User guide
Table Of Contents
- About This Manual
- Overview
- Technology Enhancements in Release 5.1
- SIP Implementation
- Multi-Site Enterprise Management - AltiEnterprise Manager
- Multi-Chassis Gateway Support (ACM only)
- Capacity Improvements
- Voice Processing New Features and Enhancements
- PBX New Features and Enhancements
- Call Center New Features and Enhancements
- Client Application New Features and Enhancements
- IP Phone New Features and Enhancements (IP600, IP710, IP705)
- Key System Features
- Technology Enhancements in Release 5.1
- Software Installation & License Registration
- Getting Around AltiWare Administrator 5.1
- System Configuration
- Setting General Parameters
- Setting a System Number Plan
- Setting Business Hours
- Routing Calls on Holidays
- Configuring System Speed Dialing
- Defining System Call Restrictions
- Creating Account Codes
- Setting up Call Reports
- Country-Relevant Settings
- Audio Peripheral Configuration
- Activity
- Feature Profiles
- CT-Bus Configurations
- Voice Mail Configuration
- Auto Attendant Configuration
- Board Configuration
- Using the Triton Resource Board
- Using the Triton MeetMe Conference Board
- Configuring the Quantum Board
- Configuring the Triton Analog Station Board
- Configuring the Triton Analog Trunk LS/GS and LS Boards
- Configuring the Triton VoIP Board
- Configuring the Triton T1/E1 Board
- Configuring Virtual Boards SIPSP and H323SP
- Configuring the MAX Board
- Configuring the Virtual MobileExt Board
- Trunk Configuration
- Trunks Out of Service
- Channel Identification
- Opening the Trunk Configuration Window
- Selecting Trunks to Set Attributes
- Configuring One or Multiple Trunks
- Setting General Trunk Attributes
- H323 Tie Trunk Properties
- SIP Tie Trunk Properties
- SIP Trunk Properties
- Triton T1/E1 Trunk Properties
- Triton Analog Trunk GS/LS Properties
- Quantum Trunk Properties
- Incoming Call Routing
- Outgoing Call Blocking
- In Call Routing Configuration
- Out Call Routing Configuration
- Extension Configuration
- Setting Up IP Extensions
- AltiGen IP Phone Configuration
- Mobile Extension Configuration
- Hunt Group Configuration
- Paging Group Configuration
- Line Park Configuration
- Workgroup Configuration
- Managing and Using MeetMe Conference
- Network Configuration Guidelines for VoIP
- Enterprise VoIP Network Management
- Understanding VoIP Bandwidth Requirements
- Opening AltiEnterprise Manager
- Setting VoIP Codec Profiles
- Assigning Codec Profiles to IP Addresses
- Defining IP Networks
- Defining the IP Dialing Table
- The Multi-site VoIP Domain
- Working with Servers in the VoIP Domain
- Managing VoIP Domain Users
- Configuring Global Least Cost Routing
- When Information May Be Out of Sync
- System Report Management
- Tools and Applications
- E1-R2 and E1 ISDN PRI Installations
- Required Service Parameters
- Network Ports
- Technical Support & Product Repair Services
- Troubleshooting
- Index
Chapter 8: Setting Up IP Extensions
AltiWare ACC 5.1 Administration Manual 159
• SIP Extension Channel—Establishes a logical channel relationship with other
analog and MobileExt ports (displayed on the SIPSP board configuration, Channel
Mapping List).
• SIP Extension Channel Activation—Associates an extension with a SIP Extension
channel when IP phones register to the system (displayed in the Extension View
window).
• Media Channel—an RTP channel connects system-to-phone, or phone-to-phone,
system-to-system to carry the digitized voice stream. The codec resource on the
VoIP board will be allocated dynamically based on connection types. If both end
devices are IP phones, the media channel can be connected from IP phone to IP
phone using the IP phone’s codec, except when the following is true:
• H.323 tie-trunk is used
• SIP trunk is used
• codecs at two end devices are mismatched
• extension has Agent setting checked
• voice recording is enabled at the IP extension
• a NAT router exists between AltiServ and remote IP phone
SIP supports a direct connection of the voice stream between SIP phones. H.323 tie-
trunks still require the voice stream to connect to the server.