Polycom Sewing Machine 3725 70003 009F User Manual

®
ReadiVoice  
Administration &  
Maintenance Guide  
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Contents  
What is the ReadiVoice System? . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1  
How ReadiVoice Conferencing Works . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1  
Starting and Joining 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  
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Contents  
Running the dbMon.plScript . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 148  
Contents of dbMon.plLog File . . . . . . . . . . . . . . . . . . . . . . . . . . . . 148  
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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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Figures  
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  
The Operator Statistics Requests page . . . . . . . . . . . . . . . . . . . . . . . . . 78  
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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System Configuration page . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 118  
The SNMP logging settings in .odprocrc . . . . . . . . . 141  
Critical Logs page . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 145  
[DTMF_CMDS]section of default ive.inifile . . . . . . . . . . . . . . . . . . . . 157  
Prompt Sets page . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 168  
Checking the format of .wavfiles . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 180  
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Tables  
purgeCDRcommand options . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 106  
purgeCDRexamples . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 107  
Alarm section parameters of .odprocrcfile . . . . . . . . . . . . . . . . . . . . 134  
SNMP logging parameters of the .odprocrcfile . . . . . . . . . . . . . . . . 141  
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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.  
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.  
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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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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:  
Simplified URI  
The URI with the protocol name omitted (similar to omitting the “http:” in  
a Web address). For instance:  
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 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.  
For more information on this feature, see Appendix D.  
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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  
Administration interface are described in Chapter 7, “Customizing &  
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).  
The System Administration home page appears (Figure 2-1). A  
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  
appears (Figure 2-2).  
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Figure 2-1 System Administration home page  
Figure 2-2 Login dialog box  
3
Enter a user name and password authorized for Administration access.  
Then click OK.  
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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.  
The Bridges page appears (Figure 2-3 on page 20).  
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  
described on page 23.  
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.  
The Bridges page appears (Figure 2-3).  
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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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.  
The Bridges page appears (Figure 2-3 on page 20).  
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  
specified on the Bridges page as Trans DNIS Length (see “Maintaining  
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.  
The Bridges page appears (Figure 2-3 on page 20).  
2
Find the bridge to which you want to assign translation numbers and  
click its Translations link.  
The Translation Numbers page appears (Figure 2-4). If there are no  
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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6
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.  
The Bridges page appears (Figure 2-3 on page 20).  
2
Find the bridge whose translation numbers you want to change and click  
its Translations link.  
The Translation Numbers page appears (Figure 2-4 on page 23). It  
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.  
The Bridge Groups page appears (Figure 2-5). It lists existing bridge  
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.  
The Routing Lists page appears (Figure 2-6). It lists existing 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.  
The Edit Routing List page appears (Figure 2-7). It lists the bridge groups  
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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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.  
The Number Groups page appears (Figure 2-8).The list shows the existing  
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.  
The Access Classes page appears (Figure 2-9). The list shows the existing  
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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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.  
The Access Phone Numbers page appears (Figure 2-10). The Search panel  
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.  
The Access Phone Numbers page appears (Figure 2-10).  
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  
of the page in the Numbers panel (Figure 2-11).  
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,  
repeat 2 using different search criteria until you’ve retrieved the correct  
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 Numbers 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.  
The Access Phone Numbers page appears (Figure 2-10).  
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  
of the page in the Numbers panel (Figure 2-11).  
3
4
In the Numbers panel, find the access number whose subscribers you  
want to reassign and click its Change button (blue “c”).  
The Modify Existing Access Phone Number page appears (Figure 2-12).  
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 Numbers 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 numbers access class:  
1
In the navigation bar, click Access Numbers.  
The Access Phone Numbers page appears (Figure 2-10).  
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  
of the page in the Numbers panel (Figure 2-11).  
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  
subscribers (Figure 2-13). You can reassign them with the procedure on  
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.  
The Modify Existing Access Phone Number page appears (Figure 2-12).  
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 numbers hidden number:  
1
In the navigation bar, click Access Numbers.  
The Access Phone Numbers page appears (Figure 2-10).  
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  
of the page in the Numbers panel (Figure 2-11).  
3
4
Find the access number whose hidden number you want to change and  
click its Change button (blue “c”).  
The Modify Existing Access Phone Number page appears (Figure 2-12).  
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.  
The Access Phone Numbers page appears (Figure 2-10 on page 36).  
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  
of the page in the Numbers panel (Figure 2-11).  
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  
subscribers (Figure 2-13). You can reassign them with the procedure on  
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.  
The Application Providers page appears (Figure 2-14), listing any  
providers already defined in the system.  
Figure 2-14 Application Providers page  
2
Click Add Provider.  
The Add Provider page appears (Figure 2-15 on page 45).  
Enter the data for the provider. See Table 2-2 for field descriptions.  
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.  
The Application Providers page appears (Figure 2-14).  
2
Find the provider that you want to edit and click its Edit button (purple  
“e”).  
The Add Provider page appears (Figure 2-15).  
Modify the data as necessary. See Table 2-2 for field descriptions.  
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.  
The Application Providers page appears (Figure 2-14).  
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 Group page appears (Figure 2-16). The List panel shows  
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  
existing group you selected, and opens the Edit Group page (Figure 2-17).  
4
Edit the group information as described in “Editing a Subscriber Group,”  
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”).  
The Edit Group page appears (Figure 2-17).  
3
Enter or modify the data in the Group Information area. See Table 2-3 for  
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). Table 2-6 describes the settings.  
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
In the Conference Options section (Figure 2-20), configure Account  
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.  
Table 2-7 on page 54 describes the settings. Remember, the group’s  
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  
Provisioning Options. See Table 2-7.  
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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.  
The Conference Continuation setting determines whether subscribers  
can override the default continuation behavior for a specific conference.  
If you set Subscriber Configurable to No and Conference  
Continuation to No, then subscribers can neither change the default  
behavior nor override it; the Auto Continuation Setting (On or Off)  
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  
conferences. Doing so doesn’t change the default (Auto Continuation)  
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  
default behavior nor override it; the Auto Continuation Setting (On or  
Off) governs all the subscriber’s conferences. See “How Continuation  
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.  
This field has no effect if Dial Out Permission is set to No.  
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.  
This field has no effect if Dial Out Permission is set to No.  
Recorder Dial Out  
Determines whether a subscriber can choose to record a conference.  
Conference recording is available only if your system is configured  
properly and can dial out to a recording service or device. See “Enabling  
Conference Recording” on page 137 and ask your Polycom Global  
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.  
On After a configurable interval (set by Conference Termination  
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  
participants. Has no effect if Conference Termination Option is set to  
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  
Table 2-8 shows how Auto Continuation and Conference Continuation  
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.  
The Subscriber Group page appears (Figure 2-16 on page 47).  
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.  
The Quick Provisioning Setup page (Figure 2-21 and Figure 2-22) lets you  
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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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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Figure 2-22 Quick Provisioning Setup page (bottom)  
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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  
Groups” on page 46. You can, however, change a group administrator’s  
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 Passwords page appears (Figure 2-23). It lists the authorized users of  
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  
step 8). For a group provisioner, the group selected here is the only one to  
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.  
The Operator Registered Subscriber Groups page appears (Figure 2-24).  
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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 Users Password  
1
In the navigation bar, click Passwords.  
The Passwords page appears (Figure 2-23). It lists the authorized users of  
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.  
The Passwords page appears (Figure 2-23).  
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.  
The Invalid Subscriber Password Table page appears (Figure 2-25).  
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.  
The Invalid Subscriber Password Table page appears (Figure 2-25).  
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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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).  
The ReadiVoice internal home page appears (Figure 3-1). The navigation  
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  
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  
The System Status page (Figure 3-2) appears when you click the Show System  
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.  
For descriptions of the CDR data, see Appendix B.  
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  
page presents an expanded search panel (Figure 3-5), where you can specify  
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  
Report panel shows the data for one conference. Figure 3-5 shows the results  
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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Figure 3-5 Retrieving CDR data by conference  
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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  
data for the conference (Figure 3-6). The Conference Report panel at the top  
contains information about the conference and its participants. The  
Participant Features panel contains records of billable feature usage. For  
descriptions of the features that may be listed, see Table B-3, “Feature  
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  
the System Configuration page (see “Changing System Configuration  
Settings” on page 118). If it is, the ReadiVoice system creates an operator log  
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.  
The Requests page appears (Figure 3-7). It provides you with three scope  
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  
depending on the choice you made. Figure 3-8 shows the search fields that  
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  
(Figure 3-7) and has the same three options in the navigation bar (For All  
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  
depending on the choice you made. Figure 3-10 shows the search fields  
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  
(Figure 3-11). It shows:  
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  
individual bridge. See Figure 3-12.  
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.  
The SNMP Monitor Properties dialog box opens (Figure 3-13).  
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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  
system and for each individual bridge (Figure 3-14).  
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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.  
The Histograms window appears (Figure 3-15). It displays conference and  
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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Figure 3-15 Histograms window of SNMP Monitor  
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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.  
The initial Channel State Monitor page appears (Figure 3-16).  
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  
the bridge (Figure 3-17). The cable icon in the lower-right corner shows the  
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.  
Channel State Monitor opens an applet window for the card (Figure 3-18).  
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  
details for the selected span (Figure 3-19). The buttons on the right let you  
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  
details for the selected channel (Figure 3-20). If the channel is off hook, the  
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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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.  
“Backing Up Logical Logs” on page 96 describes how to make sure that  
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.  
See page 97.  
2
Check ReadiVoice processes:  
racheck  
The following processes should  
be running:  
Wait for all conferences to end.  
Run rastop and then rastart.  
odproc  
csc  
See page 111.  
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  
then run rastart. See page 111  
and page 112.  
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  
ontape -c. See page 96 and  
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.  
See page 102.  
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.  
The procedures in “Daily Maintenance Tasks” on page 97 assume that you’re  
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.  
Backing Up Logical Logs” describes how to set up continual tape backup of  
the logical logs. The procedures in “Daily Maintenance Tasks” on page 97  
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.  
In a new XTerm window, use the onstatcommand (see “Checking  
Informix Dynamic Server” on page 99) to verify that the system is  
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  
shown in Figure 4-1. Additional processes may also be listed.  
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  
web entry. Figure 4-3 shows an example.  
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  
database. Figure 4-4 shows an example.  
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.  
Once all the log files reach this condition, Informix stops! Reread “Logical Logs” on  
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  
shown in Figure 4-6.  
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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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  
capacity used for each partition. In the example shown in Figure 4-7, for  
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!  
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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  
example in Figure 4-8, the PID of the first defunct process is 123.  
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  
the loadtable.plscript referred to in “Restoring from a Manual Backup  
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.  
The System Configuration page appears (Figure 5-1).  
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.  
Overuse of the VIP Conference option (see page 56) can have a negative impact  
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  
Trans DNIS Length setting for the bridges (see “Maintaining Bridge  
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  
Trans DNIS Length setting for the bridges (see “Maintaining Bridge  
Information” on page 18) must be exactly three more than 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  
tested and supports only the combinations shown in Table 5-3.  
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).  
10 If necessary, restart the ReadiVoice application (see “Stopping and  
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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  
with Prompt Sets” on page 166 and ). The four voice prompt files are:  
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  
Sets” on page 166 and Appendix A. Contact your Polycom Global Services  
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  
use the recorderSetupStringvariables (see Table 5-7 on page 138), such as  
{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  
Table” on page 110 for details of how to use this script.  
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  
information, see Appendix A.  
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.  
Figure 5-2 shows the default configuration lines in the /etc/remotefile for a  
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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Configuring the ReadiVoice System  
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  
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  
.
Open the .odprocrcfile for editing and locate the alarm section (Figure 5-3).  
Referring to Table 5-4, change the values that you want to change, being  
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.  
The ReadiVoice Monitoring Tool starts running. See Figure 5-4.  
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  
file format requirements, see Appendix A.  
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  
configuration determined in 2. See “Recording Configuration Variables in  
4
5
Save your changes to .odprocrc. Then, as root, stop and restart the  
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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6
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  
For the default prompt text and file format requirements, see Appendix A.  
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  
variables listed in Table 5-7 to define a setup string that will uniquely  
identify each recording of each conference.  
See the example shown in Figure 5-5 on page 140 for one implementation.  
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  
Figure 5-5 shows an example of .odprocrcconfiguration settings for  
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  
configuration shown in Figure 5-5, the system sends (with DTMF digits  
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  
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  
described in Table 5-8. Figure 5-6 shows a sample [snmpMgr]section.  
3
Save your changes to .odprocrc. Then, as root, stop and restart the  
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  
Figure 7-1 shows the default ive.inifile’s [DTMF_CMDS]section. For each  
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  
alternatives are disabled. See “Detailed Information for Specific Touchtone  
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.  
The default ive.inifile (Figure 7-1 on page 157) enables the two-command  
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.  
The default ive.inifile (Figure 7-1 on page 157) enables the two-command  
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.  
The default ive.inifile (Figure 7-1 on page 157) enables the two-command  
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.  
The default ive.inifile (Figure 7-1 on page 157) enables the two-command  
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.  
The default ive.inifile (Figure 7-1 on page 157) enables only the private roll  
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.  
You’ll also have to change the relevant voice prompts (see Appendix A).  
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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  
customize them, see Appendix A.  
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 companys  
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  
created in 6 and group#.pngis the name of the customized image). Then,  
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  
Prompt sets are sets of .wavaudio files (in the WAVE format; see “Details  
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!  
See “Voice Prompt File Reference” on page 182 for a list of the default voice  
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  
Initial Greetings” on page 181 for a description of how symbolic links are used  
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.  
The Prompt Sets page appears (Figure 7-2), listing the available prompt  
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  
location where you put them in 5 into the subdirectory you created in 7,  
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  
for the custom greeting files you created. See “About Initial Greetings” on  
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.  
See Appendix A for descriptions of the customizable files (listed above),  
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
Create the DNIS-specific symbolic links (see “About Initial Greetings” on  
page 181) by 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).  
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  
this information as well. See “Working with Prompt Sets” on page 166 for  
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  
language. The next section, “Building Indexed WAVE Files, describes how to  
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  
directory specified in the genvoicecommand (see page 174).  
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  
WAVE files. They’re described in the “Indexed WAVE File Specifications”  
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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Diagnostics and Troubleshooting  
This chapter describes how to perform some diagnostic and troubleshooting  
tasks.  
Viewing Critical Logs  
The Critical Logs page (Figure 6-1) appears when you click the View Critical  
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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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  
“Checking Solaris Disk Space” on page 103). If it’s more than 90% full, remove  
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  
Bridges page in System Administration (see page 18).  
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  
rings. Skip to step 12.  
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  
step 10.  
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 all channels on the span, enter 1. Then skip to step 12.  
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  
or µ-law format. Figure A-1 shows an example of checking two .wavfiles.  
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  
Messages” on page 169), the tnl_mklnscript creates a symbolic link for each  
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  
ReadiVoice system. Table A-1 lists the default voice prompt files in  
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]  
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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]  
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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]  
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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  
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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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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]  
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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  
“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]  
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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]  
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Voice Prompts and Call Flows  
Call Flow Diagrams  
The following pages contain call flow diagrams for the ReadiVoice system.  
Figure A-2 provides a legend that explains some of the symbols and  
conventions used in the diagrams.  
The diagrams themselves are arranged into several groups or categories, as  
listed in Table A-2 below.  
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  
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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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ReadiVoice Administration & Maintenance Guide  
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  
216  
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Voice Prompts and Call Flows  
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  
Proprietary & Confidential  
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ReadiVoice Administration & Maintenance Guide  
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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ReadiVoice Administration & Maintenance Guide  
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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Voice Prompts and Call Flows  
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  
Proprietary & Confidential  
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ReadiVoice Administration & Maintenance Guide  
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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Voice Prompts and Call Flows  
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  
Proprietary & Confidential  
223  
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ReadiVoice Administration & Maintenance Guide  
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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Voice Prompts and Call Flows  
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  
Proprietary & Confidential  
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ReadiVoice Administration & Maintenance Guide  
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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Voice Prompts and Call Flows  
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  
Proprietary & Confidential  
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ReadiVoice Administration & Maintenance Guide  
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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Voice Prompts and Call Flows  
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  
Proprietary & Confidential  
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ReadiVoice Administration & Maintenance Guide  
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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Voice Prompts and Call Flows  
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  
Proprietary & Confidential  
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ReadiVoice Administration & Maintenance Guide  
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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Voice Prompts and Call Flows  
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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ReadiVoice Administration & Maintenance Guide  
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  
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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  
‘Hor ‘ ‘).  
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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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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  
‘Hor ‘ ‘).  
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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  
If you enable the SNMP logging function (see “Enabling SNMP Logging” on  
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  
Access Phone Numbers page 36, 38  
AccOptChanges table 110, 128  
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  
subscriber group settings 54, 56  
Critical Logs page 145  
cron 94, 108, 110  
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  
voice prompts 166, 171, 179  
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  
dbMon.pl 147, 148  
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-in testing 149, 152  
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  
hold extender message 130, 187  
I
indexed WAVE files 171  
Informix  
archives 93  
backup 93  
checking logical logging 100  
checking users 99  
dbspace 99  
logical logs 94, 96  
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  
extender, music hold 130, 187  
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  
large conference music hold extender 130, 187  
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  
mplex 97, 111  
multiple providers 43  
logical logs  
music hold extender 130, 187  
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  
odproc 97, 111  
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  
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Index  
P
call processing 6  
configurations 5  
system diagram 6  
page  
Access Classes 33  
purging  
Access Phone Numbers 36, 38  
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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routing list 29  
status  
conference 72  
restarting  
system 71  
ReadiVoice processes 111  
Solaris 112  
stopping  
ReadiVoice processes 111  
restoring  
Solaris 112  
manual backup 115, 116  
purged CDR records 108  
Subscriber Group page 47  
rlogcap 97, 111  
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  
tapes 97, 102  
tar 115, 116  
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  
tnl_mkln (make link) command 122, 168, 170  
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 messages 169, 181  
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  
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282  
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