User manual
Table Of Contents
- Table of Contents
- VoicePlus Overview
- Defining Standard Audio Only Conferences and Reservations
- Monitoring On Going Conferences
- General Monitoring
- Participants Queue
- Conference Level Monitoring
- Participant Level Monitoring
- Operations Performed During On Going Conferences
- Management Functions Overview
- Participant Level Operations
- Making Dial-Out Connections
- Changing Participant Connection Types (Dial-In/Dial-Out)
- Disconnecting Participants from Conferences
- Naming Undefined Dial-in Participants
- Changing the Disconnected Participant’s Properties
- Moving a Participant from one Conference to Another
- Designating an Exclusive Speaker
- Changing Participant’s Status to Conference Chairperson
- Designating a VIP Participant
- Adjusting Participant’s Broadcasting and Listening Volume
- Muting and Unmuting Participant’s Audio
- Enabling/Disabling Auto Gain Control (AGC)
- Modifying the Participant’s User Defined Properties
- Conference Level Operations
- Adding New Participants to a Conference
- Muting Dial-In Participants Upon Connection
- Adding Remarks During an On Going Conference
- Locking and Unlocking a Conference
- Managing Question-and-Answer Sessions
- Managing Voting Sessions
- Placing a Conference On Hold
- Modifying Conference General Parameters
- Changing the Conference Duration
- Ending a Conference before its Scheduled Termination Time
- Rescheduling Conference Reservations
- Deleting Recurring Reservations
- Printing Conference Data
- Managing Conferences Using DTMF Codes
- Meeting Rooms and Entry Queues
- IVR and Entry Queue Services
- Attended Conferencing
- Requirements for an Attended Conference
- Defining an Operator Conference
- Setting the Participants Connection to the Conference to Attended Mode
- Participants Queue Management
- Managing Attended Participants from the Browser, Status and Monitor Panes
- Recording
- Appendix A: Glossary
Chapter 2 - Defining Standard Audio Only Conferences and Reservations
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7. Define the following parameters:
Table 2-12: H.323 and SIP Participant Properties - Identification
Field/Option Description
Participant IP Enter the IP address of the participant’s endpoint.
• For H.323 participants:
— In a Dial-out connection, enter the IP address
or alias of the endpoint to be called by the
MCU. If you entered only an alias the
gatekeeper can resolve the participant’s alias
into an IP address.
— In a Dial-in connection, the participants IP
address or alias are entered. The IP address
or alias is used to identify and route the
participant to the appropriate conference.
• For SIP endpoints, Participant IP is optional as
these participants can be located by using their
URI addresses.
Signaling Port Indicates the signaling port used for participant
connection.
For H.323 the default port is 1720.
For SIP, the default port is 5060.
SIP Address
SIP Only
Enter the endpoint address in the format:
[user name]@[domain].
Note: The SIP URI adheres to URI rules: no spaces
or special characters such as commas, quotation
marks, inverted tags and so forth either in the user
name or in the domain part.