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
MGC Manager User’s Guide - VoicePlus Edition
2-61
Address Type
SIP Only
Select the format in which the SIP address is written:
• SIP URI – Uses the format of an E-mail address,
typically containing a user name and a host
name: sip:[user]@[host]. For example,
sip:dan@polycom.com.
• TEL URI – Used when the endpoint does not
specify the domain that should interpret a
telephone number that has been input by the
user. Rather, each domain through which the
request passes would be given that opportunity.
As an example, a user in an airport might log in
and send requests through an outbound proxy in
the airport, not by the user's home domain.
Alias Name
H.323 Only
If you are using the endpoint’s alias and not the IP
address, first select the type of alias and then enter
the endpoint’s alias as registered with the
gatekeeper. Enter the alias name using the naming
conventions appropriate to the selected Alias type.
Alias Type
H.323 Only
Select the appropriate alias type before entering an
alias name:
• H.323 ID (alphanumeric ID)
• E.164 (digits 0-9, *, #)
• URL ID (URL style address)
• Email ID (email address format)
• Transport ID (IP address: port number)
• Party Number (identical to the E.164 format)
Notes: Although all types are supported, the
gatekeeper capabilities dictate the Alias Type to be
used, mainly H.323 ID and E.164. Refer to your
gatekeeper documentation for information on its
supported alias types.
Table 2-12: H.323 and SIP Participant Properties - Identification (Continued)
Field/Option Description