesterday.
The MIB data is updated hourly.
9
To close the Histograms window, click OK.
10 To stop running SNMP Monitor entirely, select Exit from the SNMP
Monitor window’s Configuration menu, close your browser, or navigate
away from the host page.
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Using the Channel State Monitor
In a PSTN system, you can use the Channel State Monitor tool to monitor the
teleconferencing bridge resources of the system (cards, spans, and channels)
and to enable or disable telephony spans.
To run Channel State Monitor:
1
In the navigation bar of the internal user home page, click Channel State
Monitor. If prompted, enter your system administration user name and
password.
2
Click the Display button (purple “D”) of the bridge for which you want
to see span and channel information.
The Java applet loads, which may take a few moments, and displays
summary status information for all the processor (VCE or HMod) cards in
state of the network connection between the CACS and the bridge; it
blinks if the connection is lost.
Figure 3-16 Channel State Monitor page, listing bridges in the system
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Figure 3-17 Channel State Monitor’s summary information for a bridge
3
Click on a card to see detailed information about that card.
The top row contains a status indicator for each span:
Green — span is up and connected.
Red — span is in red alarm (connection to far end lost).
Yellow — span is in yellow alarm.
Blue — span is in blue alarm (busied out).
Black (or dark gray) — span has been disabled.
Below each span indicator is a column of status indicators for its channels:
Green — channel is off hook.
Gray — channel is on hook.
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Figure 3-18 Applet window with details for the selected card
4
To see additional information about a span, click its status indicator in the
card window.
Another applet window appears, displaying configuration and status
change the span’s status:
Hard Span Disable — Takes the span out of service.
Enable Span — Restores a disabled span to service. This takes some time
(typically, about 30 seconds).
Put Into Blue Alarm — Busies out the span.
Put Into Green Alarm — Restores a busied-out span to service. This takes
some time (typically, about 30 seconds).
Close — Closes the applet window for the span.
Disabling a span or putting it into blue alarm immediately hooks all channels on that
span.
Caution!
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Figure 3-19 Applet window with details for the selected span
5
To see additional information about a channel, click its status indicator in
the card window.
Another applet window appears, displaying configuration and status
information includes the handle of the channel process and the subscriber
ID of the associated conference process, if any.
Figure 3-20 Applet window with details for the selected channel (port)
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4
Maintaining the ReadiVoice System
This chapter describes how to maintain your ReadiVoice Intelligent Voice
Conferencing System. It includes information about backup procedures and
other routine maintenance tasks. It assumes that you are knowledgeable about
Solaris UNIX, Sun servers, Informix databases, and networking.
The maintenance procedures are essential to proper operation of the ReadiVoice
system. Failure to perform these tasks as recommended may result in critical failure
of the ReadiVoice system.
Caution!
When the Polycom installs and configures your system, we set up certain
automated processes, including transaction logging for the Informix database
and monthly purging of ReadiVoice log files.
Nevertheless, you must perform some maintenance tasks on a regular basis in
order to check your system, keep it operating properly, and ensure that you
can recover from a hardware or software failure. These tasks are described in
this chapter. Topics include:
•
weekly tasks described in more detail later.
•
archiving and logical logging, and helps you understand backup
strategies and options.
•
•
Informix Dynamic Server’s transaction logs are being backed up to tape on
an ongoing basis.
on page 103 provide detailed procedures for the tasks summarized in the
quick maintenance checklist.
•
•
records from the database.
that you perform only when needed.
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Quick Maintenance Checklist
Table 4-1 summarizes the daily and weekly maintenance tasks, which are
covered in more detail later in the chapter.
Table 4-1
Summary of daily and weekly maintenance tasks
Resolving Problems or Getting
More Information
Step
Daily:
1
Action/Command
Expected Results
Stop logical log backup and
replace tapes.
2
Check ReadiVoice processes:
racheck
The following processes should
be running:
Wait for all conferences to end.
Run rastop and then rastart.
odproc
csc
megasock
sockcap
vbootp
mplex
instore
You must be root.
sua(IP systems only)
3
4
Check Informix dbspace:
onstat -d
At least 50,000 free pages in
Call Polycom Global Services.
Wait for all conferences to end.
/dev/info/dbspace1
.
Check Informix users:
At least four informix, one web
,
onstat -u
and three rootusers are active.
Run rastop, reboot the Sun, and
You must be root.
5
6
7
Check Informix logging:
onstat -l
All logical logs are flagged
F------or U-B---- except the
current log file, which is flagged
Start logical log backup with
U---C-L
.
Restart logical log backup and
check status of its tape drive:
ontape -c
Device busy.
Insert tape, restart ontape, or
troubleshoot drive. See
mt -f /dev/rmt/1 status
Check status of archive tape
drive:
Device at load point.
Insert tape or troubleshoot drive.
mt -f /dev/rmt/0 status
Weekly:
8
Check Solaris disk space:
df -k
All disk partitions at less than
90% of capacity.
Remove old, unneeded files. See
9
Check for defunct processes:
ps -ef | grep ‘<defunct>’ |
grep -v grep
No defunct processes listed.
Kill defunct processes. See
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Understanding Informix Backup
The ReadiVoice system actually uses two Informix databases. One, cdrcnow
,
consists of all the CDR (call detail record) tables. The other, cnow, contains
everything else, including your subscriber data. When we refer to the Informix
database here, we mean the entire Informix Dynamic Server dbspace,
including both of these databases.
Informix backup involves two distinct processes:
•
•
Periodically creating a tape backup (called an archive) of the database.
Continually logging all database transactions to disk and, optionally,
backing up these transaction logs (called logical logs) to tape.
All ReadiVoice CACS servers have two 4mm tape drives, so one can be
dedicated to each of these processes.
The /usr/informix/onconfig.conferencenow configuration file includes parameters
specifying the archive tape device (TAPEDEV) and log tape device (LTAPEDEV).
By default, both are set to /dev/null. To enable archives and logical log backups,
you must change these to the correct tape drive. Typically, /dev/rmt/0 is used for
archives and /dev/rmt/1 is used to continually back up logical logs.
Caution!
The sections below provide an overview of each process and some guidelines
for setting up your backup procedures. For more detailed information, consult
the IBM Informix Backup and Restore Guide for Informix Dynamic Server 9.3 and
other Informix documentation or call your Polycom Global Services
representative.
IBM provides two data recovery systems with Informix Dynamic Server, the
ON-Bar system and the ontapeutility. Although they perform similar
functions, they’re not compatible with each other or interchangeable. For more
information, see the IBM Informix Backup and Restore Guide. This manual
discusses only the ontapeutility.
At the time that this is being written, all IBM Informix documentation is
available in PDF form at:
Archives
Archiving creates a complete (level 0) or incremental (level 1 or 2) backup of
the Informix database on tape.
We don’t recommend incremental archives. The ReadiVoice database is
typically small enough that a level 0 archive goes quickly (under an hour), and
it offers greater security and a much simpler restore process. This manual
assumes that all archives are level 0.
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In the event of database corruption, a crash, or catastrophic disk failure, you
can restore the database from the most recent archive tape to the state it was in
when the archive was created. Logical logs (see next section) then enable you
to recover the transactions that took place after the most recent archive.
How often should you archive your database? It depends on the size and
activity level of your system, whether you have the optional disk mirroring
feature, and your comfort level with regard to risk.
We recommend, for most circumstances, nightly archives. The command
ontape -s -L 0creates a full level 0 archive on the archive tape device. The
process prompts for user input, so you must deal with this in the crontabfile
if you want to schedule the task.
Each morning, remove the tape, label it, and replace it with the next one. Use
ten tapes (or fourteen if you operate on weekends) in rotation. Consider
rotating the oldest five (or seven) to an off-site facility for protection against an
extreme disaster at your site.
Validate your archives and, from time to time, test your ability to restore from them.
Tapes wear out over time. Label them with the date of first use and replace them
every six months.
Caution!
Combining the most recent nightly archive with the subsequent logical logs
(on the logical log backup tape and/or on disk), you have a very high
probability of recovering all database information right up to the point of
failure. Even if the logical logs on disk are lost and the logical log backup tape
fails, you lose only one day’s data. If an archive tape fails, you can still recover
all data by using the previous day's archive tape together with the logical logs.
archiving the database nightly.
Logical Logs
Informix records all database transactions in its logical log files as soon as
they’re complete. By default, your system is set up to use thirty log files of 1
Mbyte each. You can change the number and size of the log files in
/usr/informix/onconfig.conferencenow. The logical logs provide a
complete record of all database activity.
When a log is full, Informix starts writing to the next one. If all logs are full,
Informix halts. Therefore, the logical logs must be backed up to tape (or to
/dev/null) and marked as available again on an ongoing basis.
If your database crashes, you can restore it to the point of the last archive using
the most recent archive tape. Then, you can recover all subsequent
transactions using the logical log tape together with the log file on disk that
was being used at the time of the crash. This recovers all data up to and
including the last completed transaction.
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Even in the event of a catastrophic disk failure, the most recent archive tape
and current logical log tape let you recover all the transactions up to the point
where the last logical log file was written to tape. You never lose more than the
transactions in the logical log file being used at the time of the crash (assuming
no tape failure).
You can adjust the size of the logical log files so that they fill up and are written
to tape at a comfortable interval. For example, if it takes an average of eight
hours for each logical log to fill up, you may want to reduce their size so that
you don’t risk losing so much data in the event of a disk failure.
We strongly recommend backing up logical logs continually to tape. If you
don’t, and the logical logs on disk are unrecoverable due to a disk failure, then
you lose all transactions since the last archive. Since the log files must be sent
either to tape or to /dev/null, backing them up to tape costs you only the
price of the tapes and the time needed to change tapes and check the status of
the process periodically.
We recommend using at least three tapes in rotation for logical log backup.
Change tapes on at least a weekly basis and more often if necessary. Consider
rotating the oldest tape(s) to off-site storage for extreme disaster recovery.
assume that you’re backing up logical logs to tape.
Validate your logical log backups and, from time to time, test your ability to restore
from them. Don’t let the logical log backup tape become full. If it does, Informix
stops backing up logical logs, and when all logical logs are full, the database halts.
Tapes wear out over time. Label them with the date of first use and replace them
every six months.
Caution!
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Backing Up Logical Logs
This procedure assumes the CACS is easily accessible. If it’s not, you can:
•
•
Use ontape -ato perform a one-time backup of all the full logical logs
until you can get to the site to restart continual backup.
Use a wrapper script to run ontape -cfrom a nohupprocess, directing
output to a log file. If you do this, be sure you check the log file frequently.
Informix Dynamic Server must be configured properly to back up logical log files to
tape. If you choose not to back up logical logs to tape, they must be directed to
/dev/null to prevent Informix from halting when all the log files become full. Before
Caution!
To start logical log backup:
1
2
3
Insert a tape into the drive designated for logical log backups (we assume
here that it’s /dev/rmt/1).
Log into the CACS server as root, open an XTerm window, and become
user informix by entering su – informix
Type ontape -cand press ENTER.
Informix responds:
.
Performing continuous backup of logical logs.
Please mount tape 1 on /dev/rmt/1 and press Return to continue.
4
5
Press ENTER.
Informix backs up the currently full log files and marks them as once again
available for logging. As long as the process continues to run, it backs up
each log as it becomes full and marks it as available again.
You can minimize this XTerm window if you want, but don’t quit. If you
do, the ontape process stops, and your logical logs are no longer being
backed up.
operating properly.
Whenever you reboot the Sun, be sure to restart logical log backup. Insert a new
tape when you restart if you don’t want the logical logs already backed up to the
existing tape to be overwritten. Don’t let the logical log backup tape become full! If it
does, Informix stops backing up logical logs, and when all logical log files are full,
the database halts.
Caution!
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Daily Maintenance Tasks
This section describes tasks you should perform daily, assuming that you
archive nightly and back up logical logs to tape. Perform them first thing in the
morning or any time after the archive process is complete.
Stopping Logical Log Backup and Replacing Tapes
1
In the XTerm window running ontape -c, press CTRL+C.
The logical log backup process stops.
Leave this XTerm window open (or running minimized) so that later you
can restart logical log backup in it. See “Restarting Backup and Verifying
2
Eject and replace the archive tape and, if necessary, the logical log tape.
Never try to unload a tape cartridge while the tape is in motion as indicated by the
LED. Doing so can permanently damage the tape.
Caution!
Checking the ReadiVoice Processes
The ReadiVoice processes are:
odproc— main ReadiVoice CACS process
csc— conference state controller
megasock— manages communications
mplex — runs Operator application
sockcap — captures and records log files
instore — collects and stores health and status information
vbootp — bootstraps the cards in a bridge
sua — call control (IP systems only)
ReadiVoice behavior is unpredictable unless all required processes are running.
Caution!
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To check the ReadiVoice processes:
1
Open a Telnet session to the CACS as user cnow and switch user to root
(or log into the CACS as root and use an XTerm window).
2
Change directories to /rahome and enter ./racheckfor a brief
summary, or ./checkmore detail.
The system should show the required ReadiVoice processes running, as
You can also check the processes using the ReadiVoice Administration
3
If all required processes aren’t running, follow the procedure in
Figure 4-1 Checking the ReadiVoice processes
SunOS 5.8
login: cnow
Password:
Last login: Thu Feb 17 16:23:05 from :0
Sun Microsystems Inc.
$ su root
SunOS 5.8
Password:
# cd /rahome
# ./racheck
330 ?
230 ?
677 ?
674 ?
340 ?
345 ?
669 ?
0:01 csc
0:03 vbootp
0:01 mplex
0:13 odproc
0:00 megasock
0:00 instore
0:00 sockcap
#
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Checking Informix Dynamic Server
1
In the same XTerm window in which you’re logged in as root, switch user
to informix and enter tcshto set the Informix environment.
2
Check the database space by entering onstat -d
.
Informix displays information about the disk space that it controls.
Figure 4-2 shows an example.
3
If /dev/info/dbspace1has fewer than 50,000 free pages, call your
Polycom Global Services representative.
Figure 4-2 Checking the Informix database space
su informix
tcsh
ragnar </rahome> cnow> onstat -d
Informix Dynamic Server Version 9.30.UC1
34960 Kbytes
-- On-Line -- Up 12 days 19:27:51 --
Dbspaces
address number
ba92150 1
ba92790 2
flags
1
1
fchunk
1
2
nchunks flags
owner
informix rootdbs
informix dbspace1
name
1
1
N
N
2 active, 2047 maximum
Chunks
address chk/dbs offset
size
free
bpages
flags pathname
ba92210 1
ba926b0 2
1
2
0
0
525183
1024000 821333
403104
PO-
PO-
/dev/info/rootdbs
/dev/info/dbspace1
2 active, 2047 maximum
ragnar </rahome> cnow>
4
5
Check the Informix users by entering onstat -u.
Informix displays information about the users logged into the database.
With the database online and the ReadiVoice processes running, the list
should include at least four informix entries, three root entries, and one
If the required users aren’t listed, stop the ReadiVoice processes, reboot
the Sun, and restart the ReadiVoice processes. See “Stopping and
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Figure 4-3 Checking Informix users
ragnar </rahome> cnow> onstat -u
Informix Dynamic Server Version 9.30.UC1 -- On-Line -- Up 12 days 19:29:52 --
34960 Kbytes
Userthreads
address flags
sessid
user
tty
wait
tout locks nreads
nwrites
ba94018 ---P--D 1
ba944cc ---P--F 0
ba94980 ---P--F 0
ba94e34 ---P--F 0
ba973d4 Y--P--- 1172
ba97888 Y--P--- 1074
ba97d3c ---P--D 13
ba981f0 Y--P--- 1075
ba986a4 Y--P--- 1073
ba98b58 Y--P--- 1183
informix -
informix -
informix -
informix -
0
0
0
0
0
0
0
0
0
0
0
0
1
1
0
1
1
1
14
0
0
0
0
0
0
0
0
4209
3577
1619
0
0
271
0
15
0
9
web
root
-
4
bbddf18 0
bb91e30 0
0
bca7b28 0
bb91760 0
bbad500 0
informix -
0
root
root
patrol
4
4
-
0
17 active, 128 total, 25 maximum concurrent
ragnar </rahome> cnow>
6
Check logging by entering onstat -l.
Informix displays information about physical and logical logging of the
In the flags column, logical logs that have not been used (if any) are
marked F------. Logs that have been used and backed up, and therefore
are available for reuse, are marked U-B----. The log to which Informix is
currently writing is marked U---C-L. Logical logs that are full, but have
not been backed up are marked U------
.
If more than one entry is marked U------, the logical logs aren’t being backed up.
page 94 if necessary and be sure you’re set up properly for logical log backup (or
are redirecting the log files to /dev/null). Then, restart logical log backup as
Caution!
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Figure 4-4 Checking logical logging
ragnar </rahome> cnow> onstat -l
Informix Dynamic Server Version 9.30.UC1 -- On-Line -- Up 21 days 10:14:39 --
34944 Kbytes
Physical Logging
Buffer bufused bufsize numpages numwrits pages/io
P-1 0
16
10436
923
phyused %used
0 0.00
11.31
phybegin physize phypos
10003f
500
441
Logical Logging
Buffer bufused bufsize numrecs numpages numwrits recs/pages pages/io
L-3 0 16 81863 15255 10598 5.4 1.4
Subsystem numrecs Log Space used
OLDRSAM 81863 10880128
address number
a58d5b8 1
a58d5d4 2
a58d5f0 3
a58d60c 4
a58d628 5
a58d644 6
a58d660 7
a58d67c 8
a58d698 9
a58d6b4 10
a58d6d0 11
a58d6ec 12
a58d708 13
a58d724 14
a58d740 15
a58d75c 16
a58d778 17
a58d794 18
a58d7b0 19
a58d7cc 20
a58d7e8 21
a58d804 22
a58d820 23
a58d83c 24
a58d858 25
a58d874 26
a58d890 27
a58d8ac 28
a58d8c8 29
a58d8e4 30
flags
uniqid
begin
size
used
500
500
500
500
500
500
500
500
500
500
500
500
500
500
500
265
500
500
500
500
500
500
500
500
500
500
500
500
500
500
%used
U-B---- 91
U-B---- 92
U-B---- 93
U-B---- 94
U-B---- 95
U-B---- 96
U-B---- 97
U-B---- 98
U-B---- 99
U-B---- 100
U-B---- 101
U-B---- 102
U-B---- 103
U-B---- 104
U-B---- 105
U---C-L 106
U-B---- 77
U-B---- 78
U-B---- 79
U-B---- 80
U-B---- 81
U-B---- 82
U-B---- 83
U-B---- 84
U-B---- 85
U-B---- 86
U-B---- 87
U-B---- 88
U-B---- 89
U-B---- 90
100233
100427
10061b
10080f
100a03
100bf7
100deb
100fdf
1011d3
1013c7
1015bb
1017af
1019a3
101b97
101d8b
101f7f
102173
102367
10255b
10274f
102943
102b37
102d2b
102f1f
103113
103307
1034fb
1036ef
1038e3
103ad7
500
500
500
500
500
500
500
500
500
500
500
500
500
500
500
500
500
500
500
500
500
500
500
500
500
500
500
500
500
500
100.00
100.00
100.00
100.00
100.00
100.00
100.00
100.00
100.00
100.00
100.00
100.00
100.00
100.00
100.00
53.00
100.00
100.00
100.00
100.00
100.00
100.00
100.00
100.00
100.00
100.00
100.00
100.00
100.00
100.00
ragnar </rahome> cnow>
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Restarting Backup and Verifying Tape Drive Status
1
In the CACS XTerm window in which you stopped ontape(see
logical log backup by entering ontape -c
.
Informix responds:
Performing continuous backup of logical logs.
Please mount tape 1 on /dev/rmt/1 and press Return to continue.
2
Press ENTER.
Informix begins backing up all the currently full log files and marks them
as once again available for logging. As long as the process continues to
run, it backs up each log as it becomes full and marks it as available again.
You can minimize this XTerm window if you want, but don’t quit. If you
quit this XTerm window, the ontapeprocess stops, and your logical logs
are no longer being backed up.
3
In a new XTerm window, check the status of the logical log backup tape
device by entering mt -f /dev/rmt/1 status
.
The system should indicate that the tape drive is in use, as shown in
Figure 4-5 Checking the status of the logical log tape drive
ragnar </rahome> informix> mt -f /dev/rmt/1 status
/dev/rmt/1: Device busy
ragnar </rahome> informix>
4
Check the archive tape device by entering mt -f /dev/rmt/0 status.
The system should indicate that the tape drive is at the load point, as
Figure 4-6 Checking the status of the archive tape drive
ragnar </rahome> informix> mt -f /dev/rmt/0 status
HP DDS-3 4MM DAT tape drive:
sense key(0x0)= No Additional Sense
file no= 0 block no= 0
ragnar </rahome> informix>
residual= 0
retries= 0
This concludes the routine daily maintenance. You may want to check logical
logging status again later (by entering onstat -l), after Informix has had time
to fill a logical log file, to confirm that the logs are being backed up.
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Maintaining the ReadiVoice System
Weekly Maintenance Tasks
This section describes procedures that you should perform at least weekly and
perhaps more often for a large and busy system.
Checking Solaris Disk Space
1
Open a Telnet session to the CACS as user cnow and switch user to root
(or log into the CACS as root and use an XTerm window).
2
Check disk space by entering df -k
.
The system displays file system statistics, including the percentage of
instance, /usris at 92% of capacity and requires attention.
Figure 4-7 Checking disk space
# df -k
Filesystem
/proc
/dev/dsk/c0t0d0s0
/dev/dsk/c0t0d0s5
fd
/dev/dsk/c0t0d0s6
/dev/dsk/c0t1d0s5
/dev/dsk/c0t1d0s0
/dev/dsk/c0t1d0s1
swap
kbytes
0
48023
used
0
31869
avail capacity Mounted on
0
11352
78796
0
71789
75301
0%
74%
92%
0%
14%
10%
56%
40%
35%
/proc
/
/usr
/dev/fd
/var
1015695 875958
0
92231
92231
0
11219
7707
/opt
492351 246332 196784
380311 133557 208723
459800 158808 300992
/rahome
/web
/tmp
#
You can also check the disk space from an operator/maintenance station
using the ReadiVoice Administration interface. See “Checking System
3
If any partition other than root (/) is at or above 90% of capacity, you
should remove old, unneeded files to free up space. Call your Polycom
Global Services representative if you need help in determining what you
can safely remove.
If you run out of space on any disk partition, Solaris ceases to operate!
Caution!
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Checking for Defunct Processes
1
As root, enter ps -ef | grep '<defunct>' | grep -v grep
.
The system lists any defunct processes (processes that have no parent). If
there are none, the prompt simply reappears. The example in Figure 4-8
shows three defunct processes.
Figure 4-8 Checking for defunct processes
# ps -ef|grep '<defunct>'|grep -v grep
jhaug
root 14919
jhaug 5807 5806 0
123
122 0
377 0
0:00 <defunct>
0:00 <defunct>
0:01 <defunct>
#
2
If there are defunct processes, note their process IDs (PIDs). The PID is
the first number after the process owner’s name. For instance, in the
3
4
5
Kill each defunct process by entering kill PID, where PIDis the PID
number for the process.
Enter ps -ef | grep '<defunct>' | grep -v grepagain to confirm
that the processes were killed.
If the basic kill command failed to kill a process, enter kill -9 PID
.
Then check again to confirm that it was killed.
Be very careful when using the kill command and especially when using kill -9. Be
certain that you enter the correct PID. Never kill an sqlturbo process or you may
bring down the database.
Caution!
6
Restart any processes that you had to kill, but that should be running.
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Database Maintenance Tasks
Without proper maintenance, your ReadiVoice databases will grow ever
larger, slowing the system. It’s imperative that you routinely remove (purge)
old records. Polycom provides several scripts you can use for this purpose.
You can run these scripts manually, but you’ll probably want to schedule them
to run automatically, using the cron task scheduler. The sections that follow
describe these scripts. Read through these sections to determine which scripts
you need to run and how to run and schedule them.
Run only the scripts described in this section or elsewhere in this manual. Other
scripts in the file system must not be run without express instructions from your
Polycom Global Services representative.
Caution!
Purging Outdated Call Detail Records (CDRs)
Your ReadiVoice installation includes a script, purgeCDR, that lets you easily
remove old CDR data from your database. You can run the purgeCDRscript:
•
•
Manually from the command line (called interactive mode).
Automatically at periodic intervals, using the cron task scheduler.
Either way, the purgeCDRscript:
•
Purges from the database all the CDRs that are older than a cutoff date or
a number of days that you specify. By default, it purges only conferences
with the processed flag set to 1, but you can tell it to ignore the flag.
•
By default, writes purged records to a text file named
YYYYMMDDhhmmsscdr.log(where YYYYMMDDhhmmss is the run date of the
purge) and then archives the file as YYYYMMDDhhmmsscdr.log.gz. The
archives are stored in /rahome/cdr/archives. Archiving can be turned
off.
When you run the purgeCDRscript from the command line, you can view a list
of conferences to be purged without actually purging them.
CDR Purge Script Command Syntax and Examples
The purgeCDRscript resides in /rahome/cdr. When you run purgeCDR, it calls
another script, purgeCDR.pl, in the same directory. This second script
performs the actual purge process; purgeCDRcan monitor the CPU usage and
halt purgeCDR.plas needed in order to minimize the impact on your system.
We strongly discourage running purgeCDR.pl directly on a live conferencing
system. Doing so may impede your ReadiVoice conferencing services.
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The purgeCDRcommand syntax is:
purgeCDR [-h] [-d mm/dd/yyyy] [-n numOfDays] [-i] [-u] [-m maxCpuUsage] [-rt numOfMinutes]
[-cs numOfSubscribers] [-v] [-ip] [-na]
Table 4-2 describes the options (command-line arguments).
Table 4-2
purgeCDR command options
Option
-h
Description
Display help.
Cutoff date. CDRs of conferences that ended before this date are purged. Must be
in mm/dd/yyyy format, where mm is month, dd is day, and yyyy is year. This option is
only useful for running the script manually (interactive mode). Either it or the next
must be specified.
-d
Cutoff number of days. CDRs of conferences that ended more than this many days
ago are purged. May be any integer. Use this option to run the script automatically
at scheduled intervals. Either it or the -d(date) option must be specified.
-n
-i
Run in interactive mode (manually at the command line as a one-time purge). The
script prompts you to confirm before purging the records. This option must be
specified in order to use the -uoption.
Updates database statistics (indexes) upon completion. The script must be run by
user cnow to use this option.
-u
-m
Maximum CPU usage. The purge process halts temporarily if CPU usage (in
percent) by other processes exceeds this value. Valid values are 0 to 100. Doesn’t
check CPU usage if not specified. When running purgeCDR on a live conferencing
system, we strongly recommend using this option to minimize the impact on your
system.
Maximum run time in minutes. Default is 120.
-rt
-cs
Number of conferences to delete per batch. Default is 4000. Valid values are 50 to
4000.
Display a list of conferences to be purged without purging them.
-v
If you use this option with cutoff parameters that return a very large number of
records (>50,000), the script may terminate with an “out of memory” error.
Ignore the processedflag in the cdr_post_conftable. If this option isn’t specified,
-ip
-na
the script deletes only conferences with the processedflag set to
1.
No archive. Specify this option to skip writing the records to a file before purging.
Both the cutoff date and number of days options use midnight as the
boundary. If you specify a cutoff date of 08/21/2003, the last conference
purged will be one that ended at 11:59:59 on 08/20/2003. If you specify the
number of days as 30, and 30 days ago was 08/14/2003, the last conference
purged will be one that ended at 11:59:59 on 08/13/2003.
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Table 4-3 gives examples of the command with various options and values and
describes the result of running the command.
Table 4-3
purgeCDR examples
Command
Description
Displays a list of all conferences that ended before July 29,
2005, but does not purge them. Since -ip isn’t specified, it
purgeCDR -d 07/29/2005 -v
lists only those with the processed flag set.
Issued at command line (interactive mode). Purges all CDR
records associated with conferences flagged as processed
that ended before July 29, 2005. Runs for up to 60 minutes.
purgeCDR -d 07/29/2005 -i -rt 60
purgeCDR -n 5 -i -m 20 -ip -na
Issued at command line (interactive mode). Purges all CDR
records associated with conferences that ended more than
five (5) days ago. Doesn’t check the processed flag or create
an archive file. Process halts temporarily if CPU usage by
other processes exceeds 20 percent.
Suitable for scheduling in cron (non-interactive mode).
Purges all CDR records associated with conferences that
ended more than five days ago. Process halts temporarily if
CPU usage by other processes exceeds 40 percent. Deletes
the conferences 1000 at a time. Doesn’t check the
processed flag. Updates statistics when finished.
purgeCDR -n 5 -m 40 -cs 1000 -ip -u
Running a One-Time Purge of CDRs
To manually purge CDRs from the command line:
1
Open a Telnet session to the CACS as user cnow and switch user to root
(or log into the CACS as root and use an XTerm window).
2
3
Change to the /rahome/cdrdirectory.
Type the purgeCDRcommand with the options you want and press ENTER.
To see which CDRs will be purged, but not purge them, use the -v(view)
option.
4
At the prompt to continue, type
process to complete.
y
and press ENTER. Then wait for the
If you’re purging records (not using the -voption), the script removes the
out-of-date records from the database, writes them to a text file named
YYYYMMDDhhmmsscdr.log(where YYYYMMDDhhmmss is the run date of the
purge) and then archives the file as YYYYMMDDhhmmsscdr.log.gz
.
5
When the script has finished, verify that the records were deleted and
that the backup file was created.
The backup file is in /rahome/cdr/archives
.
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Scheduling Automatic Purges of CDR Records
To schedule purgeCDRto run periodically and automatically in the cron task
scheduler:
1
2
3
Decide how many days you want to retain CDR records in the database.
Use this number as the argument of the -noption. If you’re not flagging
conferences as processed, specify the -ipoption.
Decide what the maximum CPU usage should be before the purgeCDR
script suspends operation. Use this number as the argument of the -m
option.
For your records, write the command on the line below as you want it to
run. For instance, if you want to purge records more than 30 days old,
suspend the process when CPU usage exceeds 40%, and ignore the
processed flag, write the command as purgeCDR -n 30 -m 40 -ip
.
_____________________________________________________
Don’t use the -d -i, or -voptions, which aren’t suitable for automatic
,
scheduling.
4
5
Decide how often you want to run the purge process. We recommend a
frequency from weekly to daily, depending on your ReadiVoice system’s
conferencing and database usage.
In the root user’s crontabfile, schedule the command to run at the time
and frequency you want. See the Solaris documentation or the crontab
and cron man pages for information on scheduling tasks with cron.
Restoring Purged CDR Records
Should the need arise, you can restore purged CDRs to your database using
To restore all the CDRs from an archive file created by purgeCDR
:
1
Open a Telnet session to the CACS as user cnow and switch user to root
(or log into the CACS as root and use an XTerm window).
2
3
4
Change directories to /rahomeand enter ./rastop
.
Set the Informix environment by entering tcsh
.
Change directories to /rahome/cdr/archivesand find the archive file
containing the records you want to restore (its name is
YYYYMMDDhhmmsscdr.log.gz, where YYYYMMDDhhmmss is the run date of
the purge).
5
Unzip the archive, extracting YYYYMMDDhhmmsscdr.log.
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6
Restore the data from the CDR backup file to the cdrcnowdatabase by
entering (all on one line):
/rahome/dbcdr/loadtable.pl DB=cdrcnow <
/rahome/cdr/archives/YYYYMMDDhhmmsscdr.log
Replace YYYYMMDDhhmmsscdr.logwith the name of the backup file.
7
8
Change directories to /rahomeand enter ./rastart.
Log into the Administration interface and use the CDR Data pages to
verify that the records you wanted to restore are there (see “Viewing
Purging Subscriber Records Flagged for Deletion
When you change a subscriber’s status to deleted, either in the Provisioning
interface or with a stored procedure call, the ReadiVoice system doesn’t
actually delete the subscriber from the database; it merely flags the record for
deletion. To actually delete such records from the database, you must run the
subDelete.plscript.
The subDelete.pl script immediately and permanently deletes all subscriber
records whose status is set to 2 (deleted) in the database. This script does not
create a backup of the deleted records. Be sure you really want to delete these
subscribers. Be sure your provisioning procedures are appropriate (that is, the
guidelines for setting a subscriber’s status to deleted take into account the use of
this script).
Caution!
To permanently and irrevocably delete all subscriber records flagged for
deletion:
1
2
3
Open a Telnet session to the CACS as user cnow.
Change to the /rahome/database/scriptsdirectory.
Type ./subDelete.pland press ENTER.
All subscriber records flagged for deletion are permanently deleted from
the database.
You can schedule this script as a cron task. But, remember that purged subscriber
records, unlike purged CDRs, can’t be recovered. Therefore, approach the
automatic deletion of subscriber records with caution.
Caution!
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Purging the CacsEventUpdate Table
If you use the Provisioning Stored Procedure Interface (PSPI) for provisioning,
you need to purge the CacsEventUpdatetable from time to time to keep it
from growing too large.
Several stored procedures don’t load the new data into call router memory
immediately. Instead, they put the new data into the CacsEventUpdatetable;
the CACS then updates the call router with the data from the temporary table
every ten seconds.
For more information about PSPI, see the ReadiVoice PSPI Reference.
If you use stored procedure calls (SPCs) that write records to this table, then
you must purge those records regularly.
Your ReadiVoice installation includes a script, cleanCacsEvtUpdate.pl, that
purges the processed records from the CacsEventUpdatetable. You can use
the cron task scheduler to run this script periodically.
Using this script improperly may impede your ReadiVoice conferencing services.
Your Polycom Global Services representative can help you determine the best
frequency, time, and run time parameter for this script.
Caution!
The cleanCacsEvtUpdate.plscript resides in /rahome/database/scripts
.
The syntax is:
cleanCacsEvtUpdate.pl [-rt nn]
The -rtparameter lets you specify a run time, nn, in minutes. If you omit it,
the run time defaults to 10 minutes. The script terminates at the end of the run
time, even if it hasn’t purged all processed records from the table.
Purging the AccOptChanges Table
If your system has the Account Options Updates feature enabled (see
AccOptChangestable from time to time to keep it from growing too large.
If the Account Options Updates feature is enabled, when subscribers change
their account options (using the account options menu in the call flow), the
ReadiVoice system records those changes in the AccOptChangestable (in
addition to updating the subscriber records).
You can use the GetAccOptChangesstored procedure to retrieve these account
options update records and update your customer database. The stored
procedure flags the records you retrieve as processed.
For more information about this stored procedure, see the ReadiVoice PSPI
Reference.
Your ReadiVoice installation includes a script, cleanAccOptChanges.pl, that
purges the processed records from the AccOptChangestable. You can use the
cron task scheduler to run this script periodically.
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The cleanAccOptChanges.plscript resides in /rahome/database/scripts
.
The syntax is:
cleanAccOptChanges.pl [-rt nn]
The -rtparameter lets you specify a run time, nn, in minutes. If you omit it,
the run time defaults to 10 minutes. The script terminates at the end of the run
time, even if it hasn’t purged all the processed records from the table.
The cleanAccOptChanges.plscript suspends the purge process temporarily
if CPU usage by other processes exceeds 30%. Nevertheless, it’s best to
schedule execution for times of low load. Your Polycom Global Services
representative can help you choose the best frequency, time, and run time
parameter for this script.
Infrequent Maintenance Tasks
This section describes procedures that you perform only as needed, not on a
regular basis.
Stopping and Restarting ReadiVoice Processes
The ReadiVoice processes are:
odproc— main ReadiVoice CACS process
csc— conference state controller
megasock— manages communications
mplex — runs Operator application
sockcap — captures and records log files
instore — collects and stores health and status information
vbootp — bootstraps the cards in a bridge
sua — call control (IP systems only)
ReadiVoice behavior is unpredictable unless all required processes are
running. Under certain circumstances, you should stop all ReadiVoice
processes and then restart them. These include:
•
•
If a ReadiVoice process isn’t running.
If you need to change the date and time on the Sun server.
In this case, you should also reboot the Sun. See “Rebooting the Sun
•
If you change the system configuration settings for routing, confidence, or
hidden number use.
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Stopping and restarting the ReadiVoice processes affects current users as
described below.
Caution!
It’s best to stop and restart ReadiVoice processes only when there are no active
conferences. In an emergency, however, you may have to do this while
conferences are running. If so, keep the following in mind:
•
Anyone using the Operator or Moderator interface is isolated from the
conferencing system for the time it takes to stop and restart (about one
minute).
•
New calls can’t be handled until the core processes have restarted.
Depending on your system configuration, callers hear either a busy signal
or a message stating that the system isn’t available and asking them to call
back.
•
Existing conferences continue to operate normally. The participants are
unaffected unless they request an operator or try to use the Moderator
interface during the stop/restart operation.
To stop and restart the ReadiVoice processes:
1
Open a Telnet session to the CACS as user cnow and switch user to root
(or log into the CACS as root and use an XTerm window).
2
3
4
5
Change directories to /rahome.
Enter ./rastop. If the prompt doesn’t reappear, press ENTER.
Enter ./rastart. If the prompt doesn’t reappear, press ENTER.
To verify that the required processes are running, enter ./racheck
.
Rebooting the Sun Server
Under certain circumstances, you should reboot the server. These include:
•
•
If you change the date and time on the Sun server.
If the system exhibits erratic or unpredictable behavior.
Rebooting the Sun server affects current ReadiVoice users as described in the
previous section and below.
Caution!
Naturally, rebooting the server also includes stopping and restarting the
ReadiVoice processes. If conferences are running, the same considerations
apply. But, the entire process takes longer, perhaps five to ten minutes. In
addition, the bridges don’t have access to files on the server during the reboot.
This may affect existing conferences if, for example, sound files not already
cached on the bridge are needed.
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To reboot the Sun server:
1
2
As root, stop the ReadiVoice processes as described above.
To perform a warm reboot in 30 seconds, enter shutdown -g30 -i6 -y
.
Solaris begins shutting down in 30 seconds without prompting for
verification. The entire reboot process takes several minutes.
Omit -yto get a confirmation prompt. Change the value after -gto change
the delay. See the man page for the shutdown command for more options.
3
Once Solaris has finished rebooting, restart the ReadiVoice processes as
described above.
Rebooting a Bridge
Under certain circumstances and with the guidance of Polycom Global
Services, you may need to reboot a bridge.
A cold boot means that you turn off power to the entire bridge and then turn
it back on.
A warm boot means that power to the bridge is always on while you
individually re-start various components. You can re-start many components
either through the software or through reset switches on the components
themselves.
Due to the number of cards, it’s impractical to perform a warm boot of an
entire bridge. Individual cards can be warm booted, which might be preferable
if the problem and its resolution can be localized. Rebooting individual cards
permits the remaining cards to continue to service conferences and share
resources.
You should have Polycom Global Services reboot individual VCE or HMod
cards. Global Services representatives can do this remotely. They can also
determine whether a card is in use and can set the card as least-preferred for
all new conferences and for resource sharing between cards. This allows the
card to be rebooted without affecting any conferences on the bridge.
To reboot (cold boot) a bridge:
1
On the System Administration interface’s Bridges page, change the status
2
Wait for all conferencing activity to end.
Rebooting a bridge terminates all conferencing activity! Reboot only when there are
no active conferences on the bridge.
Caution!
3
Using the power switch on the rear of the bridge cabinet, turn off power
to the bridge.
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The VCE/HMod cards in the bridge must be able to access files on the
server in order to boot and to initiate various processes. If you’re also
rebooting the server, be sure it has finished booting before restoring power
to the bridge.
4
Wait at least fifteen seconds. Then, turn on the bridge.
The bridge powers up and the VCE/HMod cards boot from the server.
Changing the ReadiVoice Password
From time to time, you may want to change the ReadiVoice cnow user
password. It’s defined in several places, so you must change:
•
•
The login password for cnow on the server (and on a boot server, if used).
The password for both the cnow user and the SYSTEM (Telnet) cnow user in
the ReadiVoice application (several places) and in the vbootp.dbbridge
configuration file.
Your ReadiVoice installation includes a shell script, cnow_password_change
that lets you easily make all the password changes for the ReadiVoice
application and bridges.
,
Perform this procedure only during maintenance periods when there are no active
conferences.
Caution!
To change the cnow password:
1
Open a Telnet session to the CACS as user cnow and switch user to root
(or log into the CACS as root and use an XTerm window).
2
Change directories to /rahome/binand enter cnow_password_change
The script prompts you twice to enter the cnow user password.
Enter and confirm the new password.
.
3
4
The script prompts you twice to enter the SYSTEM cnow user password.
Enter and confirm the new password. Be sure it’s the same password you
entered in the previous step.
The script encrypts the new passwords and updates them in various
application configuration files and processes, the database, and the
vbootp.dbfile (from which the bridges will be updated).
5
6
If your system includes one or more boot servers, repeat the preceding
steps on the boot server(s).
One at a time, reboot the bridges.
Especially for InnoVox 4000 bridges, it’s important to wait until a bridge
finishes booting before rebooting the next bridge.
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Manually Backing Up and Restoring
You may want to manually back up your ReadiVoice system at times, such as
prior to an upgrade. The backed-up data lets you to revert to the previous
version if there’s a problem with the upgrade.
The procedure described here backs up the databases, along with the contents
of the /rahomedirectory, which contains most of your configuration files.
Software upgrades automatically bring forward all database information
during the upgrade process. If you need to revert to an earlier version of the
ReadiVoice software, you must restore databases that match the earlier
software version.
Backing Up Manually
1
2
Place a tape in the archive tape drive (typically, /dev/rmt/0).
In an Xterm window, as root, set the Informix environment by entering
tcsh
.
3
Verify that the tape is ready (see “Restarting Backup and Verifying Tape
4
5
Change directories to /rahome/database.
Dump the ReadiVoice tables to a file by entering (on one line):
/rahome/database/dumptable.pl DB=cnow >
/rahome/dbdata_mmddyyyy.sql
Replace mmddyyyywith the current date.
6
Dump the CDR tables to a file by entering (on one line):
/rahome/database/dumptable.pl DB=cdrcnow >
/rahome/cdrdata_mmddyyyy.sql
Replace mmddyyyywith the current date.
7
8
9
Change to the root directory and back up the /rahomedirectory to tape
by entering tar cvf /dev/rmt/0 /rahome
When the backup is finished, verify the contents of the tape by entering
tar tvf /dev/rmt/0
.
.
Eject the tape, write-protect it, and label it with the date and the
ReadiVoice version number.
Label all backups with a version number! If you downgrade to a previous
version of the ReadiVoice software, you must also restore databases matching
that version. The loadtable.plprocess fails if the data being restored doesn’t
match the ReadiVoice version installed.
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Restoring from a Manual Backup Tape
1
In an XTerm window, as root, stop the Web server process by entering
sh /etc/rc3.d/S91rvhttpd stop
.
2
3
Change directories to /rahomeand enter ./rastop
.
Extract the backup data from tape by entering tar xvf /dev/rmt/0
.
The system restores the backup of the /rahomedirectory, including the
database dump files you created there.
4
5
Switch user to cnow. Set the Informix environment by entering tcsh
Change directories to /rahome/database. Then, drop the existing cnow
database by entering make drop
Create a new empty cnowdatabase by entering make create
Change directories to /rahome/cdrdb. Then, drop the existing cdrcnow
database by entering make drop
Create a new empty cdrcnowdatabase by entering make create
.
.
6
7
.
.
8
9
.
Restore the data from the dump file to the cnowdatabase by entering (on
one line):
/rahome/database/loadtable.pl DB=cnow <
/rahome/dbdata_mmddyyyy.sql
Replace mmddyyyywith the date of the backup.
10 Restore the data from the CDR dump file to the cdrcnowdatabase by
entering (on one line):
/rahome/dbcdr/loadtable.pl DB=cdrcnow <
/rahome/cdrdata_mmddyyyy.sql
Replace mmddyyyywith the date of the backup.
You must use the loadtable.plscript in /rahome/dbcdr, not the one in
/rahome/database, to restore the CDR data.
Caution!
11 Exit from the cnow login (returning you to the root prompt) and restart
the Web server process by entering sh /etc/rc3.d/S91rvhttpd start
.
12 Change directories to /rahomeand enter ./rastart
.
13 To verify that the required processes are running, enter ./racheck
.
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5
Configuring the ReadiVoice System
This chapter describes how to change your ReadiVoice system’s configuration
settings and which capabilities, features and options are enabled.
You can perform some of these tasks in the System Administration interface,
but many of them involve editing configuration files and working at the UNIX
command prompt. Before undertaking these tasks, you should be
knowledgeable about Solaris UNIX, Sun servers, personal computers,
software configuration, telephony, and networking.
If your ReadiVoice system is covered by warranty or a service contract, don’t
attempt to move, reinstall, or reconfigure the system or make any other hardware or
network changes without first consulting Polycom Global Services. Doing so may
void your warranty. Even in the absence of such coverage, we strongly urge you to
contact Polycom before proceeding. We can give you a quote for assistance
ranging from remotely supporting your planned change to sending a support
specialist to your site to implement the change for you.
Caution!
Before you proceed with any configuration change, review your installation
and upgrade records and log files to verify that your desired configuration is
compatible with your current configuration.
For descriptive simplicity, this chapter assumes that all components relating to
SCP or SIP call control, databases, and Web servers reside on a single Sun
workstation that serves as the CACS. If you have a larger system, these
components, if used, may be distributed among multiple workstations.
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Changing System Configuration Settings
Polycom sets your system configuration options to the correct values for your
installation. Before making any changes, contact your Polycom Global Services
representative.
Caution!
1
In the System Administration navigation bar, click System
Configuration.
Figure 5-1 System Configuration page
2
Make any changes to the first five settings that you want to make. The
settings are:
Wait Time — Determines how long (in seconds) the first conference
participant can wait on hold for the subscriber to dial in and start the
conference. Additional participants may wait for less time depending on
when they arrive for the conference. The default is 600 (ten minutes). The
maximum value is 32,767.
Max Participants — The maximum number of participants permitted in a
conference. The default is 8. The maximum value is 300.
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Min Code Length — The the minimum number of digits permitted in an
access code, subscriber password, or participant password. The default is
4. The maximum value is 20.
Confidence — the Statistical Port Management Confidence Factor determines
how aggressively the system attempts to manage port utilization. It can be
set to any integer value from 0 to 100. The default is 100, which is the most
conservative setting. At this setting, the system allocates every new
conference 100% of the ports that it’s entitled to use (the Max Participants
setting for the subscriber account). This guarantees that every running
conference can always accommodate its entire subscribed capacity.
A lower value lets you “oversubscribe” the system’s capacity on the
premise that not all users will access the system simultaneously or use
their entire subscription size. For best overall results, Polycom
recommends a value between 85 and 95.
Before changing this setting, contact your Polycom Global Services
representative for help in choosing an appropriate value. We have
statistics tables, based on an average conferencing profile, for calculating
an appropriate confidence level. Or, Voyant can statistically model your
actual conferencing history to tune your system for maximum efficiency.
This modeling typically requires about six to nine months of usage data.
on your system’s port utilization efficiency regardless of the Confidence setting.
Caution!
3
4
Select On or Off for Log Operator Data. If you turn on this option, then
every night at midnight the ReadiVoice system stores operator data
collected the previous day. You can view this data from the Operator
Select On or Off for Hidden Number Usage. If, for internal routing
purposes, your carrier maps your system’s access phone numbers to
different numbers, select On and use the Access Numbers pages to
associate every access phone number to its own hidden (routing) number.
Otherwise, or if this is an IP system, select Off.
5
Select a supported routing mode, translation number type, and call flow
for your system. See page 121 for the supported configurations.
Don’t change these settings without first contacting Polycom Global Services. If
these settings are wrong, callers can’t reach your system! Furthermore, changing
your routing mode requires additional changes to both your telephone network
connections and your ReadiVoice configuration.
Caution!
Contact your Polycom Global Services representative if you have any questions
about how calls are routed to your system or want to discuss changes to your
system’s call routing configuration.
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If you change Routing Mode from Non-Routed System Private to
Non-Routed System Shared, delete all existing (private) access phone
numbers and make sure that the Provisioning pages have the Access Code
For ReadiVoice IP, Routing Mode must always be set to Routed System.
Routing Mode
The routing mode determines how calls are routed among bridges:
— Non-Routed System Shared — Each trunk group terminates on a
specific bridge, so the access phone number determines which bridge
receives a call. Each access phone number can be used by multiple
subscribers.
— Non-Routed System Private — Each trunk group terminates on a
specific bridge, so the access phone number determines which bridge
receives a call. Each access phone number can be used by only one
subscriber, uniquely identifying the subscriber.
— Routed System — Uses an Intelligent Network Call Routing (INCR)
system, such as SS7, to tell the network how to route a call. For each
number group, a routing list specifies which bridge or bridges the
conferences in that group can use and the priority order of those
bridges.
Translation Number Type
In a routed PSTN system, when someone dials a ReadiVoice access
number, the originating switch queries the CACS, which returns a
translation number to tell the telephone network how to route the call. In a
non-routed or IP system, each bridge is provisioned with a block of
dummy translation numbers.
The Translation Number Type setting dictates how translation numbers
are assigned. The following settings are available:
— Random Translation Number — The system uses translation
numbers from the pool of numbers assigned to a bridge. All
non-routed systems must use this setting.
— Fixed Translation Number: The system uses a predetermined
translation number for each subscriber. The translation number is
stored in the ExternalIdfield of the subscriber record.
If you select this setting, then the ExternalIdfield for each
subscriber must be unique (that is, no two subscribers on this
ReadiVoice system may have the same external ID). In addition, the
the external ID.
— Fixed Translation Number with 3-digit code: The system also uses a
predetermined translation number for each subscriber (stored in the
ExternalIdfield of the subscriber record). But it appends a 3-digit
number (internally-formed) to each translation number when the
translation number is used by the call router. This setting is used for
leg-based billing.
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If you select this setting, then the ExternalIdfield for each
subscriber must be unique (that is, no two subscribers on this
ReadiVoice system may have the same external ID). In addition, the
number of digits in the external ID.
Call Flow Options
ReadiVoice has the following call flow options:
— Traditional Private: With this option selected, subscribers can be
provisioned with private access numbers. Not available if Routing
Mode is set to Non-Routed System Shared.
— Traditional Shared: With this option selected, subscribers can be
provisioned with shared access numbers. Not available if Routing
Mode is set to Non-Routed System Private.
— Two Passcode Private: With this option selected, subscribers can be
provisioned with private access numbers and participants can be
required to enter a participant password. Not available if Routing
Mode is set to Non-Routed System Shared.
— Two Passcode Shared: With this option selected, subscribers can be
provisioned with shared access numbers. Not available if Routing
Mode is set to Non-Routed System Private.
Supported Call Flow
Combinations
Voyant has tested and supports the following routing mode and call flow
combinations:
— Non-Routed System Shared: Only the shared call flow options are
available. Voyant has tested and supports the combinations shown in
Table 5-1
Supported call flows on non-routed shared systems
Call Flow Options
Translation Number Type
Traditional Shared
Random
Random
Traditional Shared +
Two Password Shared
— Non-Routed System Private: Only the private call flow options are
available. Voyant has tested and supports the combinations shown in
Table 5-2
Supported call flows on non-routed private systems
Call Flow Options
Traditional Private
Two Password Private
Translation Number Type
Random
Random
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— Routed System: The Traditional Shared, Traditional Private, and
Two Passcode Private call flow options are available. Voyant has
Table 5-3
Supported call flows on routed systems
Call Flow Options
Traditional Shared
Two Password Private
Translation Number Type
Random
Fixed Translation Number
Traditional Shared +
Two Password Private
Fixed Translation Number with
3-digit code
6
7
Select Yes or No for the Validate on Acc Code option. This option isn’t
available if Routing Mode is Non-Routed System Private.
— Yes: The ReadiVoice system identifies a caller’s conference using the
access code only.
— No: The ReadiVoice system identifies a caller’s conference using the
DNIS/access code pair.
To implement your changes, click Commit Configuration.
The system confirms your changes and provides a link back to the System
Configuration page. If necessary, a note reminds you that you must
restart the application.
8
9
Click the link to return to the System Configuration page and verify that
the settings are correct.
If you changed any of the routing or call flow options, you must run the
tnl_mklnscript:
a
b
c
Log into the CACS as root and, in an XTerm window, enter tcsh.
Change to the /rahome/bindirectory.
Run the make link script by entering tnl_mkln
.
d
One at a time, reboot the bridge(s).
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Configuring the ReadiVoice System
Changing the Talker Update Frequency
Through the Operator and Moderator interfaces (and the APIs), the
ReadiVoice system can report which two channels in a conference have talk
slots. By default, talker information is updated every two seconds, but this
interval can be lengthened to reduce the load on the system.
Perform the following procedure after operational hours. Don’t reduce the interval to
less than two seconds.
Caution!
To change the talk slot update interval for the system:
1
Open a Telnet session to the CACS and log in as cnow (or log directly into
the CACS as root and switch user to cnow).
2
Open the.odprocrcfile for editing, and find the [bridgeInt]section.
The updateIntervalFrequency keyword sets the talker update
frequency. By default, it’s set as follows:
[bridgeInt]
...
updateIntervalFrequency = 2
3
4
To reduce the update frequency (lengthen the interval), change the value
to an integer greater than 2. Then, save the.odprocrcfile.
As root, stop and restart the ReadiVoice system as described in “Stopping
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Configuring Shortened Dial-Out Call Flow
In the standard dial-out call flow, after the subscriber enters and confirms the
phone number to dial, the system supports four commands for processing the
call. The help menu played to the subscriber (do_long_cmd_help.wav
)
describes the options as follows:
“After the call is answered, to connect the line into the conference, press
star 1. To connect the line and continue dialing, press star 2. To disconnect
the line, press star 3. To disconnect the line and continue dialing, press star
4.”
To provide faster and simpler dial-outs, your system can be configured to
reduce these four options to two. In the shortened dial-out configuration, the
help menu (do_short_cmd_help.wav) describes the options as follows:
“Once the call is answered, to place the participant into the conference,
press star 1. To disconnect the participant, press star 2.”
This is a system-wide configuration. If the shortened call flow is in effect, you
can shorten it further by also turning off dialed number confirmation (so that
the system doesn’t repeat the dialed number back to the subscriber, who must
press the # key to confirm).
Perform the following procedure after operational hours.
Caution!
To configure your system to use the shortened dial-out call flow:
1
If you’ve customized or added prompt sets, review the voice prompt files
used in the shortened dial-out call flow configuration and, if necessary,
install replacements or customized versions of these files (see “Working
do_part_disconnected.wav
do_part_joined.wav
do_short_cmd_help.wav
do_short_number_prompt.wav
Contact your Polycom Global Services representative for help with voice
prompts.
2
3
Open a Telnet session to the CACS and log in as cnow (or log directly into
the CACS as root and switch user to cnow).
Change to the /rahome/bridge/scriptsdirectory and remove the long
dial-out script by entering:
rm inDialSub.x
4
Establish a symbolic link to the shortened dial-out script by entering:
ln -s inDialSubShort.x inDialSub.x
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5
6
To also turn off dialed number confirmation:
a
b
As user cnow, open the ive.inifile for editing.
In the [Dialout]section, add the following line:
ConfirmDialNum = 0
c
Save and close the file.
Reboot the bridges.
Disabling Waiting Room Notifications
If your subscribers use the Waiting Room feature (which puts callers to a
locked conference on hold until admitted by the subscriber), the ReadiVoice
system notifies the subscriber when a caller enters the waiting room and
periodically reminds the subscriber that callers are waiting to be admitted.
If your customers find these waiting room notification messages too intrusive,
you can configure your system not to play them. In that case, however, your
customers should be instructed to use the Moderator whenever a conference
is locked with Waiting Room on. The Moderator interface shows that callers
are waiting and facilitates the processing of those callers.
Perform the following procedure after operational hours.
Caution!
To disable waiting room notifications:
1
Review the voice prompt files that mention the Waiting Room
notifications feature and have replacements or customized versions
recorded. The five voice prompt files are:
cf_in_conf_subs_cmd_help.wav
wr_cmd_help.wav
_wr_new_caller.wav
wr_notification_off.wav
wr_notification_on.wav
For more information about voice prompts, see “Working with Prompt
representative for help with voice prompts.
2
As user cnow, install the replacement .wavfiles.
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3
As user cnow, open the ive.inifile for editing and find this section near
the end:
[MiscConfig]
WRAnnounceDefault = 1
4
5
Change the value after the WRAnnounceDefaultkeyword from
Then, save and close the ive.inifile.
1
to
0.
Reboot the bridges so that they read the updated ive.inifile and load
the new .wavfiles.
Configuring Dial-Out Billing
The dial-out billing feature (also known as unattended dial-out, or UDO,
billing) facilitates billing when subscribers dial out to additional participants
without operator assistance.
With this feature enabled, you can have the ReadiVoice system append (or
prepend) one or more identifying strings of DTMF digits to the dialed phone
number when a subscriber dials out from a conference. The switch that
receives the dial-out must be configured to recognize, strip off, and properly
process these digits for billing purposes.
Perform the following procedure after operational hours.
Caution!
To enable dial-out billing:
1
Determine the format of the dial-out string that your switch will expect
and recognize.
You can define two additional DTMF strings for each subscriber, and
either or both can be outpulsed before or after the dialed number. You can
define pauses, if necessary, between the dialed number and the prepended
or appended digits. The total length of the string or strings to be
appended/prepended is limited to 24 DTMF digits.
2
Determine how and where these identifier numbers will be stored for
each subscriber.
The subscriber record provides two fields specifically for this purpose,
DialOutPreFixand DialOutPostFix. You can use either or both.
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3
4
If necessary, update your existing subscriber records with the required
identifiers.
You may want to do this using the Provisioning Stored Procedure
Interface (PSPI). See the ReadiVoice PSPI Reference.
As user cnow, open the .odprocrcfile for editing and, in the
[bridgeInt]section, configure the dialOutSetupStringparameter to
meet your requirements (see “Dial-out Setup String Example”).
5
6
As root, restart the ReadiVoice application (see “Stopping and Restarting
Make sure that future subscribers will be provisioned properly:
— If you’re using the Dial Out Prefix and Dial Out Postfix fields, make
sure they’re available in the Provisioning interface (see “Setting Up
— Let your ReadiVoice provisioners know how you want them to define
these two fields when adding or updating a subscriber. The two fields
together must not be more than 24 characters.
— Make sure provisioners know that, to use this feature, subscribers
must have Dial-Out Permission enabled.
Dial-out Setup String Example
To enable the Dial-Out Billing feature, you must add a line such as this to your
.odprocrcfile’s [bridgeInt]section:
[bridgeInt]
...
dialOutSetupString = {phoneNum},,,{dialoutPreFix},,,{dialOutPostFix}
The example shows the variables specific to Dial-Out Billing, but you can also
{subId}or {externId}. Each comma is a 20-millisecond pause.
The Dial-Out Billing setup string example would out-pulse three values in
succession:
1
2
3
The dialed phone number, {phoneNum}, followed by a 60-millisecond
pause.
The value in the subscriber’s DialOutPreFixfield, {dialoutPreFix}
followed by another 60-millisecond pause.
,
The value in the subscriber’s DialOutPostFixfield, {dialOutPostFix}
.
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Enabling Account Options Updating
With Account Options Updating enabled, you can use the Provisioning Stored
Procedure Interface (PSPI) to retrieve records of the account options changes
that your subscribers make (using the account options menu in the call flow).
This lets you update your customer management database through PSPI.
At this time, the Account Options Updating mechanism only records changes
made via DTMF commands; changes made through the Moderator aren’t
available.
This feature works as follows:
1
When a subscriber uses the account options menu (prior to conference) to
change a feature setting on his or her account, the ReadiVoice system
writes a record of that change to a special table, AccOptChanges(in
addition to updating the subscriber record, of course).
The AccOptChangesrecord identifies the subscriber, the time of the
change, the subscriber field that was changed, and its old and new values.
2
3
Using the GetAccOptChangesstored procedure (see the ReadiVoice PSPI
Reference), you retrieve the new options update records from the
AccOptChangestable from time to time. This flags those records as
processed.
Periodically, you must run the cleanAccOptChanges.plpurge script to
delete the processed records from the AccOptChangestable.
Don’t attempt to perform these tasks unless you’re an experienced Informix
database and UNIX system administrator. Perform the following procedure after
operational hours.
Caution!
To use this feature:
1
2
3
Review the GetAccOptChangesstored procedure description in the
ReadiVoice PSPI Reference and ensure that you have a process in place for
retrieving options update records from the AccOptChangestable.
In cnow’s crontabfile, schedule the cleanAccOptChanges.plpurge
script to run every 24 to 48 hours. See “Purging the AccOptChanges
As user cnow, open the.odprocrcfile for editing, and find the
[bridgeInt]section. To enable the Account Options Updating feature,
set the dtmfOptChanges variable as follows:
[bridgeInt]
...
dtmfOptChanges = 1
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4
5
As root, stop and restart the ReadiVoice system as described in “Stopping
Use the GetAccOptChangesstored procedure to retrieve records from the
AccOptChangestable periodically. To determine how frequently to call
this stored procedure, balance your need for up-to-date account options
data with other demands on your database and ReadiVoice system.
6
Test the system to make sure the GetAccOptChangesstored procedure
and cleanAccOptChanges.plpurge script are running properly. Verify
that account options updates are being:
— Written to the AccOptChanges table.
— Retrieved by the stored procedure.
— Purged by the purge script.
If any of these elements is not running properly, disable this feature
immediately by repeating steps 3 and 4, but this time setting the
dtmfOptChanges variable to zero. Then, troubleshoot your
implementation of this configuration.
If you enable the Account Options Updating feature, continue to monitor the
AccOptChanges table as part of your regular ReadiVoice database maintenance
routine to make sure the GetAccOptChanges stored procedure and
cleanAccOptChanges.pl purge script are running properly. If left unmanaged, the
AccOptChanges table could slow down database operations and hinder
ReadiVoice performance.
Caution!
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Using the Music Hold Extender Message
If many participants are waiting on hold when the subscriber arrives, there
may be a short delay (a few seconds) between the time when they stop hearing
the hold music and the time when they’re placed into conference. You can
configure the system to play a special message when the subscriber arrives if
there are more than a specified number of callers on hold. The message lets
participants know the system is in the process of putting them into conference.
The default prompt set includes the voice prompt file that this feature uses,
cf_music_hold_to_conf.wav. It says, “The subscriber has joined the
conference. Your conference will now begin. Please stand by.” If you want
different wording, you can create your own version of this file. For more
Perform the following procedure after operational hours.
Caution!
To use this “extender” message feature:
1
If you have your own version of the cf_music_hold_to_conf.wavfile,
put it into the appropriate subdirectory (adpcmor g711) of
/rahome/bridge/sound/1, depending on format.
2
As user cnow, open the ive.inifile for editing, find the [SETUP]section,
and look for an entry that reads:
PlaySubArrivedWave = 0
The default value of zero disables this feature.
3
Change the value after the PlaySubArrivedWavekeyword from
number of waiting callers you want to trigger the extender message.
0
to the
If you set it to ., the message always gets played. If you set it to 10, then
1
nine or fewer callers won’t trigger the message, but ten or more will.
4
5
Save and close the ive.inifile.
Reboot the bridges so that they read the updated ive.inifile and load
the new .wavfile.
6
To test the feature:
a
Have enough participants dial into a conference to trigger the message
(at least the number you set in ive.ini).
b
Have the subscriber dial in, and verify that the participants hear the
correct message.
The bridge plays the cf_music_hold_to_conf.wavfile to the participants
immediately after identifying the subscriber.
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Configuring the ReadiVoice System
Configuring Remote Alarm Notifications
The ReadiVoice system can notify someone when an alarm occurs. The
ReadiVoice installation script provides the opportunity to enter one or more
pager numbers, email addresses, or both, to which you want remote alarm
notifications sent. This section describes how to finish setting up the pager and
email notifications, test the configuration, and run the ReadiVoice Monitoring
Tool that enables the notifications.
Configure and test alarm notifications only after operational hours.
Caution!
Pager Notification Setup
All ReadiVoice systems include one modem (and cable) to permit dial-up
remote maintenance. A second modem, dedicated to the remote alarm
function, is optional.
However, remote alarm configuration doesn’t require a dedicated modem for
pager notification. You can configure remote alarm notification to use the
same modem as the dial-up remote maintenance function. Keep in mind,
however, that alarm notifications to pagers will fail if that modem is in use at
the time.
By default, the remote alarm function is configured in the system files to use
the dedicated remote alarm modem connected to serial port B. Install this
modem using the cable provided or your own full 25-pin serial cable.
The /etc/remotefile specifies the serial port to use and the modem-specific
settings. If you need to change any of the default settings, see the remote(4)
man page for complete details on the configuration settings in the
/etc/remotefile.
system using the Polycom-supplied dedicated modem.
Figure 5-2 The default alarm modem settings in the /etc/remote file
pager:\
:pn=@:tc=cuaa:
cuaa:\
:el=^D^U^C^S^Q^O@:du:at=hayes:ie=#$%:oe=^D:br#38400
:tc=dialers:
The /etc/phonesfile specifies the number or numbers to call for pager alerts.
When you enter pager numbers during the installation, the install script
creates an entry in this file for each. You can edit this file manually to finish
configuring the entries, make changes, or if you skipped alarm configuration
during the installation.
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Each line in the /etc/phonesfile takes the form
pager PHONENUMBER,,,,,,_ALARMCODE_
where PHONENUMBER is the complete phone number needed to reach the pager,
including any digits required to access an outside line. If the pager service
requires a PIN code, include it as well.
The PHONENUMBER string can contain valid DTMF keys (the digits 0 through 9,
#, and *) and commas.
Commas represent modem-specific delays, typically one second each. If the
modem must dial an 8 or 9 to access an outside line, you may need to insert a
comma or two after that number to allow enough time for accessing the line.
Also, different paging service providers may require more or less delay to
complete a call or process a PIN code. Therefore, you must adjust the number
of commas properly for each pager entry. This may be a trial-and-error
When an alarm occurs, the placeholder _ALARMCODE_is replaced with the
actual numeric alarm code, which the modem sends as DTMF digits. Don’t
change the keyword pageror the placeholder _ALARMCODE_. Modify only the
dialing string between these two items.
Here are two sample pager entries (the second includes a PIN code):
# Page George in New York office
pager 9,,18005551212,,,,,_ALARMCODE_
# Page local support through on call pager number.
pager 8,6466672,,,1455#,,,,,,,_ALARMCODE_#
Some paging services ask callers to press the pound key after entering the
number to be sent. As the second example shows, you can accommodate this
by putting a pound sign after _ALARMCODE_
.
Email Notification Setup
You can have the ReadiVoice system send alarm notifications to designated
email addresses, either instead of or in addition to telephone numbers.
Assuming that your Sun is set up properly for UNIX mail, you can send mail
to any known user (such as root) on your network. To send an email
elsewhere (such as an Internet email address), your network must have an
SMTP server with Internet access. Contact your network specialist about
configuring your network for email notification.
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The /rahome/bin/emailfile specifies the addresses to which to send email
alerts. When you enter email addresses during the installation, the install
script creates an entry in this file for each. You can edit this file manually to
make changes or if you skipped alarm configuration during the installation.
Each line in the file contains a single email address. For instance, it might
contain the following:
root
cnow
george_w@confservices.com
Testing Remote Alarm Notification
To verify that remote alarm notification is working properly and to adjust the
timing in the pager notifications, you need to generate alarm notifications for
testing. You can do so at the command prompt. To create a test alarm, sending
the alarm code 123456789, enter this command:
/rahome/bin/alarm.sh 123456789
The pagers you entered should receive the page. Depending on the paging
service, this may take a few minutes.
To test the entire process, including alarm code generation and both email and
pager notification, create loss of capacity events. The simplest approach is to
remove spans manually until you reach the capacity threshold for the bridge
or system. For instance, with the default threshold setting, you can generate a
bridge capacity alarm for a 480-port T1 bridge by removing four or more
spans.
If you want to generate a system capacity alarm for a multiple-bridge system,
you may want to temporarily increase the threshold setting first.
The default settings for alarm monitoring work well for most situations, but
you can modify them if necessary. The sections that follow describe how to
change the alarm settings, run the ReadiVoice Monitoring Tool, and interpret
the alarm codes.
The ReadiVoice Monitoring Tool must run continuously in order to provide
alarm monitoring and remote alarm notification.
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Changing the Alarm Settings
To change the default alarm settings:
1
2
3
4
Establish a Telnet connection to the Sun server and log in as cnow.
Change directory to /rahome/bin
.
careful not to change anything else in the file.
5
Save and close the .odprocrcfile.
Figure 5-3 The alarm settings in the .odprocrc file
[alarm]
system_id = 01
check_mibs = 0
interval = 900
sysmax_capacity = 480
systreshold = 80
brgmax_capacity_1 = 480
brgtreshold_1 = 80
brgmax_capacity_2 = 240
brgtreshold_2 = 80
[]
// [alarm]
Table 5-4
Alarm section parameters of .odprocrc file
Keyword
Description
Uniquely identifies your ReadiVoice system. Don’t change this number.
system_id
check_mibs
Boolean. If this is set to 1, at each polling interval, the ra_monitor.pl
script checks the MIB variables listed in the ra_monitor.mibsfile and
prints them to standard output (the terminal window in which it’s running).
Interval in seconds between polling for alarms. The default value of 900
(15 minutes) works well for large or busy systems. For a single-bridge or
moderately busy system, you can reduce this to 500 – 700 seconds.
interval
For short-term testing or troubleshooting purposes, you can set this
value to less than 20 seconds; but, you may notice that the ReadiVoice
system slows down at this setting.
Maximum capacity (number of ports) of your system. Don’t change this
number.
sysmax_capacity
brgmax_capacity
Maximum capacity (number of ports) of an individual bridge. Don’t
change this number.
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Table 5-4
Alarm section parameters of .odprocrc file (continued)
Keyword
Description
Percentage of maximum system capacity at which a capacity alarm is
triggered. The default value is 80, which means an alarm occurs if the
number of available ports in your system drops to 80% or less of
sysmax_capacity. Change this number to the threshold you want to use
for system capacity alarms.
systreshold
Percentage of maximum bridge capacity at which a capacity alarm is
triggered. The default value is 80, which means an alarm occurs if the
number of available ports in an individual bridge drops to 80% or less of
brgmax_capacity. Change this number to the threshold you want to use
for bridge capacity alarms.
brgtreshold
Continuously Monitoring for Alarms
To monitor your system continuously for alarms:
1
On the Sun server, log in as root, open an XTerm window, and switch
user to cnow. You can’t use a Telnet session for this purpose.
2
3
Change directories to /rahome/bin.
Type ./ra_monitor.pland press ENTER.
Leave this window open as long as you want to monitor the system. If you
close this XTerm session, monitoring stops.
Alternatively, you can run the monitoring tool as a background process by
entering nohup ./ra_monitor.pl &. Then you can close the XTerm window.
Figure 5-4 ReadiVoice Monitoring Tool running
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Interpreting the Alarm Codes
When an alarm condition occurs, the ReadiVoice Monitoring Tool sends a
numeric alarm code to designated pager numbers and email addresses. The
numeric code consists of from six to ten digits that identify the source of the
alarm and its nature. The tables below help you understand the alarm codes.
Table 5-5 describes the format for the alarm codes.
Table 5-5
Alarm code format
First 2 digits
Next 2 digits
Next 2 digits
Next 2 digits
Next 2 digits
System ID (customer specific).
Error code.
Number of bridges logged in.
Bridge with problem; only used with error code 06.
Bridge capacity threshold; only used with error code 06.
Table 5-6 shows the possible error code values (the third and fourth digits in
the alarm code).
Table 5-6
Error code descriptions
Code
01
Description
CACS is down.
02
Mplex server is down (can’t run Operator interface).
03
SNMP daemon is down (no monitoring because system status isn’t known).
Non-busyout bridge logged out (bridge crash).
04
05
System capacity has dropped below threshold (by default, 80% of maximum).
06
Bridge capacity has dropped below threshold (by default, 80% of maximum). This
indicates spans in alarm state.
For example, if you receive an alarm code of 0806020180, it tells you that on
your system (08), there is a bridge capacity alarm (06). Two bridges are logged
in (02) and Bridge 1 is the one in alarm state (01). Its capacity has dropped
below the threshold of 80% (80).
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Configuring the ReadiVoice System
Enabling Conference Recording
You can enable your subscribers to record their conferences by configuring
your ReadiVoice system to connect to a recording device or service using a
dial-out connection.
Recording can be initiated using a designated touch-tone command or a
button on the Moderator interface. The sections that follow describe how to
configure and enable a recording connection and how it works.
Configure and test conference recording only after operational hours.
Caution!
Enabling Conference Recording Connections
To set up your system for conference recording connections:
1
Verify that the voice prompt files used for recording are appropriate for
your business needs or update them as necessary. The voice prompt files
Polycom recommends professional recording of all voice prompt files. For
2
3
Determine the telephone number that the ReadiVoice system can use to
connect to the external recording device or service and the DTMF key
sequence that the ReadiVoice system can use to initiate recording.
As user cnow, edit the .odprocrcfile to set up the recorderPhoneand
recorderSetupStringparameters to implement the recording
4
5
Save your changes to .odprocrc. Then, as root, stop and restart the
ReadiVoice system (see “Stopping and Restarting ReadiVoice Processes”
on page 111) to load the new settings.
Test the recording capability:
a
b
c
Set up or open a test subscriber account.
In Provisioning, turn on Recorder Dial Out for the test account.
If you used the {externId}variable in the setup string, make sure the
Subscriber External ID is populated.
d
e
f
Save the changes to the test account.
Start a conference as the test subscriber.
Use the touchtone command for your system to start recording.
Speak into the conference to create a recording that you can verify.
End the conference.
g
h
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7
Verify that the test conference was recorded by listening to the recording.
If the test conference was recorded, continue to the next step. If not,
review the previous steps or call your Polycom Global Services
representative for assistance.
Instruct provisioners to turn on Recorder Dial Out for subscribers who
need to be able to record conferences.
Voice Prompts for Conference Recording
The voice prompt files used during the recording process are:
rec_change_failed.wav
rec_conf_full.wav
rec_not_enabled.wav
rec_part_join_reminder
rec_rejoin_reminder.wav
rec_setup_failed.wav
rec_start_prompt.wav
rec_started.wav
rec_stop_prompt.wav
rec_stopped.wav
rec_subs_join_reminder
rec_wait.wav
Recording Configuration Variables in the .odprocrc File
To enable conference recording, the recorderPhoneand
recorderSetupStringvalues in the [modules] [bridgeInt]section of
.odprocrcmust be specified:
•
For recorderPhone, specify the telephone number of the recording device
or service.
•
For recorderSetupString, use valid DTMF keys (0-9, *, and #) and the
identify each recording of each conference.
Table 5-7
Variable
Variables you can use in recorderSetupString
Description
Subscriber ID.
{subId}
Contents of External ID field from SUBSCRIBERINFO table.
Credit card number from BILLINGINFO table.
{externId}
{ccnum}
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Table 5-7
Variable
Variables you can use in recorderSetupString
Description
Credit card expiration month.
{ccmonth}
{ccyear}
{confStart}
{now}
Credit card expiration year.
Conference start time in UNIX seconds.
Time in UNIX seconds when the recording device or
service is dialed.
Subscriber’s access code from ASSIGNEDPHONES table.
{accesscode}
Conference security code set by the subscriber for the
conference, if any.
{securitycode}
Access phone number that the subscriber used to dial
into the system (empty if the system dialed out to the
subscriber).
{accessnum}
{partpin}
Participant password from SUBSCRIBERINFO table (if
subscriber has two-password call flow; otherwise,
field is empty).
Subscriber password from SUBSCRIBERINFO table.
Subscriber group ID from SUBSCRIBERINFO table.
{subpin}
{subgroupId}
{subscribernote1}
Contents of first optional subscriber note field
(Subscriber User Field A in Provisioning) from
SUBSCRIBERDETAIL table.
Contents of second optional subscriber note field
(Subscriber User Field B in Provisioning) from
SUBSCRIBERDETAIL table.
{subscribernote2}
Contents of first optional billing note field (Billing User
Field A in Provisioning) from the BILLINGINFO table.
{billingnote1}
{billingnote2}
{confId}
Contents of second optional billing note field (Billing
User Field B in Provisioning) from BILLINGINFO table.
Conference ID (from CDR_POST_CONF table) of
conference from which the recorder dial-out is being
initiated.
Since the recorderSetupStringis outpulsed as DTMF digits, data referenced
by the variables that you use must be numeric. If a retrieved variable string
contains a non-numeric character, the system omits the entire string for that
variable from the outpulsed data.
In particular, be aware that the ReadiVoice system performs no validation on
the two subscriber note fields or the two billing note fields during the
provisioning process. In most system configurations, it also doesn’t validate
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the External ID field. If you intend to use any of these fields for recorder setup
information, it's up to you to ensure that they contain numeric data and that
your provisioning process populates the fields properly.
How Conference Recording Works
conference recording.
Figure 5-5 Example of conference recording settings in the .odprocrc file
[modules]
[bridgeInt]
...
recorderPhone = 5551212
recorderSetupString =1,,4,,5*{accesscode}#*{confStart}##
...
[]
[]
In a system using the example settings shown, the following sequence occurs
when a subscriber selects the recording option:
1
The subscriber selects the option to record the conference in one of two
ways:
— Entering the touchtone command (default is *2).
— Clicking the Record button in the Moderator.
2
3
The ReadiVoice system dials the phone number specified by
recorderPhone. In the example, this is 5551212.
The system sends the DTMF tones specified by recorderSetupString
substituting actual DTMF tones for the variables. With the example
replacing the variables described in brackets):
,
1 (pause) 4 (pause) 5 * [subscriber’s access code] # *
[start time of conference] # #
In this example, the system uses the DTMF sequence 1,,4,,5 to maneuver
through an IVR menu, with pauses between DTMF keys to allow the IVR
platform to process them (each comma represents a 20-millisecond pause).
It sends * and # to delimit the fields and ## to mark the end of the setup
string to the recording device. This is one possible implementation.
4
5
Conference recording begins.
Recording ends when the conference ends or when the recorder
telephone line is disconnected.
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Configuring the ReadiVoice System
Enabling SNMP Logging
The SNMP logging function writes system usage and performance data to log
files on an ongoing basis. For more information about SNMP, logging, and the
log files, see .
Perform the following procedure only after operational hours.
Caution!
To turn on SNMP logging:
1
2
As cnow, open .odprocrcfor editing.
In the [snmpMgr]section of the file, edit the logging parameters as
3
Save your changes to .odprocrc. Then, as root, stop and restart the
ReadiVoice system (see “Stopping and Restarting ReadiVoice Processes”
on page 111) to load the new settings.
Table 5-8
SNMP logging parameters of the .odprocrc file
Parameter
Value/Description
1
0
Enables SNMP logging.
Disables SNMP logging.
logPrint
n
Polling interval, in seconds, at which log file records are
written (minimum interval is 600 seconds).
logInterval
snmplogfile
filename.log
The full path and name of the SNMP log file (default is
/rahome/bin/snmp/snmp.log).
filename.log
histogramlogfile
The full path and name of the histogram log file (default is
/rahome/bin/snmp/histogram.log).
Figure 5-6 The SNMP logging settings in .odprocrc
[machines]
...
[snmpMgr]
...
logPrint = 1
loginterval = 600
snmplogfile = /rahome/bin/snmp/snmp.log
histogramlogfile = /rahome/bin/snmp/histogram.log
...
[]
...
[]
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Changing the Web Server Port
This procedure allows you to change the Web server port used by the
ReadiVoice HTML pages (Administration, Provisioning, Operator login, and
Moderator) from port 80 to something else.
This procedure doesn’t modify the port used by Java applets, such as the
Operator, Channel State Monitor, and SNMP Monitor.
By default, the script moves the Web server to port 40004, but you can specify
any port.
Perform the following procedure only after operational hours.
Caution!
To change the Web server port:
1
Log into the ReadiVoice CACS as root and change to the /rahome/utils
directory.
2
3
4
Make sure the webportscript has 755 file permissions.
Type./webportand press ENTER to run the script.
When prompted for what port to use, do one of the following:
— Press ENTER to accept the default, 40004
.
— Type a different port number and press ENTER.
When prompted for the host name, do one of the following:
— Press ENTER to accept the host name shown in brackets.
— Type a different host name and press ENTER.
5
6
Watch the messages that scroll by as the script stops the Web server,
changes the port, and restarts the Web server on the new port.
If the script succeeds, it displays a final message like this:
Startup: listening to http://host, port portnum as web
In place of host and portnum, you’ll see the host name and port number
you specified.
If there is an error binding to the port you specified, the script displays a
message like this:
Unable to bind to port portnum - error 125
This means that the port is already in use by another process.
7
If the port you specified is already in use, check the /etc/servicesfile to
see which ports are in use, choose another one, and rerun the script. If
you need help choosing a port or if you see other error messages, call
your Polycom Global Services representative.
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8
9
If the script succeeded, test the URL that includes the new Web server
port number by pointing a browser to:
http://hostip:portnum/index2.html
Use the IP address of the SUN (CACS) for hostip and the port number
you specified for portnum. For example, an actual URL might be:
http://127.0.0.1:40004/index2.html
Ensure that users of this system access the new URL.
Enabling Operators to Unlock Conferences
If your operators use the ReadiVoice Operator application for Windows
(which we refer to simply as the Windows Operator), they can lock
conferences. By default, however, the Windows Operator is configured so that
they can’t unlock conferences.
If you want your operators to be able to unlock conferences, you can easily
enable this feature. Note that you can enable operator unlock for a specific
operator station (PC), but not for a specific operator or operators. To enable
this feature globally, you must make the change on each operator station.
To enable operator unlock on an operator station:
1
Log into the operator station as a user with administrator rights and
make sure the Windows Operator isn’t running.
2
In Windows Explorer (or another file manager), navigate to the Windows
Operator installation folder. Typically, this is:
C:\Program Files\Polycom ReadiVoice\WinOp.Installer\
3
Find the Operator.exe.configfile and open it in Notepad or another
text editor.
This is an XML file that defines a number of key-value pairs. The one you
need to change looks like this:
<add key="WinOp.Client.AllowUnlock" value="false" />
Change the valueof the key to true, so that the line reads:
<add key="WinOp.Client.AllowUnlock" value="true" />
Save and close the file. Log out of the operator workstation.
4
5
The next time an operator logs in and starts the Windows Operator, it
reads the modified configuration file and enables conference unlock.
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7
Customizing & Branding Your
ReadiVoice System
This chapter describes some of the customizing and branding options in the
ReadiVoice system. It assumes that you know Solaris UNIX, Sun servers,
personal computers, software configuration, telephony, and networking.
Customizing Touchtone Commands
The touchtone (DTMF, or dual tone multi-frequency) commands that the
ReadiVoice system recognizes are defined in the bridge initialization file for
the system, /rahome/bridge/scripts/ive.ini
.
Before changing your system’s touchtone commands, read:
•
•
•
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Rules and Guidelines for Customizing Commands
entry, the touchtone key sequence is to the left of the equal sign, and the
command to which it’s assigned is to the right. Entries preceded by two slashes
(//) are commented out (disabled). You can enable or disable commands, or
change the touchtone key sequences that invoke the commands.
Any changes you make to the entries in the [DTMF_CMDS]section of the
ive.inifile absolutely must obey the following rules:
•
Change only the left side of the equal sign, which specifies the DTMF key
sequence assigned to a command. The command names, beginning with
eCMD_, must not be modified in any way.
•
Each key sequence must be unique (not assigned to more than one
command). If the same key sequence is assigned to two commands, one of
them must be commented out (disabled).
•
•
Each key sequence must consist of from one to three DTMF keys.
Only characters that appear on standard telephone keypads are valid
DTMF keys. They are
0
,
1
,
2
,
3
,
4
,
5
,
6
,
7
,
8
,
9
,
*
, and
#.
When you make DTMF command changes, you also need to change the
affected voice prompts. For instance, any change in commands will require at
least a new “help” prompt, which describes the available commands. See
Appendix A for detailed information about the voice prompts used in the
ReadiVoice system.
Procedure for Customizing Commands
To change the DTMF key sequences associated with commands or to enable or
disable commands:
1
Check to determine which voice prompt (.wav) files are impacted by the
command change you want to make. Replace the .wavfiles if necessary.
We recommend that .wavfiles be recorded professionally. Contact your
Polycom Global Services representative if you want help in obtaining
appropriate .wavfile recordings.
2
3
Log into the CACS as root and switch user to cnow.
Open the ive.inifile for editing and find the [DTMF_CMDS]section near
the end of the file.
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Figure 7-1 [DTMF_CMDS] section of default ive.ini file
//////////////////////////////////////////////////////////////////////
// The following entries configure the DTMF commands for users in
// conference.
// NOTE: All DTMF commands must be unique, duplicate values will cause
// ive.ini read error
//////////////////////////////////////////////////////////////////////
[DTMF_CMDS]
** = eCMD_HELP
*0 = eCMD_OP_REQUEST
00 = eCMD_CONF_REQUEST
*1 = eCMD_DIAL_OUT
*2 = eCMD_CONF_RECORD
// *2 = eCMD_ROLL_CALL_CONF
*3 = eCMD_CONF_CONFIG
// *4 = eCMD_LOCK_TOGGLE
*4 = eCMD_CONF_LOCK
*5 = eCMD_CONF_UNLOCK
// *6 = eCMD_MUTE_TOGGLE
*6 = eCMD_MUTE
*7 = eCMD_UNMUTE
*8 = eCMD_CONF_CONTINUE
*9 = eCMD_ROLL_CALL_PRIV
*# = eCMD_PART_COUNT
## = eCMD_MUTE_ALL
// ## = eCMD_MUTE_ALL_TOGGLE
99 = eCMD_UNMUTE_ALL
// #1 = eCMD_LISTEN_ONLY_TOGGLE
#1 = eCMD_LISTEN_ONLY
#2 = eCMD_UNLISTEN_ONLY
#5 = eCMD_WR_PROCESS
#6 = eCMD_WR_ANNOUNCE_TOGGLE
#8 = eCMD_CONF_END_CONF
1 = eCMD_WR_PART_TO_CONF
2 = eCMD_WR_PART_IGNORE
3 = eCMD_WR_PART_DISCONNECT
[]
4
To disable a command, comment out its line by putting two slashes (//)
at the beginning of the line.
Don’t delete lines that you don’t want to use. Comment them out so that
they’re ignored, but you don’t lose the syntax information. This allows
you to enable them easily later by just removing the slashes.
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5
To enable a disabled command, remove the two slashes (//) at the
beginning of the line.
Some commands are mutually exclusive. They can be enabled only if the
Features” for discussions of these commands.
Caution!
6
To change the DTMF key sequence that invokes a command, edit the two
characters to the left of the equal sign for that command.
7
8
When you’re finished making changes, save and close the ive.inifile.
Reboot the bridges so that they read the updated ive.inifile and load
the new .wavfiles.
9
Set up a test conference and verify that the touchtone commands you
modified work as intended and that the voice prompts are correct.
10 Make sure your teleconferencing users are aware of the changes.
Detailed Information for Specific Touchtone Features
This section provides more detailed information about the following
touchtone command features and how they can be implemented:
•
•
•
•
•
Lock/Unlock
Conference locking and unlocking can be controlled either with two separate
commands or with a single command that toggles the lock state of the
conference.
option; *4locks a conference and *5unlocks it. The relevant lines look like
this:
//*4 = eCMD_LOCK_TOGGLE
*4 = eCMD_CONF_LOCK
*5 = eCMD_CONF_UNLOCK
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With the standard toggle option, pressing *4locks a conference and pressing
*4again unlocks it. To enable this configuration, change the relevant lines to
look like this:
*4 = eCMD_LOCK_TOGGLE
//*4 = eCMD_CONF_LOCK
//*5 = eCMD_CONF_UNLOCK
If you enable the eCMD_LOCK_TOGGLE command, you must disable both the
eCMD_CONF_LOCK and eCMD_CONF_UNLOCK commands.
Caution!
You can, of course, change the DTMF key sequence(s) used in either
configuration, but you’ll also have to change the relevant voice prompts (see ).
Mute/Unmute
Muting and unmuting of individual lines can be controlled either with two
separate commands or with a single command that toggles the mute state of
the channel.
option; *6mutes the line of the person entering the command and *7unmutes
it. The relevant lines in ive.inilook like this:
//*6 = eCMD_MUTE_TOGGLE
*6 = eCMD_MUTE
*7 = eCMD_UNMUTE
With the standard toggle option, pressing *6mutes the line of the person
entering the command and pressing *6again unmutes it. To enable this
configuration, change the relevant lines in ive.inito look like this:
*6 = eCMD_MUTE_TOGGLE
//*6 = eCMD_MUTE
//*7 = eCMD_UNMUTE
If you enable the eCMD_MUTE_TOGGLE command, you must disable both the
eCMD_MUTE and eCMD_UNMUTE commands.
Caution!
You can, of course, change the DTMF key sequence(s) used in either
configuration, but you’ll also have to change the relevant voice prompts (see ).
You can also customize the system’s mute confirmation behavior. See
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Mute All / Unmute All
Muting and unmuting of all participants can be controlled either with two
separate commands or with a single command that toggles the mute state of
the conference.
option; ##mutes all the ordinary participants (not subscriber or operator) in
the conference and 99unmutes them. The relevant lines in ive.inilook like
this:
## = eCMD_MUTE_ALL
// ## = eCMD_MUTE_ALL_TOGGLE
99 = eCMD_UNMUTE_ALL
With the standard toggle option, pressing ##mutes all the ordinary
participants and pressing ##again unmutes them. To enable this
configuration, change the relevant lines in ive.inito look like this:
// ## = eCMD_MUTE_ALL
## = eCMD_MUTE_ALL_TOGGLE
// 99 = eCMD_UNMUTE_ALL
If you enable the eCMD_MUTE_ALL_TOGGLE command, you must disable both
the eCMD_MUTE_ALL and eCMD_UNMUTE_ALL commands.
Caution!
You can, of course, change the DTMF key sequence(s) used in either
configuration, but you’ll also have to change the relevant voice prompts (see ).
You can also customize the system’s mute confirmation behavior. See
Listen Only / Unlisten Only
Turning listen only mode on and off can be controlled either with two separate
commands or with a single command that toggles the listen-only state of the
conference.
option; #1puts the conference into listen only mode (all ordinary participants
are muted and can’t unmute themselves) and #2turns off listen only mode
(enabling ordinary participants to speak again). The relevant lines in ive.ini
look like this:
// #1 = eCMD_LISTEN_ONLY_TOGGLE
#1 = eCMD_LISTEN_ONLY
#2 = eCMD_UNLISTEN_ONLY
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With the standard toggle option, pressing #1turns on listen only mode and
pressing #1again turns it off. To enable this configuration, change the relevant
lines in ive.inito look like this:
#1 = eCMD_LISTEN_ONLY_TOGGLE
// #1 = eCMD_LISTEN_ONLY
// #2 = eCMD_UNLISTEN_ONLY
If you enable the eCMD_LISTEN_ONLY_TOGGLE command, you must disable
both the eCMD_LISTEN_ONLY and eCMD_UNLISTEN_ONLY commands.
Caution!
You can, of course, change the DTMF key sequence(s) used in either
configuration, but you’ll also have to change the relevant voice prompts (see ).
Private and Conference Roll Call
The ReadiVoice system supports two different implementations of the roll call
feature, which plays the recorded names of participants, along with a
participant count:
•
Private roll call plays the roll call only to the requesting line. It’s available
to both the subscriber and participants.
•
Conference roll call plays the roll call to the entire conference. It’s available
to only the subscriber.
call option, leaving the conference roll call command commented out
(disabled). The relevant lines in ive.inilook like this:
// *2 = eCMD_ROLL_CALL_CONF
...
*9 = eCMD_ROLL_CALL_PRIV
You can have either or both of these commands enabled. Of course, no two
commands can use the same DTMF key sequence, so if you want to enable the
conference roll call option you must do one of the following:
•
•
•
Disable the private roll call command and assign its key sequence (*9) to
the conference roll call command.
Disable the conference record command so that the private roll call
command can use its key sequence (*2).
Assign some other unused key sequence (for instance, #9) to the
conference roll call command.
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Customizing Confirmation Sounds
The [CustomSounds]section in ive.inicontrols how the system confirms the
following changes:
•
•
•
•
Mute and unmute of an individual line.
Mute all and unmute all.
Listen only on and off.
Waiting room announcements/reminders on and off.
Each of these changes can be set to a tone confirmation (two-tone sequence), a
specific confirmation.wavfile, or nothing at all. Confirmations of mute and
unmute are played to the individual line. Confirmations of the other changes
are played to the conference.
The default settings provide tone confirmations for mute and unmute and for
the toggling of waiting room announcements/reminders, but don’t provide
any confirmations for the global mute and listen only commands. The relevant
lines in ive.inilook like this:
[CustomSounds]
[Mute]
MUTE_SOUND = TONE
UNMUTE_SOUND = TONE
MUTE_ALL_SOUND = NONE
UNMUTE_ALL_SOUND = NONE
LISTEN_ONLY_SOUND = NONE
UNLISTEN_ONLY_SOUND = NONE
[]
[WaitingRoom]
CONF_ANNOUNCE_ON = TONE
CONF_ANNOUNCE_OFF = TONE
[]
[]
To change any of these settings, change the value after the equal sign. The
following values are valid:
TONEto play a two-tone DTMF sequence.
NONEto play no sound at all (no confirmation).
filename.wavto play the voice prompt file specified.
If you specify a .wavfile to play, the ReadiVoice system looks for that file first
in the directory containing the subscriber’s prompt set (for instance, this is
/rahome/bridge/sound/2/adpcmfor Prompt Set 1 on a system with
InnoVox 480 bridges). If the file isn’t there, it looks in the directory that
contains the default prompt set installed with the ReadiVoice system
(/rahome/bridge/sound/1/adpcmon a system with InnoVox 480 bridges).
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The default prompt set installed with the ReadiVoice system contains six .wav
files with spoken messages designed to work with these commands:
global_muted.wav
global_unmuted.wav
private_muted.wav
private_unmuted.wav
wr_notification_off.wav
wr_notification_on.wav
For more information about these files, their default contents, and how to
Customizing the Moderator Interface
By default, the ReadiVoice Moderator displays a Polycom logo image file on
both the login/logout page and the Moderator interface page. You can
customize the Moderator in two ways:
•
Replace the Polycom logo image file to brand both the login/logout page
and the Moderator interface page with your company logo.
•
Install additional image files to customize the Moderator interface page
for specific subscriber groups.
Branding the Moderator with Your Company Logo
The default image file is the image file used for everyone except the subscriber
groups that you customize.
The default image file is:
/rahome/ns-home/docs/HTMLMOD/default/images/moderator/corporate_logo.png
Its size is 35x146 pixels (HxW).
In order to install a new image file, you must be logged into the CACS server
as root.
To replace the existing (Polycom logo) image file with your own company’s
default image file:
1
Create the replacement image file you want to use as the default file.
This procedure is simplest if the replacement image is a .pngfile of the
same or similar size. But, you can use a .gifor .jpgfile. Other file formats
are also possible (such as .swf), but may impose browser limitations or
require plug-ins. If your replacement differs in size from the one we
provide, be sure to test it to ensure that the appearance is acceptable.
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2
3
4
5
Use file transfer protocol (ftp) to copy the replacement image file to a
local directory on the CACS server. Be sure to use binary transfer mode.
Open a Telnet session to the CACS server, log in as cnow, and switch user
to root.
Make a backup of the existing default image file in case you need to
restore it.
If you’re replacing the existing image file with another .pngfile, rename
your replacement file corporate_logo.pngand copy it to the
/rahome/ns-home/docs/HTMLMOD/default/images/moderator/
directory, overwriting the existing file.
This is by far the easiest solution, and it doesn’t require any special
precautions when your ReadiVoice system is upgraded.
6
If your replacement file is a different file format, or if you don’t want to
use the same file name for some other reason, do the following:
a
Copy the replacement file to a location in /rahome/ns-home/docs/
/rahome/ns-home/docs/HTMLMOD/default/images/moderator/is
(
OK).
b
c
Change directories to /rahome/ns-home/docs/HTMLMOD/default/
.
Open moderator.htmlfor editing and find the string
corporate_logo.pngin two different places (in both places, it’s the
srcparameter in an <IMG>tag).
d
Change both occurrences of corporate_logo.pngto the name of the
replacement file. Change the path to the replacement file if necessary.
The web server’s document root directory is /rahome/ns-home/docs/
,
so the path to the replacement file must be relative to that directory.
e
Save and close moderator.html
.
If you customize the Moderator by editing the moderator.htmlfile,
remember that the HTML file is likely to be replaced whenever you
upgrade your ReadiVoice system. Therefore, you may have to repeat this
process each time your system is upgraded.
7
In your browser, access the Moderator to verify that the new image file
appears.
Customizing the Moderator for a Specific Subscriber Group
You can further customize the Moderator interface for a specific subscriber
group by installing a group-specific image file. On the Moderator interface
page (but not the login page), the group-specific file replaces the default image
file for members of that group.
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To customize the Moderator for a specific subscriber group:
1
In the System Administration Subscriber Groups page, determine the
Group ID of the subscriber group.
2
Create the image file you want to use for this subscriber group. Give it a
clearly identifiable name, such as group#.png, where is the Group ID of
#
the subscriber group you want to customize.
We recommend using a .pngfile of the same size as the default, but you
can use other formats, such as .gifor .jpg. If your replacement differs in
size from the default, be sure to test it to ensure that the appearance is
acceptable.
3
4
5
Use file transfer protocol (ftp) to copy the replacement image file to a
local directory on the CACS server. Be sure to use binary transfer mode.
Open a Telnet session to the CACS server, log in as cnow, and switch user
to root.
If it doesn’t already exist, create the directory
/rahome/ns-home/docs/moderator2/group_custom. Make sure the
directory is owned by root:weband has permissions set to 750
Change to the /rahome/ns-home/docs/moderator2/imagesdirectory
and create a new subdirectory named group#, where is the Group ID of
the subscriber group you want to customize. Make sure the directory is
owned by root:weband has permissions set to 750
.
6
#
.
7
8
Copy the custom image file for the group to the new group#
subdirectory. Make sure the file is owned by root:weband has
permissions set to 640
From the /rahome/ns-home/docs/moderator2directory, copy the
default_group.inifile to group_custom/group#.ini, where is the
Group ID of the subscriber group you want to customize. Make sure the
file is owned by root:weband has permissions set to 640
.
#
.
9
Change to the group_customdirectory and open the new group#.ini
file for editing.
10 Replace the reference to images/workspaceimage.jpgwith
images/group#/group#.png(where group#is the subdirectory you
save and close the file.
11 In your browser, log into the Moderator as a subscriber from the group
you customized and verify that the new image appears.
Repeat the above procedure to customize another subscriber group.
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Working with Prompt Sets
for the messages and prompts played to subscribers and participants. When
you create a subscriber group, you select a prompt set for the group. The
ReadiVoice system supports the use of up to 20 prompt sets. This permits you,
for example, to provide customized foreign language prompts for certain
customers.
Custom welcome messages (the initial greetings that callers hear) are handled
differently from the prompt set messages. See “Customizing Greetings and
The installation process puts the default voice prompt files into the following
locations:
•
•
•
/rahome/bridge/sound/1/adpcmcontains the .wavfiles in ADPCM
format (for InnoVox 480 bridges).
/rahome/bridge/sound/1/g711contains the .wavfiles in µ-law format
(for InnoVox 4000 bridges).
/rahome/bridge/sound/2contains the same two format-specific
subdirectories and their files. The appropriate subdirectory (based on
your bridge type) is the source for Prompt Set 1.
If you want to add a second prompt set, you must create the directory
/rahome/bridge/sound/3, and put the new .wavfiles into the appropriate
format-specific subdirectory (adpcmor g711) for your system. The third set’s
files must be in the appropriate subdirectory of /rahome/bridge/sound/4
,
and so on.
When it responds to events by playing specific voice prompt files, the ReadiVoice
system expects to find files with the specific names listed. When you create and
Caution!
install new prompt sets or replace prompts in existing prompt sets, be sure the files’
names are correct. If the system can’t find a file in the specific prompt set directory,
it looks in the appropriate subdirectory of the default directory (/rahome/bridge/sound/1).
Don’t change or delete any of the files in this directory!
prompt files, their contents, and their length.
When you install additional prompt sets, you must run the tnl_mklnscript to
create the appropriate symbolic links for the initial greetings. See “About
for greetings and what the tnl_mklnscript does.
The Prompt Sets page in the System Administration interface lets you add
new prompt sets to the ReadiVoice system, rename prompt sets, or delete
prompt sets from the system (deleting a prompt set doesn’t remove the files
from the disk). To change the prompts themselves, you must replace the .wav
files for the prompts you want to change.
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For InnoVox 480 bridges, the .wavfiles must be in 4-bit IMA ADPCM format
(8000 samples/second, 16-bit mono, compressed 4:1). For InnoVox 4000 bridges,
they must be in 8-bit µ-law format (8000 samples/second, 16-bit mono or higher).
For the highest sound quality, prompt set files should be recorded in a
professional sound studio. If you’re upgrading from an InnoVox 480 bridge
and want the same prompt set, create the µ-law encoded .wavfiles from the
original recordings. Converting a 4-bit IMA ADPCM file into 8-bit µ-law won’t
provide noticeable quality improvements.
Adding a Prompt Set
To add a prompt set:
1
In the System Administration navigation bar, click Prompt Sets.
sets. The Path column shows the directory path where the ReadiVoice
system expects to find each prompt set’s files.
The path shown doesn’t include the format-specific subdirectory (adpcm
or g711) that contains the actual files, but points to its parent directory.
2
In the empty field at the bottom of the Name column, type a name for the
new prompt set. Then click the Add button.
The system confirms that the prompt set has been added and provides a
link for returning to the Prompt Sets page.
3
4
Click the link to reload the Prompt Sets page.
The Prompt Sets page appears, with the new entry at the end of the list.
Make a note of the directory path assigned to the new set.
The remaining steps require you to have access to the CACS server or
establish a Telnet session to it in order to install the files.
5
6
Use file transfer protocol (ftp) to copy the new prompt set’s .wavfiles to
a local directory on the CACS server. Be sure to use binary transfer mode.
On the CACS server, create the directory assigned to the new set. For
instance, if this is the second prompt set, you’d create the directory
/rahome/bridge/sound/3.
7
In the new prompt set directory, create the format-specific subdirectory
needed for your system (adpcmor g711). If the new prompt set uses some
of the same .wavfiles as an existing prompt set, it’s easiest to copy this
directory and its contents from the existing set.
8
9
Copy or move the new .wavfiles for this prompt set from the temporary
overwriting any existing files that you’re replacing.
Repeat the preceding steps to add another prompt set.
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10 When you’re finished adding prompt sets, create the symbolic links for
running the tnl_mklnscript:
a
b
c
Log into the CACS as root and, in an XTerm window, enter tcsh
.
Change to the /rahome/bindirectory.
Run the make link script by entering tnl_mkln
.
d
Reboot the bridge(s).
Figure 7-2 Prompt Sets page
Renaming a Prompt Set
To change the name of a prompt set:
1
In the navigation bar, click Prompt Sets.
The Prompt Sets page appears, listing the available prompt sets.
Edit the prompt set’s name. Then click its Commit button (blue “c”).
The system confirms that the prompt set has been modified.
2
Repeat the above procedure to rename another prompt set.
Deleting a Prompt Set
To remove a prompt set from your ReadiVoice system:
1
In the navigation bar, click Prompt Sets.
The Prompt Sets page appears, listing the available prompt sets.
Find the prompt to remove and click its Delete button (red “–”).
The system confirms that the prompt set has been deleted.
2
Repeat the above procedure to delete another prompt set.
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Deleting the prompt set from within System Administration only makes it
unavailable to the ReadiVoice system. It doesn’t delete the .wavfiles from the
CACS server or remove the directory that contains them.
Customizing Greetings and Related Messages
In a ReadiVoice non-routed system and a ReadiVoice IP system, the bridge
plays the initial greeting. You can configure ReadiVoice to use up to 50 custom
greetings, so that it selects the appropriate greeting based on the DNIS (dialed
number identification service) digits that the network delivers to the bridge.
Custom welcome messages aren’t available in a routed (INCR) system because
the network that routes calls to the bridges plays the initial greeting (welcome
message) that callers hear. The carrier stores and plays this message.
The DNIS-specific initial greeting files you need depend on your system
configuration:
•
•
•
If your system is Non-routed Private, create DNIS-specific versions of the
helo_inbound.wavfile.
If your system uses only the Traditional Shared call flow, create
DNIS-specific versions of the cr_access_code_prompt.wavfile.
If your system also uses the Two-Password Shared call flow, create
DNIS-specific versions of the pw_prompt.wavfile.
When you run the tnl_mklnscript, it creates the DNIS-specific symbolic links
page 181 for a description of how symbolic links are used for greetings and
what the tnl_mklnscript does.
In addition to the initial greeting, several related prompts can also be
customized based on DNIS. The DNIS-customizable files are:
cf_call_not_completed
cr_access_code_wrong_.wav
_cr_access_code_wrong.wav
cr_conf_ending.wav
helo_inbound.wav
cf_conf_full_disconnect.wav
cr_access_code_collection_failure.wav
cr_access_code_good.wav
cr_access_code_prompt.wav
cr_access_code_wrong.wav
pw_prompt.wav
This feature is called Welcome on DNIS, and it works as follows:
1
During the call-setup protocol, the network delivers the call’s DNIS
number to the bridge.
2
The bridge looks in /rahome/bridge/sound/greetings/adpcmor
/rahome/bridge/sound/greetings/g711(depending on bridge type)
for a .wavfile that starts with the DNIS digits received and matches the
needed event.
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For example, if the system is Non-routed Private, and the bridge receives
the digits 1111, it plays 1111helo_inbound.wavto the caller if the file
exists. Another caller, whose DNIS digits are 2222, would hear
2222helo_inbound.wavif the file existed.
3
If the DNIS-specific .wavfile doesn’t exist, the bridge plays the
corresponding non-customized greeting (helo_inbound.wav, for
example) in /rahome/bridge/sound/1/adpcmor
/rahome/bridge/sound/1/g711(depending on bridge type).
To implement this feature:
1
Create the custom messages you want to use in the required format, and
add the DNIS digits to the beginning of each standard file name.
including the wording of the default versions. The appendix also describes
the bridge-specific file formats.
To use the same greeting (or other customized message) for several
DNISes or access numbers, create multiple copies of the same .wavfiles,
each named with its DNIS.
The custom file names must match exactly the DNIS digits delivered to the bridge.
Typically, on a non-routed system, a DNIS is a four-digit number. If your system
receives more DNIS digits, however, the file names must include all the digits.
Caution!
2
Place the new appropriately-named files into the correct format-specific
subdirectory (adpcmor g711) of the/rahome/bridge/sound/greetings/
directory.
ReadiVoice uses the default greeting-related files (from the appropriate
subdirectory of /rahome/bridge/sound/1) for any DNIS that doesn’t
have its own .wavfiles. To use your own default greetings, replace the
supplied default greeting files with ones of the same names containing the
messages you want.
3
4
a
b
c
Log into the CACS as root and, in an XTerm window, enter tcsh.
Change to the /rahome/bindirectory.
Run the make link script by entering tnl_mkln
.
d
Reboot the bridge(s).
Inform provisioners of the system of the specific access numbers that
you’ve associated with custom welcome messages.
If you’ve created corresponding custom prompt sets, give provisioners
more information.
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Creating Your Own Indexed WAVE Files
ReadiVoice uses an indexed WAVE file to store the short sound clips that are
played in various combinations to speak numbers, dates, and times. Indexed
WAVE files can also be used for other purposes.
Using the indexed WAVE file format, you can support several languages by
supplying the necessary sound clips and building separate files for each
create indexed WAVE files.
For more detailed information about WAVE files in general and indexed
Building Indexed WAVE Files
Polycom’s indexed WAVE file is the concatenation of several individual
WAVE (.wav) files into a single file. Individual sound clips are accessed by
using an index or offset into the file.
For InnoVox 480 bridges, both the source WAVE files and the output indexed
WAVE file store the audio data using 4-bit IMA ADPCM format
(8000 samples/second, 16-bit mono, compressed 4:1). For InnoVox 4000
bridges, both the source WAVE files and the output indexed WAVE file store
the audio data using 8-bit µ-law format (8000 samples/second, 16-bit mono).
The two types of indexed WAVE files you can build are:
Predefined — contains sound clips for numbers, dates, times, etc. The number
of sound clips in the file is language specific and may vary from language to
language. Grammar rules for the language are applied during playback.
Generic — contains a fixed set of sound clips used to say words or phrases for
the application. No grammar rules are applied during playback.
Both types of indexed WAVE files are built the same way, using the genvoice
utility. You specify the key file to use, the directory location for the sound clips,
and the output file name. Only setting up the key file is different.
Key File Guidelines
Follow these formatting guidelines when creating key files:
•
•
Each entry must be on one line (carriage returns aren’t legal white space).
Lines containing a pound sign (
as comment lines).
#) in the first column are ignored (treated
•
•
Blank lines are ignored.
Identifiers can be separated by any amount of white space. White space is
defined as tabs and spaces.
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•
Only in predefined key files, the first line contains LANG=followed by the
upper-case English name of the desired language. These names are
converted into an appropriate integer for the indexed WAVE file.
Recognized languages are:
— ENGLISH
— GERMAN
— JAPANESE
— FRENCH
— SPANISH
— CHINESE
— SWEDISH
•
Each key entry on one line consists of:
— An index number and a semicolon (
— The entry name and a semicolon.
— The name of the WAVE file.
;).
•
•
The index numbers in the predefined key files are fixed by the application.
Don’t change them.
The name identifiers are for the benefit of the humans using the file,
providing meaningful information about the entries. They aren’t used by
the application.
•
If a WAVE file name includes a path, it’s relative to the source file
Setting Up a Predefined Key File
The key files for predefined indexed WAVE files look like this:
LANG=language
0; name_00; waveFile_00
.
.
.
x; name_xx; waveFile_xx
Don’t change the leading index numbers (x) or name identifiers (name_xx).
When you change predefined key files, edit only the waveFile_xx fields so
that they point to the correct sound files for the key entries. If necessary,
include a path that’s relative to the directory you specify in the genvoice
command to build the indexed WAVE file (see “Running the Build Process”
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The example below shows part of a German predefined key file:
LANG=GERMAN
0; number_0; null.wav
1; number_1; eins.wav
2; number_2; zwei.wav
.
.
.
107; 30th; dreizigste.wav
108; der; der.wav
109; ein; ein.wav
110; eine; eine.wav
Due to grammatical differences, the number of entries needed may vary from
language to language. This is why Polycom supplies the predefined key files
for the supported languages.
Creating a Generic Key File
An application can take advantage of indexed WAVE files for outgoing voice
prompts. The format of the key files is identical to the format of predefined key
files except that no language is specified.
You create a key file in order to build an application indexed WAVE file. It
should look like this, without a line specifying the language:
0; name_00; waveFile_00
.
.
.
x; name_xx; waveFile_xx
Your key file can contain any number of entries, and, of course, you can specify
any name (name_xx) you like for each entry. As with the predefined key file,
edit the waveFile_xx field for each entry to reference the correct sound file.
The example below shows part of a a generic key file:
0; welcome to the system;cf_you_subs_prompt.wav
1; howdy y’all; howdy.wav
.
.
.
107; goodbye y’all; goodbye.wav
Because no grammar rules are employed during playback, the number of
entries in generic key files can be the same from language to language.
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Running the Build Process
To build an indexed WAVE file, use the genvoiceutility in the
/rahome/bridge/binaries/tools/directory as follows:
1
2
Create or edit the key file as described above.
Be sure that all the input WAVE files are located in the proper directory
or directories.
3
At the command prompt, enter:
genvoice -i inputFile -d voiceDir -o outputFile[-v]
inputFile specifies the key file to use. If the key file isn’t in the current
directory, inputFile must specify its path. InnoVox provides predefined key
files for each language that it supports.
voiceDir specifies the directory where the source WAVE files are located. If
the source files aren’t in this directory, the entries in the key file must contain
the relative path from this directory.
outputFile specifies the file name for the indexed WAVE file being created.
If outputFile doesn’t specify a path, the file is created in the current directory.
-vspecifies that the WAVE file is generated with µ-law encoding. If -visn’t
specified, IMA ADPCM encoding is used.
InnoVox 480 bridges require IMA ADPCM encoded WAVE files.
InnoVox 4000 bridges require µ-law encoded WAVE files. If you’re upgrading
from an InnoVox 480 bridge and want the same prompt set, you should create
the µ-law encoded WAVE files from the original recordings at
8000 samples/sec., 16-bit mono (or higher quality). Converting a 4-bit IMA
ADPCM into 8-bit µ-law doesn’t provide noticeable quality improvements.
The genvoiceutility displays an appropriate error message and terminates if:
•
•
•
A source file is in the wrong format or can’t be found.
The key file is in the wrong format or can’t be found.
The number of entries found in the predefined key file doesn’t match the
expected number for the specified language. This check doesn’t apply to
generic key files.
An indexed WAVE file can be played using any program that supports
WAVE, but the indices will be ignored. Some sound editors may not be able to
play the file correctly because of the compression used.
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Details About the WAVE File Format
A WAVE file stores digital (sampled) audio data in Microsoft’s Resource
Interchange File Format (RIFF). RIFF is a variant of Electronic Arts’ Interchange
File Format (IFF). Unlike IFF, the Intel-oriented RIFF uses Little Endian byte
order. For more information about RIFF, visit the Microsoft Developer
RIFF files consist of logical units of data called chunks. The first chunk in a RIFF
file is the RIFF chunk. All other chunks in the file are subchunks of the RIFF
chunk.
At a minimum, all WAVE files contain:
•
The top-level RIFF chunk, which consists of three parts:
— The chunk identifier, which is the four-character code RIFF
.
— The size of the data member chunks, a DWORD(four-byte integer). The
size is everything after the first eight bytes (the chunk identifier and
the size DWORDitself).
— The data member, which begins with a DWORDthat specifies the form
type of the chunk (in this case WAVE). All other chunks in the file are
considered part of the data member of the RIFF chunk.
•
•
A Format subchunk that describes the data (sample rate, bit resolution,
type of compression, and so forth).
A Data subchunk that contains the actual sampled audio data.
But, RIFF (like IFF) is extensible. WAVE files can contain, and developers can
define, additional types of chunks to provide additional data and
functionality.
Polycom added two new chunks to the WAVE file format to create its indexed
section.
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Indexed WAVE File Specifications
In Polycom’s indexed WAVE files, the Data chunk consists of a set of
individual sound files that are concatenated together. An Index chunk enables
Polycom applications to locate the individual sound clips and quickly piece
them together to play phrases, numbers, dates, and so forth. The sound clips
can be common words, digits, alphabetic letters, or even short phrases.
An indexed WAVE file has:
•
•
The chunk identifier, which is the four-character code RIFF.
The size of the data member chunk, a DWORD(4-byte integer). The size is
everything after the first eight bytes (the chunk identifier and the size
DWORDitself).
•
The data member is the largest portion of the file. It consists of:
— A DWORDthat specifies the form type of the chunk (in this case WAVE).
— A Format subchunk that describes the data (sample rate, bit
resolution, type of compression, and so forth).
— A Language chunk identifying the language of the sound clips.
— An Index chunk containing all of the offset information. The Index
chunk contains:
»
»
»
The size (DWORD) of the Index chunk.
The number (DWORD) of sound clips in the WAVE file.
The offset location (DWORD) and length in bytes (DWORD) of each
sound clip in the file.
— The Data subchunk contains the actual sampled audio data, or sound
clips. These are the data subchunks pulled out of individual WAVE
files and concatenated here.
The specifications for these chunks use Microsoft’s RIFF notation. Look up
RIFF notation on the Microsoft Developer Network
Language Chunk
The Language chunk identifies the language associated with the indexed
WAVE file. This lets applications using the file apply the appropriate
grammar, where required, for building number or date phrases correctly in
that language. An indexed WAVE file can have only one language identifier.
Only predefined indexed WAVE files use the language field. You specify other
languages using the appropriate predefined key file. Supported languages are:
•
•
•
CHINESE
ENGLISH
FRENCH
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•
•
•
•
GERMAN
JAPANESE
SPANISH
SWEDISH
The Language chunk consists of the chunk identifier (lang), the size of the data
member (DWORD), and the two-byte integer (WORD) language field. In RIFF
notation:
<lang-ck> ->
lang(
<size:DWORD>
// chunk ID
// data member size
// language
<language:WORD>)
Index Chunk
The Index chunk identifies the starting position (offset) and length of each
segment (sound clip) in the file. The chunk begins with the chunk identifier
bmak(for “bridge multi-access key”). This is followed by a 4-byte integer
(
DWORD) containing the length in bytes of the Index chunk’s data member (that
is, the size of the chunk, not including the chunk ID and size fields).
The data member itself begins with a 4-byte integer (DWORD) specifying the
number of index entries. This is followed by the indexPairstruct containing
the actual index data. In RIFF notation:
<bmak-ck> ->
bmak(
<size:DWORD>
// chunk ID
// size of index chunk
<numIdxPairs:DWORD> // number of index pairs
<indexPair> …)
// index pair list
<indexPair> ->
struct
{
DWORD offset;
// Absolute offset of
// segment, in bytes.
DWORD length;
}
// Length of segment, in bytes.
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6
Diagnostics and Troubleshooting
This chapter describes how to perform some diagnostic and troubleshooting
tasks.
Viewing Critical Logs
Logs link in the ReadiVoice home page’s navigation bar. It gives you access to
critical troubleshooting information that may require corrective action.
Processing the log files imposes significant processor and memory burdens on
your computer and can take some time. Log files are typically useful only to
trained technicians, and even trained technicians typically need assistance
from Polycom Global Services.
Figure 6-1 Critical Logs page
On this page, select BRIDGE LOGS and a date. and then click Commit Query.
ReadiVoice searches the raw log files for the date you entered and retrieves
critical entries.
When the system is operating normally, the critical logs should be nearly
empty. If the system is in a testing mode or other abnormal state that generates
large raw log files, your query may take a very long time.
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The critical log file shows you critical entries about bridge processes, such as:
•
•
•
•
•
Bridge up or down.
Span trouble (doesn’t apply to IP systems).
Port capacity changes.
Bridge clock source changes or failures.
Abnormal bridge task deletion or suspension.
Figure 6-2 shows a bridge critical log page. Typically, a properly running
system has few or no entries.
Each critical log entry contains a link to the raw log file from which it came. To
investigate an entry further, click its link. The system retrieves a portion of the
raw log file that shows the entry in context (from 100 lines before to 100 lines
after the entry).
Figure 6-2 Bridge critical log page
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Diagnostics and Troubleshooting
Database Troubleshooting
This section suggests some methods for diagnosing the database and
investigating database problems.
Checking for Informix Server Engine Errors
To determine if an error has occurred in the Informix server engine, check the
/usr/informix/online.logfile for erroror panicmessages, using the grep
command to search for those words.
With the standard configuration, Informix temporarily stores results data in
the /tmpdirectory. Lack of sufficient free space in this directory can be the
cause of server engine errors.
Use the dfcommand to check the disk space in the /tmpdirectory (see
unneeded files to free up space for database data. Call your Polycom Global
Services representative if you need help in determining what you can safely
remove.
Checking for Slow SQL Queries
If you suspect the database is slowing down the system, check for an SQL
query that takes much more time than the others:
1
In a CACS XTerm window, log in as root, switch user to informix, and set
the Informix environment by entering tcsh
.
2
Enter onstat –g ses
.
Informix displays a list of sessions and their memory usage.
3
4
Locate the session that uses the most memory and note its session ID.
Enter onstat –g sql
.
Informix displays a list of active queries and their session IDs.
5
Check the list of active queries for the session you identified in 3. If it’s
listed, enter onstat –g sql ID(where IDis the session ID number of
the session you identified in 3).
Informix displays the running query.
6
Contact your Polycom Global Services representative to determine if this
query is taking more memory than it should.
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Logging Database Activity with the Database Monitoring Script
Your ReadiVoice installation includes a Perl script, dbMon.pl, that logs
Informix database information (such as number of active sessions and amount
of memory used per session) into a file that can be used later for analysis.
If your system seems to be slowed by SQL queries that are taking inordinately
long, it may be helpful to run this script and provide the resulting log file to
your Polycom Global Services representative.
Use this script only if you are trained and knowledgeable in working with Solaris,
Informix, and the ReadiVoice system.
Caution!
The dbMon.plscript:
•
•
•
Resides in the /rahome/database/scriptsdirectory.
Must be run by root.
Should normally be run in the background.
The script creates a log file named YYYYMMDDhhmmssdbMon.log, where
YYYYMMDDhhmmss is the system date and time (using a 4-digit year and 24-hour
time) when the script is run. For example, if you run dbMon.plon August 14th,
2002, at 2:22:37 PM, it creates a log file named 20020814142237dbMon.log
.
The log files are written to the /rahome/database/scripts/archives/
directory.
Running the dbMon.plScript
The syntax for running the dbMon.plscript is:
dbMon [-r n &]
To use the runtime option (-r), replace
the script to run. If you omit this option, the script runs for the default duration
of 5 minutes. Use to make it a background process.
n
with the number of minutes you want
&
For example, enter dbMon.pl –r 20 &at the command prompt to run the
script in the background for 20 minutes.
Contents of dbMon.plLog File
The dbMon.plscript writes data to the log file at one-second and ten-second
intervals.
Logged every second:
Active SQL statements: This section shows the session IDs and the statement
types (SELECT
,
UPDATE
,
INSERTor UPDATE STAT) of the SQL statements that are
currently running.
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Active sessions: This section shows the session IDs, user types (such as root,
web, or cnow), and total used memory of active database connections.
With this information, you can identify queries that are being run constantly
against the database and determine the approximate time (to the nearest
second) that it takes to run these queries.
Logged every ten seconds:
Number of reads/writes per user thread: This section shows the number of
read and write operations performed on the database by each thread.
Processes waiting on latches, locks and transactions: This section shows the
process IDs of processes waiting on latches, locks, or transactions.
With this information, you can identify the most active session (based on the
memory usage) and the processes that are being suspended or waiting on
locks or latches.
Telephony Troubleshooting
Your ReadiVoice system includes two ways of testing the function of the
bridges:
•
•
Dial-out testing, using the telephony test utility, diagnose
.
Dial-in testing, using the LOG_CHAN_DEBUGparameter in the ive.inifile.
These procedures are normally used during installation or upgrades, but may
be run as needed, as long as the affected bridges are not being used for live
conferencing.
Dial-Out Channel Testing
The diagnosetelephony test utility is run from the command line. You can
specify these parameters:
•
•
The ID of the bridge you want to test.
The equipment on the specified bridge that you want to test as:
— Any range of cards, spans, and channels.
— A specific card, span, and channel.
— Any combination of ranges and specific cards, channels, or spans.
In an IP system, there are no spans or channels as such. You can only
specify the cards that you want to test.
•
The phone number you want the bridge to call to test the equipment you
specified.
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When you run the diagnoseutility:
1
The bridge you’re testing calls you at the number you specified, using the
first bridge resource to be tested.
2
When you answer the call, the bridge plays three numbers that identify
the card, span, and channel on which the call is being made.
In an IP system, the bridge plays a single number indicating the card on
which the call is being made.
3
4
When you hang up, the bridge calls you on the next resource to be tested
and repeats the process until it completes the test.
For each test call completed, the utility writes a line to a log file.
The log file is specified in the diagnosticlogfileparameter of the
.odprocrcfile. The default setting is:
diagnosticlogfile = /rahome/bin/TTU.log
If this file exists, the utility appends the new test call records to it.
Otherwise, it creates the file, adds a heading, and writes the records.
To start dial-out testing:
1
2
3
Open a Telnet session to the CACS as user cnow, and switch user to root.
Change to the /rahome/utilsdirectory.
Type ./diagnoseand press ENTER.
You see:
To quit, press ’q’:
To Go To Previous Step, Press ’p’:
Enter the ID of the Bridge to Test:
4
Type the ID of the bridge and press ENTER. To find bridge IDs, go to the
You see:
Enter the Phone Number to Dial Out To:
5
Enter the telephone number at which you want the bridge to call you
exactly as it should be dialed. Type only the digits, with no parentheses
or dashes, and then press ENTER.
You see:
Enter Range of Cards:
From:
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6
Enter the logical number of the first card you want to test (card
numbering begins with zero).
You see:
Enter Range of Cards:
To:
7
Enter the logical number of the last card you want to test. To test only the
card you entered in step 6, enter the same number again.
If this is an IP system, The script displays a blank line, and your phone
If this is a PSTN system, you see:
Press -1- To Test All Spans On A Board Or -2- For Individual
8
Choose one of these options:
— To test both spans on the selected card or cards, enter 1. Then skip to
— To test a single span, enter
2.
You see:
Enter the Span Port Number:
9
Enter the number of the span you want to test (
You see:
0
or 1).
Press -1- To Test All Channels On A Span Or -2- For A Particular
Channel:
10 Choose between these options:
— To test a single channel, enter
2.
You see:
Enter the Channel:
11 Enter the number of the channel you want to test.
The script displays a blank line, and your phone rings.
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12 Answer the phone, listen to the message, and then hang up.
For each call, you hear three numbers indicating the card, span, and
channel being used for the call (or, in an IP system, a single number
indicating the card).
The details of each test call are recorded in the log file, so you need only
note a channel if it has audio problems, such as static.
13 Choose one of these options:
— Continue answering the phone each time it rings, listening to the
message and then hanging up, until the bridge stops calling you.
— To stop the dial-out test process before it completes, type CTRL + C. This
also stops the updating of the diagnostic log file.
14 View the diagnostic log file (by default, /rahome/bin/TTU.log).
A line for each test call shows the card, channel, and span numbers, and
the outcome of the call. For a successful test of channel 13 on span 0, card
3 of the bridge you tested, the log file would contain the line:
3/0/13
SUCCESS
In an IP system, the line for each test call shows the bridge and card
numbers and the outcome of the call. For a successful test of card 0 of
bridge 1, the log file would contain the line:
1/0
SUCCESS
If you find telephony problems, such as any outcome other than “SUCCESS,” print
the file for your records or save the file under a new name or in a safe location. To
enable Polycom Global Services to help you, be sure have this file available to send
us.
Caution!
Dial-In Channel Testing
A dial-in channel test script runs automatically when you set the
LOG_CHAN_DEBUGparameter in the ive.inifile to . By default, this parameter
is set to
1
0
.
Don’t enable dial-in channel testing on a live conferencing bridge or bridges. When
dial-in testing is turned on, callers hear a debug message (stating the card, span,
and channel numbers—card only for IP—on which their call has reached the
bridge) before the standard welcome message.
Caution!
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To start dial-in channel testing:
1
Remove the bridge or bridges from service and wait until all conferencing
activity has finished.
2
3
Open the ive.inifile for editing.
In the [Debug]section, set LOG_CHAN_DEBUGto
The edited line should read:
1.
LOG_CHAN_DEBUG = 1
4
5
Save and close the edited ive.inifile.
Reboot the bridge or bridges so that the processors reread the ive.ini
file and begin running in channel-debug mode.
6
Call the ReadiVoice system and listen for the board, span, and channel
numbers to be played before the welcome message (or, in an IP system, a
single number indicating the card).
7
8
As you use the system, make a note of the specific channels (or cards), if
any, associated with static or other audio problems.
When you’re finished testing and want to make the system available for
live conferencing, reopen the ive.inifile that you modified and restore
the LOG_CHAN_DEBUGsetting to
The edited line should read:
LOG_CHAN_DEBUG = 0
0.
9
Save and close the edited ive.inifile.
10 Reboot the bridge or bridges so that the processors reread the ive.ini
file and stop running in channel-debug mode.
11 Call the ReadiVoice system and verify that the standard welcome
message plays without a debug message preceding it.
The system can now be restored to normal conferencing service.
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A
Voice Prompts and Call Flows
This appendix describes the default voice prompts installed with the
ReadiVoice system, how they’re used, and the ReadiVoice call flows.
Overview
You can use multiple sets of voice prompts in the ReadiVoice system, which
you can customize for specific needs. Before you start customizing your voice
prompts or installing additional voice prompt sets, it helps to familiarize
yourself with the default voice prompt set.
The installation process puts the default voice prompt files into the following
locations:
•
•
•
/rahome/bridge/sound/1/adpcmcontains the .wavfiles in ADPCM
format (for InnoVox 480 bridges).
/rahome/bridge/sound/1/g711contains the .wavfiles in µ-law format
(for InnoVox 4000 bridges).
/rahome/bridge/sound/2contains the same two format-specific
subdirectories and their files. The appropriate subdirectory (based on
your bridge type) is the source for Prompt Set 1.
If you want to add a second prompt set, you must create the directory
/rahome/bridge/sound/3, and put the new .wavfiles into the appropriate
format-specific subdirectory (adpcmor g711) for your system. The third set’s
files must be in the appropriate subdirectory of /rahome/bridge/sound/4
,
and so on.
The Prompt Sets page in the Administration interface lets you add new
prompt sets to the ReadiVoice database (making them available for
provisioning), rename prompt sets, or delete prompt sets from the ReadiVoice
database (making them unavailable, but not removing the files themselves).
To change the prompts themselves, you must replace the .wavfiles for the
prompts you want to change. Use the information in this appendix to help you
determine the appropriate text for your customized messages.
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The default prompt set includes a file, silence.wav, containing only a very
brief silence. If you want to replace a specific prompt with nothing (that is, you
don’t want any message to play in the situation where that prompt is called),
you can replace it with a copy of silence.wavrenamed to replace the .wavfile
that you don’t want played.
The format of the .wavfiles depends on the type of bridges in your system:
InnoVox 480 bridges: All .wavfiles must be in 4-bit IMA ADPCM format
(8000 samples/second, 16-bit mono, compressed 4:1). If your sound editor
application offers multiple format options, use the DVI option when saving.
InnoVox 4000 bridges: All .wavfiles must be in 8-bit µ-law format
(8000 samples/second, 16-bit mono). If you’re upgrading from an
InnoVox 480 bridge and want the same prompt set, you should create the
µ-law encoded .wavfiles from the original recordings at 8000 samples/sec.,
16-bit mono (or higher quality). Converting a 4-bit IMA ADPCM into 8-bit
µ-law won’t provide noticeable quality improvements.
You can use the filecommand to determine whether a .wavfile is in ADPCM
Figure A-1 Checking the format of .wav files
jaba </rahome> cnow> file bridge/sound/1/adpcm/0.wav
bridge/sound/1/adpcm/0.wav:
WAVE file, ADPCM format (mono)
jaba </rahome> cnow> file bridge/sound/1/g711/0.wav
bridge/sound/1/g711/0.wav:
jaba </rahome> cnow>
WAVE file, G711 format (mono)
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About Initial Greetings
In a non-routed system, the bridge plays the initial greeting to callers. The file
that’s appropriate depends on your system configuration:
•
•
If your system is Non-routed Private, then the initial greeting is always the
helo_inbound.wav file, which doesn’t prompt the caller to enter anything.
If your system is Non-routed Shared, then, depending on the call flow, the
caller could have either an access code (Traditional Shared) or a subscriber
or participant password (Two-Password Shared). And the system may
have both those call flows enabled.
The default prompt set includes two greeting files with different prompts:
•
cr_access_code_prompt.wavspecifically asks for an access code. It’s
appropriate if only the Traditional Shared call flow is enabled.
•
pw_prompt.wavis more generic, asking for a “passcode.” It’s appropriate
if both Traditional Shared and Two-Password Shared are enabled, since
the system doesn’t know initially which call flow applies to a caller.
The bridge doesn’t actually look for either of those files. Instead, it looks for the
cr_code_prompt.wavfile, which is a symbolic link to one of the above.
During system installation, the tnl_mklnscript must be run. Among other
things, it sets up cr_code_prompt.wavas a symbolic link to the appropriate
prompt, depending on system configuration:
•
If your system uses only the Traditional Shared call flow, the symbolic link
points to the cr_access_code_prompt.wavfile.
•
If your system also uses the Two-Password Shared call flow, the symbolic
link points to the pw_prompt.wavfile.
If you ever change those call flow settings, add prompt sets, or set up custom
greetings, you must run the tnl_mklnscript again so that it can change the
symbolic links, if necessary, or add them for the new prompt sets or custom
greetings.
In the case of custom greetings (see “Customizing Greetings and Related
DNIS for which it finds the target file.
For instance, if the /rahome/bridge/sound/greetings/adpcmdirectory
contains a file named 1234gencode_prompt.wav(and the Two-Password
Shared call flow is enabled), the tnl_mklnscript creates a symbolic link to it
named 1234pass_access_code.wav
.
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Voice Prompt File Reference
This section documents the default voice prompt files installed in the
alphabetical order.Use it as a reference when contemplating new voice prompt
sets and to determine which voice prompts may need to be customized.
When it responds to events by playing specific voice prompt files, ReadiVoice
expects to find files with the filenames listed. When you create and install new
prompt sets or replace prompts in existing prompt sets, be sure the files’ names are
correct.
Caution!
Table A-1
All voice prompts, alphabetized
Configuration or
Feature
Length
(sec.)
Required Filename
Default Prompt
“Zero”
All
All
All
All
All
All
All
All
All
All
All
All
All
All
All
All
All
0.4
0.5
0.6
0.7
0.5
0.8
0.7
0.8
0.9
1.0
0.8
1.0
0.4
0.6
0.4
0.6
0.5
0.wav
“One”
1.wav
“Ten”
10.wav
11.wav
12.wav
13.wav
14.wav
15.wav
16.wav
17.wav
18.wav
19.wav
2.wav
“Eleven”
“Twelve”
“Thirteen”
“Fourteen”
“Fifteen”
“Sixteen”
“Seventeen”
“Eighteen”
“Nineteen”
“Two”
“Twenty”
“Three”
20.wav
3.wav
“Thirty”
30.wav
4.wav
“Four”
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Table A-1
All voice prompts, alphabetized (continued)
Configuration or
Feature
Length
(sec.)
Required Filename
40.wav
Default Prompt
“Forty”
All
All
All
All
All
All
All
All
All
All
All
0.5
0.6
0.6
0.4
0.5
0.6
0.6
0.5
0.5
0.8
0.7
6.4
“Five”
5.wav
“Fifty”
50.wav
“Six”
6.wav
“Sixty”
“Seven”
“Seventy”
“Eight”
“Eighty”
“Nine”
60.wav
7.wav
70.wav
8.wav
80.wav
9.wav
“Ninety”
90.wav
“I’m sorry, the system is currently unable
to change this option, please try again
later.”
Account Options
menu
ao_change_failure.wav
“To return to conference, press star.”
Announcement
setting process
1.8
6.4
ao_conf_return_prompt.wav
ao_failure.wav
“I’m sorry, the system is currently unable
to change this option, please try again
later.”
Account Options
menu
“Account Options feature overview…”
Account Options
menu
2.7
7.0
ao_feature_overview.wav
ao_initial_menu.wav
ao_main_menu_.wav
“To start or join your conference, press 1.
To change Account Options, press 2.”
Account Options
menu
“Account Options. Changes to your
account options affect both the current
conference and future conferences,
except that quick start and listen only
changes affect only future conferences.
You may change your account options at
any time by returning to this menu.”
Account Options
menu
“For an overview of Account Options,
press 9.”
Account Options
menu
2.5
_ao_main_menu.wav
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Table A-1
All voice prompts, alphabetized (continued)
Configuration or
Length
(sec.)
Required Filename
Default Prompt
Feature
“This option is not currently available.”
Account Options
process
ao_none_avail.wav
“I’m sorry, this feature is not enabled.”
Account Options
menu
2.5
1.2
1.3
1.3
1.3
ao_not_enabled.wav
ao_press_1.wav
“Press 1.”
“Press 2.”
“Press 3.”
“Press 4.”
“Press 5.”
“Press 6.”
“Press 7.”
“Press 8.”
Account Options
menu
Account Options
menu
ao_press_2.wav
Account Options
menu
ao_press_3.wav
Account Options
menu
ao_press_4.wav
Account Options
menu
ao_press_5.wav
Account Options
menu
ao_press_6.wav
Account Options
menu
ao_press_7.wav
Account Options
menu
ao_press_8.wav
“To return to the previous menu, press
star.”
Account Options
menu
3.7
3.0
3.7
ao_previous_menu_prompt.wav
ao_wrong_key.wav
cc_active_reminder.wav
“I’m sorry, that entry is not valid.”
Account Options
menu
“The conference will be allowed to
continue after you disconnect.”
Continuation
[Sub call flow]
“To change the Auto Continuation
option…”
Continuation
3.0
cc_auto_change_prompt_.wav
_cc_auto_change_prompt.wav
cc_auto_off.wav
[Autocontinuation
setting]
<silence>
Continuation
[Autocontinuation
setting]
“Auto Continuation is off.”
Continuation
2.9
[Autocontinuation
setting]
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Table A-1
All voice prompts, alphabetized (continued)
Configuration or
Length
(sec.)
Required Filename
Default Prompt
Feature
“Auto Continuation is on.”
Continuation
2.9
cc_auto_on.wav
[Autocontinuation
setting]
“Auto Continuation turns continuation on
for all of your conferences, so they can
continue after you disconnect.”
Continuation
cc_auto_overview.wav
[Account Options
help]
“To turn Auto Continuation off, press 1.”
“To turn Auto Continuation on, press 1.”
“I’m sorry, this feature is not enabled.”
Continuation
4.5
4.5
2.5
cc_auto_set_off_prompt.wav
cc_auto_set_on_prompt.wav
cc_change_not_enabled.wav
[Autocontinuation
setting]
Continuation
[Autocontinuation
setting]
All
[In conf sub]
“The subscriber has disconnected. The
conference will now end.”
Continuation off
4.1
8.3
cc_conf_end_disconnect.wav
cc_off.wav
“The conference will end when you
disconnect. To allow the conference to
continue after you disconnect, press star
8.”
Continuation
available
[In conf sub]
“The conference will be allowed to
continue after you disconnect. To set the
conference to end when you disconnect,
press star 8.”
Continuation
[In conf sub]
8.9
4.2
cc_on.wav
“I’m sorry, your call could not be
completed. Please try your call again.”
All
cf_call_not_completed.wav
cf_caller_count_.wav
[Dial-in]
“There are…”
Conference Entry
with Count
[Sub call flow]
[Part call flow]
“…participants in the conference.”
Conference Entry
with Count
_cf_caller_count.wav
[Sub call flow]
[Part call flow]
“I’m sorry, the conference you are
attempting to join is full.”
All
3.6
cf_conf_full_disconnect.wav
[Sub call flow]
[Part call flow]
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Table A-1
All voice prompts, alphabetized (continued)
Configuration or
Length
(sec.)
Required Filename
Default Prompt
Feature
“Thank you. Your conference will now
begin.”
All
2.5
cf_conf_start.wav
[Sub call flow]
“Thank you.”
All (default part
disconnect
message)
0.7
cf_default_non_part_
disconnect.wav
“You have been disconnected by the
system. Please try your call again.”
All (default non-part
disconnect
message)
4.4
4.4
cf_default_part_
disconnect.wav
“You have been disconnected by the
system. Please try your call again.”
All
cf_eip_disconnect.wav
[Sub call flow]
[Part call flow]
“You are the first participant to join the
conference, please stand by.”
Quick Start
4.6
6.1
cf_first_caller.wav
[Part call flow]
“For a menu of available commands,
press star star at any time during the
conference.”
All
cf_in_conf_help_cmd.wav
[Sub call flow]
“The following conference commands are
available:
All
22.0
cf_in_conf_part_cmd_
help.wav
[Part - in-conf dtmf]
To request an Operator, press star zero.
To request an Operator join your
conference, press zero zero.
To mute your line, press star 6.
To un-mute your line, press star 7.
To hear a list of conference participants,
press star 9.
Participant count, press star pound.”
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Table A-1
All voice prompts, alphabetized (continued)
Configuration or
Feature
Length
(sec.)
Required Filename
Default Prompt
“The following conference commands are
All
cf_in_conf_subs_cmd_
help.wav
available to the subscriber:
[Sub - in-conf dtmf]
To request an operator, press star zero.
To request an operator join your
conference, press zero zero.
To dial out, press star 1.
To add a recorder to conference, press
star 2.
To change the conference entry and exit
announcement options, press star 3.
To lock your conference, press star 4.
To unlock your conference, press star 5.
To mute your line, press star 6.
To unmute your line, press star 7.
To allow the conference to continue after
you disconnect, press star 8.
To hear a list of conference participants,
press star 9.
To hear participant count, press star
pound.
To mute all but the subscriber, press
pound pound.
To unmute all, press 99.
To place all callers in Listen Only except
the subscriber, press pound 1.
To turn off Listen Only, press pound 2.
To process callers in the Waiting Room,
press pound 5.
To toggle the waiting room notifications
announcement setting on or off, press
pound 6.”
“Someone has already joined this
conference using the subscriber
password. You'll join the conference as a
participant.”
All
cf_late_subscriber.wav
[Part call flow]
<hold music>
All
30.0
cf_music.wav
[Part call flow]
“The subscriber has joined the
conference. Your conference will now
begin. Please stand by.”
Large conference
music hold
extender
cf_music_hold_to_conf.wav
[Part call flow]
“You will now be placed into conference.”
“You will now be placed into conference.”
All
2.4
2.4
cf_non_part_join.wav
cf_part_join.wav
[Sub call flow]
All
[Part call flow]
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Table A-1
All voice prompts, alphabetized (continued)
Configuration or
Length
(sec.)
Required Filename
Default Prompt
Feature
“The subscriber has not yet arrived.
Please stand by.”
Quick Start off
[Part call flow]
4.3
cf_part_music_wait.wav
“Please wait.”
All
cf_please_standby.wav
[Newcomer, dial-in
or dial-out]
“You have been disconnected by the
conference moderator. Good-bye.”
Dial-out
3.7
4.6
6.6
cf_subs_disconnect.wav
cf_subs_first_caller.wav
cf_you_subs_prompt.wav
[Dial-out call flow]
“You are the first participant to join the
conference, please stand by.”
All
[Sub call flow]
“The subscriber has not yet arrived. If you
are the subscriber, please press the star
key now.”
Prompt for
subscriber
[Dial-in, trad.
shared]
“I’m sorry, your call could not be
completed. Please try your call again.”
Shared Access
System
4.2
4.4
0.7
4.6
cr_access_code_collection_
failure.wav
[Dial-in, shared]
“You have been disconnected by the
system. Please try your call again.”
Shared Access
System
cr_access_code_
disconnect.wav
[Dial-in,shared]
“Thank you.”
Shared Access
System
cr_access_code_good.wav
cr_access_code_prompt.wav
[Dial-in, shared]
“Welcome to the Conferencing Center.
Please enter your access code followed
by the pound sign.”
Shared Access
System
[Dial-in, trad.
shared]
“I’m sorry, that entry is not valid.”
“I’m sorry, your entry…”
Shared Access
System
cr_access_code_wrong.wav
cr_access_code_wrong_.wav
_cr_access_code_wrong.wav
cr_access_number_wrong.wav
[Dial-in, shared]
Shared Access
System
1.9
5.2
[Dial-in, shared]
“… is not valid. Please enter the valid
digits followed by the pound sign.”
Shared Access
System
[Dial-in, shared]
“I’m sorry. This access number is not
correct.”
Private Access
System
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Table A-1
All voice prompts, alphabetized (continued)
Configuration or
Feature
Length
(sec.)
Required Filename
Default Prompt
Not a prompt, but a symbolic link to either
cr_access_code_prompt.wavor
Shared Access
System
cr_code_prompt.wav
pw_prompt.wav
.
[Dial-in, shared]
“This conference is not currently available.
Please try your call again later.”
All but routed and
IP
5.0
3.6
5.0
4.4
9.5
4.0
3.5
13.6
cr_conf_ending.wav
cr_conf_full.wav
cr_system_busy.wav
csn_disconnect.wav
[Dialin]
“I’m sorry, the conference you are
attempting to join is full.”
All but routed and
IP
[Dialin]
“This conference is not currently available.
Please try your call again later.”
All but routed and
IP
[Dialin]
“You have been disconnected by the
system. Please try your call again.”
Conference
Security Code
[CSC setting]
“You must set up a conference security
code. Please enter a 4 to 9 digit code
followed by the pound sign.”
Conference
Security Code
csn_mandatory_setup_
prompt.wav
[CSC setting]
“A conference security code will not be
required for this conference.”
Conference
Security Code
csn_none_set.wav
csn_prompt.wav
[CSC setting]
“Please enter the conference security
code followed by the pound sign.”
Conference
Security Code
[Part CSC flow]
“To bypass setting up a conference
security code press star now. To set up a
conference security code please enter a 4
to 9 digit code followed by the pound
sign.”
Conference
Security Code
csn_setup_prompt.wav
[CSC setting]
“Your conference security code is…”
“…To change this entry, press star now.”
“I’m sorry, that entry is not valid.”
Conference
Security Code
2.0
2.8
3.0
csn_verify_prompt_.wav
_csn_verify_prompt.wav
csn_wrong.wav
[CSC setting]
Conference
Security Code
[CSC setting]
Conference
Security Code
[Part CSC flow]
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Table A-1
All voice prompts, alphabetized (continued)
Configuration or
Length
(sec.)
Required Filename
Default Prompt
Feature
“I’m sorry, your entry…”
Conference
Security Code
1.9
5.2
8.6
csn_wrong_.wav
[Part CSC flow]
“… is not valid. Please enter the valid
digits followed by the pound sign.”
Conference
Security Code
_csn_wrong.wav
[Part CSC flow]
“I’m sorry, the conference security code
must be 4 to 9 digits long.”
Conference
Security Code
csn_wrong_length.wav
ct_auto_disconnect.wav
[CSC setting]
“This conference has exceeded the time
allowed for a single-participant
conference. You will now be disconnected.
Goodbye.”
Conference
Termination
"Are you there? If you wish to remain in
conference, please press any key now."
Conference
Termination
ct_auto_prompt.wav
“The subscriber has not arrived. Please try
your call again later.”
Quick Start off
[Music hold]
4.4
ct_conf_timeout_
disconnect.wav
“The conference will continue.”
All
ct_manual_canceled.wav
ct_manual_disconnect.wav
ct_manual_end.wav
[Sub end conf]
"The subscriber has ended the
conference. Goodbye."
All
[Sub end conf]
“The conference has ended. All
participants will be disconnected.”
All
[Sub end conf]
“To end this conference, press 1; to
continue this conference, press 2.”
All
ct_manual_prompt.wav
do_auto_disconnect.wav
[Sub end conf]
“You have been disconnected by the
system. Please try your call again.”
Dial-out
[Dial-out part call
flow]
"The dial out was unsuccessful."
Dial-out
do_auto_join_failure.wav
do_auto_join_prompt.wav
do_bad_num_.wav
[Dial-out part call
flow]
“You are being called to join a
teleconference. Press 1 to join.”
Dial-out
4.5
[Dial-out part call
flow]
<silence>
Dial-out
[Sub dial-out]
190
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Voice Prompts and Call Flows
Table A-1
All voice prompts, alphabetized (continued)
Configuration or
Feature
Length
(sec.)
Required Filename
Default Prompt
“… is not valid. Please enter the vaild
Dial-out
1.8
_do_bad_num.wav
digits followed by the pound sign.”
[Sub dial-out]
<silence>
Dial-out
do_blast_dial_subname_
announce_.wav
[Dial-out part call
flow]
"... is calling you to join a teleconference.
Please press one to join."
Dial-out
_do_blast_dial_subname_
announce.wav
[Dial-out part call
flow]
"No additional people can be added to this
call."
Dial-out
3.6
0.8
1.0
0.8
do_conf_full.wav
do_dialing.wav
do_extplay_.wav
_do_extplay.wav
[Sub dial-out]
“Dialing.”
Dial-out
[Sub dial-out]
“Extension…”
Dial-out
[Sub dial-out]
“…please.”
Dial-out
[Sub dial-out]
<sound of telephone ringing >
Dial-out (IP system
only)
do_local_ringback.wav
do_long_cmd_help.wav
“After the call is answered, to connect the
line into the conference, press star 1. To
connect the line and continue dialing,
press star 2. To disconnect the line, press
star 3. To disconnect the line and continue
dialing, press star 4.”
Dial-out
[Sub dial-out]
“I’m sorry, your call could not be
completed. Please try your call again.”
Dial-out
4.2
2.5
do_not_completed.wav
do_not_enabled.wav
[Sub dial-out]
“I’m sorry, this feature is not enabled.”
“The number you dialed is…”
Dial-out not
enabled
[Sub dial-out]
Dial-out
1.5
4.0
9.3
do_num_verify_prompt_.wav
_do_num_verify_prompt.wav
do_number_prompt.wav
[Sub dial-out]
“Press pound to proceed, or press star to
change this number.”
Dial-out
[Sub dial-out]
“To dial out, please dial the area code and
number that you wish to connect to the
conference, followed by the pound sign.
To return to the conference, press star.”
Dial-out
[Sub dial-out]
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Table A-1
All voice prompts, alphabetized (continued)
Configuration or
Length
(sec.)
Required Filename
Default Prompt
Feature
“The participant was disconnected.”
Dial-out (shortened
call flow)
do_part_disconnected.wav
[Sub dial-out short]
“The participant has entered the
conference.”
Dial-out (shortened
call flow)
do_part_joined.wav
[Sub dial-out short]
“Press pound to proceed with dialing.”
Dial-out
2.8
do_proceed_prompt.wav
do_short_cmd_help.wav
[Sub dial-out]
“Once the call is answered, to place the
participant into the conference, press star
1. To disconnect the participant, press star
2.”
Dial-out (shortened
call flow)
[Sub dial-out short]
“To enter the conference press star 1. To
dial out to another participant, dial the
number followed by the pound sign.”
Dial-out (shortened
call flow)
do_short_number_prompt.wav
[Sub dial-out short]
“An anonymous participant has joined the
conference.”
Name Announce
[Entry noise]
3.3
3.3
6.4
eea_anon_join.wav
“An anonymous participant has left the
conference.”
Name Announce
[Exit noise]
eea_anon_leave.wav
eea_change_failure.wav
“I’m sorry, the system is currently unable
to change this option, please try again
later.”
Entry/Exit
Announcement
changeable
[Announce setting]
“I’m sorry, this feature is not enabled.”
“I’m sorry, this feature is not enabled.”
Entry/Exit
Announcement not
changeable
2.5
eea_change_in_conf_not_
enabled.wav
[Sub in conf]
Roll Call off
2.5
3.9
eea_change_not_enabled.wav
eea_change_prompt.wav
[Roll call setting]
"To change the conference entry and exit
announcement setting, press 2."
Entry/Exit
Announcement
changeable
[Roll call setting]
"Participants joining or leaving the
conference are announced by name."
Entry/Exit
Announcement
changeable
eea_name_entry.wav
[Announce setting]
192
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Voice Prompts and Call Flows
Table A-1
All voice prompts, alphabetized (continued)
Configuration or
Feature
Length
(sec.)
Required Filename
Default Prompt
“I’m sorry, Participant Name Record must
be on to choose this option.”
Entry/Exit
Announcement
changeable
6.1
eea_name_entry_no_rc.wav
[Announce setting]
<silence>
Name Announce
[Entry announce]
eea_name_join_.wav
_eea_name_join.wav
eea_name_leave_.wav
_eea_name_leave.wav
eea_overview.wav
“…has joined the conference.”
<silence>
Name Announce
[Entry announce]
Name Announce
[Exit announce]
“…has left the conference.”
Name Announce
[Exit announce]
“The conference entry and exit
Entry/Exit
Announcement
changeable
13.8
8.2
announcement setting determines what
will be heard when participants join and
leave the conference. The choices are
tones, name announce or silence.”
[Account Options
help]
“To select name announce, press 1. To
select tones, press 2. To select silence,
press 3.”
Entry/Exit
Announcement
changeable
eea_set_menu.wav
[Announce setting]
“Participants joining or leaving the
conference are not announced.”
Entry/exit
announcement
changeable
eea_silent_entry.wav
eea_tone_entry.wav
[Announce setting]
“Participants joining or leaving the
conference are announced by tones.”
Entry/Exit
Announcement
changeable
[Announce setting]
<beeps>
Tone Announce
0.1
0.1
0.8
eea_tone_join.wav
eea_tone_leave.wav
error.wav
<beeps>
Tone Announce
All
“error”
“The conference has been muted.”
Custom
global_muted.wav
mute/unmute
confirmation (see
[Sub in conf]
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Table A-1
All voice prompts, alphabetized (continued)
Configuration or
Length
(sec.)
Required Filename
Default Prompt
Feature
“The conference has been un-muted.”
Custom
global_unmuted.wav
mute/unmute
confirmation (see
[Sub in conf]
“Hello and welcome to the Conferencing
Center.”
Private number
initial greeting on a
fixed access
system.
3.2
3.2
helo_inbound.wav
[Trad. Private,
2-Passwd. Private]
“Hello and welcome to the Conferencing
Center.”
Dial-out
helo_outbound.wav
“hundred”
All
0.7
1.2
hundred.wav
“The conference has been locked.”
Conference Lock
[Sub in conf]
lck_conf_locked.wav
“The conference has been unlocked.”
Conference Lock
[Sub in conf]
1.8
lck_conf_unlocked.wav
“I’m sorry, the conference you are
attempting to join has been locked. You
have been disconnected by the system.
Please try again.”
Conference Lock
[Part call flow]
lck_locked_disconnect.wav
“Listen Only on Entry is off.”
Listen Only Entry
[LO entry setting]
lo_entry_off.wav
lo_entry_on.wav
“Listen Only on Entry is on.”
Listen Only Entry
[LO entry setting]
“Listen Only on Entry places participants
into conference in Listen Only mode.”
Listen Only Entry
lo_entry_overview.wav
[Account Options
help]
“To turn Listen Only on entry off, press 1.”
“To turn Listen Only on entry on, press 1.”
Listen Only Entry
[LO entry setting]
lo_entry_set_off_prompt.wav
lo_entry_set_on_prompt.wav
lo_part_join.wav
Listen Only Entry
[LO entry setting]
“You are being placed into conference in
listen only mode.”
Listen Only Entry
[Participant call
flow]
194
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Voice Prompts and Call Flows
Table A-1
All voice prompts, alphabetized (continued)
Configuration or
Feature
Length
(sec.)
Required Filename
Default Prompt
“This is a Listen Only conference.”
“I’m sorry, this feature is not enabled.”
<beeps>
Listen Only Entry
[Sub call flow]
lo_subs_join.wav
Mute unavailable
2.5
mute_not_enabled.wav
mute_off.wav
[In conf part or sub]
All
[In conf part or sub]
<beeps>
All
mute_on.wav
[In conf part or sub]
“You are being placed into conference in
muted mode.”
Mute All on
mute_part_join.wav
[Participant call
flow]
Indexed WAVE file containing all recorded
All
numbers.wav
“An operator wants to enter your
conference. Please press star 5 to unlock
it.”
Operator available
[In conf sub]
8.0
op_conf_doorbell.wav
<beep>
Played to operator
when leaving conf
0.2
4.0
op_conf_leave.wav
op_disconnect.wav
“You have been disconnected by the
operator. Good-bye.”
Operator available
[Oper wait]
<beep>
Played to operator
when
0.2
op_disconnect_from_chan.wav
disconnecting from
channel
“Your request will be answered by the next
available operator. To cancel your request,
press star zero.”
Operator request
via Moderator
7.4
2.3
op_gui_req.wav
[In conf sub]
“Your operator request has been
canceled.”
Operator request
canceled via
Moderator
op_gui_req_cancel.wav
[In conf sub]
“Please stand by for an operator.”
<beep>
Operator available
[Oper wait]
2.3
0.2
op_hold.wav
Played to operator
when connecting to
channel or conf
op_listen.wav
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Table A-1
All voice prompts, alphabetized (continued)
Configuration or
Length
(sec.)
Required Filename
Default Prompt
Feature
<hold music>
Operator available
[Oper wait]
30.0
4.4
4.2
7.4
op_music.wav
“You have been disconnected by the
system. Please try your call again.”
All (no oper avail)
[Oper wait]
op_no_op_disconnect.wav
op_not_available.wav
op_req.wav
“I’m sorry, there are no operators available
to service your request.”
Operator request
[In conf oper req]
“Your request will be answered by the next
available operator. To cancel your request,
press star zero.”
Operator request
[In conf oper req]
“Your operator request has been
canceled.”
Operator request
[In conf oper req]
2.3
2.5
op_req_cancel.wav
“I’m sorry, this feature is not enabled.”
Operator request
unavailable to part
op_req_for_subs_only.wav
[In conf oper req]
“I’m sorry, this feature is not enabled.”
Operator request
unavailable
2.5
0.4
op_req_not_enabled.wav
[In conf oper req]
“Pound”
All
pound.wav
“Press pound.”
“Press zero.”
“Press 9.”
All
All
All
All
All
press_#.wav
press_0.wav
press_9.wav
“Press any key.”
“Press star.”
press_anykey.wav
press_star.wav
private_muted.wav
“Your line is muted.”
Custom
mute/unmute
confirmation (see
[In conf part or sub]
“Your line has been unmuted.”
Custom
private_unmuted.wav
pt_operator.wav
mute/unmute
confirmation (see
[In conf part or sub]
“An operator…”
Roll Call (private)
[In conf part or sub]
1.2
196
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Voice Prompts and Call Flows
Table A-1
All voice prompts, alphabetized (continued)
Configuration or
Length
(sec.)
Required Filename
Default Prompt
Feature
“A recorder…”
“Thank you.”
Roll Call (private)
2.0
pt_recorder.wav
[In conf part or sub]
Two-password
private call flow
pw_confirmation.wav
pw_disconnect.wav
pw_prompt.wav
[2-passwd private]
“You have been disconnected by the
system. Please try your call again.”
Two-password
private call flow
4.4
[2-passwd private]
“Thank you for calling the conference
center. Please enter your passcode
followed by the pound sign at any time
during this message.”
Two-password
private call flow
[2-passwd private]
“I’m sorry, that entry is not valid.”
Two-password
private call flow
3.0
1.9
5.2
2.6
pw_wrong.wav
[2-passwd private]
“I’m sorry, your entry…”
Two-password
private call flow
pw_wrong_.wav
[2-passwd private]
“… is not valid. Please enter the valid
digits followed by the pound sign.”
Two-password
private call flow
_pw_wrong.wav
[2-passwd private]
“To change the QuickStart setting…”
<silence>
Quick Start
qs_change_prompt_.wav
_qs_change_prompt.wav
qs_future_affecting.wav
[Account Options
help]
Quick Start
[Account Options
menu]
“…for future conferences. Current
conferences are not affected by this
change.”
Quick Start
5.9
[Quick Start setting]
“Quick Start is off.”
Quick Start
qs_off.wav
[Quick Start setting]
“Quick Start is on.”
Quick Start
qs_on.wav
[Quick Start setting]
“Quick Start allows conferences to start as
soon as the first participant dials in.
Participants don’t wait on hold for you to
arrive.”
Quick Start
6.4
qs_overview.wav
[Account Options
help]
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Table A-1
All voice prompts, alphabetized (continued)
Configuration or
Length
(sec.)
Required Filename
Default Prompt
Feature
“This is a Quick Start conference, you will
be placed into the conference.”
Quick Start
4.7
qs_part_join.wav
[Part call flow]
“To turn Quick Start off, press 1.”
“To turn Quick Start on, press 1.”
Quick Start
qs_set_off_prompt.wav
qs_set_on_prompt.wav
qs_subs_join.wav
rc_.wav
[Quick Start setting]
Quick Start
[Quick Start setting]
“This is a Quick Start conference, you will
be placed into the conference.”
Quick Start
4.7
[Sub call flow]
“The following…”
Roll Call (private)
[In conf part or sub]
“…participants are in conference.”
Roll Call (private)
[In conf part or sub]
_rc.wav
“The number of anonymous participants
is…”
Roll Call (private)
2.0
2.0
rc_anon_count_.wav
rc_anonymous.wav
[In conf part or sub]
“An anonymous participant.”
Name Announce
(Mod or oper clicks
Play Name)
[In conf sub]
“To change Roll Call settings…”
<silence>
Roll Call
1.8
rc_change_prompt_.wav
_rc_change_prompt.wav
[Account Options
menu]
Roll Call
[Account Options
menu]
“The following…”
Roll Call (conf)
[In conf sub]
rc_conf_.wav
“…participants are in conference.”
Roll Call (conf)
[In conf sub]
_rc_conf.wav
“The number of anonymous participants
is…”
Roll Call (conf)
[In conf sub]
2.0
2.0
rc_conf_anon_count_.wav
rc_identify_prompt.wav
rc_off.wav
“After the tone, please state your name,
followed by the pound sign. <bong>”
Roll Call
[Name Record]
“Roll Call is off.”
Roll Call
[Roll Call setting]
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Voice Prompts and Call Flows
Table A-1
All voice prompts, alphabetized (continued)
Configuration or
Feature
Length
(sec.)
Required Filename
Default Prompt
"Roll call is on. Participants will be
prompted to record their names."
Roll Call
rc_on.wav
[Roll Call setting]
“Roll Call prompts participants to record
their names as they join a conference.
During the conference, any participant can
press star 9 to hear the roll call replayed
privately.”
Roll Call
rc_overview.wav
[Account Options
help]
“I’m sorry, we did not get your name.”
Roll Call
2.0
rc_rerecord_prompt.wav
rc_set_off_prompt.wav
rc_set_on_prompt.wav
rec_change_failed.wav
rec_conf_full.wav
[Name Record]
“To turn Roll Call off, press 1.”
Roll Call
[Roll Call setting]
"To turn roll call on so participants are
prompted to record their names, press 1."
Roll Call
[Roll Call setting]
“I’m sorry, your call could not be
completed. Please try your call again.”
Recorder Dial Out
[Recording]
“I’m sorry. Your conference is full. A
recording connection cannot be added.”
Recorder Dial Out
[Recording]
5.7
2.5
“I’m sorry, this feature is not enabled.”
Recorder Dial Out
unavailable
rec_not_enabled.wav
[Recording]
“This conference is being recorded.”
“This conference is being recorded.”
Recorder Dial Out
[Part call flow]
2.3
2.3
5.7
2.3
5.7
3.0
2.3
rec_part_join_reminder.wav
rec_rejoin_reminder.wav
rec_start_prompt.wav
rec_started.wav
Recorder Dial Out
[In conf rejoin]
“To start the conference recording, press
1. To cancel, press star.”
Recorder Dial Out
[Recording]
“This conference is being recorded.”
Recorder Dial Out
[Recording]
“To stop the conference recording, press
1. To cancel, press star.”
Recorder Dial Out
[Recording]
rec_stop_prompt.wav
rec_stopped.wav
“This conference is no longer being
recorded.”
Recorder Dial Out
[Recording]
“This conference is being recorded.”
Recorder Dial Out
[Sub call flow]
rec_subs_join_reminder.wav
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Table A-1
All voice prompts, alphabetized (continued)
Configuration or
Length
(sec.)
Required Filename
Default Prompt
Feature
“Please stand by while your recording
connection is established. To cancel the
recording, press star 2.”
Recorder Dial Out
[Recording]
7.2
rec_wait.wav
“Due to technical difficulties, you will be
disconnected in…”
All
sd_chan_.wav
“…minutes.”
All
All
_sd_chan.wav
“Due to technical difficulties, you will be
disconnected now.”
sd_chan_now.wav
“Due to technical difficulties, your
conference will end in…”
All
sd_conf_.wav
“…minutes.”
All
All
_sd_conf.wav
“Due to technical difficulties, your
conference will end now.”
sd_conf_now.wav
<silence>
Not used as
silence.wav
named. Copy and
rename to replace
any prompt you
don’t want played.
“The system is unable to change your
password at this time. Please try again
later.”
Password
changeable
5.0
2.6
spw_change_failure.wav
spw_change_prompt_.wav
[PW setting]
“To change your subscriber password…”
Password
changeable
[Account Options
menu]
<silence>
Password
changeable
_spw_change_prompt.wav
[Account Options
menu]
“You have been disconnected by the
system. Please try your call again.”
Prompt for
Subscriber
4.4
6.7
spw_disconnect.wav
spw_invalid.wav
[Sub validation]
“I’m sorry, the password you entered is not
permitted. Please enter a different
Password
changeable
password followed by the pound sign.”
[PW setting]
200
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Table A-1
All voice prompts, alphabetized (continued)
Configuration or
Feature
Length
(sec.)
Required Filename
Default Prompt
“The subscriber password is the password
you use to log into the conferencing
system, to start and join your conferences,
and to change your account options.”
Password
changeable
spw_overview.wav
[Account Options
help]
“Please enter your subscriber password
followed by the pound sign.”
Prompt for
Subscriber
3.7
2.4
7.3
3.0
1.9
5.2
8.6
spw_prompt.wav
[Sub validation]
“Your new subscriber password is…”
Password
changeable
spw_repeat_.wav
spw_set_prompt.wav
spw_wrong.wav
[PW setting]
“Please enter your new subscriber
password followed by the pound sign. The
password must be 4 to 20 digits.”
Password
changeable
[PW setting]
“I’m sorry, that entry is not valid.”
Prompt for
Subscriber
[Sub validation]
“I’m sorry, your entry…”
Prompt for
Subscriber
spw_wrong_.wav
[Sub validation]
“… is not valid. Please enter the valid
digits followed by the pound sign.”
Prompt for
Subscriber
_spw_wrong.wav
[Sub validation]
“I’m sorry, the subscriber password must
be between 4 and 20 digits long. Please
re-enter your password followed by the
pound sign.”
Password
changeable
spw_wrong_length.wav
[PW setting]
“Star.”
All
0.5
0.9
star.wav
“System error.”
All
sys_error.wav
wr_change_prompt_.wav
“To change the conference lock type
setting…”
Waiting Room
[Account Options
menu]
<silence>
Waiting Room
_wr_change_prompt.wav
wr_cmd_help.wav
[Account Options
menu]
“Press pound 5 to process callers in the
waiting room. Press pound 6 to toggle
notification announcements on or off.”
Waiting Room
[Sub call flow, Sub
WR processing]
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Table A-1
All voice prompts, alphabetized (continued)
Configuration or
Length
(sec.)
Required Filename
Default Prompt
Feature
“This conference has been locked with
Waiting Room on.”
Waiting Room
[In conf sub]
wr_conf_locked.wav
“There are…”
Waiting Room
wr_count_.wav
[Sub WR
processing]
“…participants in the waiting room.”
Waiting Room
_wr_count.wav
[Sub WR
processing]
“There are no callers in the waiting room
at this time. You will now be returned to
the main conference.”
Waiting Room
wr_empty.wav
[Sub WR
processing]
“Callers in the waiting room are processed
in the order they arrived.”
Waiting Room
wr_handling_prompt.wav
wr_hold.wav
[Sub WR
processing]
“This conference is locked. Please hold
while the Subscriber is notified.”
Waiting Room
[Part lock/full
processing]
<hold music>
Waiting Room
30.0
wr_music.wav
[Part lock/full
processing]
"An anonymous participant would like to
join your call. Press pound 5 to process
callers in the waiting room. Press pound 6
to toggle caller notification
Waiting Room
wr_new_anon_caller.wav
[Sub WR
processing]
announcements on or off."
“A new caller…”
Waiting Room
wr_new_caller_.wav
_wr_new_caller.wav
[Sub WR
processing]
"…would like to join your conference.
Press pound 5 to process callers in the
waiting room. Press pound 6 to turn caller
notification off."
Waiting Room
[Sub WR
processing]
“Waiting room notification is off.”
Waiting Room
wr_notification_off.wav
wr_notification_on.wav
[Sub WR
processing]
“Waiting room notification is on.”
Waiting Room
[Sub WR
processing]
202
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Table A-1
All voice prompts, alphabetized (continued)
Configuration or
Feature
Length
(sec.)
Required Filename
Default Prompt
"Currently, your conferences start
unlocked, so participants are put directly
into conference."
Lock (Waiting
Room) available
wr_off.wav
[WR setting]
"Currently, your conferences start locked,
so participants are put into the waiting
room."
Lock (Waiting
Room) available
wr_on_entry.wav
[WR setting]
“The conference will start locked with the
waiting room on.”
Waiting Room
[Sub call flow]
wr_on_entry_conf_start.wav
wr_on_lock.wav
"Currently, your conferences start
unlocked, so participants are put directly
into the conference, but the waiting room
is available for later use."
Lock (Waiting
Room) available
[WR setting]
“The conference lock type option
determines if callers dialing into a locked
conference are disconnected or put into a
waiting room.”
Waiting Room
wr_overview.wav
[Account Options
help]
“Press pound to talk with the next caller.
Then, press 1 to join the caller to
conference. Press 2 to leave the caller in
the Waiting Room. Press 3 to disconnect
the caller. Press star to return to the main
conference.”
Waiting Room
wr_private_conf_cmd_
help.wav
[Sub WR
processing]
“You are being placed in a private
conference with an operator. Please hold.”
Waiting Room
wr_private_conf_w_op.wav
wr_private_conf_w_subs.wav
wr_set_off_prompt.wav
[Part lock/full
processing]
“You are being placed in a private
conference with the Subscriber. Please
hold.”
Waiting Room
[Part lock/full
processing]
"To start conferences unlocked with
waiting room off, press 2. Participants will
be put directly into conference."
Lock (Waiting
Room) available
[WR setting]
“To start conferences locked, so
participants are put into the waiting room,
press 1.”
Lock (Waiting
Room) available
wr_set_on_entry_prompt.wav
wr_set_on_lock_prompt.wav
[WR setting]
"To start conferences unlocked with
waiting room on, press 3. Participants will
be put directly into conference, but the
waiting room is available for later use."
Lock (Waiting
Room) available
[WR setting]
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Table A-1
All voice prompts, alphabetized (continued)
Configuration or
Length
(sec.)
Required Filename
Default Prompt
Feature
“The subscriber has returned you to the
waiting room. Please hold.”
Waiting Room
wr_sub_hold.wav
[Part lock/full
processing]
“You have been disconnected by the
conference moderator. Good bye.”
Waiting Room
wr_subs_disconnect.wav
[Part lock/full
processing]
“Sorry. You have been in the waiting room
for the maximum amount of time and will
be disconnected. Goodbye.”
Waiting Room
wr_timeout_disconnect.wav
[Part lock/full
processing]
204
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Voice Prompts and Call Flows
Call Flow Diagrams
The following pages contain call flow diagrams for the ReadiVoice system.
conventions used in the diagrams.
The diagrams themselves are arranged into several groups or categories, as
Table A-2
Figures
Guide to call flow diagrams
Category
Description
Initial Inbound Call
Flows
Initial inbound call processing for different call flow configurations.
During this processing, the system identifies the caller’s conference
and determines whether the caller is the subscriber or a participant.
Dial-out (Outbound)
Call Flows
Processes related to dial-out, including dial-out to a recording device.
Subscriber and
Participant Call Flows
Subscriber and participant call flows after the initial entry processing
and some of the subprocesses within these call flows.
Subscriber Account
Options Processes
Processes related to the subscriber’s account options menu and the
settings available from it.
In-conference
processes
Processes related to in-conference actions by the subscriber,
participants, or moderator, plus the system’s conference termination
process and other in-conference events.
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Figure A-2 Understanding the call flow diagrams
Call Flow Legend
Examples of and information about certain symbols and conventions used in the call flow
diagrams.
cr_access_code_wrong.wav
Message to the user, usually one or more .wav file names.
Message box containing both an event name (all caps) and a .wav
file name. The .wav file shown is the default, but the system can be
configured (in ive.ini) to play a different file for that event.
CONF_LOCKED
lck_locked_disconnect.wav
rc_conf_.wav
caller count (numbers.wav)
_rc_conf.wav
Messages played to the entire conference, not to a single user
(channel).
When the system waits for user input, the diagram indicates the
default timeout, but many of these are configurable. For instance,
the default timeout for codes and passwords is 10 seconds, but it
can be changed using the Passcode.TimeOut4Prompt parameter
in the ive.ini file.
Get tones
10 sec. timer
Valid
subscriber
PW?
Yes
No
Decision, usually yes/no or a/b. Sometimes a/b/c.
Named subprocess, defined in its own diagram. If this symbol
has an arrow departing it as well as entering, then the
subprocess both starts and ends at this point.
Name
Record
process
Autonomous input (DTMF tones) from subscriber leads to
process/action by system. All DTMF (touchtone) commands
shown in the call flows are the system defaults, which can be
changed in the ive.ini file.
Subscriber
unlocked
conference (*5)
Join all callers to
conference (*5)
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Figure A-3 Traditional shared call flow (initial entry)
Traditional Shared Call Flow
Start
Non-routed System Shared, Access Code required
Starts when inbound call hits bridge.
Ends at named process.
cf_call_not_
completed.wav
No
Yes
Any error?
cr_code_prompt.wav
(see note)
cr_access_code_wrong_.wav
entered code (numbers.wav)
_cr_access_code_wrong.wav
Hook
Get tones
<3
30 sec. timer
Error
Valid
access
code?
Failed
attempts?
cr_access_code_
collection_failure.wav
No
Yes
3
Ending
cr_conf_ending.wav
cr_conf_full.wav
Routing
Failure
reason?
No
Full
cr_access_code_wrong.wav
successful?
Yes
Busy/other
cr_system_busy.wav
cf_please_standby.wav
Hook on
failure?
cr_access_code_
good.wav
Yes
Hook
No
Problem
communicating w/
conf?
HR_ACCESSCODE_FAIL
cr_access_code_disconnect.wav
Yes
Operator
available?
No
No
Failed
attempts?
Yes
No
Yes
<11
Conf ending?
Unexpected
error
Operator
Wait
process
11
Subscriber in
conf?
No
Yes
sys_error.wav
error number
(numbers.wav)
HR_CONF_END_
IN_PROGRESS
cf_eip_disconnect.wav
Prompt for
subscriber?
Participant
process
Yes
No
No
cf_you_subs_
prompt.wav
Subscriber
process
Subscriber
(* entered)?
10 sec.
timer
Yes
Hook
Get tones
Note: The initial greeting, cr_code_prompt.wav, is a symbolic link to either cr_access_code_prompt.wav or
pw_prompt.wav, depending on system configuration. See “About Initial Greetings” on page 181 for details.
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Figure A-4 Traditional private call flow (initial entry)
Traditional Private Call Flow
Non-Routed System Private, Participant Password not available
Starts when inbound call hits bridge.
Ends at named process.
Start
cf_call_not_
completed.wav
Found sub/
conf?
Yes
No
Yes
Any error?
No
cr_access_number_
wrong.wav
Bad access
number?
Yes
helo_inbound.wav
No
Ending
cr_conf_ending.wav
cr_conf_full.wav
Problem
communicating w/
conf?
Other failure
reason?
No
Full
Busy/other
cf_please_standby.wav
cr_system_busy.wav
Failed
attempts?
<11
11
HR_CONF_END_IN_PROGRESS
cf_eip_disconnect.wav
Yes
Conf ending?
Hook
No
Subscriber in
conf?
Yes
sys_error.wav
error number
(numbers.wav)
No
Prompt for
subscriber?
Participant
process
Unexpected
error
No
Yes
No
cf_you_subs_
prompt.wav
Subscriber
process
Subscriber
(* entered)?
10 sec.
timer
Yes
Get tones
208
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Voice Prompts and Call Flows
Figure A-5 Two-password shared call flow (initial entry)
Two Password Shared Call Flow
Start
Non-routed System Shared, Prompt for Subscriber off,
Participant Password required
Starts when inbound call hits bridge.
Ends at named process.
cf_call_not_
completed.wav
No
Yes
Any error?
cr_code_prompt.wav
(see note)
cr_access_code_wrong_.wav
entered code (numbers.wav)
_cr_access_code_wrong.wav
Hook
Get tones
<3
30 sec. timer
Error
Failed
attempts?
Valid
password?
cr_access_code_
collection_failure.wav
No
Yes
3
Ending
cr_conf_ending.wav
cr_conf_full.wav
Routing
successful?
Failure
reason?
No
Full
cr_access_code_wrong.wav
Yes
Busy/other
cr_system_busy.wav
cf_please_standby.wav
Hook on
failure?
Yes
Hook
cr_access_code_good.wav
No
HR_ACCESSCODE_FAIL
cr_access_code_disconnect.wav
Problem
communicating
w/conf?
Yes
Operator
available?
No
Failed
attempts?
<11
No
Yes
11
No
Yes
Conf ending?
Operator
Wait
HR_CONF_END_
IN_PROGRESS
cf_eip_disconnect.wav
Yes
No
Subscriber?
Other
subscriber in
conf?
Hook
Participant
process
Yes
sys_error.wav
error number (numbers.wav)
No
Subscriber
process
Unexpected
error
Note: The initial greeting, cr_code_prompt.wav, is a symbolic link to either cr_access_code_prompt.wav or
pw_prompt.wav, depending on system configuration. See “About Initial Greetings” on page 181 for details.
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Figure A-6 Two-password private call flow (initial entry)
Two Password Private Call Flow
Non-routed System Private, Prompt for Subscriber
off, Participant Password optional
Start
Starts when inbound call hits bridge.
Ends at named process.
cf_call_not_
completed.wav
Found sub/
conf?
Yes
No
Yes
Any error?
No
cr_access_number_
wrong.wav
Bad access
number?
Yes
helo_inbound.wav
No
Ending
cr_conf_ending.wav
cr_conf_full.wav
Problem
communicating w/
conf?
No
Other failure
reason?
Full
Yes
cf_please_standby.wav
Busy/other
cr_system_busy.wav
Failed
attempts?
<11
11
HR_CONF_END_IN_PROGRESS
cf_eip_disconnect.wav
Yes
Conf ending?
Hook
No
sys_error.wav
error number
(numbers.wav)
pw_prompt.wav
pw_wrong_.wav
password (numbers.wav)
_pw_wrong.wav
Error
Unexpected
error
Get tones
10 sec. timer
<3
Failed
attempts?
Valid
No
Yes
No
pw_confirmation.wav
Subscriber?
password?
3
Yes
Hook on
failure?
HR_SUB_PASSWORD_FAIL
spw_disconnect.wav
Yes
pw_wrong.wav
Other
subscriber in
conf?
No
Yes
No
Hook
Operator
Wait
process
Operator
available?
HR_PART_PIN_FAIL
pw_disconnect.wav
Subscriber
process
Participant
process
Yes
No
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Figure A-7 Routed (INCR) call flow (initial entry)
Routed Call Flow
Routed (INCR) System – Access Code, if any, collected by switch.
Starts when inbound call hits bridge. Ends at named process.
Start
cf_please_standby.wav
<11
Problem
communicating w/
conf?
Failed
attempts?
Yes
11
No
IP system?
No
Yes
HR_CONF_END_IN_PROGRESS
cf_eip_disconnect.wav
Yes
Conf ending?
Hook
IP Call
process
No
pw_wrong_.wav
password (numbers.wav)
_pw_wrong.wav
2-PW
Private?
No
Yes
pw_prompt.wav
<3
Error
Subscriber in
conf?
No
Yes
Get tones
Failed
attempts?
10 sec. timer
Prompt for
subscriber?
Participant
process
No
3
Valid
password?
Yes
No
Yes
pw_wrong.wav
pw_confirmation.wav
cf_you_subs_
prompt.wav
HR_SUB_PASSWORD_FAIL
spw_disconnect.wav
Hook on
failure?
Yes
No
Subscriber?
No
Get tones
Hook
10 sec. timer
Yes
HR_PART_PIN_FAIL
pw_disconnect.wav
Operator
available?
No
Other
subscriber in
conf?
Subscriber
(* entered)?
No
Yes
Yes
Yes
No
Operator
Wait
process
Subscriber
process
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Figure A-8 IP Call process
IP Call Process
Applies to ReadiVoice-IP systems.
Starts and ends in Routed (INCR) Call Flow.
Start IP Call
Process
cf_call_not_
completed.wav
Yes
Any error?
No
pw_prompt.wav
Hook
cr_access_code_wrong_.wav
entered code (numbers.wav)
_cr_access_code_wrong.wav
Get tones
30 sec. timer
Error
cr_access_code_
collection_failure.wav
<3
Failed
attempts?
Routing
successful?
No subscription
Conf busy
cr_conf_busy.wav
Yes
3
Hook on
failure?
Yes
No
cr_access_code_good.wav
cr_access_code_wrong.wav
End IP Call
Process
Hook
HR_ACCESSCODE_FAIL
cr_access_code_disconnect.wav
Operator
available?
Operator
Wait
No
Yes
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Figure A-9 Recording dial-out process
Recording Process
Starts in conference when subscriber enters recording command.
Ends at Rejoin Conference process.
Start Recording
Process
Recording
enabled?
No
Yes
Yes
rec_conf_full.wav
Conf full?
No
rec_not_enabled.wav
Not
recording
Recording
state?
Recording
rec_start_prompt.wav
rec_stop_prompt.wav
Get tones
Get tones
10 sec. timer
10 sec. timer
Yes
Yes
1 entered?
1 entered?
No
No
rec_wait.wav
Stop recording
<3
No
No
<3
* entered?
* entered?
eea_name_leave_.wav
pt_recorder.wav
_eea_name_leave.wav
Failed
attempts?
Failed
attempts?
Start recording
3
3
Unexpected
error
rec_started.wav
rec_stopped.wav
eea_name_join_.wav
pt_recorder.wav
_eea_name_join.wav
rec_change_
failed.wav
Rejoin
Conference
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Figure A-10 Subscriber dial-out process (long)
Subscriber Dial-out Process – default (long) configuration
Starts in conference when subscriber presses *1. Ends at Rejoin Conference process.
Start Subscriber
DO Process
do_not_enabled.wav
No
Dial-out
enabled?
Rejoin
Conference
A
do_not_completed.wav
Yes
Yes
No
Error
do_conf_full.wav
do_number_prompt.wav
A
Conf full?
Get tones
15 sec. timer
Yes
No
do_bad_num_.wav
number (numbers.wav)
_do_bad_num.wav
Anything
entered?
No
First try?
10 sec.
timer
# entered?
Get tones
No
Yes
Yes
No
do_num_verify_.wav
number (numbers.wav)
_do_num_verify.wav
do_long_cmd_help.wav
do_proceed_prompt.wav
Rejoin
Conference
Valid length
number?
Yes
No
Yes
* entered?
Get tones
do_conf_full.wav
10 sec. timer
No capacity
Problem?
None
No
Yes
Call not completed
# entered?
do_dialing.wav
do_not_completed.wav
A
Sub connected
to outbound line
do_long_cmd_help.wav
Get tones
No
Yes
No
No
No
No
** entered?
*1 entered?
*2 entered?
*3 entered?
*4 entered?
Yes
Yes
Yes
Yes
Rejoin
Conference
Join called
party to conf
Join called
party to conf
Disconnect
line
Disconnect
line
A
A
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Figure A-11 Subscriber dial-out process (short)
Subscriber Dial-out Process – optional short configuration
Starts in conference when subscriber presses *1. Ends at Rejoin Conference process.
Start Short
Subscriber DO
Process
Rejoin
Conference
Dial-out
enabled?
No
do_not_enabled.wav
do_conf_full.wav
No
Yes
Conf full?
A
Yes
do_not_completed.wav
First or
subsequent
dial?
do_short_number_
prompt.wav
Subsequent
Error
A
Get tones
do_number_
prompt.wav
First
15 sec. timer
Yes
do_num_verify_.wav
number (numbers.wav)
_do_num_verify.wav
First or
subsequent
dial?
Anything
entered?
Yes
No
First
First try?
Get tones
10 sec. timer
No
Subsequent
No
Yes
# entered?
* entered?
Yes
Yes
No
do_short_cmd_help.wav
do_proceed_prompt.wav
Confirm
number?
Rejoin
Conference
Valid length
number?
Yes
No
Yes
*1 entered?
No
No
do_bad_num_.wav
number (numbers.wav)
_do_bad_num.wav
A
Get tones
10 sec. timer
do_short_cmd_help.wav
do_dialing.wav
No capacity
do_conf_full.wav
Call not
completed
No
Yes
# entered?
Problem?
do_not_completed.wav
None
Sub connected
to outbound line
do_part_
disconnected.wav
do_short_cmd_help.wav
A
Get tones
No
Disconnect
line
Yes
No
No
Yes
** entered?
*1 entered?
*2 entered?
Yes
Join called
party to conf
do_part_joined.wav
A
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Figure A-12 Called party dial-out process
Called Party Dial-out Process
Describes system interaction with the dialed-out line (the called party). Starts when system
accesses line to place outbound call. May be initiated via Operator or Moderator interface
or via “One Click” link. Ends at named process.
Dial specified
number
Subscriber
being called?
Play
extension?
120 sec.
timer
Far end off
hook?
Start Called Party
DO Process
Yes
No
No
No
Yes
Yes
do_auto_join_failure.wav
Hook
helo_outbound.wav
do_extplay_.wav
extension (numbers.wav)
_do_extplay.wav
BLAST_DIAL_DISCONNECT
do_auto_disconnect.wav
3
Failed
attempts?
<3
do_blast_dial_subname_announce_.wav
recorded name
_do_blast_dial_subname_announce.wav
Sub name
recorded?
Yes
Yes
Blast dial-out?
Error
Get tones
No
No
do_auto_join_prompt.wav
10 sec. timer
Anything
entered?
Called party
authorized
Supervised
dial-out?
No
Yes
No
Yes
Dial-out
Newcomer
process
Talking with
subscriber
Subscriber
talking to caller
Called party
known type?
No
Yes
Join
command
entered?
Yes
Subscriber
process
Yes
Subscriber?
Join caller to
conference
Yes
Join to conf?
No
No
No
Hook
command
entered?
Yes
No
Participant
process
Disconnect?
No
Disconnect caller
Yes
MODER_ACTION
cf_subs_disconnect.wav
Hook
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Figure A-13 Dial-out newcomer process
Dial-Out Newcomer Process
Starts in Called Party Dial-Out process when caller type (subscriber
or participant) isn’t known. Ends at named process.
Start DO
Newcomer
Process
pw_wrong_.wav
password (numbers.wav)
_pw_wrong.wav
2-PW
Private?
No
Yes
pw_prompt.wav
<3
Error
Subscriber in
conf?
No
Yes
Get tones
Failed
attempts?
10 sec. timer
Prompt for
subscriber?
Participant
process
No
3
Valid
password?
Yes
No
Yes
pw_wrong.wav
pw_confirmation.wav
cf_you_subs_
prompt.wav
HR_SUB_PASSWORD_FAIL
spw_disconnect.wav
Hook on
failure?
Yes
No
Subscriber?
No
Get tones
Hook
Yes
10 sec.
timer
HR_PART_PIN_FAIL
pw_disconnect.wav
Operator
available?
No
Other
subscriber in
conf?
Subscriber
(* entered)?
No
Yes
Yes
Yes
No
Operator
Wait
process
Subscriber
process
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Figure A-14 Subscriber call flow (after initial entry)
Subscriber Process
Starts after initial entry process (inbound call flow or called party dial-out process).
Named processes are defined in their own flow diagrams.
Subscriber
Validation
process
Account
Options
process
Conference
Security
Code process
Name
Record
process
Subscriber
Start
No
Yes
Yes
Yes
Sub PW
validated?
CSC
available?
Acct Opts
available?
Yes
No
No
No
Roll Call?
lo_subs_join.wav
cf_non_part_join.wav
No
No
Conf ending?
Conf full?
Yes
Yes
Yes
Yes
Locked w/
WR?
Listen only
mode?
Callers in
conf?
No
No
HR_CONF_END_
IN_PROGRESS
cf_eip_disconnect.wav
cf_conf_full_
disconnect.wav
No
Yes
cf_conf_start.wav
wr_on_entry_conf_start.wav
wr_cmd_help.wav
Hook
qs_subs_join.wav
Yes
cf_in_conf_help_
cmd.wav
Callers in
conf?
No
Yes
Quick Start?
No
QS or entry
w/count?
Yes
cf_subs_first_caller.wav
No
Entry with
count?
Auto
continuation?
No
No
Yes
cf_caller_count_.wav
count (numbers.wav)
_cf_caller_count.wav
Yes
cc_active_reminder.wav
Yes
Recording?
rec_subs_join_reminder.wav
No
Subscriber
in
conference
Entry
Announce
218
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Figure A-15 Participant call flow (after initial entry)
Participant Process
Starts after initial entry process (inbound call flow or called party dial-out process) or in
Subscriber Validation process. Named processes are defined in their own flow diagrams.
Participant
Start
Name
Record
process
Yes
Roll Call?
Music
Hold
process
No
No
Late
subscriber?
No
Subscriber
present?
Conf locked
w/WR?
No
No
Yes
Quick Start?
Yes
Yes
Yes
cf_late_subscriber.wav
Dialed in
(incl. one click)
or out?
In
Subscriber
present?
Out
No
Yes
qs_part_join.wav
Quick Start?
Yes
No
Locked/
Full Conf
process
Participant
CSC
process
Yes
Listen Only?
No
Yes
No
lo_part_join.wav
Listen Only?
Yes
Yes
No
Mute All?
mute_part_join.wav
Mute All?
cf_part_join.wav
No
cf_caller_count_.way
count (numbers.wav)
_cf_caller_count.wav
Conf Entry
with Count?
First into
conf?
No
Yes
Yes
No
Participant
in
conference
Entry
Announce
No
cf_first_caller.wav
Recording?
Yes
rec_part_join_reminder
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Figure A-16 Subscriber validation process
Subscriber Validation
Starts in Subscriber Call Flow or Music Hold.
Ends in Subscriber Call Flow.
Start Subscriber
Validation
spw_prompt.wav
spw_wrong_.wav
password (numbers.wav)
_spw_wrong.wav
Error
Get tones
10 sec. timer
<3
Valid
subscriber
PW?
Failed
attempts?
3
No
Yes
spw_wrong.wav
Other
subscriber in
conf?
Yes
No
Hook on
failure?
Yes
No
Participant
Call Flow
End Subscriber
Validation
HR_SUB_PASSWORD_FAIL
spw_disconnect.wav
Operator
available?
No
Yes
Hook
Operator
Wait
process
220
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Figure A-17 Conference security code process (subscriber)
Subscriber Conference Security Code Process
Starts and ends in Subscriber Call Flow.
Start Sub CSC
Process
CSC already
set?
Yes
No
CSC optional
or required?
Optional
Required
csn_mandatory_
setup_prompt.wav
csn_setup_prompt.wav
Get tones
Get tones
10 sec. timer
10 sec. timer
<5
Failed
attempts?
Anything
entered?
Digits plus #
entered?
No
No
5
Yes
Yes
csn_none_
set.wav
csn_wrong_
length.wav
*
No
* or #?
4-9 digits?
#
Yes
csn_verify_prompt_.wav
code (numbers.wav)
_csn_verify_prompt.wav
csn_wrong_
length.wav
Preceded by
4-9 digits?
No
Yes
Get tones
csn_verify_prompt_.wav
code (numbers.wav)
_csn_verify_prompt.wav
10 sec. timer
Yes
* entered?
Yes
Get tones
5 sec. timer
No
No
* entered?
End Sub CSC
Process
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Figure A-18 Conference security code process (participant)
Participant Conference Security Code Process
Starts and ends in Participant Call Flow.
Start Part
CSC Process
Conf
security code
required?
Yes
csn_prompt.wav
csn_wrong_.wav
code (numbers.wav)
_csn_wrong.wav
No
Get tones
10 sec. timer
<3
Failed
attempts?
Yes
No
Correct?
3
csn_wrong.wav
Hook on
CSC failure?
Yes
No
Operator
Wait
process
Operator
available?
Yes
No
HR_CONF_PASSCODE_FAIL
csn_disconnect.wav
End Part
CSC Process
Hook
222
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Figure A-19 Music hold process
Music Hold Process
Starts and ends in Participant Call Flow.
Start Music Hold
Process
Problem
communicating
w/conf?
HR_CONF_END_SUB_LEAVE
cc_conf_end_disconnect.wav
Yes
Yes
Yes
Yes
cf_part_music_wait.wav
No
Conf full?
cf_conf_full.wav
No
Hook
HR_CONF_END_
IN_PROGRESS
cf_eip_disconnect.wav
Conf ending?
No
HR_CONF_LOCKED
lck_locked_disconnect.wav
No
cf_music.wav (repeats)
Conf locked?
Hold time
expired?
HR_CONF_END_TIMEOUT
ct_conf_timeout_disconnect.wav
Yes
No
Many callers
waiting?
No
Conf started?
Yes
Yes
No
Play
hold extender
message?
End Music Hold
Process
No
Get tones
Yes
cf_music_hold_to_conf.wav
Subscriber
Validation
process
Prompt for
sub on and *
entered?
No
Yes
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Figure A-20 Name record process
Subscriber/Participant Name Record Process
Starts and ends in Subscriber or Participant Call Flow.
Start Name
Record Process
rc_identify_prompt.wav
Recording
rc_rerecord_prompt.wav
<3
Anything
recorded?
Failed
attempts?
No
Yes
3
End Name
Record Process
224
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Figure A-21 Locked/full conference process (participant)
Locked/Full Conference Process
Starts and ends in Participant Call Flow. A high-level view of the parallel subscriber
process is shown for reference. See Sub WR Process diagram for details.
Full
Start Locked/Full
Process
Conf full or
ending?
cf_conf_full.wav
Ending
HR_CONF_END_IN_PROGRESS
cf_eip_disconnect.wav
No
Hook
Waiting
Room?
HR_CONF_LOCKED
lck_locked_disconnect.wav
Yes
No
Locked?
No
Yes
wr_hold.wav
Sub
unlocked conf
(*5)?
wr_music.wav
(repeats)
Subscriber
unlocked
conference (*5)
Yes
Join all callers to
conference (*5)
No
Talking with
subscriber
Subscriber
entered #5 and
then #
Join to
subscriber
(#)?
No
Subscriber
action
Yes
Join caller to
subscriber (#)
Subscriber
processed caller
wr_private_conf_w_subs.wav
Join to
conf
(1)?
Yes
No
Join caller to
conference (1)
Yes
Yes
Yes
1 entered?
Return to WR
(2)?
Yes
No
No
No
Disconnect
(3)?
wr_sub_hold.wav
Return caller to
WR (2)
Yes
2 entered?
wr_subs_
disconnect.wav
Hold time
expired?
No
No
Yes
Disconnect caller
(3)
3 entered?
End Locked/
Full Process
wr_timeout_
disconnect.wav
Hook
Operator picks
caller in WR for
private conf
Join caller to
operator
Operator
assistance
wr_private_conf_w_op.wav
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Figure A-22 Operator wait process
Operator Wait Process
Starts in Traditional Shared, Two Password Shared, Two Password Private, Routed,
IP Call, Subscriber Validation, or Participant Conference Security Code.
Ends where it started.
Start Operator
Wait Process
Operator
logged in?
No
Hook
op_no_op_disconnect.wav
Yes
op_hold.wav
op_music.wav
(repeats)
Operator
assistance
OPER_ACTION
op_disconnect.wav
Operator
answers?
No
Yes
Yes
Return caller
to hold?
Disconnect
caller?
Yes
No
No
End Operator
Wait Process
226
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Figure A-23 Account options process
Account Options Process
Starts and ends in Subscriber Call Flow.
Start Account
Options Process
ao_initial_menu.wav
ao_none_avail.wav
Get tones
10 sec. timer
<5
Digits
entered?
Failed
attempts?
No
ao_wrong_key.wav
5
Yes
End Account
Options Process
1
Other key
1 or 2?
No
2
Any option
configurable?
AO Menu
process
Yes
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Figure A-24 Account options menu
Account Options Menu Process
Starts and ends in Account Options Process.
The first configurable option
is assigned the 1 key (that is,
ao_press_1.wav is played).
The next is assigned the 2
key (ao_press_2.wav is
played), and so on.
Start AO Menu
Process
ao_main_menu_.wav
spw_change_prompt_.wav
ao_press_<n>.wav
_spw_change_prompt.wav
All configurable
options played ?
Is password
configurable?
No
Yes
Yes
Yes
Yes
Yes
Yes
Yes
No
_ao_main_menu.wav
ao_previous_menu_prompt.wav
rc_change_prompt_.wav
ao_press_<n>.wav
_rc_change_prompt.wav
Is roll call
configurable?
No
Get tones
Failed
attempts?
<5
10 sec. timer
qs_change_prompt_.wav
ao_press_<n>.wav
_qs_change_prompt.wav
Is quick start
configurable?
5
Anything
entered?
No
No
End AO Menu
Process
Yes
cc_auto_change_prompt_.wav
ao_press_<n>.wav
_cc_auto_change_prompt.wav
Is autocontinue
configurable?
Yes
* (star key)?
No
No
Account
Options
Help
lo_entry_change_prompt_.wav
ao_press_<n>.wav
_lo_entry_change_prompt.wav
Is listen only entry
configurable?
Yes
9?
No
No
ao_not_
enabled.wav
Other valid
key?
wr_change_prompt_.wav
ao_press_<n>.wav
_wr_change_prompt.wav
No
Is waiting room
configurable?
Yes
Setting change
process
corresponding to
key entered
228
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Figure A-25 Account options help
Account Options Help Process
Start AO Help
Process
Starts and ends in Account
Options Menu process.
ao_feature_overview_.wav
Is password
configurable?
Yes
Yes
Yes
Yes
Yes
Yes
Yes
spw_overview.wav
No
Is roll call
configurable?
rc_overview.wav
qs_overview.wav
No
Is quick start
configurable?
No
Is autocontinue
configurable?
cc_auto_overview.wav
lo_entry_overview.wav
wr_overview.wav
No
Is listen only entry
configurable?
No
Is waiting room
configurable?
No
Is announce
configurable?
eea_overview.wav
No
End AO Help
Process
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Figure A-26 Announcement setting process
Announcement Setting Process
Starts either from roll call setting process or from subscriber
entering DTMF command (default is *3) while in conference.
Ends where started.
Start Announce
Setting Process
Tone
Name
eea_tone_entry.wav
Announce type?
eea_name_entry.wav
Silence
eea_silence_entry.wav
eea_set_menu.wav
ao_previous_menu_
prompt.wav
ao_conf_return_
prompt.wav
Yes
No
Called from roll call
setting?
Get tones
10 sec. timer
Other key
Failed
attempts?
Anything
entered?
<3
No
ao_wrong_key.wav
Yes
3
Roll call
enabled?
No
1
No
* (star key)?
1, 2, or 3?
eea_name_
entry_no_rc.wav
Yes
2 or 3
Yes
End Announce
Setting Process
Set selected
announcement
option
Set
announcement
to Tone
Unexpected
error
eea_change_failure.wav
230
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Figure A-27 Autocontinuation setting process
Autocontinuation Setting Process
Starts and ends in account options menu.
Start
Autocontinue
Setting Process
cc_auto_off.wav
cc_auto_set_on_prompt.wav
cc_auto_on.wav
cc_auto_set_off_prompt.wav
Is autocontinue
on?
No
Yes
ao_previous_menu_prompt.wav
Get tones
ao_wrong_key.wav
10 sec. timer
No
Failed
attempts?
Anything
entered?
<3
No
Yes
No
* (star key)?
1?
3
Yes
Yes
Toggle
autocontinue
setting
End
Autocontinue
Setting Process
Unexpected
error
ao_change_failure.wav
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Figure A-28 Listen only entry setting process
Listen Only Entry Setting Process
Starts and ends in account options menu.
Start LO Entry
Setting Process
lo_entry_off.wav
lo_entry_set_on_prompt.wav
lo_entry_on.wav
lo_entry_set_off_prompt.wav
LOE
Off
LOE
On
LOE setting?
ao_previous_menu_prompt.wav
Get tones
ao_wrong_key.wav
10 sec. timer
No
Failed
attempts?
Anything
entered?
<3
No
Yes
No
* (star key)?
1?
3
Yes
Yes
Toggle LOE
setting
End LO Entry
Setting Process
Unexpected
error
ao_change_failure.wav
232
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Figure A-29 Password setting process
Password Setting Process
Starts and ends in account options menu.
Start Password
Setting Process
spw_set_prompt.wav
ao_previous_menu_prompt.wav
spw_invalid.wav
Get tones
spw_wrong_length.wav
10 sec. timer
<3
No
No
Failed
attempts?
Anything
entered?
Valid
password?
Valid number
of digits?
No
Yes
No
Yes
* (star key)?
3
Yes
Yes
spw_repeat_.wav
pw (numbers.wav)
_spw_repeat.wav
Password
changed
End Password
Setting Process
ao_change_failure.wav
spw_change_failure.wav
Unexpected
error
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Figure A-30 Quick start setting process
Quick Start Setting Process
Starts and ends in account options menu.
Start Quick Start
Setting Process
Is QS on for
subscription?
No
Yes
qs_off.wav
qs_on.wav
Is QS
setting same for
subscription and
conf?
No
qs_future_affecting.wav
qs_set_on_prompt.wav
Yes
Is QS on for
subscription?
No
Yes
qs_set_off_prompt.wav
ao_previous_menu_prompt.wav
Get tones
ao_wrong_key.wav
10 sec. timer
No
Failed
attempts?
Anything
entered?
<3
No
Yes
No
* (star key)?
1?
3
Yes
Yes
Toggle Quick
Start setting
End Quick Start
Setting Process
Unexpected
error
ao_change_failure.wav
234
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Figure A-31 Roll call setting process
Roll Call Setting Process
Starts and ends in account options menu.
Start Roll Call
Setting Process
rc_off.wav
rc_set_on_prompt.wav
rc_on.wav
rc_set_off_prompt.wav
No
Yes
Is Roll Call on?
Is entry
announcement
configurable?
No
Yes
ao_previous_menu_
prompt.wav
eea_change_prompt.wav
ao_wrong_key.wav
Get tones
10 sec. timer
<3
Other key
Failed
attempts?
Anything
entered?
Change roll call
setting
No
Yes
No
1
* (star key)?
1 or 2?
3
Yes
2
End Roll Call
Setting Process
Is entry
announcement
configurable?
No
eea_change_not_enabled.wav
Yes
ao_change_failure.wav
Announce
Setting
Unexpected
error
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Figure A-32 Waiting room setting process
Waiting Room Setting Process
Starts and ends in account options menu.
Start WR Setting
Process
wr_off.wav
wr_set_on_entry_prompt.wav
wr_set_on_lock_prompt.wav
wr_on_lock.wav
wr_set_on_entry_prompt.wav
wr_set_off_prompt.wav
WR
Off
WR
On
WR setting?
WR On, Conf Locked
wr_on_entry.wav
wr_set_off_prompt.wav
wr_set_on_lock_prompt.wav
ao_previous_menu_prompt.wav
Get tones
Failed
attempts?
Anything
entered?
<3
No
10 sec. timer
ao_wrong_key.wav
Yes
3
No
Set selected
Waiting Room
option
No
Yes
* (star key)?
1, 2, or 3?
Yes
End Announce
Setting Process
Unexpected
error
ao_change_failure.wav
236
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Figure A-33 Entry announce process
Entry Announce Process
Describes what the system plays to the conference (shaded boxes) as someone enters the
conference. Starts and ends in Subscriber, Participant, or Rejoin Conference process.
Start Entry
Announce
Process
Name
announce?
No
Yes
No
Anonymous?
Yes
eea_name_join_.wav
recorded name
_eea_name_join.wav
Tone
announce?
Yes
eea_tone_join.wav
eea_anon_join.wav
No
End Entry
Announce
Process
Error playing
.wav file(s)?
No
Yes
19 (DTMF)
When an operator joins, instead
of a recorded name, the system
plays pt_operator.wav.
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Figure A-34 Exit announce process
Exit Announce Process
Describes what the system plays to the conference (shaded boxes) as someone leaves
the conference. Starts in conference and ends with leaving party gone (temporarily, as
when subscriber departs to process WR or speak with operator, or permanently, as when
someone is disconnected from the bridge).
Start Exit
Announce
Process
Name
announce?
No
Yes
No
Anonymous?
Yes
eea_name_leave_.wav
recorded name
_eea_name_leave.wav
Tone
announce?
Yes
eea_tone_leave.wav
eea_anon_leave.wav
No
End Exit
Announce
Process
Error playing
.wav file(s)?
No
Yes
91 (DTMF)
When an operator leaves,
instead of a recorded name, the
system plays pt_operator.wav.
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Figure A-35 Waiting room process (subscriber)
Subscriber WR Process
Starts with subscriber in conference, which is locked with
waiting room. After process reaches end, it restarts if
any callers remain in waiting room or new caller arrives.
Start Sub WR
Process
New caller
Caller(s)
in WR?
Reminder
time?
No
Yes
No
Announce
type?
Name
None
Yes
Tone
wr_count_.wav
count (numbers.wav)
_wr_count.wav
Caller
anonymous
?
Reminders
on?
No
Yes
No
Yes
57 (DTMF)
Yes
wr_new_anon_
caller.wav
Toggle WR
Reminders
process
Reminders
on?
No
Get tones
wr_new_caller_.wav
<recorded name>
_wr_new_caller.wav
10 sec. timer
Yes
Yes
#5 entered
(process
callers)?
#6 entered
(toggle notif.)?
Reminder
time?
Yes
No
No
No
wr_empty.wav
wr_handling_prompt.wav
wr_private_conf_cmd_help.wav
No
Caller(s)
Yes
in WR?
10 sec.
timer
Get tones
No
Yes
* entered?
wr_private_conf_
cmd_help.wav
Join subscriber
to caller
Yes
No
# entered?
Invalid key
entered?
Yes
Get tones
No
No
No
No
No
1 entered?
2 entered?
3 entered?
* entered?
Yes
Yes
Yes
Yes
Join caller to
conference
Leave caller in
waiting room
Return sub to
conference
Disconnect caller
End Sub WR
Process
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Figure A-36 Toggle waiting room reminders process
Toggle Waiting Room Reminders Process
Starts in conference when subscriber enters command to toggle waiting room
reminders (default is #6). Ends at Rejoin Conference process.
Start Toggle
WR Reminders
Process
Reminders
on?
No
Yes
Toggle
acknowledge
type?
Toggle
acknowledge
type?
Prompt
Prompt
None
None
wr_notification_on.wav
or custom .wav file
wr_notification_off.wav
or custom .wav file
Tones
Tones
18 (DTMF)
81 (DTMF)
Rejoin
Conference
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Figure A-37 Roll call process
Roll Call Process
Starts in conference when participant or subscriber enters roll call command.
Ends in conference or at Rejoin Conference process.
Start Roll Call
Process
Roll Call
enabled?
Private or
conf?
Yes
Yes
Conference
Subscriber?
No
No
Private
Any
recorded
names?
Any
recorded
names?
cf_caller_count_.wav
count (numbers.wav)
_cf_caller_count.wav
Yes
Yes
No
No
rc_.wav
count (numbers.wav)
_rc_.wav
rc_conf_.wav
count (numbers.wav)
_rc_conf.wav
recorded names
(pt_operator.wav for operator;
pt_recorder.wav for recorder)
recorded names
(pt_operator.wav for operator;
pt_recorder.wav for recorder)
Any
anonymous
callers?
Any
anonymous
callers?
No
No
Yes
Yes
rc_anon_.wav
count (numbers.wav)
_rc_anon.wav
rc_conf_anon_.wav
count (numbers.wav)
_rc_conf_anon.wav
Rejoin
Conference
End Roll Call
Process
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Figure A-38 Operator request process
Operator Request Process
Starts in conference when participant or subscriber enters operator request
command (via DTMF or Moderator/API). Ends at Rejoin Conference process.
Start Operator
Request Process
DTMF or
Mod/API?
Oper reqs
enabled?
Subscriber or
Moderator?
Oper Req for
sub only?
Yes
No
Yes
Yes
Yes
Mod/API
No
Yes
No
DTMF
op_req_for_subs_only.wav
DTMF or
Mod/API?
Pending
oper req?
DTMF or
Mod/API?
Mod/API
Mod/API
op_gui_req_cancel.wav
DTMF
No
DTMF
op_req_cancel.wav
Cancel request
Any oper
available?
DTMF or
Mod/API?
Mod/API
op_req_not_enabled.wav
op_gui_req.wav
No
DTMF
op_req.wav
Send request
op_not_available.wav
Rejoin
Conference
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Figure A-39 Rejoin conference process
Rejoin Conference Process
When someone in conference enters a command, the system usually disconnects
that person’s line (channel) temporarily from the conference (although the conference
can still be heard). When the requested interaction or command execution is finished,
this process governs how the line is returned to the conference. It ends with the
subscriber or participant back in conference and the system waiting for the next
command.
Rejoin is silent when returning from:
Conference termination prompt
Start Rejoin Conf
Process
Private roll call
Private roll call cancellation
A command initiated via the
Moderator interface
Yes
No
Silent rejoin?
Recording start and end (but not
cancellation before recording is
established)
Conference ACM (if available)
Yes
No
mute_part_join.wav
lo_part_join.wav
cf_part_join.wav
Muted?
Yes
Listen only?
No
Back from
dial-out or oper
1-to-1?
Yes
Yes
Recording?
rec_rejoin_reminder.wav
No
No
Entry
Announce
No
Silent rejoin?
Yes
End Rejoin Conf
Process
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Figure A-40 End conference process (subscriber)
Subscriber End Conference Process
Starts in conference when subscriber enters end conference command (default is #8).
Ends at Rejoin Conference process or with conference termination.
Start Sub End
Conf Process
ct_manual_prompt.wav
Error
Get tones
10 sec. timer
Yes
No
1 entered?
ct_manual_end.wav
End conference
Hook all
ct_manual_canceled.wav
Rejoin
Conference
Parallel participant process (experienced by each participant)
Subscriber
entered #8
and then 1
ct_manual_disconnect.wav
Hook
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Figure A-41 In-conference commands by subscriber
In-conference Subscriber/Moderator Commands
All start in conference when subscriber enters indicated DTMF (touchtone) command
(defaults shown) or corresponding Moderator command. Also shown here is notification
of subscriber that an operator wants to join a locked conference.
Subscriber
entered **
Rejoin
Conference
cf_in_conf_subs_cmd_help.wav
Subscriber
entered *0/00
(or Moderator)
Operator
Request
process
Oper clicks Ring
for admittance to
locked conf
Rejoin
Conference
op_conf_doorbell.wav
Subscriber
Dial-out
process
Long
Subscriber
Long or
short?
entered *1
Short
Short Sub
Dial-out
process
Subscriber
entered *2
(or Moderator)
Recording
process
Subscriber
entered *3
(or Moderator)
Announce
changeable?
No
eea_change_in_conf_not_enabled.wav
Yes
Announce
Setting
process
Rejoin
Conference
Subscriber
entered *4
(or Moderator)
Waiting
room?
No
lck_conf_locked.wav
Rejoin
Conference
Yes
wr_conf_locked.wav
wr_cmd_help.wav
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Figure A-42 In-conference commands by subscriber (continued)
In-conference Subscriber/Moderator Commands
All start in conference when subscriber enters indicated DTMF (touchtone) command
(defaults shown) or corresponding Moderator command.
Subscriber
entered *5
(or Moderator)
Rejoin
Conference
lck_conf_unlocked.wav
Subscriber
entered *6
(or Moderator)
Rejoin
Conference
Mute
Tones
sound type?
No
mute_on.wav
Playing error?
Custom file
Yes
custom .wav file
35 (DTMF)
Subscriber
entered *7
(or Moderator)
Rejoin
Conference
Unmute
Tones
sound type?
No
mute_off.wav
Playing error?
Custom file
Yes
custom .wav file
53 (DTMF)
cc_change_not_
enabled.wav
No
Subscriber
entered *8
(or Moderator)
Turn on conf
continuation
Continuation
changeable?
No
cc_on.wav
Rejoin
Conference
Turn off conf
continuation
Continuation
on?
Yes
Yes
cc_off.wav
Subscriber
entered *9
(or Moderator)
Roll Call
process
Subscriber
entered *#
(or Moderator)
Rejoin
Conference
conferee count
None
Subscriber
entered ##
(or Moderator)
Mute
sound type?
Rejoin
Conference
Tones
Mute all
35 (DTMF)
Custom file
custom .wav file
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Figure A-43 In-conference commands by subscriber (continued)
In-conference Subscriber DTMF and Moderator Commands
All start in conference when subscriber enters indicated DTMF (touchtone) command
(defaults shown) or corresponding Moderator command.
None
Subscriber
entered 99
(or Moderator)
Mute
sound type?
Rejoin
Conference
Tones
Tones
Tones
Unmute all
Listen only all
LO all off
53 (DTMF)
Custom file
custom .wav file
None
LO
sound type?
Subscriber
entered #1
Rejoin
Conference
98 (DTMF)
Custom file
custom .wav file
None
LO
sound type?
Subscriber
entered #2
Rejoin
Conference
89 (DTMF)
Custom file
custom .wav file
Subscriber
entered #5
Sub WR
process
Sub
Subscriber
entered #6
Toggle WR
Reminders
process
Sub End
Conference
process
Subscriber
entered #8
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Figure A-44 In-conference commands by participant
In-conference Participant Commands
All start in conference when participant enters indicated
DTMF (touchtone) command (defaults shown).
Participant
Rejoin
Conference
cf_in_conf_part_cmd_help.wav
entered **
Operator
Request
process
Participant
entered *0/00
Participant
entered *6
Mute
sound type?
Rejoin
Conference
Tones
No
mute_on.wav
Playing error?
Custom file
Yes
custom .wav file
35 (DTMF)
Participant
entered *7
Unmute
sound type?
Rejoin
Conference
Tones
No
mute_off.wav
Playing error?
Custom file
Yes
custom .wav file
53 (DTMF)
Participant
entered *9
Roll Call
process
Participant
entered *#
Rejoin
Conference
conferee count
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Figure A-45 Conference termination process
Conference Termination Process
Starts in conference when system determines that conference has exceeded the
maximum time for a one-person (or configured minimum) conference. Ends at
Rejoin Conference process or with conference termination.
Start Conf
Termination
Process
Refer to
operator?
Operator
logged in?
Yes
Yes
No
No
Operator
assistance
ct_auto_prompt.wav
Wait at least 30
seconds
Get tones
10 sec. timer
<3
Let
conference
continue?
Failed
attempts?
Any tones
entered?
Rejoin
Conference
No
Yes
Yes
3
No
CONF_TERM_TIME_OUT
ct_auto_disconnect.wav
OPER_ACTION
op_disconnect.wav
End conference
Hook all
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Figure A-46 Other in-conference events
Miscellaneous Events
In-conference events experienced by those indicated in circumstances shown.
Participants only:
Subscriber
entered #8
and then 1
ct_manual_disconnect.wav
Hook
Subscriber and participants:
Channel or
conference?
Minutes to
shutdown
1 or
more
Shutdown
necessary
Channel
Conf
<1
sd_chan_.wav
minutes (numbers.wav)
_sd_chan.wav
sd_chan_now.wav
Minutes to
shutdown
1 or
more
<1
sd_conf_.wav
minutes (numbers.wav)
_sd_conf.wav
CARD_SHUTDOWN
sd_disconnect.wav
Yes
Time up?
sd_conf_now.wav
No
Hook
Yes
Time up?
Hook
No
Subscriber only:
Operator seeks
admittance to
locked conf
Rejoin
Conference
op_conf_doorbell.wav
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B
CDR Data Reference
This appendix describes post-conference data found in the Call Detail Records
(CDRs). CDRs provide the billing information for your ReadiVoice system.
CDR Processing Database
The CDR database is composed of five Informix tables. These tables are kept
in a DBSPACE within the Informix storage area. By convention, this DBSPACE
is called dbspace1. Since the storage of CDR information is not a core system
component (that is, not necessary for conference functionality), it’s designed to
have minimal impact on other Conference Allocation and Control System
(CACS) components. This includes all Informix references and queries.
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Informix CDR Tables
Five Informix tables hold information from completed conferences:
•
•
•
•
•
Conference Information (cdr_post_conf
Participant Information (cdr_post_part
Feature Information (cdr_post_state
Conference End Verification (cdr_end_conf
ACM Data (cdr_acm_data
)
)
)
)
)
The tables that follow describe the contents of each Informix table. Fields in
bold type have indexes.
Table B-1
Conference Information (cdr_post_conf)
Field Name
Field Type
Description
Unique billing ID or customer reference
number. In the CNOW database, this is
SUBSCRIBERID from the SUBSCRIBERINFO
table.
customer_id
INTEGER
Unique conference ID number.
Forced to zero (0).
conf_id
sub_id
INTEGER
SMALLINT
INTEGER
In the CNOW database, this is GROUPID from
the SUBSCRIBERINFO table.
resv_id
Not applicable.
chairperson_name
conf_actual_start
conf_actual_end
service_provider
CHAR(30)
INTEGER
INTEGER
CHAR(20)
Start time of conference in UNIX seconds.
End time of conference in UNIX seconds.
In the CNOW database, this is
CARRIERNAME from the CARRIERS table.
Not applicable.
reservation_op_id
participant_num
CHAR(6)
Number of actual participants who called into
the conference.
SMALLINT
Forced to zero (0).
Forced to zero (0).
voting_duration
q_a_duration
processed
SMALLINT
SMALLINT
SMALLINT
ReadiVoice sets this flag to zero (0). The
ReadiVoice CDR purge program will delete
CDRs that have this flag set to 1 by CDR
fetching programs.
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CDR Data Reference
Table B-1
Conference Information (cdr_post_conf) (continued)
Field Name
Field Type
Description
Title of subscriber. In the CNOW database, this
is TITLE from the SUBSCRIBERDETAIL table.
reserver_title
CHAR(10)
First name of subscriber. In the CNOW
database, this is FIRSTNAME from the
SUBSCRIBERINFO table.
reserver_first
CHAR(30)
CHAR(30)
CHAR(30)
Last name of subscriber. In the CNOW
database, this is LASTNAME from the
SUBSCRIBERINFO table.
reserver_last
reserver_phone
Phone number of subscriber. In the CNOW
database, this is PHONE from the
SUBSCRIBERDETAIL table.
Forced to zero (0).
resv_begin
resv_end
INTEGER
INTEGER
SMALLINT
CHAR(4)
CHAR(55)
CHAR(15)
INTEGER
Forced to zero (0).
Number of lines authorized for subscription.
Forced to “AUMM”.
resv_lines
conf_type_code
oper_notes
adver_code
billing_id
Not applicable.
Not applicable.
In the CNOW database, this is BILLINGID from
the SUBSCRIBERINFO table.
Forced to zero (0).
Forced to zero (0).
Forced to zero (0).
Forced to zero (0).
Forced to zero (0).
Forced to zero (0).
Forced to zero (0).
Forced to zero (0).
Forced to zero (0).
Forced to zero (0).
Forced to zero (0).
r_auto_lo_conf
r_fax_confirm
r_mon_100
SMALLINT
SMALLINT
SMALLINT
SMALLINT
SMALLINT
SMALLINT
SMALLINT
SMALLINT
SMALLINT
SMALLINT
SMALLINT
r_announce
r_announce_late
r_music
r_roll_call
r_smart_poll
r_operator_tone
r_conf_record
r_auto_dial
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Table B-1
Field Name
Conference Information (cdr_post_conf) (continued)
Field Type
SMALLINT
SMALLINT
SMALLINT
SMALLINT
SMALLINT
CHAR(30)
Description
Forced to zero (0).
r_security_enable
r_pre_notify
r_full_duplex
u_no_show
Forced to zero (0).
Forced to zero (0).
Forced to zero (0).
Forced to zero (0).
u_cancelled
Optional External ID B (an alternative ID) for
subscriber. In the CNOW database, this is
ExternalIdB from the SUBSCRIBERINFO
table.
externalid_b
Identifies the bridge on which the conference
was running.
bridge_id
card_num
INTEGER
SMALLINT
CHAR(1)
Identifies the card on which the conference
was running.
Identifies the status of the CDR. Possible
values:
cdr_status
‘‘
The conference ended normally.
‘?‘
The CDR is incomplete (i.e., the
conference is still active).
‘H‘
‘A‘
The conference ended abnormally
(e.g., the card died).
The CDR was cleaned by a CDR
cleaner (when the card running the
conference dies all of the calls from
that conference will have this status
unless they already had a status of
‘H‘or ‘ ‘).
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CDR Data Reference
Table B-2
Participant Information (cdr_post_part)
Field Name
Field Type
Description
Conference ID (key to cdr_post_conf table)
(indexed with part_id).
conf_id
INTEGER
Forced to zero (0).
sub_id
SMALLINT
CHAR(20)
Dial-in = A number
Dial-out = B number
subscriber_number
B number (DNIS).
Not applicable.
connect_number
billing_number
part_actual_start
CHAR(20)
CHAR(20)
INTEGER
Start time for participant in UNIX seconds
(when participant connects to bridge).
End time for participant in UNIX seconds
(when participant disconnects from bridge).
part_actual_end
INTEGER
Enum:
access_method
SMALLINT
0 = ACC_CAMM (subscriber dial-in).
1 = ACC_PAMM (participant dial-in).
4 = ACC_COMM (operator-initiated).
6 = ACC_CODS (subscriber dial-out).
7 = ACC_PODS (participant dial-out).
Participant ID number (indexed alone and with
conf_id).
part_id
INTEGER
Forced to zero (0).
part_privacy
SMALLINT
SMALLINT
Enum:
part_privilege
0 = ordinary participant
1 = subscriber
3 = operator
4 = recorder
If subscriber record, the subscriber’s TITLE
from the SUBSCRIBERDETAIL table (CNOW
database).
part_title
CHAR(10)
If participant, operator, or recorder record,
blank.
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Table B-2
Field Name
Participant Information (cdr_post_part) (continued)
Field Type
Description
If subscriber record, one of the following:
part_first
CHAR(30)
•
•
ACM Pins value, if any
Blank, if participant name updated (via
API, Operator, or Moderator) and no ACM
Pins
•
Otherwise, FIRSTNAME from
SUBSCRIBERINFO table
If participant record, one of the following:
•
•
ACM Pins value, if any
Blank, if participant name updated (via
API, Operator, or Moderator) and no ACM
Pins
•
Otherwise, “Participant”
If operator or recorder record, “Operator” or
“Recorder,” respectively.
If subscriber record, one of the following:
part_last
CHAR(30)
•
Updated name, if participant name
updated (via API, Operator, or Moderator)
•
Otherwise, LASTNAME from
SUBSCRIBERINFO table
If participant record, one of the following:
•
Updated name, if participant name
updated (via API, Operator, or Moderator)
and no ACM Pins
•
Otherwise, blank
If operator or recorder record, blank.
Bridge ID of bridge to which participant
connected.
bridge_id
INTEGER
Forced to 1.
bridge_leg_id
leg_type
SMALLINT
SMALLINT
SMALLINT
SMALLINT
Forced to zero (0).
Forced to zero (0).
gmt_offset
in_out_flag
Enum:
0 = dial-in
1 = dial-out
Forced to zero (0).
Forced to zero (0).
Forced to zero (0).
Forced to zero (0).
r_op_assist
SMALLINT
SMALLINT
SMALLINT
SMALLINT
r_price_req
r_person_person
r_listen_only
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CDR Data Reference
Table B-2
Participant Information (cdr_post_part) (continued)
Field Name
Field Type
SMALLINT
SMALLINT
INTEGER
Description
Forced to zero (0).
port_group
Forced to zero (0).
Forced to zero (0).
res_port_group
line_number
card_num
Identifies the card on which the conference
was running.
SMALLINT
Identifies the status of the CDR. Possible
values:
cdr_status
CHAR(1)
‘ ‘
‘?‘
The call ended normally.
The CDR is incomplete (i.e., the call
is still active).
‘H‘
‘A‘
The call ended abnormally (e.g., the
card died).
The CDR was cleaned by a CDR
cleaner (when the card running the
conference dies all of the calls from
that conference will have this status
unless they already had a status of
‘H‘or ‘ ‘).
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Table B-3
Field Name
conf_id
Feature Information (cdr_post_state)
Field Type
INTEGER
Description
Conference ID (key to cdr_post_conf table).
Participant ID (key to cdr_post_part table).
part_id
INTEGER
Type of feature used:
Duration Eventsa
feature
SMALLINT
8 = oper_request(operator request –
private)
9 = oper_confreq(operator request –
conference)
10 = dial_out(subscriber dial-out)
11 = conf_lock(conference locked)
15 = conf_record(conference recorded)
21 = chan_in_pre_acm(channel in pre-conf
ACM)
22 = chan_in_ic_acm(channel in in-conf
ACM)
Common Eventsb
4 = chan_join(someone joined the
conference)
12 = chan_mute(channel muted)
13 = chan_unmute(channel unmuted)
14 = virt_gavel(conference muted)
17 = roll_call(roll call enabled)
18 = conf_continue(conference
continuation enabled)
19 = quick_start(Quick Start conference)
20 = stream(not supported)
23 = op_req(operator request – private)
24 = op_confreq(operator request –
conference)
25 = cancel_req(cancel operator request)
28 = listen_only(channel is in listen only
mode)
29 = chan_in_wr(channel is in waiting room)
Reserved Events
0-3, 5-7, 16, 26, 27
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Table B-3
Feature Information (cdr_post_state) (continued)
Field Name
Field Type
Description
Participant ID of the operator answering the
request. Set for duration events 8 and 9;
otherwise N/A.
oper_id
INTEGER
Start timestamp of feature in UNIX seconds.
start_time
end_time
INTEGER
INTEGER
End timestamp of feature in UNIX seconds (set
to 0 for common events).
a. Duration events have both a start_time and end_time.
b. Common events require a start_time; end_time is always zero (0).
Table B-4
Conference End Verification (cdr_end_conf)
Field Name
Field Type
Description
Conference ID of conference that has all of its
CDRs stored in the database.
conf_id
INTEGER
Available for use as a water mark for
processing.
water_mark
SERIAL
The ACM Data table is available for storing additional data collected by ACM
(Application Control Mode) applications that you develop. The ACM
application is responsible for sending the collected data to the ACM manager
in the form key:value. See the ReadiVoice SDK for information about
developing ACM applications that can write to or read from this table.
Table B-5
ACM Data (cdr_acm_data)
Field Name
acm_index
Field Type
Description
SERIAL
Record index. Available for use as a water
mark for processing.
conf_id
INTEGER
Conference ID of conference from which this
ACM data was collected (indexed alone and
with part_id).
part_id
INTEGER
INTEGER
CHAR(50)
CHAR(50)
Participant ID of the participant associated with
this ACM data, if any.
start_time
acm_name
acm_value
Start timestamp in UNIX seconds when the
ACM data was collected.
Name of ACM data. This is the keysent by the
ACM application.
Collected ACM data. This is the valuesent by
the ACM application.Can be blank if no data
was collected.
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C
SNMP Events and Alarms
This appendix contains reference information for ReadiVoice system and
bridge events and their SNMP variables. It also describes SNMP logging,
which produces log files of usage and performance data, and the ReadiVoice
Monitoring Tool, which provides remote (pager or email) notification of alarm
events.
SNMP Reference
The ReadiVoice system supports the Simple Network Management Protocol
(SNMP). It provides two built-in methods of accessing SNMP data about the
system:
•
The SNMP Monitor applet lets you monitor the system in real time,
viewing data from the SNMP Management Information Base (MIB). See
•
The SNMP logging function lets you send system usage and performance
data to log files on an ongoing basis. To enable SNMP logging, see “Using
To do more with SNMP, including making configuration changes, you can use
standard network administration tools, such as HP Openview from Hewlett
Packard or similar tools from companies such as Sun Microsystems or IBM.
Figure C-1 provides an overview of how such a network management tool
interfaces with the ReadiVoice SNMP module and SNMP agent. The tables in
the following section show the fields in the SNMP MIB tables.
By default, all SNMP variables are the standard SNMP data type UInteger32
(range of values: 0…4294967295).
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Figure C-1 SNMP overview
ReadiVoice System
SNMP Agent
Voyant-Written Custom Agent
Extension for ReadiVoice
Other Modules
SNMP Module
Requests To Get Information Or
Change Configuration
System Configuration
Information
Alarms, Errors, and
Status Information
Asynchronous Notification of Changed
Information or Alarms
Network Management Station
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Contents of the SNMP MIB
The MIB contains four tables:
sysTable— system information
sysHistogramTable— system histogram (statistical) information
brgTable— bridge information
brgHistogramTable— bridge histogram (statistical) information
The tables that follow list all the fields in each MIB table.
Table C-1
The SNMP MIB’s system information table
Field
Description
Number of active bridges.
sysNumBridgesActive
sysNumModeratorDialOuts
sysNumOperatorDialOuts
sysNumDTMFDialOuts
sysNumDialsIn
Number of dial-outs made by moderator.
Number of dial-outs made by operator.
Number of dial-outs made by DTMF.
Number of dial-ins.
Number of attempts to join full conferences.
sysNumJoinFullFails
sysAlarmStatus
System alarm:
0
= false;
1
= true.
Number of active conferences.
sysNumConfsActive
sysModeratorUsedPrct
sysPortMaxCapacity
sysNumPortsReserved
sysNumPortsActive
sysNumOperatorsActive
sysNumModeratorsActive
sysOperQueueLength
Percent of conferences using moderator.
Total number of available ports in the system.
Number of reserved ports in the system.
Number of active ports in the system.
Number of operators logged into the system.
Number of moderators logged into the system.
Length of operator queue.
Number of ports reserved for operator voice paths in the system.
Number of operators with voice path established in the system.
SS7 routing failures – timeout.
sysNumOperatorsPortsReserved
sysNumOperatorsPortsActive
sysCrTimeout
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Table C-1
The SNMP MIB’s system information table (continued)
Field
Description
SS7 routing failures – no capacity.
sysCrNoCapacity
SS7 routing failures – no DB entry (bad number).
SS7 routing failures – out of translation number.
sysCrNoDBEntry
sysCrOutOfTranslationNum
Table C-2
The SNMP MIB’s system histogram table
Field
Description
Hour-based time (index into the table).
sysTime
Average size of the conferences for the system.
Maximum number of the conferences for the system.
Maximum number of reserved ports for the system.
Maximum number of active ports for the system.
Average wait time in the operator queue.
sysConfAvgSize
sysMaxConfActive
sysMaxPortsReserved
sysMaxPortsActive
sysOperQueueAvgWait
sysOperQueueMaxWait
Maximum wait time in the operator queue.
Table C-3
The SNMP MIB’s bridge information table
Field
Description
Bridge identifier (index into the table).
brgID
Bridge status:
brgStatus
0 = unknown
1 = bridgeUp
2 = bridgeLogout
3 = bridgeAlarm
4 = bridgeNew
5 = bridgeDeleted
Bridge maintenance status:
0 = unknown
brgMaintenanceStatus
1 = bridgeBusyOut
2 = bridgeInService
Number of dial outs made by moderator.
Number of dial outs made by operator.
brgNumModeratorDialOuts
brgNumOperatorDialOuts
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Table C-3
The SNMP MIB’s bridge information table (continued)
Field
Description
Number of dial outs made by DTMF.
brgNumDTMFDialOuts
brgNumDialsIn
Number of dial ins.
Number of active conferences.
brgNumConfsActive
brgPortMaxCapacity
brgNumPortsReserved
brgNumPortsActive
brgNumOperatorsActive
brgOperQueueLength
brgOperQueueWait
brgOperPortsReserved
brgOperPortsActive
brgIpAddress
Total number of available ports in the bridge.
Number of reserved ports in the bridge.
Number of active ports in the bridge.
Number of active operators in the system.
Length of operator queue.
Wait time in the operator queue.
Number of operator ports reserved on this bridge.
Number of operator voice paths established on this bridge.
Bridge IP address.
Fan alarm:
0
= false;
1
0
= true.
brgFanAlarm
Power supply alarm:
= false;
1
= true.
brgPowerSupplyAlarm
Table C-4
The SNMP MIB’s bridge histogram table
Field
Description
Bridge identifier (index into the table).
brgId
Hour-based time (index into the table).
brgTime
Average size of the conferences for the bridge.
Maximum number of the conferences for the bridge.
Maximum number of reserved ports for the bridge.
Maximum number of active ports for the bridge.
brgConfAvgSize
brgMaxConfActive
brgMaxPortsReserved
brgMaxPortsActive
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Table C-4
The SNMP MIB’s bridge histogram table (continued)
Field
Description
Average wait time in the operator queue.
brgOperQueueAvgWait
brgOperQueueMaxWait
cfgTimeFormat
Maximum wait time in the operator queue.
Time format:
0
= 12-hour clock;
1
= 24-hour clock.
Using the SNMP Log Files
page 141), it writes system usage and performance data to log files on an
ongoing basis. This data can be useful for trend analysis and capacity
planning.
SNMP logging creates two log files:
•
•
The SNMP log file contains selected bridge and system data.
The histogram log file contains system usage data on an hour-by-hour
basis.
Both log files are comma-separated-value (CSV) text files. The first line of each
file contains the field names. You can import the files into a spreadsheet or
other application for manipulation and analysis.
Every night, the SNMP logging function closes the current SNMP log file,
archives it as filename_archivedate.gz, and begins a new SNMP log file.
Once a week on Sunday night, it closes and archives the current histogram log
file (naming the archive in the same way) and begins a new one.
Contents of SNMP Log File
Figure C-2 shows a sample of an SNMP log file for a three-bridge system.
Because it’s impossible to fit an entire line (row) from the file on one line,
continuations of a single line are shown indented.
The first line in the sample contains the field names. The second and third lines
contain two example records. Note that their time stamps are ten minutes
apart, the polling interval used.
Table C-5 lists the fields in the SNMP log file records, along with brief
descriptions and the data from the two example records shown in the sample.
Note that, after the system data fields, the bridge data fields are repeated for
each additional bridge in the system. In the three-bridge example shown here,
these fields repeat twice (for bridges 2 and 3).
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Figure C-2 Sample SNMP log file
DATE/TIME,BRIDGEID,CONFERENCES,MAX_CAPACITY,ACTIVE_PORTS,ALLOCATED_PORTS,DIAL_INS,DTMF_DIAL_
OUTS,MODERATOR_DIAL_OUTS,OPERATOR_DIAL_OUTS,OPERATORS,OP_QUEUE,SYS_CONFERENCE,SYS_MAX_
CAPACITY,SYS_ACTIVE_PORTS,SYS_ALLOCATATED_PORTS,SYS_DIAL_INS,SYS_DTMF_DIAL_OUTS,SYS_
MODERATOR_DIAL_OUTS,SYS_OPERATOR_DIAL_OUTS,SYS_MODERATOR,SYS_MODERATOR_USAGE,SYS_OPERATORS,
SYS_OP_QUEUE_LENGTH,SYS_TIMEOUTS,SYS_NO_CAPACITY,SYS_NO_DATABASE_ENTRY,SYS_OUT_OF_
TRANSLATIONS_NUMBERS,BRIDGEID,CONFERENCES,MAX_CAPACITY,ACTIVE_PORTS,ALLOCATED_PORTS,DIAL_
INS,DTMF_DIAL_OUTS,MODERATOR_DIAL_OUTS,OPERATOR_DIAL_OUTS,OPERATORS,OP_QUEUE,BRIDGEID,
CONFERENCES,MAX_CAPACITY,ACTIVE_PORTS,ALLOCATED_PORTS,DIAL_INS,DTMF_DIAL_OUTS,MODERATOR_
DIAL_OUTS,OPERATOR_DIAL_OUTS,OPERATORS,OP_QUEUE
Tue May 2 16:30:11 2000,1,1,0,480,0,0,0,0,0,0,0,0,3,1440,12,20,12,0,0,0,0,0,0,0,0,
0,0,0,2,1,3,480,12,20,12,0,0,0,0,0,3,1,0,480,0,0,0,0,0,0,0,0
Tue May 2 16:40:12 2000,1,1,2,480,4,8,2,1,1,0,0,0,5,1440,16,22,14,1,1,0,2,40,0,0,0,
0,0,0,2,1,3,480,12,14,12,0,0,0,0,0,3,1,0,480,0,0,0,0,0,0,0,0
Table C-5
SNMP log file fields
Field
Description
Example 1
Example 2
Date/time stamp of record
(local time on CACS server)
Tue May 2
16:30:11 2000
Tue May 2
16:40:12 2000
DATE/TIME
Bridge ID
1
1
1
1
BRIDGEID
Bridge status, either 0 (off line)
or 1 (on line)
BRIDGE_STATUS
Number of active conferences
for the bridge
0
2
CONFERENCES
Port capacity available
480
0
480
4
MAX_CAPACITY
ACTIVE_PORTS
Ports in use – active for the
bridge
Ports reserved for the bridge
0
0
8
2
ALLOCATED_PORTS
DIAL_INS
Number of dial-ins for the
bridge
DTMF dial-outs for the bridge
0
0
1
1
DTMF_DIAL_OUTS
Moderator (GUI) dial-outs for
the bridge
MODERATOR_DIAL_OUTS
Operator (GUI) dial-outs for the
bridge
0
1
0
0
1
0
OPERATOR_DIAL_OUTS
OPERATORS
Number of operators for the
bridge
Operator queue length
(number) for the bridge
OP_QUEUE
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Table C-5
SNMP log file fields (continued)
Field
Description
Example 1
Example 2
Number of active conferences
for the system
3
5
SYS_CONFERENCE
Port capacity available
(number of ports in system)
1440
12
20
12
0
1440
16
22
14
1
SYS_MAX_CAPACITY
SYS_ACTIVE_PORTS
SYS_ALLOCATATED_PORTS
SYS_DIAL_INS
Ports in use – active across the
system
Ports in use – reserved across
the system
Number of dial-ins for the
system
Number of dial-outs for the
system
SYS_DTMF_DIAL_OUTS
SYS_MODERATOR_DIAL_OUTS
SYS_OPERATOR_DIAL_OUTS
SYS_MODERATOR
Number of moderator dial-outs
for the system
0
1
Number of operator dial-outs
for the system
0
0
Number of moderators on the
system
0
2
Percentage of conferences
using the moderator GUI
0
40
1
SYS_MODERATOR_USAGE
SYS_OPERATORS
Number of operators for the
system
1
Operator queue length for the
system
0
0
SYS_OP_QUEUE_LENGTH
SS7 routing failures – timeouts
0
0
0
0
SYS_TIMEOUTS
SS7 routing failures – no
capacity
SYS_NO_CAPACITY
SS7 routing failures – no DB
entry
0
0
0
0
SYS_NO_DATABASE_ENTRY
SS7 routing failures – out of
translation numbers
SYS_OUT_OF_TRANSLATIONS_NUMBERS
Bridge ID
2
1
2
1
BRIDGEID
Bridge status, either 0 (off line)
or 1 (on line)
BRIDGE_STATUS
Number of active conferences
for the bridge
3
3
CONFERENCES
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Table C-5
SNMP log file fields (continued)
Field
Description
Example 1
Example 2
Port capacity available
480
480
MAX_CAPACITY
ACTIVE_PORTS
Ports in use – active for the
bridge
12
20
12
12
14
12
Ports in use – reserved for the
bridge
ALLOCATED_PORTS
DIAL_INS
Number of dial-ins for the
bridge
DTMF dial-outs for the bridge
0
0
0
0
DTMF_DIAL_OUTS
Moderator (GUI) dial-outs for
the bridge
MODERATOR_DIAL_OUTS
Operator (GUI) dial-outs for the
bridge
0
1
0
0
1
0
OPERATOR_DIAL_OUTS
OPERATORS
Number of operators for the
bridge
Operator queue length
(number) for the bridge
OP_QUEUE
Bridge ID
3
1
3
1
BRIDGEID
Bridge status, either 0 (off line)
or 1 (on line)
BRIDGE_STATUS
Number of active conferences
for the bridge
03
0
CONFERENCES
Port capacity available
480
0
480
0
MAX_CAPACITY
ACTIVE_PORTS
Ports in use – active for the
bridge
Ports in use – reserved for the
bridge
0
0
0
0
ALLOCATED_PORTS
DIAL_INS
Number of dial-ins for the
bridge
DTMF dial-outs for the bridge
0
0
0
0
DTMF_DIAL_OUTS
Moderator (GUI) dial-outs for
the bridge
MODERATOR_DIAL_OUTS
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Table C-5
SNMP log file fields (continued)
Field
Description
Example 1
Example 2
Operator (GUI) dial-outs for the
bridge
0
0
OPERATOR_DIAL_OUTS
Number of operators for the
bridge
1
0
1
0
OPERATORS
OP_QUEUE
Operator queue length
(number) for the bridge
Contents of Histogram Log File
Figure C-3 shows a sample of a histogram log file. The first line in the sample
contains the field names. The remaining lines are example records. The
logging function writes a record to the histogram log every hour on the hour.
Table C-6 lists the fields in the histogram log records and describes them
briefly. Note that the data is for the entire system (all bridges) and is a snapshot
as of the time stamp, not an average for the one-hour period.
Figure C-3 Sample histogram log file
DATE/TIME,SYS_MAX_PORTS_RESERVED,SYS_CONF_AVG_SIZE,SYS_MAX_CONF_ACTIVE,SYS_MAX_PORTS_ACTIVE
06/05/00 0:00,30,5,5,24
06/05/00 0:00,30,5,5,24
06/05/00 0:00,30,5,5,24
06/05/00 0:00,30,5,5,24
06/05/00 0:00,30,5,5,24
06/05/00 0:00,30,5,5,24
06/05/00 0:00,30,5,5,24
06/05/00 0:00,30,5,5,24
06/05/00 0:00,30,5,5,24
Table C-6
Histogram log file fields
Field
Description
Date/time stamp of record (local time on CACS server)
DATE/TIME
Number ports reserved for the entire system
SYS_MAX_PORTS_RESERVED
SYS_CONF_AVG_SIZE
Average size (number of ports) of all currently active
conferences
Number of currently active conferences
SYS_MAX_CONF_ACTIVE
SYS_MAX_PORTS_ACTIVE
Number of currently active conferences ports
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D
ReadiVoice-IP with IP Tributaries
ReadiVoice-IP systems can be configured to enable the IP Tributaries feature.
This feature provides alternative access methods for initiating and entering a
conference. These methods perform authentication and call routing, and thus
can bypass elements of the front-end call flow.
To use this feature, you must provide an authentication mechanism separate
from the ReadiVoice system. The authentication mechanism puts one or more
pieces of the caller’s identifying information into the SIP INVITEmessage used
to access the system.
What are IP Tributaries?
IP tributaries are defined as part of the TOheader within the SIP INVITE
message sent to the ReadiVoice CACS. The following sections describe the
content of this TOheader.
Single Number,
Traditional Access
In the ReadiVoice single number, traditional access call flow, the SIP INVITE
message has a TOheader in the following format:
TO: sip:accessnumber@host
When the header has this format, the caller is routed to the ReadiVoice IVR
system and asked to enter identifying information. The caller who keys in this
information is then routed to the conference and enters the traditional
ReadiVoice call flow.
Access Code Included
In this type of SIP INVITE, the TOheader includes the access code. It has the
following format:
TO: sip:accessnumber@host;AC=accesscode
When the header has this format, the caller is routed directly to the conference
(rather than being prompted for an access code) and enters the traditional
ReadiVoice call flow.
In this case, the access code has already been validated.
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Password Included—
One-Password Call Flow
ReadiVoice-IP has three possible IP tributaries for one-password call flows.
Clear Text Password
In this type of SIP INVITE, the TOheader includes the access code and
password. It has the following format:
TO: sip:accessnumber@host;AC=accesscode;PCODE=password
When the header has this format, the caller enters the traditional ReadiVoice
call flow after the password prompt (rather than being prompted for an access
code and password). If the password within the header is invalid, the system
sends a 401 Unauthorizedresponse to the INVITE
.
PCODE=""is valid if the participant password field in the database is empty.
Per Call Authorization
In this type of SIP INVITE, the TOheader includes the access code and
verification of authorization. It has the following format:
TO: sip:accessnumber@host;AC=accesscode;AUTH=TRUE
When the header has this format and the SIP protocol authorization is valid,
the caller enters the traditional ReadiVoice call flow after the password
prompt (rather than being prompted for an access code and password). If the
authorization within the header is invalid, the system sends a 401
Unauthorizedresponse to the INVITE
.
Trusted Proxy
In this type of SIP INVITE, the TOheader includes the access code and
participant type. It has one of the following formats:
TO: sip:accessnumber@host;AC=acccesscode;PTYPE=SUBSCRIBER
or
TO: sip:accessnumber@host;AC=acccesscode;PTYPE=PARTICIPANT
When the header has one of these formats and the INVITEcomes from a
trusted proxy, the caller enters the traditional ReadiVoice call flow after the
password prompt (rather than being prompted for an access code and
password). If the proxy is not a trusted proxy, the system sends a 401
Unauthorizedresponse to the INVITE. If the PTYPEspecified is not valid, the
system sends a 400 Bad Requestresponse to the INVITE
.
Password Included—
Two-Password Call Flow
ReadiVoice-IP has three possible IP tributaries for two-password call flows.
Clear Text Password
In this type of SIP INVITE, the TOheader includes the password, but the access
code is not included or is empty. It has one of the following formats:
TO: sip:accessnumber@host;AC="";PCODE=password
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or
TO: sip:accessnumber@host;PCODE=password
When the header has one of these formats, the caller enters the traditional
ReadiVoice call flow after the password prompt (rather than being prompted
for a password). If the password within the header is invalid, the system sends
a
401 Unauthorizedresponse to the INVITE.
PCODE=""is also valid if the participant password field in the database is
empty.
Per Call Authorization
In this type of SIP INVITE, the TOheader includes the verification of
authorization, but the access code is not included or is empty. It has one of the
following formats:
TO: sip:accessnumber@host;AC="";AUTH=TRUE
or
TO: sip:accessnumber@host;AUTH=TRUE
When the header has one of these formats, the caller enters the traditional
ReadiVoice call flow after the password prompt (rather than being prompted
for a password). If the authorization within the header is invalid, the system
sends a 401 Unauthorizedresponse to the INVITE
.
Trusted Proxy
In this type of SIP INVITE, the TO header includes the participant type, but the
access code is not included or is empty. It has one of the following formats:
TO: sip:accessnumber@host;AC="";PTYPE=SUBSCRIBER
or
TO: sip:accessnumber@host;AC="";PTYPE=PARTICIPANT
or
TO: sip:accessnumber@host;PTYPE=SUBSCRIBER
or
TO: sip:accessnumber@host;PTYPE=PARTICIPANT
When the header has one of these formats and the INVITE comes from a
trusted proxy, the caller enters the traditional ReadiVoice call flow after the
password prompt (rather than being prompted for a password). If the proxy is
not a trusted proxy, the system sends a 401 Unauthorizedresponse to the
INVITE. If the PTYPEspecified is not valid, the system sends a 400 Bad
Requestresponse to the INVITE.
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Implementing IP Tributaries
To implement IP tributaries:
1
2
Voyant technical support must modify the .suarcconfiguration file and
enable the IP tributary configuration parameters.
You must create the mechanism (email, outside IVR system, website, etc.)
for collecting the correct information and formatting the URI of the
INVITE sent to the CACS.
You may implement multiple IP tributaries; none of these methods are
mutually exclusive.
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Index
Symbols
.odprocrc file 138
translation numbers 23
user (internal) 63
Administration home page 17
administration interface 16
A
Administration login dialog box 17
alarm conditions
changing settings 134
interpreting codes 136
monitoring for 135
access
managing 63
phone numbers 35
translation numbers 22
access class
access number assignments 40
adding 33
alarm configuration 131
application control mode 142
Application Providers page 43
application service providers (ASPs) 43
archives 93
defined 33
deleting 34
renaming 34
Access Classes page 33
access phone numbers
adding 36
changing access class 40
changing hidden numbers 41
deleting 42
ASPs
adding 43
deleting 46
editing 45
Auto Continuation 54
moving to another bridge 37
private, defined 35
reassigning subscribers 38
shared, defined 35
B
backup
archives 93
explained 93
logical logs 96
manual 115
nightly 93
working with 35
account option changes and PSPI 128
Add Provider page 45
adding
access class 33
access phone numbers 36
bridge 18
restarting 102
stopping 97
billing information fields 51
bridge
access number assignments 37
adding 18
bridge group 25
invalid subscriber passwords 66
number group 31
prompt set 167
assigning translation numbers 23
changing translation numbers 24
critical log page 146
deleting 21
provider 43
information 18
modifying 20
routing list 27
subscriber group 47
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monitoring 86
Solaris disk space 103
checklist, maintenance 92
cleanAccOptChanges.pl 128
commands, touchtone 2
company information fields 50
components of system 5
rebooting 113
routing lists 25
troubleshooting channels 149
bridge groups
adding 25
deleting 26
conference
managing 25
continuation settings, interaction 58
controlling 2
modifying 26
Bridge Groups page 26
bridges 5
Bridges page 20
recording 137
starting 2
status 72
Conference Allocation and Control System
C
(CACS) 5
CacsEvtUpdate table 110
call detail record (CDR) 251
call flow
Conference Continuation 56
Conference Entry with Count 57
conference options fields 54
Conference Status page 72
Conference Termination Option 57
Conference Termination Time 57
confidence factor 119
configuration, system 118
configuring
conference recording 137
provisioning interface 60
remote alarm notification 131
ConfirmDialNum in ive.ini 124
continuation
effect on call flow 12
setting interactions 58
Critical Logs page 145
customer support xvii
customizing
initial greetings 169
diagrams 205
shortened for dial out 124
call flow diagram, basic 13
call flow fields 52
call flow, basic 12
capacity alarms 131
CDR
by conference 74
by subscriber 73
database 251
for one conference 76
overview 72
purging old records 105
tables 252
changing
access number assignments 38
access numbers’ access class 40
access numbers’ bridge 37
bridge configuration 20
bridge group 26
hidden numbers 41
Moderator interface 163
mute/unmute confirmation 162
touchtone commands 155
internal user’s password 65
number group 32
provider record 45
provisioning labels 60
routing list 28
D
translation numbers 24
daily maintenance 97
database
archiving 93
backing up manually 115
CDR 251
monitoring 148
restoring 116
channel states 86
checking
defunct processes 104
Informix dbspace 99
Informix users 99
logical logging 100
ReadiVoice processes 97
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Index
troubleshooting 147
F
file system, checking 103
files
defunct processes 104
default prompt set 182
indexed WAVE 171
initial greeting 181
deleting
access class 34
access phone number 42
bridge 21
Fixed Access 5
bridge group 26
internal user 66
invalid subscriber passwords 67
number group 32
prompt set 168
fixed translation number 120
fixed translation number with 3-digit code 120
G
GetAccOptChanges 128
Global Services xvii
greetings, custom 169
greetings, initial 181
group information fields 49
group operator 64
groups, subscriber 46
provider 46
routing list 29
subscriber group 59
df -k 103
diagram
call flow 13
ReadiVoice system 6
signaling flow 8
Dial Out Permission 56
Dial Out Postfix 56
Dial Out Prefix 56
Dial-Out Billing configuration 126
Dial-Out Callflow, Shortened 124
dial-out testing 149
disk space
Informix 99
Solaris 103
DNIS, welcome on 169
document
conventions xvi
introduction xv
purpose xv
H
heart beat 19
hidden numbers, changing 41
histogram log file 270
I
indexed WAVE files 171
Informix
archives 93
backup 93
checking logical logging 100
checking users 99
dbspace 99
manual backup 115
restoring from manual backup 116
tables, CDR data 252
downgrade 115
DTMF commands 2
customizing 155
Intelligent Network Call Routing (INCR) 5
interfaces, overview of 3
internal users 3
DTMF-initiated changes, updating 128
E
Edit Group page 48
interval, heart beat 19
Edit Routing List page 28
introduction
to guide xv
editing
to ReadiVoice system 1
Invalid Subscriber Password Table page 67
invalid subscriber passwords 66
provider 45
subscriber group 48
email notification 132
Entry/Exit Announcement 55
IP
call processing 10
configurations 9
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system diagram 9
IP Tributaries, overview 14
ive.ini
ConfirmDialNum 124
Waiting Room notifications 125
bridge resources 86
conference information 72
database (dbMon.pl) 148
operator information 77
system 69
system status 71
monitoring tools
K
Channel State Monitor 86
ReadiVoice Monitoring Tool 135
SNMP Monitor 81
killing processes 104
L
moving
listen only touchtone commands 160
lock/unlock touchtone commands 158
logging, SNMP 266
access numbers to another bridge 37
subscribers to another access number 38
multiple providers 43
logical logs
mute all touchtone commands 160
mute/unmute
confirmations, custom 162
touchtone commands 159
backing up 96
explained 94
restarting backup 102
stopping backup 97
login dialog box 17
N
M
Name Record 55
non-routed system 5
maintenance
checklist 92
number group
daily 97
adding 31
Informix backup 93
infrequent tasks 111
weekly 103
managing
bridge groups 25
defined 30
deleting 32
managing 30
modifying 32
Number Groups page 31
bridge Information 18
invalid subscriber passwords 66
routing lists 27
O
system access 63
manual
one-click conference 12
onstat -d 99
onstat -l 100
onstat -u 99
opening ReadiVoice Administration 16
operator
registered subscriber groups 64
statistics 77
Operator Registered Subscriber Groups page 65
Operator Request Available 57
operator/maintenance stations 3
orphan processes 104
conventions xvi
purpose xv
media servers 5
Moderator
customizing 163
Moderator interface 2
Modify Existing Access Phone Number page 39
modifying
access number assignments 38
bridge 20
bridge group 26
number group 32
routing list 28
overview
call process 6, 10
user interfaces 3
translation number assignments 24
monitoring
278
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Index
P
call processing 6
configurations 5
system diagram 6
page
Access Classes 33
purging
Add Provider 45
AccOptChanges table 110
CacsEvtUpdate table 110
CDR records 105
Application Providers 43
Bridge Groups 26
subscribers 109
Bridges 20
Conference Status 72
Edit Group 48
Q
Edit Routing List 28
quick maintenance checklist 92
quick provisioning setup 60
Quick Provisioning Setup page 61
quick start
effect on call flow 12
settings 54
Invalid Subscriber Password Table 67
Modify Existing Access Phone Number 39
Number Groups 31
Operator Registered Subscriber Groups 65
Passwords 64
Prompt Sets 168
Quick Provisioning Setup 61
Routing Lists 27
R
Subscriber Group 47
racheck 97
random translation number 120
rastart 111
rastop 111
ReadiVoice
basic call flow 12
call flow diagram 13
management and operation 3
monitoring 69
Monitoring Tool 135
system components 5
System Administration home 17
System Status 71
Translation Numbers 23
pager notification 131
Passwords page 64
passwords, invalid 66
phone numbers, access 35
port, Web server 142
private access number 35
process overview 6, 10
ReadiVoice-IP
processes
call processing 10
checking 97
stopping, restarting 111
configurations 9
dialed phone numbers, URIs, and SIP 11
prompt sets
adding 167
default 182
deleting 168
overview 166
renaming 168
system diagram 9
ReadiVoice-PSTN
call processing 6
configurations 5
INCR signaling flow diagram 8
system diagram 6
providers
rebooting
bridge 113
Sun 112
adding 43
deleting 46
editing 45
Recorder Dial Out 56
provider information fields 43
working with multiple 43
recorderPhone, .odprocrc file 138
recorderSetupString, .odprocrc file 138
recording, conference 137
provisioning labels, changing 60
provisioning setup 60
PSPI
remote alarm configuration 131
account-option changes 128
renaming
access class 34
prompt set 168
required cron tasks 110
PSTN
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ReadiVoice Administration & Maintenance Guide
routing list 29
status
conference 72
restarting
system 71
ReadiVoice processes 111
Solaris 112
stopping
ReadiVoice processes 111
restoring
Solaris 112
purged CDR records 108
Subscriber Group page 47
roll call touchtone commands 161
routed system 5
routing lists 25
deleting 29
managing 27
modifying 28
renaming 29
subscriber groups
adding 47
deleting 59
editing 48
group information fields 49
working with 46
subscriber passwords, invalid 66
subscribers
reassigning access number 38
Routing Lists page 27
Sun server, rebooting 112
support xvii
system
components 5
configuration 118
diagram 6
management and operation 3
monitoring 69
status 71
System Administration home page 17
system administration interface 16
System Administration login dialog box 17
System Configuration page 118
System Status page 71
S
scripts, purge 105
servers 5
Session Initiation Protocol (SIP) 11
setting up provisioning 60
shared access number 35
Shortened Dial-Out Callflow 124
shutdown 112
signaling flow 8
SNMP.
SIP 11
SNMP
data types 261
logging 266
T
tables
MIB 263
AccOptChanges 110
CacsEvtUpdate 110
CDR 252
overview 262
reference 261
sample histogram log file 270
sample log file 267
SNMP MIB 263
talker update frequency 123
telephony troubleshooting 149
testing
SNMP logging 141
SNMP Monitor
Perfmeter tab 82
Properties dialog box 83
using 81
space
Informix dbspace 99
Solaris file system 103
cards, spans, channels 149
dial-in 152
dial-out 149
three-digit code on fixed translation number 120
timer, heart beat 19
touchtone commands 2
starting a conference 2
Statistical Port Management Confidence Factor
statistics, operator 77
280
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Index
customizing 155
listen only 160
lock/unlock 158
mute all / unmute all 160
mute/unmute 159
roll call 161
unmute all touchtone command 160
unmute touchtone command 159
update frequency, talker 123
updating account options changes 128
user (internal)
adding 63
changing password 65
deleting 66
managing access 63
touchtone initiated changes, updating 128
translation number type
fixed 120
fixed with 3-digit code 120
random 120
users, Informix 99
translation numbers
V
adding 23
VIP Conference 56
voice prompt fields 52
voice prompts and call flows 179
voice prompts, default 182
changing 24
defined 22
managing 22
Translation Numbers page 23
troubleshooting
database (dbMon.pl) 147
telephony 149
W
Waiting Room
disabling notifications 125
two-password call flow, overview 14
WAVE files, indexed 171
U
Web, controlling conference from 2
Web-server port, changing 142
weekly maintenance 103
Welcome on DNIS 169
UDO. See Dial-Out Billing.
unattended dial out (UDO). See Dial-Out Billing.
Universal Resource Identifier (URI) 11
unlock touchtone command 158
Proprietary & Confidential
281
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ReadiVoice Administration & Maintenance Guide
282
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