Cisco 8831 Administration Manual page 11

Get Started
This chapter contains basic information on Cisco Unified IP Conference Phone 8831 for Third-Party Call
Control. This chapter contains the following sections:
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Overview of the Conference Phone
The Cisco Unified IP Conference Phone 8831 for Third-Party Call Control is a full-featured VoIP
(Voice-over-Internet Protocol) phone that provides voice communication over an IP network. It provides
all the features of traditional business phones, such as call forwarding, redialing, speed dialing,
transferring calls, and conference calling. The Conference Phone is targeted for solutions centered on
3rd-Party SIP-based IP PBX.
For more information on phone features, see the data sheets for this product.
Network Configurations
The Cisco Unified IP Conference Phone 8831 for Third-Party Call Control is used as a part of a SIP
network as it supports Session Initiation Protocol (SIP).
This document describes some common network configurations; however, your configuration can vary,
depending on the type of equipment used by your service provider.
Other SIP IP PBX Call Control Systems
The Cisco Unified IP Conference Phone 8831 for Third-Party Call Control are compatible with other
SIP IP PBX call control systems, such as BroadSoft, MetaSwitch, and Asterisk. Configuration of these
systems is not described in this document. For more information, see the documentation for the SIP PBX
system to which you are connecting the Cisco Unified IP Conference Phone 8831 for Third-Party Call
Control.
Overview of the Conference Phone, page 1-1
Network Configurations, page 1-1
Use the Web-Based Configuration Utility, page 1-2
View Phone Information, page 1-4
Wireless Microphone Region Setting, page 1-5
Cisco Unified IP Conference Phone 8831 for Third-Party Call Control Administration Guide
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