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 6: Trunk Configuration
AltiWare ACC 5.1 Administration Manual 107
SIP Trunk Properties
Traditionally telecom trunks are from your local carrier’s PSTN switch and the dial tone
is provided via either analog trunks or T1/PRI digital trunks. A new type of service called
“IP Dial Tone,” which allows you to dial a long distance call at a lower rate, is available.
IP Dial Tone is delivered through your IP data network, and the service provider can be
anywhere in the world, as long as the VoIP data packets can be routed properly.
If you have SIP-based IP dial tone service from an Internet Telephony Service Provider
(ITSP), you need to configure SIP trunk channels to connect the service. Before you
start, note the following:
• An AltiGen SIP Trunking channel is licensed. You need to buy and register a license
to be able to configure this option.
• AltiGen does not guarantee the voice quality of the SIP dial tone coming from your
service provider. You need to work with your data service and SIP trunking service
provider to make sure adequate QoS is provisioned for your WAN service.
• AltiGen does not guarantee SIP trunk implementation will work with all SIP Dial Tone
service providers. You need to verify that your SIP Dial Tone service provider
supports the following:
• G.711, G.723.1, G.729 codec
• RFC 2833 for DTMF tone delivery
• SIP MD5 authentication with SIP registration
• If AltiServ is behind NAT, verify that your SIP SP can support this configuration.
When subscribing to SIP Dial Tone service, typically your service provider will provide
you with the information required in the configuration dialog box shown in Figure 5 on
page 108. Enter these service parameters to each SIP trunk channel configuration
individually.
Note: This is signal only trunks. Make sure you have enough IP resource boards to cover
your needs.
Important: You need to add the SIP Trunk service provider’s IP address to the IP Device
Range in AltiEnterprise Manager and select the proper codec profile for this
service. See “Assigning Codec Profiles to IP Addresses” on page 278. Failure
to do this step may cause no voice path, even if the SIP Trunk channel shows
the call is connected.
Configuring a SIP Trunk
To open a trunk configuration dialog box for a SIP trunk, do one of the following:
• In the Trunk Configuration window, select a SIP trunk type, click the Trunk
Properties button, then click the SIP Trunk Configuration button.
• In the Board View window, double-click a SIPSP board type, click the Board
Configuration button, then click the SIP Trunk Configuration button.