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Alvarion
SIP R2J, Voice Gateway
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B
Appendix B - Default Telephony
Parameters
In This Appendix:
This appendix provides the default se
ttings for various telephony parameters.
Contents
Main
Document History
Page
Legal Rights
Trade Names
Statement of Conditions
Warranties and Disclaimers
Exclusive Warranty
Disclaimer
Limitation of Liability
Disposal of Electronic and Electrical Waste
Important Notice
About This Manual
Contents
Chapter 1 - System Description
Chapter 2 - Installation
Chapter 3 - Using the Web Configuration Server
Appendix A - Internal Class 5 Services
Appendix B - Default Telephony Parameters
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1.1 Introducing the Voice Gateway
1.2 Specifications
1.2.1 Telephony and Fax Services
1.2.2 Security
1.2.3 Voice Quality
1.2.4 Configuration and Management
1.2.5 Bridge Functionality
1.2.6 Mechanical 1.2.7 Electrical
1.2.8 Connectors
1.2.9 Regulatory Standards Compliance
1.2.10 Environmental
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2.1 Installation Requirements
2.1.1 Packing List
2.1.2 Additional Installation Requirements
2.2 Front and Rear Panel Components
2.2.1 Connectors
2.2.2 Reset to Factory Default Configuration
2.2.3 LEDs
2.3 Installation and Commissioning
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2.4 Notes on Using the Voice Gateways in Alvarion's Systems
2.4.1 BreezeMAX System (Version 1.5 and higher)
2.4.2 BreezeACCESS VL System (Version 3.1)
Chapter 3 - Using the Web Configuration Server
In This Chapter:
Page
3.2 Accessing the Web Configuration Server
3.3 Using the Web Configuration Server
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3.4 Home Menu - Product Info Page
Page
3.5 WAN Menu
3.5.1 WAN Status Page
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3.5.2 WAN Configuration Page
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3.6 VLAN Tagging Menu
3.6.1 VLAN Tagging Page
3.6.2 Adding and Deleting VLANs
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3.6.4 VLAN Configuration Example 1
Untagged
POTS
3.6.5 VLAN Configuration Example 2
3.6.5.1 VG-1 Configuration
3.6.5.2 VG-2 Configuration
3.6.6 VLAN Configuration Example 3
3.6.6.1 Method 1
3.6.6.2 Method 2
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3.7 Telephone Menu
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The jitter buffer options are common to both lines (if applicable):
The following settings are available for each line:
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The SIP Extensions page includes the following components:
3.7.2 SIP Extensions Page
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3.7.3 NAT Traversal Configuration Page (SIP Only)
3.7.4 STUN Client Configuration Page (SIP only)
3.7.5 ToS Page
3.7.6 Line Configuration Page
3.7.7 Dial Plan Schemes
Page
3.7.7.1 Hotline
3.7.7.2 Adding/Removing Prefixes
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3.9 System Menu
3.9.1 Set Security Password Page
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3.9.2 Localization Page
The Localization page includes the following components:
3.9.3 SNMP Configuration Page
The SNMP Configuration page includes the following components:
3.9.4 Service Access Configuration Page
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3.10 Upgrade Page
3.10.1 Downloader Result Codes (hexadecimal)
3.10.1.1 Examples
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3.13 Parameters Summary
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A.1 Actions and Keypad Sequences
A.2 Using the Class 5 Services
A.2.1 Call Waiting
A.2.2 Call Inquiry
A.2.3 Call Alteration
A.2.4 Call Drop
A.2.5 3-Party Conference 1
Description: One on-going call active, place a new call to a third party and start conference.
Description: Available only when there are no calls active/in progress.
A.2.6 3-Party Conference 2
Description: Two on-going calls active, mix them into a conference session.
A.2.8 Call Forward
Description: Available only when there are no calls active/in progress.
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C.1 Metering Support
C.2 Sending VoIP Performance Data to a Remote System
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Glossary
