AudioCodes 310HD Specifications

AudioCodes 310HD Specifications
AudioCodes Series of High Definition IP Phones
HD VoIP
Administrator’s Manual
310HD / 320HD / 420HD IP Phones
Version 2.0.0
Administrator's Manual
Contents
Table of Contents
1 Introduction ....................................................................................................... 15 Configuration Tools ................................................................................................17 2 IP Phone User Interface .................................................................................... 19 2.1 Accessing the Administration Menu ....................................................................... 19 2.1.1 310HD IP Phone ......................................................................................................19 2.1.2 320HD IP Phone ......................................................................................................20 2.1.3 420HD IP Phone ......................................................................................................21 2.2 Changing Display Language .................................................................................. 22 2.2.1 310HD IP Phone ......................................................................................................22 2.2.2 320HD IP Phone ......................................................................................................23 2.2.3 420HD IP Phone ......................................................................................................24 3 Web Interface ..................................................................................................... 25 3.1 Accessing Web Interface ....................................................................................... 25 3.2 Getting Started with the Web ................................................................................. 26 4 Configuration File.............................................................................................. 27 4.1 4.2 4.3 4.4 4.5 5 Introduction ............................................................................................................ 27 File Syntax ............................................................................................................. 27 Linking Multiple Files .............................................................................................. 27 Downloading Configuration File from IP Phone ..................................................... 28 Creating Configuration Files using VoIProvision Utility .......................................... 29 4.5.1 Introduction ..............................................................................................................29 4.5.2 Configuration File Format ........................................................................................29 4.5.3 Global Configuration File .........................................................................................29 4.5.4 VoIProvision Utility Overview ...................................................................................29 4.5.5 CSV File ...................................................................................................................30 4.5.6 Template File ...........................................................................................................30 4.5.7 Generated Configuration Files .................................................................................30 4.5.8 Starting the VoIProvision Utility ...............................................................................30 4.5.9 Usage.......................................................................................................................31 TR-069 Management ......................................................................................... 33 Automatic Provisioning ..........................................................................................35 6 Introduction ....................................................................................................... 37 7 Updating (Manually) Configuration File .......................................................... 39 8 Setting up Network for Auto Provisioning ...................................................... 41 9 Obtaining Firmware and Configuration ........................................................... 43 9.1 Methods ................................................................................................................. 43 9.1.1 DHCP .......................................................................................................................43 9.1.1.1 Configuring DHCP Option 160 .................................................................47 9.1.1.2 Configuring DHCP Options 66 and 67 .....................................................47 9.1.2 Static DNS Record (Generic Domain Name) ..........................................................48 9.1.3 SIP SUBSCRIBE and NOTIFY Messages ..............................................................50 Version 2.0.0
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9.2 Protocols for Acquiring Files .................................................................................. 53 Quick Setup..............................................................................................................55 10 Quick Setup ....................................................................................................... 57 Networking ...............................................................................................................59 11 Introduction ....................................................................................................... 61 12 Configuring Date and Time .............................................................................. 63 12.1 Configuring Date and Time .................................................................................... 63 12.2 Configuring Network Time Protocol Server ............................................................ 67 12.3 Configuring Network Time Protocol Server via DHCP ........................................... 69 13 IP Network Settings........................................................................................... 71 13.1 Configuring Static IP Address ................................................................................ 71 13.1.1 Configuring the Static IP Address using Phone User Interface ...............................71 13.1.1.1 310HD IP Phone ......................................................................................71 13.1.1.2 320HD IP Phone ......................................................................................72 13.1.1.3 420HD IP Phone ......................................................................................74 13.1.2 Configuring IP Network Settings using Web and Configuration File .......................76 13.2 Configuring Partial DHCP ...................................................................................... 78 14 LAN and PC Port Settings ................................................................................ 81 15 VLAN Settings ................................................................................................... 83 16 Enabling Telnet Access .................................................................................... 85 VoIP Settings ...........................................................................................................87 17 SIP Settings ....................................................................................................... 89 17.1 Configuring General SIP Settings .......................................................................... 89 17.2 Proxy and Registration ........................................................................................... 93 17.2.1 Configuring General Proxy and Registration Settings .............................................93 17.2.2 Configuring Proxy Redundancy ...............................................................................96 17.3 Configuring SIP Timers .......................................................................................... 99 17.4 Configuring SIP QoS ............................................................................................ 101 17.5 Configuring SIP Reject Code ............................................................................... 102 18 Dialing Settings ............................................................................................... 103 18.1 Configuring General Dialing Settings ................................................................... 103 18.2 Tones ................................................................................................................... 105 18.2.1 Uploading Ring Tones ...........................................................................................107 18.2.2 Regional Settings for Call Progress Tones............................................................109 18.3 Configuring Default Audio Device ........................................................................ 112 18.4 Configuring DTMF ................................................................................................ 113 18.5 Configuring Digit Maps and Dial Plans................................................................. 114 19 Media Settings ................................................................................................. 117 19.1 Configuring Media Streaming............................................................................... 117 Administrator's Manual
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19.2 Configuring RTP Port Range and Payload Type ................................................. 118 19.3 Configuring RTP QoS .......................................................................................... 119 19.4 Configuring Codecs.............................................................................................. 120 20 Voice Settings ................................................................................................. 123 20.1 Configuring Jitter Buffer ....................................................................................... 123 20.2 Configuring Silence Compression ........................................................................ 124 20.3 Configuring Echo Cancellation ............................................................................. 125 21 Lines/SIP Accounts Settings .......................................................................... 127 21.1 Configuring Lines/SIP Accounts using Phone Interface....................................... 127 21.1.1 310HD IP Phone ....................................................................................................127 21.1.2 320HD IP Phone ....................................................................................................129 21.1.3 420HD IP Phone ....................................................................................................130 21.2 Configuring Lines using Web and Configuration File ........................................... 132 22 Supplementary Services................................................................................. 135 22.1 Selecting the Application Server .......................................................................... 135 22.2 Configuring Call Waiting....................................................................................... 136 22.3 Configuring Call Forwarding................................................................................. 137 22.4 Configuring Three-way Conferencing .................................................................. 139 22.5 Configuring Automatic Dialing .............................................................................. 140 22.6 Configuring Automatic Answer ............................................................................. 141 22.7 Configuring Do Not Disturb .................................................................................. 142 22.8 Configuring Call Pickup ........................................................................................ 143 22.9 Configuring Message Waiting Indication .............................................................. 144 22.10 Configuring Busy Lamp Field ............................................................................... 146 22.11 Configuring Advice of Charge .............................................................................. 148 22.12 Enabling Electronic Hook Switch ......................................................................... 149 23 Volume Levels ................................................................................................. 151 23.1 23.2 23.3 23.4 23.5 23.6 Configuring Automatic Gain Control..................................................................... 151 Configuring Tone Volume .................................................................................... 152 Configuring Ringer Volume .................................................................................. 153 Configuring Speakerphone (Hands Free) Volume ............................................... 154 Configuring Handset Volume ............................................................................... 156 Configuring Headset Volume ............................................................................... 158 Advanced Phone Settings ....................................................................................161 24 Locking and Unlocking the IP Phone ............................................................ 163 24.1 Locking and Unlocking the 310HD IP Phone ....................................................... 163 24.2 Locking and Unlocking the 320HD IP Phone ....................................................... 164 24.2.1 Single-Line Mode ...................................................................................................164 24.2.2 Multi-Line Mode .....................................................................................................165 24.3 Locking and Unlocking the 420HD IP Phone ....................................................... 166 24.3.1 Single-Line Mode ...................................................................................................166 24.3.2 Two-Line Mode ......................................................................................................167 Version 2.0.0
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24.4 Using the Configuration File ................................................................................. 168 25 Phone Directory............................................................................................... 169 25.1 Corporate Directory .............................................................................................. 169 25.1.1 LDAP-based Corporate Directory ..........................................................................169 25.1.2 Loading a Text-based Corporate Directory File ....................................................172 25.2 Local Phone Directory .......................................................................................... 174 26 Speed Dial ........................................................................................................ 175 26.1 26.2 26.3 26.4 Adding a Speed Dial ............................................................................................ 175 Deleting a Speed Dial .......................................................................................... 177 Saving Speed Dial File Configuration to a File..................................................... 177 Creating a Speed Dial File for Configuration File ................................................. 177 Security Settings ...................................................................................................179 27 Transport Layer Security and Certificate Management ............................... 181 27.1 SIP over TLS Overview ........................................................................................ 181 27.1.1 Obtaining Certificate Authority File ........................................................................181 27.1.2 Appending Private and Public Key Data................................................................182 27.2 Configuring TLS ................................................................................................... 183 27.2.1 Configuring TLS using the Configuration file .........................................................183 27.2.1.1 Configuring TLS using a URI ................................................................ 183 27.2.1.2 Configuring TLS by Loading File Content ............................................. 185 27.2.2 Manually Configuring TLS using the Web .............................................................187 28 Using the Encryption Tool .............................................................................. 189 28.1 Encrypting Configuration Files ............................................................................. 189 28.2 Encrypting Passwords in Configuration File......................................................... 189 29 Secure Real-time Transport Protocol ............................................................ 191 30 Configuring 802.1x .......................................................................................... 193 30.1 Configuring 802.1x using Phone User Interface .................................................. 193 30.1.1 310HD IP Phone ....................................................................................................193 30.1.1.1 EAP-MD5 Mode .................................................................................... 194 30.1.1.2 EAP-TLS Mode ..................................................................................... 195 30.1.2 320HD IP Phone ....................................................................................................196 30.1.2.1 EAP-MD5 Mode .................................................................................... 196 30.1.2.2 EAP-TLS Mode ..................................................................................... 197 30.1.3 420HD IP Phone ....................................................................................................198 30.1.3.1 EAP-MD5 Mode .................................................................................... 198 30.1.3.2 EAP-TLS Mode ..................................................................................... 199 30.2 Configuring 802.1x using Web and Configuration File ......................................... 200 30.2.1 EAP MD5 Mode .....................................................................................................200 30.2.2 EAP TLS Mode ......................................................................................................201 31 HTTPS Management........................................................................................ 203 31.1 Web Interface ....................................................................................................... 203 31.2 Secure Provisioning ............................................................................................. 203 Maintenance ...........................................................................................................205 Administrator's Manual
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32 Changing User Login Credentials.................................................................. 207 33 Restarting Phones........................................................................................... 209 33.1 Restarting Phone using Phone User Interface ..................................................... 209 33.1.1 310HD IP Phone ....................................................................................................209 33.1.2 320HD IP Phone ....................................................................................................210 33.1.3 420HD IP Phone ....................................................................................................210 33.2 Restarting Phone using Web ............................................................................... 211 34 Restoring Phone Defaults .............................................................................. 213 34.1 Restoring Factory Defaults using Phone Interface............................................... 213 34.1.1 310HD IP Phone ....................................................................................................213 34.1.2 320HD IP Phone ....................................................................................................214 34.1.3 420HD IP Phone ....................................................................................................214 34.2 Restoring Factory Defaults using the Web........................................................... 215 Status and Monitoring ...........................................................................................217 35 Viewing Network Status.................................................................................. 219 35.1 LAN Status ........................................................................................................... 219 35.2 Port Status ........................................................................................................... 220 35.3 802.1x Status ....................................................................................................... 221 36 Viewing VoIP Status ........................................................................................ 223 36.1 Phone Status........................................................................................................ 223 36.2 Line Status ........................................................................................................... 224 36.3 Current Call Information ....................................................................................... 225 37 Viewing Call History ........................................................................................ 227 38 Viewing Phone Version Number .................................................................... 229 38.1 Viewing Phone Version Number using the Web .................................................. 229 38.2 Viewing Firmware Version using Phone Interface ............................................... 230 38.2.1 310HD IP Phone ....................................................................................................230 38.2.2 320HD IP Phone ....................................................................................................231 38.2.3 420HD IP Phone ....................................................................................................232 Diagnostics and Troubleshooting ........................................................................233 39 Testing Phone Hardware ................................................................................ 235 39.1 310HD IP Phone BIT............................................................................................ 235 39.1.1 Test Preparation ....................................................................................................235 39.1.2 Keypad Test ...........................................................................................................236 39.1.3 LED Test ................................................................................................................237 39.1.4 Handset Test .........................................................................................................238 39.1.5 Headset Test .........................................................................................................239 39.1.6 Speaker Test .........................................................................................................240 39.1.7 MAC Address and Serial Number Verification Test ..............................................241 39.1.7.1 MAC Address and Serial Number Verification Test .............................. 241 39.2 Testing 320HD IP Phone BIT ............................................................................... 242 39.2.1 Test Preparation ....................................................................................................242 Version 2.0.0
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39.2.2 39.2.3 39.2.4 39.2.5 39.2.6 39.2.7 39.2.8 Keypad and Off-Hook/On-Hook Test.....................................................................243 Green-Color LEDs Test .........................................................................................245 Red-Color LEDs Test.............................................................................................246 Handset Test .........................................................................................................247 Headset Test .........................................................................................................249 Speaker Test .........................................................................................................251 MAC Address and Serial Number Verification Test ..............................................253 39.3 Testing 420HD IP Phone BIT ............................................................................... 253 40 Recovering Firmware ...................................................................................... 255 41 System Logging (Syslog) ............................................................................... 257 42 Phone User Interface Error Messages........................................................... 259 43 Media Debug Information ............................................................................... 261 44 Port Mirroring .................................................................................................. 263 Appendices ............................................................................................................265 A Server-Specific Configuration ........................................................................ 267 B Typical Setup Scenario Example ................................................................... 271 C SIP Support (RFC, Headers) ........................................................................... 273 D IP Phone Specifications.................................................................................. 279 Administrator's Manual
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List of Figures
Figure 2-1: Language Web Field – 310HD.............................................................................................22 Figure 2-2: Language Web Field – 320HD.............................................................................................23 Figure 2-3: Language Web Field – 420HD.............................................................................................24 Figure 3-1: Web Login Window ..............................................................................................................25 Figure 3-2: Web Interface Areas ............................................................................................................26 Figure 5-1: TR-069 Web Fields ..............................................................................................................33 Figure 7-1: Configuration File Web Fields ..............................................................................................39 Figure 7-2: Load New Configuration File Web Field ..............................................................................39 Figure 9-1: Automatic Provisioning Web Fields .....................................................................................43 Figure 9-2: Automatic Provisioning - DHCP Option 160 Web Fields .....................................................47 Figure 9-3: Automatic Provisioning - DHCP Options 66/67 Web Fields ................................................47 Figure 9-4: Static DNS Record Web Fields ............................................................................................48 Figure 10-1: Quick Setup Web Fields ....................................................................................................57 Figure 12-1: Date and Time Web Fields ................................................................................................63 Figure 12-2: NTP and Time Settings Web Fields ...................................................................................67 Figure 12-3: NTP and Time Settings Web Fields ...................................................................................69 Figure 13-1: Network Settings Web Fields .............................................................................................76 Figure 14-1: Port Mode Web Fields .......................................................................................................81 Figure 15-1: VLAN Web Fields ...............................................................................................................83 Figure 16-1: Telnet Web Field ................................................................................................................85 Figure 17-1: SIP General Web Fields ....................................................................................................89 Figure 17-2: SIP Proxy and Registrar Web Fields .................................................................................93 Figure 17-3: Proxy Redundancy .............................................................................................................96 Figure 17-4: SIP Timers .........................................................................................................................99 Figure 17-5: Quality of Service Page ...................................................................................................101 Figure 17-6: Reject Code Web Field ....................................................................................................102 Figure 18-1: General Dialing Web Fields .............................................................................................103 Figure 18-2: Tones Web Fields in Dialing Page ...................................................................................105 Figure 18-3: Tones Web Fields in Services Page ................................................................................105 Figure 18-4: Upload Ringing Tones Web Fields ..................................................................................107 Figure 18-5: Tones Web Fields ............................................................................................................109 Figure 18-6: Default Audio Device Web Field ......................................................................................112 Figure 18-7: DTMF Transport Mode Web Field ...................................................................................113 Figure 18-8: Digit Map and Dial Plan Web Fields ................................................................................114 Figure 19-1: Media Streaming Web Fields ...........................................................................................118 Figure 19-2: QoS Web Fields ...............................................................................................................119 Figure 19-3: Codecs Web Fields ..........................................................................................................120 Figure 20-1: Voice Web Fields .............................................................................................................123 Figure 20-2: Silence Compression Web Fields ....................................................................................124 Figure 21-1: Voice Web Fields .............................................................................................................132 Figure 22-1: Call Waiting Web Fields ...................................................................................................136 Figure 22-2: Call Forwarding Web Fields .............................................................................................137 Figure 22-3: Three-way Conferencing Web Fields ..............................................................................139 Figure 22-4: Automatic Dialing Web Fields ..........................................................................................140 Figure 22-5: DnD Web Fields ...............................................................................................................142 Figure 22-6: Call Pick Up Web Fields ..................................................................................................143 Figure 22-7: MWI Web Fields ...............................................................................................................144 Figure 22-8: BLF Support Web Fields ..................................................................................................146 Figure 22-9: AOC Support Web Fields.................................................................................................148 Figure 22-10: EHS Web Fields .............................................................................................................149 Figure 23-1: Auto Gain Control Web Fields .........................................................................................151 Figure 23-2: Tone Volume Web Field ..................................................................................................152 Figure 23-3: Ringer Volume Web Field ................................................................................................153 Figure 23-4: Hands Free Web Fields ...................................................................................................154 Figure 23-5: Additional Speaker Volume Web Field ............................................................................154 Figure 23-6: Handset Volume Settings Page .......................................................................................156 Version 2.0.0
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Figure 23-7: Headset Web Fields .........................................................................................................158 Figure 25-1: LDAP Web Fields .............................................................................................................169 Figure 25-2: Corporate Directory Web Field ........................................................................................172 Figure 25-3: Add Contact Web Fields ..................................................................................................174 Figure 26-1: Speed Dial Web Fields ....................................................................................................175 Figure 27-1: Certificate Web Fields ......................................................................................................181 Figure 27-2: Combined Private and Public key Example .....................................................................182 Figure 27-3: TLS Port and CA Certification Web Fields .......................................................................187 Figure 27-4: Private and Public Key Web Fields ..................................................................................188 Figure 29-1: Enabling SRTP Web Fields .............................................................................................191 Figure 30-1: EAP MD5 Web Fields ......................................................................................................200 Figure 30-2: EAP TLS Web Fields .......................................................................................................201 Figure 32-1: System Authorization Web Fields ....................................................................................207 Figure 33-1: Restart System Web Fields .............................................................................................211 Figure 33-2: Confirmation Box .............................................................................................................211 Figure 34-1: Restore Defaults Web Fields ...........................................................................................215 Figure 34-2: Submit Confirmation Box .................................................................................................215 Figure 35-1: LAN Information Web Fields ............................................................................................219 Figure 35-2: Port Mode Status Web Fields ..........................................................................................220 Figure 35-3: 802.1X Status Web Fields ...............................................................................................221 Figure 36-1: Phone Status Web Fields ................................................................................................223 Figure 36-2: Line Status Web Fields ....................................................................................................224 Figure 36-3: Current Call Information Web Fields ................................................................................225 Figure 37-1: Call History Web Fields ....................................................................................................227 Figure 38-1: System Information Web Fields (for example 310HD Model) .........................................229 Figure 39-1: Testing Keypad ................................................................................................................236 Figure 39-2: Testing Keypad – Characters Disappear .........................................................................236 Figure 39-3: Successful Completion of Keypad Test ...........................................................................237 Figure 39-4: Off hook Handset Test .....................................................................................................238 Figure 39-5: Recording Test .................................................................................................................238 Figure 39-6: Playing Test .....................................................................................................................238 Figure 39-7: Headset Test ....................................................................................................................239 Figure 39-8: Headset Recording Test ..................................................................................................239 Figure 39-9: Headset Playing Test .......................................................................................................239 Figure 39-10: Speaker Test ..................................................................................................................240 Figure 39-11: Speaker Recording Test ................................................................................................240 Figure 39-12: Speaker Playing Test .....................................................................................................240 Figure 39-13: MAC Address and Serial Number ..................................................................................241 Figure 39-14: Keypad Test ...................................................................................................................243 Figure 39-15: Keypad Test – Off-Hook ................................................................................................243 Figure 39-16: Keypad Test - Keys ........................................................................................................244 Figure 39-17: Green LEDs On .............................................................................................................245 Figure 39-18: Red LEDs Test ...............................................................................................................246 Figure 39-19: Handset Test ..................................................................................................................247 Figure 39-20: Handset Test - Recording ..............................................................................................247 Figure 39-21: Handset Test- Playing ....................................................................................................248 Figure 39-22: Headset Test ..................................................................................................................249 Figure 39-23: Headset Test - Recording ..............................................................................................249 Figure 39-24: Headset Test - Playing ...................................................................................................250 Figure 39-25: Speaker/Mic Test ...........................................................................................................251 Figure 39-26: Speaker/Mic Test– Recording........................................................................................251 Figure 39-27: Speaker/Mic Test – Playing ...........................................................................................252 Figure 39-28: MAC Address and Serial Number Verification Test .......................................................253 Figure 41-1: System Logging Web Fields ............................................................................................257 Figure 43-1: Recording Web Fields ......................................................................................................261 Figure 44-1: Port Mirroring ...................................................................................................................263 Figure A-1: BLF Configuration for BroadWorks Web Fields ................................................................268 Figure A-2: BLF Configuration for Application Server Type - Asterisk .................................................270 Administrator's Manual
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List of Tables
Table 2-1: LCD Display Parameters .......................................................................................................22 Table 2-2: Language Display Parameters ..............................................................................................23 Table 2-3: Language Display Parameters ..............................................................................................24 Table 4-1: Example of CSV File .............................................................................................................30 Table 5-1: TR-069 Management Parameters.........................................................................................34 Table 9-1: Automatic Update Methods Parameters ...............................................................................44 Table 9-2: Static DNS Record Parameters ............................................................................................49 Table 12-1: Daylight Saving Time Parameters.......................................................................................64 Table 12-2: NTP Server Parameters ......................................................................................................67 Table 12-3: NTP Server and GMT Parameters ......................................................................................69 Table 13-1: Network Settings Parameters .............................................................................................77 Table 13-2: Partial DHCP Parameters ...................................................................................................78 Table 14-1: Port Settings Parameters ....................................................................................................81 Table 15-1: VLAN Parameters Description ............................................................................................83 Table 16-1: Telnet Parameters ...............................................................................................................85 Table 17-1: SIP General Parameters .....................................................................................................90 Table 17-2: Proxy and Registrar Parameters .........................................................................................94 Table 17-3: SIP Proxy Server Redundancy Parameters ........................................................................97 Table 17-4: SIP Timers Parameters .....................................................................................................100 Table 17-5: SIP QoS Parameters .........................................................................................................101 Table 17-6: Reject Code Parameter .....................................................................................................102 Table 18-1: Dialing Parameters ............................................................................................................104 Table 18-2: Dial Tones Parameters .....................................................................................................105 Table 18-3: Ring Tone Parameters ......................................................................................................108 Table 18-4: Regional Parameters ........................................................................................................110 Table 18-5: Audio Device Parameter ...................................................................................................112 Table 18-6: DTMF Description .............................................................................................................113 Table 18-7: Digit Map and Dial Plan Parameters .................................................................................114 Table 19-1: Media Streaming Parameters ...........................................................................................117 Table 19-2: RTP Port Range and Payload Type Parameters ..............................................................118 Table 19-3: RTP QoS Parameters .......................................................................................................119 Table 19-4: Codec Parameters ............................................................................................................121 Table 20-1: Jitter Buffer Parameters ....................................................................................................123 Table 20-2: Silence Compression Parameters ....................................................................................124 Table 20-3: Echo Cancellation Parameter ...........................................................................................125 Table 22-1: Call Waiting Parameters ...................................................................................................136 Table 22-2: Call Forward Parameters ..................................................................................................137 Table 22-3: Three-way Conferencing Parameters ...............................................................................139 Table 22-4: Automatic Dialing Parameters...........................................................................................140 Table 22-5: Automatic Answer Parameters..........................................................................................141 Table 22-6: Do Not Disturb (Services) Parameters ..............................................................................142 Table 22-7: Call Pickup Parameters.....................................................................................................143 Table 22-8: Message Waiting Indication Parameters ..........................................................................144 Table 22-9: BLF Parameters ................................................................................................................146 Table 22-10: AOC Parameters .............................................................................................................148 Table 22-11: EHS Parameters .............................................................................................................149 Table 23-1: Automatic Gain Control Parameters .................................................................................151 Table 23-2: Tone Volume Parameter ...................................................................................................152 Table 23-3: Ringer Volume Parameters ...............................................................................................153 Table 23-4: Headset Gain Parameters .................................................................................................158 Table 24-1: Phone Lock/Unlock Parameters........................................................................................168 Table 25-1: LDAP Parameters .............................................................................................................169 Table 25-2: Provisioning Parameters ...................................................................................................173 Table 26-1: Speed Dial Parameters .....................................................................................................176 Table 27-1: SIP-over-TLS Parameters .................................................................................................183 Table 27-2: SIP over TLS File- Encryption File Content Parameters...................................................186 Version 2.0.0
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Table 29-1: SRTP Parameters .............................................................................................................191 Table 30-1: EAP MD5 Parameters .......................................................................................................200 Table 30-2: EAP TLS Parameters ........................................................................................................201 Table 32-1: Username and Password Parameters ..............................................................................207 Table 41-1: Syslog Parameters ............................................................................................................257 Table 42-1: Phone User Interface Error Messages ..............................................................................259 Table 43-1: Packet Recording Parameters ..........................................................................................261 Table 44-1: Port Mirroring Parameters .................................................................................................263 Table A-1: Busy Lamp Field Parameter ...............................................................................................267 Table C-1: Supported IETF RFC’s .......................................................................................................273 Table C-2: Supported SIP Methods .....................................................................................................275 Table C-3: Supported SIP Headers......................................................................................................276 Table D-1: 310HD, 320HD and 420HD IP Phone Specifications .........................................................279 Administrator's Manual
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Notices
Notice
This manual shows administrators how to set up and configure AudioCodes' 310HD, 320HD
and 420HD IP Phones.
Information contained in this document is believed to be accurate and reliable at the time of
printing. However, due to ongoing product improvements and revisions, AudioCodes cannot
guarantee accuracy of printed material after the Date Published nor can it accept
responsibility for errors or omissions. Updates to this document and other documents can be
viewed by registered customers at http://www.audiocodes.com/downloads.
© Copyright 2012 AudioCodes Ltd. All rights reserved.
This document is subject to change without notice.
Date Published: Oct-16-2012
Trademarks
AudioCodes, AC, AudioCoded, Ardito, CTI2, CTI², CTI Squared, HD VoIP, HD VoIP
Sounds Better, InTouch, IPmedia, Mediant, MediaPack, NetCoder, Netrake, Nuera, Open
Solutions Network, OSN, Stretto, TrunkPack, VMAS, VoicePacketizer, VoIPerfect,
VoIPerfectHD, What’s Inside Matters, Your Gateway To VoIP and 3GX are trademarks or
registered trademarks of AudioCodes Limited. All other products or trademarks are
property of their respective owners.
WEEE EU Directive
Pursuant to the WEEE EU Directive, electronic and electrical waste must not be disposed
of with unsorted waste. Please contact your local recycling authority for disposal of this
product.
Customer Support
Customer technical support and service are provided by AudioCodes’ Distributors,
Partners, and Resellers from whom the product was purchased. For Customer Support for
products purchased directly from AudioCodes, contact [email protected]
Abbreviations and Terminology
Each abbreviation, unless widely used, is spelled out in full when first used.
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Related Documentation
Document Name
300HD IP Phone Series Release Notes
310HD IP Phone Quick Guide
310HD IP Phone User’s Manual
320HD IP Phone Quick Guide
320HD IP Phone User’s Manual
420HD IP Phone User’s Manual
420HD IP Phone Quick Guide
Regulatory Information
Compliancy and Regulatory Information can be viewed at
http://www.audiocodes.com/library.
:
Note: Throughout this document, where features can be configured using the
Web interface or Configuration File, the Web parameter is displayed in
the regular font above its corresponding Configuration File parameter,
which is enclosed in square brackets in bold font type.
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1. Introduction
Introduction
This manual is intended for the system administrator responsible for setting up and
configuring the 310HD, 320HD and 420HD IP Phones.
AudioCodes' IP phones are based on AudioCodes' proprietary High Definition (HD) voice
technology, providing clarity and a rich audio experience in Voice-over-IP (VoIP) calls. The
phones are fully-featured telephones that provide voice communication over an IP network,
allowing you to place and receive phone calls, put calls on hold, transfer calls, make
conference calls, and so on.
The phone offers a wide variety of management and configuration tools:
Phone’s LCD display user interface - easy-to-use, menu-driven display screen,
providing basic phone configuration and status capabilities
Web interface - provides a user-friendly Web interface that runs on a Web browser
(Microsoft® Internet Explorer is the recommended browser).
Configuration file - text-based file (created using any plain text editor such as
Microsoft’s Notepad) containing configuration parameters and which is loaded to the
phone using the Web interface or a TFTP, FTP, HTTP or HTTPS server.
TR-069 for remote configuration and management
CLI over Telnet
For a detailed description on hardware installation and for operating the phone’s call
features, refer to the User’s Manual.
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Reader’s Notes
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Configuration Tools
310HD, 320HD and 420HD IP Phones
Reader’s Notes
Administrator's Manual
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2. IP Phone User Interface
IP Phone User Interface
The IP phone provides a Liquid Crystal Display (LCD) based screen, offering an intuitive,
menu-driven interface for configuring the phone. The administrative tasks are performed in
the phone’s Administration menu.
2.1
Accessing the Administration Menu
The procedures to access the Administration menu using the 310HD, 320HD and 420HD
IP Phones, are described below.
Notes:
2.1.1
•
The default password is 1234. To change the login password, use
the phone’s Web interface or Configuration file.
•
Once you have entered the password, the access session is applied
to all the submenus.
•
To change the Administration menu’s login password, use the
phone’s Web interface (see Section 32 on page 207), or use the
configuration file.
310HD IP Phone
The submenus of the Administration menu are password protected. When attempting to
access a submenu, the phone prompts you to enter the access password.
To access the Administration menu:
1.
In idle display, press the MENU key, and then select the Administration menu (using
the
and
Navigation keys),
2.
Press the ENTER key; you are prompted for a password:
3.
Enter the password, and then press the ENTER key; the Administration submenus
are displayed.
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2.1.2
320HD IP Phone
The Administration menu is password protected. The default password is "1234".
To access the Administration menu:
1.
In idle display, press the MENU key; the Menu list is displayed:
2.
Using the
and
Navigation keys, choose the Administration submenu, and then
press the Select softkey; you are prompted for a password:
3.
Enter your password, and then choose OK. When entering the password, you can
change between numerals and letters, by pressing the pound (#).
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2.1.3
2. IP Phone User Interface
420HD IP Phone
Accessing the Administration menu of the 420HD IP phone is identical to accessing that
of the 320HD IP phone. The phone is password protected. The default password is "1234".
To access the Administration menu:
1.
In idle display, press the MENU key; the Menu list is displayed. Press the navigation
control's lower rim to navigate to the Administration option.
2.
Press the Select softkey; you are prompted for a password.
3.
Enter your password, and then choose OK. When entering the password, you can
change between numerals and letters by pressing pound (#).
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2.2
Changing Display Language
This section describes how to change the language of the LCD display of the IP Phones.
2.2.1
310HD IP Phone
The Changing Display Language feature can be configured using the Web or Configuration
File, as described below.
To choose a language:
Using the Web:
1.
Access the ‘Language’ page (Configuration tab > Personal Settings menu >
Language).
Figure 2-1: Language Web Field – 310HD
2.
Select the language according to the parameter in the table below, and then click
Submit; the phone reboots and changes the LCD display language accordingly.
Using the Configuration File:
Table 2-1: LCD Display Parameters
Parameter
Description
Phone Display Language
[personal_settings/language]
Determines the LCD display user interface
language.
[English] English (default)
[Spanish] Spanish
[Italian] Italian
[Portuguese] Portuguese
[German] German
[Russian] Russian
[Ukraine] Ukraine
[Polish] Polish
[French] French
[/system/activation_keys/portuguese_language]
Activation key (string) required to unlock
Portuguese language support.
[personal_settings/max_directory_size]
Determines the maximum directory size.
The valid range is 0 to 700. The default
value is 700.
Note:
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To reduce the memory consumption
of the phone and to increase
performance, this value should be
optimized and set to the lowest
value according to the actual
corporate directory size.
Document #: LTRT-09910
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2.2.2
2. IP Phone User Interface
320HD IP Phone
The Changing Display Language feature can be configured using the Web or Configuration
File, as described below.
To choose a language:
Using the Web:
1.
Access the ‘Language’ page (Configuration tab > Personal Settings menu >
Language).
Figure 2-2: Language Web Field – 320HD
2.
Select the language according to the parameter in the table below, and then click
Submit; the phone reboots and changes the LCD display language accordingly.
Table 2-2: Language Display Parameters
Parameter
Description
Phone Display Language
[personal_settings/language]
Determines the LCD display user interface
language.
[English] English (default)
[Spanish] Spanish
[Russian] Russian
[German] German
[Ukraine] Ukraine
[French] French
[Italian] Italian
[Hebrew] Hebrew
[Polish] Polish
[personal_settings/lcd_contrast]
Determines the LCD contrast.
The valid range is 0 to 14. The default value
depends on the actual hardware revision.
Note: Portuguese is supported even though it may not be listed in the LCD. To
activate Portuguese, contact your AudioCodes representative.
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2.2.3
420HD IP Phone
The Changing Display Language feature can be configured using the Web or Configuration
File, as described below.
To choose a language:
Using the Web:
1.
Access the ‘Language’ page (Configuration tab > Personal Settings menu >
Language).
Figure 2-3: Language Web Field – 420HD
2.
Select the language according to the parameter in the table below, and then click
Submit; the phone reboots and changes the LCD display language accordingly.
Table 2-3: Language Display Parameters
Parameter
Description
Phone Display Language
[personal_settings/language]
Determines the LCD display user interface
language.
[English] English (default)
[Spanish] Spanish
[Russian] Russian
[German] German
[Ukraine] Ukraine
[French] French
[Italian] Italian
[Hebrew] Hebrew
[Polish] Polish
[personal_settings/lcd_contrast]
Determines the LCD contrast.
The valid range is 0 to 63. The default value
depends on the actual hardware revision.
Note: Portuguese is supported even though it may not be listed in the LCD. To
activate Portuguese, contact your AudioCodes representative.
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3. Web Interface
Web Interface
This chapter describes the phone’s embedded Web server interface that is used for
configuring the phone.
3.1
Accessing Web Interface
You can use any standard Web browser (such as Microsoft Internet Explorer) to access the
phone’s Web interface. The IP address used for accessing the Web interface is the
phone’s IP address, received from a DHCP server or manually configured (static IP
address).
To access the phone’s Web interface:
1.
Connect the LAN port of your phone to the IP network (using the Cable or ADSL
modem from your Internet Service Provider).
2.
Determine the phone’s IP address obtained from the DHCP server, using the phone’s
LCD screen as described in Section 35 on page 219 (in the ‘IP Address’ field).
3.
Open a Web browser, and then in the URL address field, enter the phone’s IP address
(for example, http://192.168.1.2 or https://192.168.1.2), as displayed below:
The Web login window appears:
Figure 3-1: Web Login Window
4.
Alternatively, if your DHCP and DNS servers are synchronized you can access the
phone Web browser by using the following method:
http://310hd-<MAC Address>.<Domain Name>
(e.g. http://310hd-001122334455.corp.YourCompany.com)
5.
Enter the User name and Password, and then click OK.
Note: The administrator’s default login user name and password are “admin”
and “1234” respectively. To change the login credentials, see Section 32
on page 207.
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3.2
Getting Started with the Web
The areas of the Web interface are shown below:
Figure 3-2: Web Interface Areas
The Web interface is composed of the following main areas:
Toolbar: displays AudioCodes logo and provides the following buttons:
•
Home: opens the Home page
•
Log off: closes the Web interface
Navigation bar: provides tabs for accessing the configuration menus:
•
Configuration: provides menus for configuring the phone.
•
Management: provides menus for various management tasks such as firmware
upgrade and changing the login username and password.
•
Status & Diagnostics: provides menus for displaying information on the status
of the phone, such as call history.
Navigation tree: tree-like, hierarchical structure of menus pertaining to the selected
tab on the Navigation bar.
Configuration pane: displays the configuration parameters pertaining to a selected
menu in the Navigation tree.
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4. Configuration File
Configuration File
This section describes the Configuration file and the parameters that you can set in the
Configuration file.
4.1
Introduction
The Configuration file can be loaded to the phone using the automatic provisioning
mechanism, or manually from your local computer using the Web interface (see Section 3
on page 25). The subsections below describe the Configuration file syntax and linking
additional Configuration files to a Configuration file.
4.2
File Syntax
The Configuration file can be created using a standard ASCII, text-based program such as
Notepad. The Configuration file is a *.cfg file with the file name being the phone’s MAC
address: <phone’s MAC address>.cfg.
The syntax of the Configuration file is as follows:
<parameter name>=<value>
Ensure that the Configuration file adheres to the following guidelines:
No spaces on either side of the equals (=) sign.
Each parameter must be on a new line.
Below is an example of part of a Configuration file:
system/type=310HD
voip/line/0/enabled=1
voip/line/0/id=1234
voip/line/0/description=310HD
voip/line/0/auth_name=1234
voip/line/0/auth_password=4321
4.3
Linking Multiple Files
The Configuration file allows you to include links (URL and/or file name) to other
Configuration files that provide additional parameter settings. This is especially useful in
deployments with multiple phones, where the phones share common configuration but
where each phone has some unique settings. In such a scenario, a phone’s Configuration
file can include unique parameter settings as well as links to additional Configuration files
with settings common to all phones.
Linking additional files is achieved by using the include function in the phone’s
Configuration file. For example, the below Configuration file provides links to additional
Configuration files (shown in bolded font):
system/type=310HD
include 310HD_<MAC>_voip.cfg
include vlan_conf.cfg
include network_conf.cfg
include provisioning_conf.cfg
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In addition, the Configuration file can provide URL paths (FTP, TFTP, HTTP, or HTTPS) to
where the additional files are located, as shown in the example below (shown in bolded
font):
system/type=310HD
include http://10.10.10.10/310HD_<MAC>_voip.cfg
include https://remote-pc/vlan_conf.cfg
include tftp://10.10.10.10/310HD_<MAC>_network.cfg
include ftp://remote-pc/provisining_conf.cfg
Note: If no URL is provided in the Configuration file, the files are retrieved
according to the provisioning information (e.g. DHCP Option 160 or
66/67).
4.4
Downloading Configuration File from IP Phone
For more information, see ‘Part VIII, Maintenance Section’.
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4. Configuration File
4.5
Creating Configuration Files using VoIProvision
Utility
4.5.1
Introduction
When installing AudioCodes’ 300HD IP Phones, the integrator or IT manager typically
wishes to configure each installed IP phone automatically. Using DHCP options or other
methods, the IP phone can be instructed to download a configuration file. This file is
typically unique to each IP phone, based on the MAC address. This MAC-specific
configuration file is generated with IP phone specific configuration parameters; such as, the
extension ID, name and authentication password.
Not all of the iPBX and SoftSwitch vendors (and especially the full solution vendors)
include provisioning in their interoperability programs. Therefore, AudioCodes, as an IP
Phone vendor, is required to provide a standalone provisioning tool that will enable the
provisioning of our phones in such environments.
AudioCodes now provides a tool that assists in the automatic generation of such
configuration files. These files can be generated for the initial configuration of the IP
phones and then later regenerated for subsequent configuration updates as desired.
4.5.2
Configuration File Format
The detailed format of the 300HD configuration files are described in section ‘Configuration
File Parameters’ in the Administrator’s Manual (LTRT-08107 320HD IP Phone
Administrator's Manual v1.4.0.pdf and LTRT-09905 310HD IP Phone Administrator's
Manual v1.4.pdf).
The following is an output example of an automatically generated MAC-specific file:
system/type=310HD
voip/line/0/enabled=1
voip/line/0/id=56832432
voip/line/0/auth_name=3423fdwer2tre
voip/line/0/auth_password=123456
include global.cfg
4.5.3
Global Configuration File
In addition to the MAC-specific files, it is recommended to maintain a single global
configuration file, which contains parameters that are common to all IP phones in the
specific site. The MAC-specific files can call the global file (using the ‘include’ method) as
illustrated in the above example. For more information, see ‘Linking Additional Files using
“Include” ’ in the Administrator’s Manual.
4.5.4
VoIProvision Utility Overview
The VoIProvision utility is a generic tool that automatically generates multiple MAC-specific
configuration files (.cfg). The utility generates a separate .cfg file for each IP phone.
In order to execute the utility, the user needs to prepare a csv file and a template file. The
csv file contains the tagged records for each IP phone and the template file maps these
tagged records to a configuration file format, which can be read by the 300HD IP Phone.
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4.5.5
CSV File
The csv file contains a list of tags and a list of the tag’s values. The first line in the file
contains the list of tags (comma-separated) and each of the other lines contains a list of
values, where each line record represents an individual IP phone.
The csv file is usually exported from the customer’s IP-PBX or some other database and
typically contains the list of IP phones (e.g. MAC, extension ID, user name and password
of each IP phone).
Table 4-1: Example of CSV File
[mac]
[name]
[id]
[password]
00908F123456
Jonathan
4071
12345
00908F123457
David
4418
12345
When opened as a text file, the csv file appears similar to the example below:
[mac],[name],[id],[password]
00908F123456,Jonathan,4071,12345
00908F123457,David,4418,12345
4.5.6
Template File
The template file defines the format of the generated configuration files, but contains tags
instead of actual values. The VoIProvision utility reads the template file and replaces each
tag with actual values from the csv file.
Example of a template file:
system/type=310HD
voip/line/0/enabled=1
voip/line/0/id=[id]
voip/line/0/auth_name=[name]
voip/line/0/auth_password=[password]
include global.cfg
4.5.7
Generated Configuration Files The generated configuration (.cfg) files use a similar format to the template file; however
the tags are replaced with the actual values that are read by the VoIProvision utility from
the csv file. One of the tags defined in the csv file, should be used as the .cfg file name (in
order for the VoIProvision utility to generate a separate .cfg file for each line record in the
csv file). Typically the tag which defines the MAC address is used as the .cfg file name.
4.5.8
Starting the VoIProvision Utility
The VoIProvision utility can run on both the Linux and Windows platforms. The
VoIProvision utility initially parses the csv file to generate the list of tags. The VoIProvision
then reads each line record of values in the csv file and for each line record, does the
following:
Parses the line record to create a list of values
Opens the template file
Generates the .cfg file name and create a new .cfg file
Reads the template file, associates the mapped tags with actual values from the csv
file and writes the result to the .cfg file
Closes the .cfg file and template file
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4.5.9
4. Configuration File
Usage
USAGE: VoIProvision <csv file> <template file> <.cfg file>
Note the following:
The first line of the csv file contains the list of tags (e.g.,
[mac],[name],[id]).
The remainder of the csv file contains a line record per .cfg file (e.g.
00908f112233,4071,Eitan).
There is no restriction on the format of the tags (e.g., [tag] or @[email protected]).
The template file defines the .cfg file format. During VoIProvision
run-time, the mapped tags in the template file are associated to actual
values that are read from the csv file.
Currently only a single tag can be defined per line record in the
template file.
The .cfg file name should represent the string of one of the predefined
tags in order to generate a separate .cfg file per csv line record (e.g.
"[mac].cfg").
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5. TR-069 Management
TR-069 Management
To connect to the remote management server, the phone’s embedded TR-069 client must
be configured. The TR-069 Management feature can be configured using the Web or
Configuration File, as described below
To configure TR-069:
Using the Web:
1.
Access the ‘TR-069’ page (Management tab > Remote Management menu >
TR-069).
Figure 5-1: TR-069 Web Fields
2.
Version 2.0.0
Configure TR-069 according to the parameters in the table below, and then click
Submit.
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Via Configuration File:
Table 5-1: TR-069 Management Parameters
Parameter
Description
Note: To add a value to these parameters, enter management/ followed by the parameter name,
equal sign and then the value (e.g. management/tr069/enabled=0).
Activate
[management/tr069/enabled]
Enables TR-069.
[0] Disable (default)
[1] Enable
Note: This feature can be enabled only if the valid
key is provided in the feature_key field.
Activation Key
[management/tr069/activation_key]
Activation key to enable TR-069 management.
ACS URL
[management/tr069/acs_url]
URL for connecting the device to the ACS.
Note: This parameter must be in the form of a valid
HTTP or HTTPS URL.
User
[management/tr069/user_name]
Username used to authenticate the device when
making a connection to the ACS.
Password
[management/tr069/password]
Password used to authenticate the device when
making a connection to the ACS.
Request Connection User
[management/tr069/connection_request/
user_name]
Username to authenticate an ACS making a
Connection Request to the device.
Request Connection Password
[management/tr069/connection_request/
password]
Password to authenticate an ACS making a
Connection Request to the device.
Periodic Inform
[management/tr069/inform/enabled]
Determines whether or not the device must
periodically send device information to the ACS
using the Inform method call.
[0] Disable
[1] Enable (default)
Periodic Inform Interval
[management/tr069/inform/interval]
The duration (in seconds) of the interval for which
the device must attempt to connect with the ACS
and call the Inform method if TR-069 is enabled
(management/tr069/inform/enabled).
The valid range is 0 to 86400. The default is 3600.
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Part II
Automatic
Provisioning
Reader’s Notes
Administrator's Manual
6
6. Introduction
Introduction
By default, the IP phone is ready for out-of-the-box deployment using its automatic
provisioning capabilities.
The IP phone offers a built-in mechanism for automatically upgrading its software image
and updating its configuration. This method is used to upgrade the phone firmware and
update its configuration, by remotely downloading an updated software image and
configuration file.
The automatic update mechanism helps you keep your software image and configuration
up-to-date, by performing routine checks for newer software versions and configuration
files, as well as allowing you to perform manual checks.
The automatic update mechanism is as follows:
Before connecting the phone, verify that the provisioning server is running and that the
firmware and configuration files are located in the correct location.
Connect your phone to the IP network, and then connect the phone to the power
outlet.
During DHCP negotiation, the phone requests for DHCP options 66/67/160 to receive
provisioning information. The DHCP server should respond with Option 160 providing
the provisioning URL or Options 66 and 67 providing the TFTP IP address and
firmware file name respectively.
The phone then checks whether new firmware is available by checking the firmware
file header. If the version is different from the one currently running on the phone, the
phone downloads the complete image and burns it to its flash memory.
If a new firmware is unavailable, the phone then checks whether a new configuration
is available. If a configuration file is available on the server, the phone downloads it
and updates the phone’s configuration after verifying that the configuration file is
related to the phone model. When a configuration update is needed, the phone might
reboot.
Notes:
Version 2.0.0
•
In the DHCP Discover message, the phone publishes its model
name in Option fields 60 and 77 (e.g. 320HD). If the administrator
wants to provide different provisioning information to different phone
models, the administrator can set up a policy in the DHCP server
according to the phone model name.
•
If the phone for some reason is powered off during the firmware
upgrade process, the phone is unusable and the recovery process
must be performed (see Section 40 on page 255).
•
You can only use firmware files with an .img extension and
configuration files with a .cfg extension.
•
To "force" the firmware or configuration file to be retrieved
immediately regardless of the "Check Period" value, click the Check
Now button on the relevant page on the Web interface.
•
An additional auto-provisioning mechanism is supported if the
provisioning environment does not provide all the required
information (e.g. DHCP options).
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7. Updating (Manually) Configuration File
Updating (Manually) Configuration File
The phone enables you to view, save, and load its configuration file to backup and restore
the current configuration.
To manually update the configuration file:
1.
Access the ‘Configuration File’ page (Management tab > Manual Update menu >
Configuration File). The current configuration file settings are displayed in the text
pane.
Figure 7-1: Configuration File Web Fields
2.
Click the Loading new Configuration File button; the following page appears:
Figure 7-2: Load New Configuration File Web Field
3.
Click the Browse button and then select the required configuration file located on your
local PC; the phone verifies that the configuration file is related to the phone model.
The configuration file is then loaded to the phone. Once loaded, the phone reboots
(indicated by a message displayed on the phone’s screen). The phone is now updated
with the new configuration.
Note: The configuration file name must have the extension *.cfg.
To save the configuration file:
In the ‘Configuration File’ page, click the Saving Current Configuration File button,
and then save the current phone configuration file to a folder on your local PC.
Note: When creating a new configuration file, ensure that the system/type
parameter in the configuration file is set to the proper phone model
(e.g.,320HD).
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8. Setting up Network for Auto Provisioning
Setting up Network for Auto Provisioning
The IP Phone supports dynamic VLAN discovery, Dynamic IP addressing (DHCP), and
NTP (as client).
Note: For manual configuration of Network Settings, see Section 13 on page
71.
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9. Obtaining Firmware and Configuration
Obtaining Firmware and Configuration
The Web interface allows you to perform the following:
Automatic update of firmware and configuration files
Manual update of firmware and configuration files
9.1
Methods
There are three different ways of configuring automatic provisioning:
DHCP
Static DNS Record
Using SIP SUBSCRIBE and NOTIFY Messages
9.1.1
DHCP
Dynamic Host Configuration Protocol (DHCP) can be used to configure automatic
provisioning. The DHCP feature can be configured using the Web or Configuration File, as
described below
To configure with DHCP:
Using the Web:
1.
Access the ‘Automatic Update page (Management tab > Automatic Update
menu > Automatic Provisioning).
Figure 9-1: Automatic Provisioning Web Fields
2.
Version 2.0.0
Configure automatic update of firmware and configuration files according to the
parameters in the table below, and then click Submit.
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Using the Configuration File:
Table 9-1: Automatic Update Methods Parameters
Parameter
Description
Note: To add a value to these parameters, enter provisioning/ followed by the parameter name
equals the value (e.g. provisioning/method=dynamic).
Provisioning Method
[provisioning/method]
Defines the provisioning method:
[Disable] Disable - Automatic update is disabled.
The phone attempts to upgrade its firmware and
configuration
[Dynamic] DHCP Options (Dynamic URL)
(default) - Using DHCP options 160 or 66/67 for
provisioning
[Static] Static URL - Using Static URL for
provisioning
DHCP Option Value
[provisioning/url_option_value]
Determines the DHCP option number to be used for
receiving the URL for provisioning.
The default value is 160.
The phone supports DHCP Option 160 for complete
URL and Options 66/67 for TFTP usage. Option 160
has the highest priority and if absent, Options 66/67
are used.
The following syntax is available for DHCP option
160:
<protocol>://<server IP address or host name>
<protocol>://<server IP address or host
name>/<firmware file name>
<protocol>://<server IP address or host
name>/<firmware file name>;<configuration file
name>
<protocol>://<server IP address or host
name>/;<configuration file name>
Where <protocol> can be one of the following: "ftp",
"tftp", "http" or "https".
For example:
ftp://192.168.2.1 – retrieved firmware file is
310HD.img and the configuration file name is
<MAC address>.cfg. For example,
001122334455.cfg
tftp://192.168.2.1/different_firmware_name.img
- retrieved firmware file is
Different_Firmware_Name.img and the
configuration file name is <MAC address>.cfg. For
example, 001122334455.cfg
http://192.168.2.1/different_firmware_name.img
; 310HD_<MAC>_conf.cfg - retrieved firmware
file is different_firmware_name.img and the
configuration file name is 310HD_<MAC
address>_conf.cfg. For example,
310HD_001122334455_conf.cfg
• https://192.168.2.1/310HD_<MAC>_conf.cfg - In
this case the retrieved firmware file is 310HD.img
and the configuration file name is 310HD_<MAC
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9. Obtaining Firmware and Configuration
Parameter
Description
Address>_conf.cfg. For example,
310HD_001122334455_conf.cfg
The following syntax is available for DHCP Options
66/67:
Option 66 must be a valid IP address or host
name of a TFTP server only.
Option 67 must be the firmware name.
If Option 67 is absent, the phone requests for the
310HD.img image file. For example:
Option 66: 192.168.2.1 or myTFTPServer
Option 67: 310HD_1.2.2_build_5.img
Notes:
This parameter is applicable only when method is
configured to "Dynamic".
It is recommended to leave the parameter at its
default value to avoid conflict with other DHCP
options settings.
Random Provisioning Time
[provisioning/random_provisioning_time]
Defines the maximum random number to start the
provisioning process.
This is used for periodic checking of firmware and
configuration files to avoid multiple devices from
starting the upgrade process at the same time. When
the device is meant to start the upgrade, the device
randomly selects a number between 1 and the value
set for random_provisioning_time and performs the
check only after the random time.
The valid range is 0-65535. The default value is 120.
Check Period
[provisioning/period/type]
Defines the period type for automatic provisioning:
[hourly] Hourly - Sets an interval in hours.
[daily] Daily (default) - Sets an hour in the day.
[weekly] Weekly - Sets a day in the week and an
hour in the day.
[powerup] On Power-up Only - The phone tries to
upgrade only after power-up.
Every (Check Period = Hourly)
The interval in hours for automatically checking for
new firmware and configuration files.
The valid range is 1 to 168. The default is 24.
Note: This parameter is applicable only when type
is configured to "hourly".
[provisioning/period/hourly/hours_interval]
Every day at
[provisioning/period/daily/time]
The hour in the day for automatically checking for
new firmware and configuration files.
The format of this value is hh:mm, where hh is hour
and mm is minutes. For example, 00:30.
The default time is 00:00.
Note: This parameter is applicable only when type is
configured to "daily".
Every (Check Period = Day)
[provisioning/period/weekly/day]
Version 2.0.0
The day in the week for automatically checking for
new firmware and configuration files.
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Parameter
Description
[Sunday] Sunday (default)
[Monday] Monday
[Tuesday] Tuesday
[Wednesday] Wednesday
[Thursday] Thursday
[Friday] Friday
[Saturday] Saturday
Note: This parameter is applicable only when type is
configured to "weekly".
Every (Check Period = Weekly)
[provisioning/period/weekly/time]
The hour in the day for automatically checking for
new firmware and configuration files.
The format of this value is: hh:mm, where hh is hour
and mm is minutes. For example: 00:30
The default time is 00:00.
Note: This parameter is applicable only when type is
configured to "weekly".
provisioning/XXX_uri
Administrator's Manual
The URI that allows the administrator to retrieve
relevant information from separate files. The URI’s
can include specific protocol and path of the
provisioning server or alternatively, the files can be
retrieved from the server whose information was
provided during provisioning (e.g. DHCP options
66/67 or 160).
For example, in the below setting, YYY.ext is
retrieved from the server whose information was
provided during provisioning (e.g. DHCP options
66/67 or 160:
provisioning/XXX_uri=YYY.ext
Alternatively, the full URL and protocol can be set as
follows:
provisioning/XXX_uri=<protocol>://<full
path>/YYY.ext
For example:
provisioning/XXX_uri=tftp://10.10.10.10/YYY.ext
provisioning/XXX_uri=ftp:// 10.10.10.10/YYY.ext
provisioning/XXX_uri=http:// 10.10.10.10/YYY.ext
provisioning/XXX_uri=https:// 10.10.10.10/YYY.ext
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9.1.1.1
9. Obtaining Firmware and Configuration
Configuring DHCP Option 160
DHCP Option Value 160 can be configured using the Web as shown below.
To configure DHCP Option 160:
Using the Web:
1.
Access the ‘Automatic Update page (Management tab > Automatic Update
menu > Automatic Provisioning).
Figure 9-2: Automatic Provisioning - DHCP Option 160 Web Fields
2.
3.
4.
5.
9.1.1.2
From the ‘Provisioning Method’ drop-down list, select DHCP Option (Dynamic
URL).
In the ‘DHCP Option Value’ field, enter “160”.
Configure the remaining parameters, and then click Submit.
After reboot, confirm that the firmware and configuration files have been updated.
Configuring DHCP Options 66 and 67
DHCP Options 66 and 67 can be configured using the Web as shown below.
To configure DHCP Options 66 and 67:
Using the Web:
1.
Access the ‘Automatic Update page (Management tab > Automatic Update
menu > Automatic Update).
Figure 9-3: Automatic Provisioning - DHCP Options 66/67 Web Fields
2.
3.
4.
5.
Version 2.0.0
From the ‘Provisioning Method’ drop-down list, select DHCP Option (Dynamic
URL).
In the ‘DHCP Option Value’ field, enter “0”.
Configure the remaining parameters, and then click Submit.
After reboot, confirm that the firmware and configuration files have been updated.
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9.1.2
Static DNS Record (Generic Domain Name)
This method is used for automatic provisioning when you are unable to manage your
DHCP server. If the provisioning server does not support using SIP SUBSCRIBE and
NOTIFY messages mechanism as described above and no response for the SIP
SUBSCRIBE message has been received, the phone tries to retrieve firmware and
configuration files using the following URL: tftp://ProvisioningServer/<Phone Model
Name>/
For example:
The phone tries to obtain the following firmware file:
tftp://ProvisioningServer/310HD/310HD.img
The phone tries to obtain the following configuration file:
tftp://ProvisioningServer/310HD/<MAC address>.cfg
(e.g. tftp://ProvisioningServer/310HD/001122334455.cfg)
It is the Administrator’s responsibility to configure a DNS entry called ProvisioningServer
on the DNS server and set it to the TFTP server IP address.
Note: If Generic Domain Name is used, the automatic provisioning mechanism
periodically tries to retrieve new firmware/configuration from Provisioning
Server domain name.
To configure Static DNS Record:
Using the Web:
1.
Access the ‘Automatic Update page (Management tab > Automatic Update
menu > Automatic Provisioning).
Figure 9-4: Static DNS Record Web Fields
2.
Configure Static DNS Record according to the parameters in the table below, and
then click Submit.
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Using the Configuration file:
Table 9-2: Static DNS Record Parameters
Parameter
Firmware URL
[provisioning/firmware/url]
Description
The static URL for checking the firmware file. The URL must
be entered using one of the following syntax options:
<protocol>://<server IP address or host name>
<protocol>://<server IP address or host name>/<firmware
file name>
Where <protocol> can be one of the following protocols: "ftp",
"tftp", "http" or "https". For example:
tftp://192.168.2.1 – retrieved firmware file is 310HD.img
ftp://192.168.2.1/Different_Firmware_Name.img - retrieved
firmware file is Different_Firmware_Name.img
Note: This parameter is applicable only when method is
configured to "Static".
Configuration URL
[provisioning/configuration/url]
Version 2.0.0
The static URL for checking the configuration file. The URL
must be entered using one of the following syntax options:
<protocol>://<server IP address or host name>
<protocol>://<server IP address or host
name>/<configuration file name>
Where <protocol> can be one of the following protocols: "ftp",
"tftp", "http" or "https". For example:
http://192.168.2.1 - configuration file name is <MAC
Address>.cfg, for example, 001122334455.cfg
https://192.168.2.1/310HD_<MAC>_conf.cfg - retrieved
configuration file name is 310HD_<MAC
Address>_conf.cfg, for example,
310HD_001122334455_conf.cfg
Note: This parameter is applicable only when method is
configured to "Static".
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9.1.3
SIP SUBSCRIBE and NOTIFY Messages
If the provisioning information (e.g. Option fields 66/67/160) is not provided by the DHCP
server, the phone sends a SIP SUBSCRIBE message to the multicast address
224.0.1.75:5060 as shown below.
Note: If the provisioning server supports using SIP SUBSCRIBE and NOTIFY
messages and the device receives the provisioning URL in the NOTIFY
message, the automatic provisioning mechanism then periodically tries to
retrieve a new firmware/configuration according to the information
provided.
SUBSCRIBE sip:224.0.1.75:5060 SIP/2.0
From: <sip:[email protected]:5060>;tag=87a5a8-25020d0a-13c450029-386d4398-66dc40c-386d4398
To: <sip:224.0.1.75:5060>
Call-ID: 8884c8-25020d0a-13c4-50029-386d4398-3e2bcb8e-386d4398
CSeq: 1 SUBSCRIBE
Via: SIP/2.0/UDP 10.13.2.37:5060;rport;branch=z9hG4bK-386d43986ad00ca2-7ca3606e
Expires: 0
Event: ua-profile;profiletype="application";model="310HD";version="1.2.1_build_6"
Max-Forwards: 70
Supported: replaces,100rel
Accept: application/url
Contact: <sip:[email protected]:5060>
User-Agent: AUDC-IPPhone/1.2.1_build_6
Content-Length: 0
The provisioning server or any other entity replies with a 200 OK message to the
SUBSCRIBE message (see below) and sends a NOTIFY SIP message with the
provisioning URL in the message body as shown below. (The provisioning URL can be in
any format as described in the Administrator's Manual).
If no response is received by the provisioning server, the phone resends SUBSCRIBE
messages for five seconds.
With the above method, the phone uses its built-in auto-provisioning mechanism while the
provisioning information is retrieved through the NOTIFY message.
The following code describes SIP 200 OK Response on the SUBSCRIBE Message:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.13.2.37:5060;rport;branch=z9hG4bK-386d43986ad00ca2-7ca3606e
Contact: <sip:10.13.2.37:5060>
To: <sip:224.0.1.75:5060>
From: <sip:[email protected]:5060>;tag=87a5a8-25020d0a-13c450029-386d4398-66dc40c-386d4398
Call-ID: 8884c8-25020d0a-13c4-50029-386d4398-3e2bcb8e-386d4398
CSeq: 1 SUBSCRIBE
Expires: 0
Content-Length: 0
The following code describes SIP NOTIFY Message with Provisioning Information.
NOTIFY sip:10.13.2.37:5060 SIP/2.0
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Via: SIP/2.0/UDP 10.13.2.37:5060;rport;branch=z9hG4bK-386d43986ad00ca2-7ca3606e
Max-Forwards: 20
Contact: <sip:10.13.4.121:5060>
To: <sip:224.0.1.75:5060>
From: <sip:[email protected]:5060>;tag=87a5a8-25020d0a-13c450029-386d4398-66dc40c-386d4398
Call-ID: 8884c8-25020d0a-13c4-50029-386d4398-3e2bcb8e-386d4398
CSeq: 1 NOTIFY
Content-Type: application/url
Subscription-State: terminated;reason=timeout
Event: ua-profile;profiletype="application";model="310HD";version="1.2.0"
Content-Length: 18
tftp://10.13.4.121
The following code describes SIP SUBSCRIBE Message to Obtain Provisioning
Information.
SUBSCRIBE sip:224.0.1.75:5060 SIP/2.0
From: <sip:[email protected]:5060>;tag=87a5a8-25020d0a-13c450029-386d4398-66dc40c-386d4398
To: <sip:224.0.1.75:5060>
Call-ID: 8884c8-25020d0a-13c4-50029-386d4398-3e2bcb8e-386d4398
CSeq: 1 SUBSCRIBE
Via: SIP/2.0/UDP 10.13.2.37:5060;rport;branch=z9hG4bK-386d43986ad00ca2-7ca3606e
Expires: 0
Event: ua-profile;profiletype="application";model="310HD";version="1.2.1_build_6"
Max-Forwards: 70
Supported: replaces,100rel
Accept: application/url
Contact: <sip:[email protected]:5060>
User-Agent: AUDC-IPPhone/1.2.1_build_6
Content-Length: 0
The provisioning server or any other entity replies with a 200 OK message to the
SUBSCRIBE message (see below) and sends a NOTIFY SIP message with the
provisioning URL in the message body as shown below. (The provisioning URL can be in
any format as described in the Administrator's Manual).
If no response is received by the provisioning server, the phone resends SUBSCRIBE
messages for 5 seconds.
With the above method, the phone uses its built-in auto-provisioning mechanism while the
provisioning information is retrieved through the NOTIFY message.
Version 2.0.0
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The following code describes SIP 200 OK Response on the SUBSCRIBE Message.
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.13.2.37:5060;rport;branch=z9hG4bK-386d43986ad00ca2-7ca3606e
Contact: <sip:10.13.2.37:5060>
To: <sip:224.0.1.75:5060>
From: <sip:[email protected]:5060>;tag=87a5a8-25020d0a-13c450029-386d4398-66dc40c-386d4398
Call-ID: 8884c8-25020d0a-13c4-50029-386d4398-3e2bcb8e-386d4398
CSeq: 1 SUBSCRIBE
Expires: 0
Content-Length: 0
The following code describes SIP NOTIFY Message with Provisioning Information.
NOTIFY sip:10.13.2.37:5060 SIP/2.0
Via: SIP/2.0/UDP 10.13.2.37:5060;rport;branch=z9hG4bK-386d43986ad00ca2-7ca3606e
Max-Forwards: 20
Contact: <sip:10.13.4.121:5060>
To: <sip:224.0.1.75:5060>
From: <sip:[email protected]:5060>;tag=87a5a8-25020d0a-13c450029-386d4398-66dc40c-386d4398
Call-ID: 8884c8-25020d0a-13c4-50029-386d4398-3e2bcb8e-386d4398
CSeq: 1 NOTIFY
Content-Type: application/url
Subscription-State: terminated;reason=timeout
Event: ua-profile;profiletype="application";model="310HD";version="1.2.0"
Content-Length: 18
tftp://10.13.4.121
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9.2
9. Obtaining Firmware and Configuration
Protocols for Acquiring Files
You can load various files to the IP phone such as CFG files, Certification Files, Speed Dial
files, and Tones files using the following protocols:
HTTP/S
TFTP
FTP
Version 2.0.0
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Part III
Quick Setup
Reader’s Notes
Administrator's Manual
10
10. Quick Setup
Quick Setup
The Web interface provides you with a Quick Setup page that allows you to configure basic
features to quickly set up your IP Phone to operational level.
Note: For descriptions of these quick setup parameters, see Part IV
(Networking) and Part V (VoIP Features).
To quickly setup your phone:
1.
Access the ‘Quick Setup’ page (Configuration tab > Quick Setup menu > Quick
Setup).
Figure 10-1: Quick Setup Web Fields
2.
Version 2.0.0
For a description of the parameters on this page, refer to the following:
•
Parameters under the LAN Setup group, see Section 13.1.2 on page 78.
•
Parameters under the SIP Proxy and Registrar group see Section 17.2 on page
93.
•
Parameters under the Line Settings group, see Section 21.2 on page 132.
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Part IV
Networking
Reader’s Notes
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11
11. Introduction
Introduction
This part describes manual network settings, if required.
Note: By default the network settings are set for automatic provisioning.
However, if you need to change them, you can do it manually as is
described in this part.
Version 2.0.0
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12
12. Configuring Date and Time
Configuring Date and Time
Generally, a phone retrieves the date and time from a Network Time Protocol (NTP) server
when it connects to the Internet. Alternatively, the date and time can be configured
manually. NTP is a protocol for distributing the Coordinated Universal Time (UTC) by
means of synchronizing the clocks of computer systems over packet-switched, variablelatency data networks.
Note: By default, the network settings are configured for automatic
provisioning. However, if you need to change them, you can do it
manually as is described in this part.
12.1
Configuring Date and Time
The date and time can be configured using the Web or Configuration File, as described
below.
To configure the date and time:
Using the Web:
1.
Access the ‘Date and Time’ page (Configuration tab > Advanced Applications
menu > Date and Time).
Figure 12-1: Date and Time Web Fields
2.
Version 2.0.0
Configure the date and time according to the parameters in the table below, and
then click Submit.
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Using the Configuration File:
Table 12-1: Daylight Saving Time Parameters
Parameter
Description
Active
[system/daylight_saving/activate]
Determines whether the phone automatically
detects the Daylight Saving setting for selected
time zones.
[Disable] Disable (default)
[Enable] Enable
Start Time
[system/daylight_saving/start_date]
This sub-section defines the starting day for the
daylight saving offset.
[month] - defines the specific month in a
year
[day] - defines the specific day in a month
[hour] - defines the specific hour in a day
[minute] - defines the specific minute in an
hour
For example: To configure the phone to start
daylight savings with a specific offset on
February 22nd at 14:30, set the following:
system/daylight_saving/start_date/month=2
system/daylight_saving/start_date/day=22
system/daylight_saving/start_date/hour=14
system/daylight_saving/start_date/minute=3
0
Start Time
[system/daylight_saving/start_date/month]
The month in a year.
The valid range is 1 to 12.
Start Time
[system/daylight_saving/start_date/day]
The day in a month.
The valid range is 1 to 31.
Start Time
[system/daylight_saving/start_date/hour]
The hour in the day.
The valid range is 0 to 23.
Start Time
[system/daylight_saving/start_date/minute]
The minute in an hour.
The valid range is 0 to 59.
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Parameter
Description
End Time
[system/daylight_saving/end_date]
This sub-section defines the ending day for the
daylight saving offset.
[month] - defines the specific month in a
year
[day] - defines the specific day in a month
[hour] - defines the specific hour in a day
[minute] - defines the specific minute in an
hour
For example: To configure the phone to end the
daylight savings on July 16th at 22:15, set the
following:
system/ntp/daylight_saving/end_date/month
=7
system/ntp/daylight_saving/end_date/day=16
system/ntp/daylight_saving/end_date/hour=2
2
system/ntp/daylight_saving/end_date/minute
=15
End Time
[system/daylight_saving/end_date/month]
The month in a year.
The valid range is 1 to 12.
End Time
[system/daylight_saving/end_date/day]
The day in a month.
The valid range is 1 to 31.
End Time
[system/daylight_saving/end_date/hour]
The hour in the day
The valid range is 0 to 23.
End Time
[system/daylight_saving/end_date/minute]
The minute in an hour.
The valid range is 0 to 59.
Offset
[system/daylight_saving/offset]
The offset value for the daylight saving.
The valid range is 0 to 180. The default offset is
60.
/system/daylight_saving/mode
Configures the daylight saving mode.
Valid values are
[Fixed]= Date is specified as: Month, Day of
month.
[DayOfWeek]= Date is specified as: Month,
Week of month, Day of week.
/system/daylight_saving/start_date/week
Relevant to “Day of week” mode:
The week of month (values 1-5) for start of
daylight saving time.
Version 2.0.0
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Parameter
Description
/system/daylight_saving/start_date/day_of_week
Relevant to “Day of week” mode:
The day of week for daylight saving time start
Valid values :
[SUNDAY]
[MONDAY]
[TUESDAY]
[WEDNESDAY]
[THURSDAY]
[FRIDAY]
[SATURDAY]
/system/daylight_saving/end_date/week
Relevant to “Day of week” mode:
The week of month (values 1-5) for end of
daylight saving time.
/system/daylight_saving/end_date/day_of_week
Relevant to “Day of week” mode:
The day of week for daylight saving time start
Valid values :
[SUNDAY]
[MONDAY]
[TUESDAY]
[WEDNESDAY]
[THURSDAY]
[FRIDAY]
[SATURDAY]
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12.2
12. Configuring Date and Time
Configuring Network Time Protocol Server
The Network Time Protocol (NTP) server can be configured using the Web or
Configuration File, as described below.
To configure the Network Time Protocol Server:
Using the Web:
1.
2.
Access the ‘Date and Time’ page (Configuration tab > Advanced Applications
menu > Date and Time).
Configure the Network Time Protocol Server according to the parameters in the
table below, and then click Submit.
Figure 12-2: NTP and Time Settings Web Fields
Using the Configuration File:
Table 12-2: NTP Server Parameters
Parameter
Description
Active
[system/ntp/enabled]
Enables the NTP server from which the phone
retrieves the date and time.
[0] Disable (default)
[1] Enable – obtains the time information from a
configured NTP server
Primary Server
[system/ntp/primary_server_address]
Defines the address of the main NTP server (this
can be a domain name, for example,
tick.nap.com.ar).
Secondary Server
[system/ntp/secondary_server_address]
Defines the address of the secondary NTP server.
Update Interval
[system/ntp/sync_time]
This sub-section defines how often the phone must
perform an update with the NTP server.
[days] - defines the number of days
[hours] - defines the number of hours
For example: To configure the phone to perform an
update with an NTP server every 1 day and 6 hours,
set the following:
system/ntp/sync_time/days=1
system/ntp/sync_time/hours=6
Update Interval
[system/ntp/sync_time/days]
The number of days.
The valid range is 0 to 7. The default of days is 0.
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Parameter
Description
Update Interval
[system/ntp/sync_time/hours]
The number of hours.
The valid range is 0 to 24. The default is 12.
Time Display Format
[system/ntp/time_display_format]
The format of the time displayed on the LCD screen.
[24Hour] (default)
[12Hour]
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12.3
12. Configuring Date and Time
Configuring Network Time Protocol Server via DHCP
If the IP Phone is set to obtain GMT offsets and NTP servers via DHCP (default), it
receives the following fields in the DHCP options:
Primary Server and Secondary Server – (Option 4 or 42).
Note: If both options (4 and 42) are received, the higher priority is given to
Option 42.
Time Zone – (Option 2)
The IP Phone sends an NTP request to the Primary NTP server. If there is no response,
the NTP request is sent to the Secondary NTP server.
After obtaining the time from the server it adds the GMT offset in Option 2 and this is the
updated system time.
To manually configure the NTP Server and GMT offset via DHCP:
Using the Web:
1.
2.
3.
Access the ‘Date and Time’ page (Configuration tab > Advanced Applications
menu > Date and Time).
From the “Obtain Time Zone From DHCP” drop-down list, select Disable.
Configure the NTP and Time Settings according to the parameters in the table
below, and then click Submit.
Note: These values will have no affect if the timezone is set to be obtained from
DHCP. If the Time Zone and NTP servers are manually set, the IP Phone
acts the same as described above, but the values are obtained from
configuration file and not from DHCP.
Figure 12-3: NTP and Time Settings Web Fields
Table 12-3: NTP Server and GMT Parameters
Parameter
[network/lan/dhcp/ntp/gmt_offset]
Version 2.0.0
Description
Enables the NTP server from which the phone
retrieves the date and time.
[0] Disable (default)
[1] Enable – obtains the time information
from a configured NTP server
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13. IP Network Settings
IP Network Settings
The following section describes how to configure IP Network Settings including:
Static IP Address
Partial DHCP
13.1
Configuring Static IP Address
The static IP address can be configured using the following:
Phone User Interface
Web and Configuration File
13.1.1 Configuring the Static IP Address using Phone User Interface
The procedures for configuring the Static IP Address on the 310HD IP Phone, 320HD IP
Phone and 420HD IP Phone using the Phone User Interface, are described below.
13.1.1.1 310HD IP Phone
The phone’s LAN connection interface can be manually defined (static IP address) or
automatically configured using a DHCP server from where the LAN IP address is obtained.
To configure the phone’s LAN connection type:
1.
Access the Network Settings screen (MENU key > Administration menu > Network
Settings > LAN Conn. Type).
The following screen appears:
2.
3.
Version 2.0.0
Choose one of the following IP addressing schemes (using the
and
Navigation
keys), and then press the ENTER key (the asterisk appears alongside your selection):
•
Static IP
•
Auto. IP (DHCP)
Press the ENTER key again. If you selected Static IP, then continue with Step 4;
otherwise, skip to Step 5.
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4.
If you selected the Static IP option, the following screen appears:
a.
Select the required network parameter (IP Address, Netmask, Gateway,
Primary DNS, and/or Secondary DNS), and then press the ENTER key; the
screen prompts you to enter the IP address of the selected network parameter
(e.g., static IP address below):
b.
c.
Enter the required value, and then press the ENTER key; the setting is applied.
Repeat steps 4.a through 4.c for each required network parameter.
5.
Press the left
below:
Navigation key until you are prompted to save settings, as shown
6.
Select the Yes option (using the
and
ENTER key; the network settings are saved.
Navigation keys), and then press the
13.1.1.2 320HD IP Phone
The phone’s LAN connection interface can be manually defined (static IP address) or
automatically provisioned using a DHCP server from where the LAN IP address is
obtained.
To configure the phone’s LAN connection type:
1.
Access the Settings screen (MENU key > Administration menu > Network Settings
submenu).
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The following screen appears:
2.
Using the
and
Navigation keys, choose LAN Connection Type, and then press
the Select softkey; the following screen appears:
3.
Using the
schemes:
and
Navigation keys, choose one of the following IP addressing
•
Static IP
•
Automatic IP (DHCP)
4.
Press the Select softkey. If you selected "Static IP" in Step 3, continue with Step 5;
otherwise, skip to Step 6.
5.
Define a static IP addressing scheme:
a.
Press the Edit softkey; the Static IP screen appears:
b.
To configure each required network parameter (i.e., "IP Address", "Netmask",
"Gateway", "Primary DNS", and "Secondary DNS"), choose the parameter using
the
and
Navigation keys, and then press the Edit softkey. Enter the new
address in dotted-decimal notation, using the following keys:
♦
Navigation keys: moves the cursor left or right in the IP address
♦
Clear softkey: deletes the digit to the left of the cursor.
Press the Save and then Apply softkey.
c.
6.
Version 2.0.0
Press the Save softkey.
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13.1.1.3 420HD IP Phone
The procedure for configuring Static IP Address on the 420HD IP Phone using the Phone
User Interface is identical to that for the 320HD IP Phone. The phone’s LAN connection
interface can be manually defined (static IP address) or automatically provisioned using a
DHCP server from where the LAN IP address is obtained.
To configure the phone’s LAN connection type:
1.
Access the Settings screen (MENU key > Administration > Network Settings).
2.
Press the Select softkey to select LAN Connection Type; the Settings prompt is
displayed:
3.
Press the navigation control's lower rim to navigate to Static IP or Automatic IP
(DHCP) IP addressing scheme.
4.
Press the Select softkey. If you selected Static IP in Step 3, continue with Step 5;
else skip to Step 6.
5.
Define a static IP addressing scheme:
a.
Press the Edit softkey; the Static IP screen appears:
b.
To configure each required network parameter, i.e., IP Address, Netmask,
Gateway, Primary DNS and Secondary DNS, choose the parameter using the
navigation control's lower or upper rim to navigate to it, and then press the Edit
softkey, for example, navigate to IP Address and press the Edit softkey:
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d.
6.
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Enter the new address in dotted-decimal notation, using the following keys:
♦
Navigation control: moves the cursor left or right in the IP address
♦
Clear softkey: deletes the digit to the left of the cursor.
Press the Save and then Apply softkey.
Press the Save softkey.
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13.1.2 Configuring IP Network Settings using Web and Configuration
File
This section describes how to configure IP Network settings using the Web.
The phone’s LAN configuration includes defining the method for obtaining an IP address.
The phone’s IP address can be static whereby the IP address is manually entered, or
automatic whereby the IP address is acquired from a DHCP server. For Automatic IP, you
can manually define some of the main parameters.
To define the phone’s LAN settings:
Using the Web:
1.
Access the ‘LAN Settings’ page (Configuration tab > Network Connections
menu > Network Settings).
Figure 13-1: Network Settings Web Fields
2.
Configure the LAN settings according to the parameters in the table below, and
then click Submit.
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Using the Configuration File:
Table 13-1: Network Settings Parameters
Parameter
Description
Note: To add a value to these parameters, enter network/ followed by the parameter name, equal
sign and then the value (e.g. network/lan_type=DHCP).
IP Type
[network/lan_type]
Defines the IP addressing method:
[STATIC] Static IP - Phone’s IP address is defined
manually
[DHCP] Automatic IP DHCP (default) - Phone’s IP
address is acquired automatically from a DHCP
server
network/lan/fixed_ip
This sub-section defines the parameters that are
relevant if lan_type is configured to "STATIC" or the
corresponding network/lan/dhcp parameter is set to
1.
IP Address
[network/lan/fixed_ip/ip_address]
The LAN IP address.
Subnet Mask
[network/lan/fixed_ip/netmask]
The subnet mask address.
Default Gateway Address
[network/lan/fixed_ip/gateway]
The IP address of the default gateway.
Domain Name
[network/lan/fixed_ip/domain_name]
The domain name.
Domain Name Server (DNS)
Primary DNS
[network/lan/fixed_ip/primary_dns]
The primary DNS server address.
Secondary DNS
[network/lan/fixed_ip/secondary_dns]
The secondary DNS server address. The phone
connects to this server if the primary DNS server is
unavailable.
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13.2
Configuring Partial DHCP
Partial DHCP can be configured with the following parameters:
Table 13-2: Partial DHCP Parameters
Partial DHCP
network/lan/dhcp
This sub-section defines the parameters to
configure if lan_type is configured to "DHCP".
Domain Name - Manual
[network/lan/dhcp/domain_name/enabled]
Enables setting the domain name manually.
[0] Disable (default)
[1] Enable
Note: If enabled,
network/lan/fixed_ip/domain_name must also
be set.
IP Address - Manual
[network/lan/dhcp/ip_address/enabled]
Enables setting the IP address manually.
[0] Disable (default)
[1] Enable
Note: If enabled,
network/lan/fixed_ip/ip_address must be set.
Subnet Mask - Manual
[network/lan/dhcp/netmask/enabled]
Enables setting the network mask manually.
[0] Disable (default)
[1] Enable
Note: If enabled, network/lan/fixed_ip/netmask
must be set.
[network/lan/dhcp/gateway/enabled]
Enables setting the default gateway manually.
[0] Disable (default)
[1] Enable
Note: If enabled, network/lan/fixed_ip/gateway
must be set.
Primary DNS - Manual
[network/lan/dhcp/primary_dns/enabled]
Enables setting the primary DNS manually.
[0] Disable (default)
[1] Enable
Note: If enabled,
network/lan/fixed_ip/primary_dns must be set.
Secondary DNS - Manual
network/lan/dhcp/secondary_dns/enabled
Enables setting the secondary DNS manually.
[0] Disable (default)
[1] Enable
Note: If enabled,
network/lan/fixed_ip/secondary_dns must be
set.
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DHCP-Related Parameters
network/lan/dhcp/ntp/server_list/enabled
Enables prioritization of the NTP server’s
information received from the DHCP server
(Option fields 42 or 4), over the static
configuration
(system/ntp/primary_server_address and
system/ntp/secondary_server_address).
[0] Disable
[1] Enable (default)
network/lan/dhcp/ntp/gmt_offset/enabled
Enables prioritization of the NTP GMT offset
information received from the DHCP server
(Option field 2), over the static configuration
(system/ntp/gmt_offset).
[0] Disable
[1] Enable (default)
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14. LAN and PC Port Settings
LAN and PC Port Settings
Port settings can be configured using the Web or Configuration File, as described below.
Note: The optional values of the Configuration file parameters are enclosed in
square brackets while its corresponding Web values are written outside
the square brackets, for example, [1] Enable.
To define the phone’s port settings:
Using the Web:
1.
Access the ‘LAN Settings’ page (Configuration tab > Network Connections
menu > LAN Settings).
Figure 14-1: Port Mode Web Fields
2.
Configure Port Mode according to the parameters in the table below, and then
click Submit.
Using the Configuration File:
Table 14-1: Port Settings Parameters
Parameter
Description
LAN Port Mode
[network/lan/port_mode]
Sets the LAN port mode.
Valid values are :
[AUTOMATIC] = Auto negotiation.
[FULL_10] = 10Mbps + full duplex
[FULL_100] = 100Mbps + half duplex
[HALF_10] = 10Mbps + full duplex
[HALF_100] = 100Mbps + half duplex
PC Port Mode
[network/pc/port_mode]
Sets the computer port mode.
See valid values above.
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15. VLAN Settings
VLAN Settings
VLAN settings can be configured using the Web or Configuration File, as described below.
To configure the phone’s VLAN settings:
Using the Web:
1.
Access the ‘Network Settings’ page (Configuration tab > Network Connections
menu > Network Settings).
Figure 15-1: VLAN Web Fields
2.
Configure the VLAN settings according to the parameters in the table below, and
then click Submit.
Using the Configuration File:
Table 15-1: VLAN Parameters Description
Parameter
Description
VLAN Discovery Mode
[network/lan/vlan/mode]
Determines the VLAN mode of operation.
[Disable] Disable
[Manual] Manual Configuration of LAN - Static
configuration of VLAN ID and priority
[CDP] Automatic Configuration of VLAN - VLAN
discovery mechanism based on Cisco Discovery
Protocol (CDP)
[LLDP] Automatic Configuration of VLAN - VLAN
discovery mechanism based on LLDP.
[CDP_LLDP] Automatic Configuration of VLAN
(default) - VLAN discovery mechanism based on
LLDP and Cisco Discovery Protocol (CDP). LLDP
protocol is with higher priority.
Period
[network/lan/vlan/period]
The time period in seconds between discovery
messages when configured to CDP, LLDP or CDP and
LLDP.
The default value is 30.
VLAN ID
[network/lan/vlan/id]
The VLAN ID.
The valid range is 0 to 4096. The default VLAN ID is 0.
VLAN Priority
[network/lan/vlan/priority]
The priority of traffic pertaining to this VLAN.
The valid range is 0 to 7 (where 7 is the highest
priority). The default VLAN priority is 0.
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16. Enabling Telnet Access
Enabling Telnet Access
Telnet access can be enabled using the Web or Configuration File, as described below.
To enable Telnet:
Using the Web:
1.
Access the ‘Telnet’ page (Management tab > Remote Management menu >
Telnet).
Figure 16-1: Telnet Web Field
2.
Enable Telnet according to the parameter in the table below, and then click
Submit.
Using the Configuration File:
Table 16-1: Telnet Parameters
Parameter
Description
Note: To add a value to these parameters, enter management/ followed by the parameter name,
equal sign and then the value (e.g. management/telnet/enabled=0).
Activate
[management/telnet/enabled]
Version 2.0.0
Enables telnet access to the phone.
[0] Disable (default)
[1] Enable
The user name and password for telnet access are according to
the parameters: system/user_name and system/password.
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VoIP Settings
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17. SIP Settings
SIP Settings
The following can be configured using the Web or Configuration File, as described below.
General
Proxy and Registration
SIP Timers
SIP QoS
17.1
Configuring General SIP Settings
SIP General settings can be configured using the Web or Configuration File, as described
below.
To configure the phone’s general SIP settings:
Using the Web:
1.
Access the ‘Signaling Protocol’ page (Configuration tab > Voice Over IP menu
> Signaling Protocols).
Figure 17-1: SIP General Web Fields
2.
Version 2.0.0
Configure the General settings according to the parameters in the table below,
and then click Submit.
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Using the Configuration File:
Table 17-1: SIP General Parameters
Parameter
Description
SIP Transport Protocol
[voip/signalling/sip/transport_protocol]
Determines the transport layer for
outgoing SIP calls initiated by the
phone.
[UDP] UDP (default)
[TCP] TCP
[TLS] TLS
TLS Port
[voip/signalling/sip/tls_port]
Defines the local TLS SIP port for
SIP messages.
The valid range is 1024 to 65535.
The default value is 5061.
/voip/signalling/sip/enable_sips
Relevant for TLS only, if enabled,
the request URI prefix will be
“sips:” otherwise, the prefix will
be “sip:”
/voip/signalling/sip/subs_no_notify_timer
Indicates the maximum time (in
milliseconds) that a subscription
waits from receiving 2xx response
for a SUBSCRIBE request, until
receiving the first NOTIFY
request. If the timer expires, the
subscription will be terminated.
SIP Local Port
[voip/signalling/sip/port]
Defines the local SIP port (UDP
or TCP) for SIP messages.
The valid range is 1024 to 65535.
The default value is 5060.
Gateway Name
[voip/signalling/sip/proxy_gateway]
Assigns a name to the phone.
The name is used as the host
part of the SIP URI in the From
header.
Notes:
Ensure that the name you
choose is the one with which
the Proxy is configured to
identify the phone.
If not specified, the phone's IP
address is used (default).
PRACK Mode
[voip/signalling/sip/prack/enabled]
Determines whether the phone
sends PRACK (Provisional
Acknowledgment) messages
upon receipt of 1xx SIP reliable
responses.
[0] Disable
[1] Enable (default)
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Parameter
Description
Enable RPORT
[voip/signalling/sip/rport/enabled]
Determines whether the phone
adds the ‘rport’ parameter to the
relevant SIP message (in the SIP
Via header).
[0] Disable (default)
[1] Enable
Include PTIME in SDP
[voip/signalling/sip/sdp_include_ptime]
Determines whether the phone
adds the PTIME parameter to the
SDP message body.
[0] Disable (default)
[1] Enable
Enable Keep Alive using OPTIONS
[voip/signalling/sip/keepalive_options/enabled]
Determines whether keep-alive is
performed using SIP OPTIONS
messages sent to the Proxy.
[0] Disable (default)
[1] Enable
Keep Alive Period
[voip/signalling/sip/keepalive_options/timeout]
Defines the Proxy keep-alive time
interval (in seconds) between
Keep-Alive messages.
The valid range is 0 to 86400.
The default value is 300.
Connect Media on 180 Response
[voip/signalling/sip/connect_media_on_180]
Determines whether the media is
connected upon receipt of SIP
180, 183, or 200 messages.
When the parameter is disabled,
media is connected upon receipt
of 183 and 200 messages only.
[0] Disable (default)
[1] Enable
Block Caller ID on Outgoing Calls
When enabled, the outgoing
INVITE message is sent with an
anonymous From header and PAsserted-Identity header.
[0] Disable (default)
[1] Enable
For example:
FROM header contains
anonymous URI:
From: "Anonymous"
sip:[email protected]
nvalid
P-Asserted-Identity header:
P-Asserted-Identity: "1001"
[email protected]
[voip/signalling/sip/block_callerid_on_outgoing_calls]
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Parameter
Description
Incoming Anonymous Call Blocking
[voip/signalling/sip/anonymous_calls_blocking]
When enabled, incoming INVITE
messages with anonymous From
header are rejected.
[0] Disable (default)
[1] Enable
For example:
From:"Anonymous"<sip:anonymo
[email protected]>
The phone responds with a SIP
403 "Forbidden" response.
[voip/signalling/sip/auth_retries]
Defines the number of times
authenticated register messages
are re-sent if 401 or 407 SIP
responses with a different
"nonce" are received.
The valid range is 0 to 100. The
default value is 4.
[voip/signalling/sip/display_name_in_registration_msg/enabled]
Sets the Display Name in the
"To" and "From" fields of the SIP
REGISTER message.
[0] Disable (default)
[1] Enable
voip/signalling/sip/semi_transfer_with_no_cancel/enabled
Determines whether semiattendant transfer is performed
without sending the SIP CANCEL
message to the remote side.
[0] Disable (default)
[1] Enable
Notes:
In this flow ("with_no_cancel"),
the Transferor's User Agent
continues the transfer as an
attended transfer even after
the Transferor hangs up. This
is the recommended flow
defined by
http://tools.ietf.org/html/draftietf-sipping-cc-transfer-03.
Existing / current behavior is
retained for backward
compatibility (disabled by
default)
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17.2
17. SIP Settings
Proxy and Registration
This section describes the process for configuring Proxy and Registration settings.
17.2.1 Configuring General Proxy and Registration Settings
Proxy and Registration settings can be configured using the Web or Configuration File, as
described below.
To configure the phone’s Proxy and Registration settings:
Using the Web:
1.
Access the ‘Signaling Protocol’ page (Configuration tab > Voice Over IP menu
> Signaling Protocols).
Figure 17-2: SIP Proxy and Registrar Web Fields
2.
Version 2.0.0
Configure Proxy and Registration according to the parameters in the table below,
and then click Submit.
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Using the Configuration File:
Table 17-2: Proxy and Registrar Parameters
Parameter
Description
Use SIP Proxy
[voip/signalling/sip/use_proxy]
Determines whether to use a SIP Proxy
server.
[0] Disable (default)
[1] Enable
voip/signalling/sip/proxy_address
The IP address or host name of the SIP
proxy server.
The default value is 0.0.0.0.
Proxy Port
[voip/signalling/sip/proxy_port]
The UDP or TCP port of the SIP proxy
server.
The valid range is 1024 to 65535. The default
value is 5060.
Enable Registrar Keep Alive
Determines whether to use the registration
keep-alive mechanism based on SIP
OPTION messages.
[0] Disable (default)
[1] Enable
Notes:
If there is no response from the server, the
timeout for re-registering is automatically
reduced to a user-defined value
(voip/signalling/sip/registration_failed_
timeout)
When the phone re-registers, the keepalive messages are re-sent periodically.
[voip/signalling/sip/registrar_ka/enabled]
Registrar Keep Alive Period
voip/signalling/sip/registrar_ka/timeout
Defines the registration keep-alive time
interval (in seconds) between Keep-Alive
messages.
The valid range is 40 to 65536. The default
value is 60.
Maximum Number of Authentication Retries
voip/signalling/sip/proxy_timeout
The SIP proxy server registration timeout (in
seconds).
The valid range is 0 to 86400. The default
value is 3600.
Registration Failed Expires
If registration fails, this parameter determines
the interval between the register messages
periodically sent until successful registration.
The valid range is 1 to 86400. The default
value is 60.
voip/signalling/sip/registration_failed_timeout
Use SIP Registrar
voip/signalling/sip/sip_registrar/enabled
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Determines whether the phone registers to a
separate SIP Registrar server.
[0] Disable (default)
[1] Enable
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Parameter
Description
Use SIP Proxy IP and Port for Registration
voip/signalling/sip/use_proxy_ip_port_for_registrar
Determines whether to use the SIP proxy’s IP
address and port for registration. When
enabled, there is no need to configure the
address of the registrar separately.
[0] Disable (default)
[1] Enable
Registrar IP Address or Host Name
voip/signalling/sip/sip_registrar/addr
The IP address or host name of the Registrar
server.
The default value is 0.0.0.0.
Registrar Port
voip/signalling/sip/sip_registrar/port
The UDP or TCP port of the Registrar server.
The valid range is 1024 to 65535. The default
value is 5060.
Use SIP Outbound Proxy
Determines whether an outbound SIP proxy
server is used (all SIP messages are sent to
this server as the first hop).
[0] Disable (default)
[1] Enable
[voip/signalling/sip/sip_outbound_proxy/enabled]
Outbound Proxy IP Address or Host Name
[voip/signalling/sip/sip_outbound_proxy/addr]
The IP address of the outbound proxy. If this
parameter is set, all outgoing messages
(including Registration messages) are sent to
this Proxy according to the Stack behavior.
The default value is 0.0.0.0.
Outbound Proxy Port
[voip/signalling/sip/sip_outbound_proxy/port]
The port on which the outbound proxy listens.
The valid range is 1024 to 65535. The default
value is 5060.
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17.2.2 Configuring Proxy Redundancy
The Redundant Proxy feature allows the configuration of a backup SIP proxy server to
increase QoS stability. Once this feature is enabled, the phone identifies cases where the
primary proxy does not respond to SIP signaling messages. In these scenarios, the phone
registers to the redundant proxy and the phone seamlessly continues normal functionality,
without the user noticing any connectivity failure or malfunction with the primary proxy.
The Redundant Proxy feature can operate in one of the following modes:
Asymmetric mode: The primary proxy is assigned a higher priority for registration
than the redundant proxy. Once the phone is registered to the primary proxy, it sends
keep-alive messages (using SIP OPTIONS messages) to the primary proxy. If the
primary proxy does not respond, the phone registers to the redundant proxy, but
continues sending keep-alive messages to the primary proxy. If the primary proxy
responds to these keep-alive messages, the phone re-registers to the primary
proxy.
Symmetric mode: Both proxies are assigned the same priority for registration. Once
the phone is registered to a proxy, it sends keep-alive messages to this proxy. The
phone switches proxies only once the proxy to which it has registered, does not
respond.
For more information see the [voip/signalling/sip/redundant_proxy/symmetric_mode]
description in the SIP Proxy Server Redundancy Parameters table below.
Proxy Redundancy can be configured using the Web or Configuration file.
To configure Proxy Redundancy:
Using the Web:
1.
Access the ‘Signaling Protocol’ page (Configuration tab > Voice Over IP menu
> Signaling Protocols).
Figure 17-3: Proxy Redundancy
2.
Configure the SIP Redundancy parameters according to the table below, and
then click Submit.
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Using the Configuration file:
Table 17-3: SIP Proxy Server Redundancy Parameters
Parameter
Description
Enables the redundant proxy mechanism.
[0] Disable (default) - Phone doesn't
use redundant proxy (default).
[1] Enable - Phone registered to
redundant proxy if the primary proxy
does not respond to SIP signaling
messages.
Use Redundant Proxy
[voip/signalling/sip/redundant_proxy/enabled]
Redundant Proxy Address
The IP address or host name of the
redundant proxy.
The default value is 0.0.0.0.
Note: This parameter is applicable only if
the parameter use_redundant_proxy is
set to 1.
[voip/signalling/sip/redundant_proxy/address]
Redundant Proxy Port
The UDP or TCP port of the redundant
proxy server.
The valid range is 1024 to 65535. The
default value is 5060.
Note: This parameter is applicable only if
the parameter use_redundant_proxy is
set to 1.
voip/signalling/sip/redundant_proxy/port
Redundant Proxy Keep Alive Period
voip/signalling/sip/redundant_proxy/keepalive_period
Switch back to Primary SIP proxy when available
[voip/signalling/sip/redundant_proxy/symmetric_mode]
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Defines the interval in seconds for
sending keep-alive messages to the
proxy.
The valid range is 0 to 300. The default
value is 60.
Note: This parameter is applicable only if
the parameter use_redundant_proxy is
set to 1.
Determines the proxy redundancy mode.
[0]- Asymmetric mode. (default)
[1] - Symmetric mode.
The Redundant Proxy feature allows the
configuration of a backup SIP proxy
server to increase QoS stability. Once
this feature is enabled, the phone
identifies cases where the primary proxy
does not respond to SIP signaling
messages. In these scenarios, the phone
registers to the redundant proxy and the
phone seamlessly continues normal
functionality, without the user noticing
any connectivity failure or malfunction
with the primary proxy.
The Redundant Proxy feature includes
two operational modes:
Asymmetric mode: the primary proxy
is assigned a higher priority for
registration than the redundant proxy.
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Parameter
Description
In the Asymmetric mode, once the
phone is registered to the primary
proxy, it sends keep-alive messages
(using SIP OPTIONS messages) to
the primary proxy. If the primary proxy
does not respond, the phone registers
to the redundant proxy, but continues
sending keep-alive messages to the
primary proxy. If the primary proxy
responds to these keep-alive
messages, the phone re-registers to
the primary proxy. Therefore, the
phone assigns the primary proxy a
higher priority for registration.
Symmetric mode: both proxies are
assigned the same priority for
registration. In the Symmetric mode,
once the phone is registered to a
proxy, it sends keep-alive messages
to this proxy. The phone switches
proxies only once the proxy to which it
has registered does not respond.
Therefore, the phone assigns both
proxies the same priority for
registration
In both modes, the following applies:
If the phone is not registered (i.e., if the
proxy server¬–redundant or primary–to
which the phone currently tries to register
does not respond), the phone attempts to
register to an alternative proxy. These
attempts continue until the phone
successfully registers.
If this feature is enabled and the user
reboots the phone, the phone registers to
the last proxy to which it was trying to
register, and not necessarily to the
primary proxy.
Note: This parameter is applicable only if
the parameter use_redundant_proxy is
set to 1.
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17.3
17. SIP Settings
Configuring SIP Timers
SIP Timers can be configured using the Web or Configuration File, as described below.
To configure the phone’s SIP timer settings:
Using the Web:
1.
Access the ‘Signaling Protocol’ page (Configuration tab > Voice Over IP menu
> Signaling Protocols).
Figure 17-4: SIP Timers
2.
Version 2.0.0
Configure the SIP Timer parameters according to the table below, and then click
Submit.
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Using the Configuration file:
Table 17-4: SIP Timers Parameters
Parameter
Description
Retransmission Timer T1
[voip/signalling/sip/sip_t1]
The time interval (in msec) between the first
transmission of a SIP message and the first
retransmission of the same message (according to
RFC 3261).
The valid range is 100 to 60000. The default value is
500.
Note: The time interval between subsequent
retransmissions of the same SIP message starts with
SipT1Rtx and is multiplied by two until SipT2Rtx. For
example (assuming that SipT1Rtx = 500 and SipT2Rtx
= 4000):
The first retransmission is sent after 500 msec.
The second retransmission is sent after 1000
(2*500) msec.
The third retransmission is sent after 2000 (2*1000)
msec.
The fourth retransmission and subsequent
retransmissions until SIPMaxRtx are sent after 4000
(2*2000) msec..
Retransmission Timer T2
[voip/signalling/sip/sip_t2]
The maximum interval (in msec) between
retransmissions of SIP messages (according to RFC
3261).
The valid range is 4000 to 60000.
The default value is 4000.
Note: The time interval between subsequent
retransmissions of the same SIP message starts with
SipT1Rtx and is multiplied by two until SipT2Rtx.
Retransmission Timer T4
[voip/signalling/sip/sip_t4]
The SIP T4 retransmission timer according to RFC
3261.
The valid range is 5000 to 60000.
The default value is 5000.
INVITE Timer
The SIP INVITE timer according to RFC 3261.
The valid range is 0 to 65535. The default value is
32000.
[voip/signalling/sip/sip_invite_timer]
Session-Expires
[voip/signalling/sip/session_timer]
Min-SE
[voip/signalling/sip/min_session_interval]
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The time (in seconds) at which an element considers
the call timed out if no successful INVITE transaction
occurs beforehand. This value is inserted into every
INVITE in the Session-Expires header unless it is
configured to 0. If the timer option tag is not part of the
supported list, the sessionExpires value is ignored.
The valid range is 0 to 65535. The default value is
1800.
The minimum value for the session interval that the
application is willing to accept.
The valid range is 0 to 65535. The default value is 90.
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17. SIP Settings
Configuring SIP QoS
SIP Quality of Service (QoS) can be configured using the Web or Configuration File, as
described below.
To configure the phone’s SIP QoS:
Using the Web:
1.
Access the ‘Signaling Protocol’ page (Configuration tab > Voice Over IP menu
> Signaling Protocols).
Figure 17-5: Quality of Service Page
2.
Configure the SIP Quality of Service parameters according to the table below,
and then click Submit.
Using the Configuration File:
Table 17-5: SIP QoS Parameters
Parameter
Description
Type of Service (ToS)
[voip/signalling/sip/tos]
QoS in hexadecimal format. This is a part of the IP header that defines
the type of routing service to tag outgoing signalling packets originated
from the phone. It informs routers that this packet must receive a
specific QoS.
The default value is 0x60.
Values can be set in decimal (e.g. 96) or hexadecimal (e.g. 0x60).
For information on Configuring RTP QoS, see Section 19.3 on page 119.
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17.5
Configuring SIP Reject Code
Reject Code can be configured using the Web or Configuration File, as described below.
To configure Reject Code:
Using the Web:
1.
Access the ‘Services’ page (Configuration tab > Voice Over IP menu >
Services > General Parameters).
Figure 17-6: Reject Code Web Field
2.
Configure the Reject Code parameters according to the table below, and then
click Submit.
Using the Configuration File:
Table 17-6: Reject Code Parameter
Parameter
/voip/services/reject_code
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Description
Configures the reject code that the IP Phone will send when
pressing the Reject softkey or while DND is activated.
Valid values are:
[CODE_603]
[CODE_486]
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18. Dialing Settings
Dialing Settings
Dialing Settings can be configured using the Web or Configuration File, as described
below.
18.1
Configuring General Dialing Settings
General Dialing settings can be configured using the Web or Configuration File, as
described below.
To define the dialing parameters:
Using the Web:
1.
Access the ‘Dialing’ page (Configuration tab > Voice Over IP menu > Dialing).
Figure 18-1: General Dialing Web Fields
2.
Version 2.0.0
Configure Dialing options according to the parameters in the table below, and
then click Submit.
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Using the Configuration File:
Table 18-1: Dialing Parameters
Parameter
Description
Dialing Timeout
[voip/dialing/timeout]
The duration (in seconds) of allowed inactivity between dialled
digits. When you work with a proxy, the number you have
dialled before the dialing process has timed out is sent to the
proxy as the user ID to be called. This is useful for calling a
remote party without creating a speed dial entry (assuming the
remote party is registered with the proxy).
The valid range is 0 to 10. The default value is 5.
Phone Number Length
[voip/dialing/phone_number_
max_size]
The maximum length of shortcut numbers that you can enter
and the maximum number of digits that you can dial
The valid range is 3 to 32. The default value is 32.
Enable Dialing Complete Key
[voip/dialing/dial_complete_k
ey/enabled]
Enables the feature for defining a key to indicate that dialing
has completed. Pressing the Dialing Complete key (defined
below) forces the phone to make a call to the dialled digits even
if there is no match in the dial plan or digit map.
[0] Disable
[1] Enable (default)
Note: This parameter is available only if the parameter
voip/dialing/dial_complete_key/enabled is set to 1.
Dialing Complete Key
[voip/dialing/dial_complete_k
ey/key]
Defines the Dialing Complete key.
The valid value is a single character. The default value is the
pound (#) key.
No Answer Call Timeout
[voip/dialing/unanswered_call
_timeout]
Timeout before the phone automatically sends a Cancel
message. When the phone makes a call and the other side
doesn’t answer, the phone sends a Cancel after this timeout
The valid range is 1 to 300. The default value is 60.
[voip/dialing/on_hook_dialing]
Defines the dialing mode when phone is on hook and no audio
device is selected (when user enters digits on idle state).
Valid values are:
[Disable] = Ignore digits press. In order to initiate a call the
user will have to select audio device by pressing speaker or
headset keys or by picking headset.
[Open_default_audio_device]= default behavior – start dialing
via default audio device (usually speaker) activated.
[Off_line_dialing] = don`t activate the default audio device
until pressing “dial” , DTMF tones will not be heard and dialing
related features ( such as “dialing timeout” , “dial complete key”
, and more ) will be disabled.
[voip/dialing/allow_calling_self
_extension]
If disabled (default), calling the self-number (user ID) will be
blocked.
If enabled, the phone will send the invite although it is for its
own extension. (In some proxies this is how you access voice
mail).
[voip/dialing/speed_dial/doubl
e_press/enabled]
Relevant for 310HD only, if enabled, the first speed dial press
will display the number and only the second press will dial.
If disabled, the first press will dial.
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18.2
18. Dialing Settings
Tones
Tones settings can be configured using the Web or Configuration File, as described below.
To configure the dial tones:
Using the Web:
1.
Access the ‘Dialing’ page (Configuration tab > Voice Over IP menu > Dialing).
Figure 18-2: Tones Web Fields in Dialing Page
2.
Access the ‘Services’ page (Configuration tab > Voice Over IP menu >
Services).
Figure 18-3: Tones Web Fields in Services Page
3.
Configure Tones options according to the parameters in the table below, and then
click Submit.
Using the Configuration File:
Table 18-2: Dial Tones Parameters
Parameter
Description
Dial Tone Timeout
[voip/dialing/dialtone_timeout]
Defines the maximum duration of the dial tone (in
seconds) after which the dial tone stops and a
reorder tone is played.
The valid range is 1 to 300. The default value is
30.
Reorder Tone Timeout
[voip/dialing/warning_tone_timeout]
Defines the maximum duration of the reorder
tone (in seconds) after which the reorder tone
stops and a howler tone is played.
The valid range is 1 to 300. The default value is
40.
Howler Tone Timeout
[voip/dialing/offhook_tone_timeout]
Defines the duration (in seconds) of the howler
tone. If the limit is exceeded, the howler tone
stops. The howler tone indicates that the phone
has been left in an off-hook state.
The valid range is 1 to 300. The default value is
120.
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Parameter
Description
Secondary Dial Tone
[voip/dialing/secondary_dial_tone/enabled]
Enables the secondary dial tone.
[0] Disable - Phone doesn't use secondary
dial tone.
[1] Enable (default) - Phone plays secondary
dial tone if the secondary dial tone key is
pressed (first digit).
Secondary Dial Tone Key
[voip/dialing/secondary_dial_tone/key_seq
uence]
Defines the secondary dial tone is played if this is
the first key pressed.
The valid range is 0 to 9. The default value is 9.
Note: This parameter is available only if the
parameter
voip/dialing/secondary_dial_tone/enabled is
set to 1.
Out of Service Behavior
[voip/services/out_of_service_behavior]
Determines whether a reorder tone is played
instead of a dial tone if you configured a
Registrar IP address and the registration failed.
[NONE] No Tone
[REORDER_TONE] Reorder Tone (default)
Stutter Tone Duration
[voip/services/msg_waiting/stutter_tone_d
uration]
Defines the duration for which a stutter tone is
played when you have unheard messages.
The valid range is 1000 to 60000.
The default value is 2500.
Automatic Disconnect
[voip/dialing/automatic_disconnect]
Determines whether the phone automatically
goes idle (i.e. on-hook) when the last remaining
call is disconnected. This is only relevant when
the speaker or headset is used.
[0] Disable
[1] Enable(default)
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18. Dialing Settings
18.2.1 Uploading Ring Tones
New Ring Tones can be uploaded using the Web or Configuration File, as described
below.
Notes:
•
The ring tone file must be in WAV file format (A/Mu-Law, 8-kHz
audio sample rate and 8-bit audio sample size or PMC 16-kHz audio
sample rate and 16-bit audio sample size, Intel PCM encoding).
•
If you want the phone to use an uploaded ring tone, you need to use
the phone’s LCD screen to select it (refer to the phone’s User’s
Manual).
To upload a ring tone:
Using the Web:
1.
Access the ‘Tones’ page (Configuration tab > Personal Settings menu >
Tones).
Figure 18-4: Upload Ringing Tones Web Fields
2.
3.
4.
Version 2.0.0
In the ‘Ringing Tone Name’ field, enter the name of the ring tone file that you
want to upload. If you do not enter a name, the phone assigns the tone’s file
name (without the .wav file extension) as the name of the tone.
Click the Browse button, navigate to the folder in which the ring tone file is
located, select the file, and then click Open; the file name and path is displayed in
the ‘File Location’ field.
Click Submit; the file is loaded to the phone and displayed in the Ring Tone list.
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To delete Ring Tones:
•
Deleting selected ring tones: select the ‘Delete’ check boxes corresponding to the
ring tone that you want to delete, and then click Submit.
Using the Configuration File:
Table 18-3: Ring Tone Parameters
Parameter
Ringing Tone Name File Location
[provisioning/ring_tone_uri]
Administrator's Manual
Description
The URI for retrieving the ring tones file. The ring tones
can be compressed to *.zip or *.tgz files and provided
to the phone during provisioning.
For example: provisioning/ring_tone_uri=tones.tgz
Notes:
The ringtone file is downloaded only after boot up,
and not periodically.
If the tones file is new, the phone updates the
information, but does not reboot.
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18. Dialing Settings
18.2.2 Regional Settings for Call Progress Tones
You can suite your phone’s call progress tones to the location in which the phone is
located. In addition, instead of using the default ring tone, you can upload a different ring
tone file and use that ring tone for indicating incoming calls. This ring tone can only be
selected using the phones LCD interface.
Follow the procedure below for selecting the region in which your phone is located. This is
important for suiting your phone for the call progress tones of the country in which the
phone is located.
To select the geographical location of your phone:
Using the Web:
1.
Access the ‘Tones’ page (Configuration tab > Personal Settings menu >
Tones).
Figure 18-5: Tones Web Fields
2.
3.
Version 2.0.0
From the ‘Current Location’ drop-down list, select the country in which your
phone is located.
Click Submit.
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Using the Configuration File:
Table 18-4: Regional Parameters
Parameter
Description
Current Location
[voip/regional_settings/selected_co
untry]
Defines the country in which your phone is located. The
behavior and parameters of analog telephones lines vary
between countries. The set of Call Progress Tones are
all location-specific. The phone automatically selects the
correct regional settings according to this setting.
Supported countries:
[Israel] Israel
[China] China
[France] France
[Germany] Germany
[Netherlands] Netherlands
[UK] UK
[Brazil] Brazil
[Italy] Italy
[Argentina] Argentina
[Portugal] Portugal
[Russia] Russia
[Australia] Australia
[USA] USA
voip/regional_settings/use_config_f
ile_values
Enables the user-defined CPT. When this parameter is
enabled, the selected_country parameter is not relevant
and the below Call Progress Tones values can be
determined by the user.
[0] - Disable (default)
[1] - Enable
Call Progress Tones (CPT)
Note: Up to 10 CPT’s can be configured (voip/regional_settings/call_progress_tones/0…9).
voip/regional_settings/call_progres
s_tones/%d/enabled
Enables the specific CPT.
[0] - Disable
[1] - Enable
voip/regional_settings/call_progres
s_tones/%d/name
Defines the name of the CPT.
voip/regional_settings/call_progres
s_tones/%d/cadence
Defines the cadence type of the tone.
[0] - Continuous signal
[1] - Cadence signal
[2] - Burst signal
voip/regional_settings/call_progres
s_tones/%d/frequency_a
Defines the low frequency (in Hz) of the tone.
The valid value range is 300 to 1980 Hz, in steps of 1 Hz.
Unused frequencies must be set to zero.
voip/regional_settings/call_progres
s_tones/%d/frequency_b
Defines the high frequency (in Hz) of the tone.
The valid value range is 300 to 3000 Hz, in steps of 1 Hz.
Unused frequencies must be set to zero.
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Parameter
Description
voip/regional_settings/call_progres
s_tones/%d/frequency_a_level
Output level of the low frequency tone (in -dBm) in Call
Progress generation.
The valid range is 0 to 63, where 63 is mute.
voip/regional_settings/call_progres
s_tones/%d/frequency_b_level
Output level of the low frequency tone (in -dBm) in Call
Progress generation.
The valid range is 0 to 63, where 63 is mute.
voip/regional_settings/call_progres
s_tones/%d/tone_on_0
tone_on_0 to tone_on_3.
If the signal is Cadence or Burst, then this value
represents the on duration. If a Continuous tone, then
this value represents the minimum detection time. The
units are in 10 msec.
The valid range is 0 to 10000.
voip/regional_settings/call_progres
s_tones/%d/tone_off_0
tone_off_0 to tone_on_3.
If the signal is Cadence, then this value represents the
off duration. The units are in 10 msec. If it is not used,
then set it to zero. If the signal is Burst, only tone_off 0 is
relevant. It represents the off time that is required from
the end of the signal to the detection time.
The valid range is 0 to 10000.
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18.3
Configuring Default Audio Device
Default Audio Device Tones settings can be configured using the Web or Configuration
File, as described below.
To configure default Audio Device:
Using the Web:
1.
Access the ‘Dialing’ page (Configuration tab > Voice Over IP menu > Dialing).
Figure 18-6: Default Audio Device Web Field
2.
Configure the parameter according to the table below and then click Submit.
Using the Configuration file:
Table 18-5: Audio Device Parameter
Parameter
Audio Device
[voip/answer_device]
Administrator's Manual
Description
Sets the default audio device to answer or initiate a new call
when no explicit audio device is set.
For example:
When pressing ANS SK.
When initiating a call by speed dial key, call history or
phone directory.
Answering talk event or auto-answer.
When starting to dial in “on hook” mode.
Valid values are:
[SPEAKER]
[HEADSET]
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18.4
18. Dialing Settings
Configuring DTMF
DTMF can be configured using the Web or Configuration File, as described below.
To configure DTMF:
Using the Web:
1.
Access the ‘Dialing’ page (Configuration tab > Voice Over IP menu > Dialing).
Figure 18-7: DTMF Transport Mode Web Field
2.
Configure the parameter according to the table below and then click Submit.
Using the Configuration file:
Table 18-6: DTMF Description
Parameter
DTMF Transport Mode
[voip/media/out_of_band_dtmf]
Version 2.0.0
Description
DTMF transport mode.
[INBAND] Inband
[RFC2833] RFC 2833 (default)
[VIA_SIP] Via SIP
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18.5
Configuring Digit Maps and Dial Plans
Digit maps and Dial plans can be configured using the Web or Configuration File, as
described below.
To configure digit map and dial plan:
Using the Web:
1.
Access the ‘Dialing’ page (Configuration tab > Voice Over IP menu > Dialing).
Figure 18-8: Digit Map and Dial Plan Web Fields
2.
Configure the parameters according to the table below, and then click Submit.
Using the Configuration file:
Table 18-7: Digit Map and Dial Plan Parameters
Parameter
Digit Map
[voip/signalling/sip/digit_map]
Administrator's Manual
Description
Enables the administrator to predefine possible formats (or
patterns) for the dialed number. A match to one of the
defined patterns terminates the dialed number.
The valid value can be up to 256 characters.
There are two main formats for the digit map configuration.
The formats are distinguished by the separator ‘;’ or ’|’.
Using ‘|’ separator: The following constructs can be
used in each numbering scheme:
Digit: A digit from "0" to "9".
DTMF: A digit, or one of the symbols "A", "B", "C",
"D", "#", or "*". Extensions may be defined.
Wildcard: The symbol "x" which matches any digit
("0" to "9").
* Range: One or more DTMF symbols enclosed
between square brackets ("[" and "]").
Sub range: Two digits separated by hyphen ("-")
which matches any digit between and including the
two. The subrange construct can only be used
inside a range construct, i.e., between "[" and "]".
Position: A period (".") which matches an arbitrary
number, including zero, of occurrences of the
preceding construct.
For example: [2-9]11|0|100|101|011xxx.|9011xxx.|1[29]xxxxxxxxx|91[2-9]xxxxxxxxx|9[29]xxxxxx|*xx|[8]xxxx|[2-7]xxx
This example includes the following rules:
[2-9]11: 911 rule: 211, 311, 411, 511, 611, 711,
811, 911 are dialed immediately
0: Local operator rule: After dialing "0" the phone
waits T seconds and then completes the call
automatically
100: Auto-attendant default extension
101: Voicemail default extension
011xxx.: International rule without prefix
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Parameter
Description
9011xxx.: International rule with prefix
1[2-9]xxxxxxxxx: LD rule without prefix
91[2-9]xxxxxxxxx: LD rule with prefix
9[2-9]xxxxxx: Local call with prefix
*xx: 2-digit star codes
[1-7]xx: A regular 3 digit extension that does not
start with 9 or 8 is dialed immediately
[2-7]xx: A regular 3 digit extension that does not
start with 9 or 8 or 1 is dialed immediately
[2-7]xxx: A regular 4 digit extension that does not
start with 9 or 8 or 1 is dialed immediately
[8]xxx: A 3 digit extension prefixed with an 8 (routes
calls directly to voicemail of extension xxx)
[8]xxxx: A 4 digit extension prefixed with an 8
(routes calls directly to voicemail of extension xxxx)
T: Refers to the Dialing Timeout.
Using ‘;’ separator: An ‘x’ in the pattern indicates any
digit. ‘;’ separates between patterns.
For example: '10x;05xxxxxxxx;4xxx'.
In this example, three patterns are defined. A number
that starts with 10 is terminated after the third digit, and
so on. If the user dials a number that does not match any
pattern, the number is terminated using the timeout or
when the user presses the pound (‘#’) key.
voip/signalling/sip/number_rules
Version 2.0.0
This parameter works in conjunction with the parameter
voip/signalling/sip/digit_map and enables translation of
specific patterns to specific SIP destination addresses. An ‘x’
represents any dialed digit. Each backslash at the right side
of the ‘=’ represents one of the dialed digits. Rules are
separated by the character ';'.
The valid value can be up to 256 characters.
For example: '4xxx=Line_\\\@10.1.2.3'
This rule issues a call to 10.1.2.3 with the SIP ID of Line_
followed by the last three digits of the dialed number.
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19. Media Settings
Media Settings
The procedures for configuring media settings are described below.
19.1
Configuring Media Streaming
The Configuring Media Streaming feature can be configured using the Configuration File
parameters as described in the table below.
Table 19-1: Media Streaming Parameters
Parameter
Description
voip/media/rtp_mute_on_hold
Mute sending RTP packets to remote in HOLD state.
[0] - Disabled. RTP packets are sent to remote end
when in HOLD state.
[1] - Enabled (default). RTP packets are not sent to
remote end when in HOLD state.
voip/media/allow_multiple_rtp
Defines whether to allow multiple RTP streams from
different remote ends to be played toward the phone in
a single call.
[0] Disable (default)
[1] Enable
voip/media/ignore_rfc_2833_packets
Defines whether to ignore playing DTMF when
RFC2833 arrives from the network.
[0] Disable
[1] Enable (default)
/voip/media/broken_connection_detection
If enabled an active call will be automatically
disconnected if no RTP packet is received within predefined time.
/voip/media/broken_connection_timeout
If no RTP packet arrives for an active call within this
timeout (in seconds), the connection will be considered
broken and the call will be disconnected.
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19.2
Configuring RTP Port Range and Payload Type
RTP Port Range and Payload Type can be configured using the Web or Configuration File,
as described below.
To configure the RTP Port Range and Payload Type:
Using the Web:
1.
Access the ‘Media Streaming’ page (Configuration tab > Voice Over IP menu >
Media Streaming).
Figure 19-1: Media Streaming Web Fields
2.
Configure the RTP Port Range and Payload Type parameters according to the
table below, and then click Submit.
Using the Configuration file:
Table 19-2: RTP Port Range and Payload Type Parameters
Parameter
Description
DTMF Relay RFC 2833 Payload Type
[voip/media/dtmf_payload]
Defines the RTP payload type used for RFC 2833 DTMF
relay packets.
The valid range is 96 to 127. The default value is 101.
RTP Port Range
[voip/media/media_port]
Defines the starting port range for Real Time Protocol
(RTP) voice transport.
The valid range is 1024 to 65535. The default value is
4000.
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19. Media Settings
Configuring RTP QoS
RTP QoS can be configured using the Web or Configuration File, as described below.
To configure RTP QoS:
Using the Web:
1.
Access the ‘Media Streaming’ page (Configuration tab > Voice Over IP menu >
Media Streaming).
Figure 19-2: QoS Web Fields
2.
Configure the RTP QoS parameters according to the table below, and then click
Submit.
Using the Configuration file:
Table 19-3: RTP QoS Parameters
Parameter
Type of Service (ToS)
[voip/media/media_tos]
Version 2.0.0
Description
QoS in hexadecimal format. This is a part of the IP header
that defines the type of routing service to tag outgoing voice
packets originated from the phone. It informs routers that this
packet must receive a specific QoS. The default value is
0xb8.
Values can be set in decimal (e.g. 184) or hexadecimal (e.g.
0xb8).
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19.4
Configuring Codecs
Codecs can be configured using the Web or Configuration File, as described below.
To define the Codecs:
Using the Web:
1.
Access the ‘Media Streaming’ page (Configuration tab > Voice Over IP menu >
Media Streaming).
Figure 19-3: Codecs Web Fields
2.
Configure the Codecs parameters according to the table below, and then click
Submit.
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Using the Configuration File:
Table 19-4: Codec Parameters
Parameter
Description
voip/codec/codec_info/%d/enabled
Determines the codecs that you want to implement and
their priority. Up to five codecs can be configured, where
the first codec (i.e., voip/codec/0/…) has the highest
priority. To make a call, at least one codec must be
configured. In addition, for best performance it is
recommended to select as many codecs as possible.
When you start a call to a remote party, your available
codecs are compared with the remote party's to
determine the codec to use. If there is no codec that both
parties have made available, the call attempt fails. Note
that if more than one codec is common to both parties,
you cannot force which of the common codecs are used
by the remote party's client. If you do wish to force the
use of a specific codec, configure the list with only that
specific codec.
The %d variable stands for the priority:
[0] - Disabled
[1] (default) - Enabled
Codec Type
[voip/codec/codec_info /%d/name]
Name of the codec. The variable %d depicts the index
number of the codec entry and its priority, where the first
codec (i.e. voip/codec/codec_info/0/name=…) has the
highest priority. The valid codec parameters are:
[G722] G.722
[PCMA] G.711 A-Law
[PCMU] G.711 Mu-Law
[G729] G.729
[G723] G.723
[RTA] (Narrowband Microsoft RT Audio)
[RTA_WB] (Wideband Microsoft RT Audio)
For example, voip/codec/codec_info/0/name=G722.
Note: Specific codecs require specific firmware files. For
more information, refer to the Release Notes.
Packetization Time
[voip/codec/codec_info /%d/ptime]
Packetization time - length of the digital voice segment
that each packet holds.
The default is 20 millisecond packets, excluding G.723
which is 30 millisecond packets.
G.723 Bitrate
[voip/codec/g723_bitrate]
Low or high bit rate for G.723.
[LOW] Low
[HIGH] High (default)
[voip/codec/g722_bitrate]
G.722 bit rate.
[G722_64K] (default)
[G722_56K]
[G722_48K]
Note: Currently, only 64bps is supported.
/system/activation_keys/amr_coder
Activation key (string) required to unlock AMR coder
(relevant for supporting firmware only).
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20. Voice Settings
Voice Settings
The procedures for configuring voice settings are described below.
20.1
Configuring Jitter Buffer
Jitter Buffer can be configured using the Web or Configuration File, as described below.
To define Jitter Buffer parameters:
1.
Access the ‘Voice’ page (Configuration tab > Voice Over IP menu > Voice).
Figure 20-1: Voice Web Fields
2.
Configure the Jitter Buffer parameters according to the table below, and then click
Submit.
Using the Configuration File:
Table 20-1: Jitter Buffer Parameters
Parameter
Description
Minimum Delay
[voip/audio/jitter_buffer/min
_delay]
The initial and minimal delay of the adaptive jitter buffer
mechanism, which compensates for network problems. The
value should be set according to the expected average jitter in
the network (in milliseconds).
The valid range is 10 to 150. The default value is 35.
Optimization Factor
[voip/audio/jitter_buffer/opti
mization_factor]
The adaptation rate of the jitter buffer mechanism. Higher values
cause the jitter buffer to respond faster to increased network
jitter.
The valid range is 0 to 13. The default value is 7.
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20.2
Configuring Silence Compression
Silence Compression can be configured using the Web or Configuration File, as described
below.
To define the Silence Compression parameters:
Using the Web:
1.
Access the ‘Voice’ page (Configuration tab > Voice Over IP menu > Voice).
Figure 20-2: Silence Compression Web Fields
2.
Configure Silence Compression parameters according to the table below, and
then click Submit.
Using the Configuration File:
Table 20-2: Silence Compression Parameters
Parameter
Description
Enable Silence Compression
[voip/audio/silence_compression/enabled]
Enables silence compression for reducing network
bandwidth consumption.
[0] Disable (default)
[1] Enable
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20. Voice Settings
Configuring Echo Cancellation
Echo Cancellation can be configured using the Configuration File, as described below.
Table 20-3: Echo Cancellation Parameter
Parameter
Description
voip/audio/echo_cancellation/enabled
Enables echo cancellation.
[0] - Disable
[1] (default) – Enable
Note: Disabling echo cancellation should be done for
testing purposes only.
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21. Lines/SIP Accounts Settings
Lines/SIP Accounts Settings
This section shows how to configure lines.
21.1
Configuring Lines/SIP Accounts using Phone
Interface
Lines/SIP Accounts can be configured on the 310HD, 320HD and 420HD IP Phones as
shown below.
21.1.1 310HD IP Phone
The SIP Accounts submenu allows you to configure parameters related to the 310HD IP
Phone’s SIP account.
To configure the Lines/SIP Accounts using phone interface:
1.
Access the SIP Accounts screen (MENU key > Administration menu > SIP
Accounts).
The following screen appears:
2.
Define the SIP proxy IP address.
3.
Select the SIP Proxy Addr. option (using the
and
press the ENTER key; the following screen appears:
4.
Select one of the following options (using the
and
Navigation keys), and then
press the ENTER key (an asterisk appears alongside the selected option):
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Disabled: Disables the SIP proxy
•
Enabled: Enables the SIP proxy
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Navigation keys), and then
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5.
Press the ENTER key again; you are prompted to enter the IP address of the proxy
server:
6.
Enter the IP address, and then press the ENTER key; you are returned to the SIP
Accounts options.
7.
Define the remaining SIP proxy parameters, by selecting (using the
Navigation keys) the required parameter:
•
and
User ID
•
Display Name: Caller ID
•
Auth. Name: Authentication name
•
Auth. Password: Authentication password
For each parameter, once you have entered the required value, press the ENTER key.
8.
Select the Save option (using the
and
Navigation keys), and then press the
ENTER key; your SIP account settings are saved.
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21.1.2 320HD IP Phone
You need to configure parameters related to the phone’s SIP account. This also allows you
to set up the phone with up to four lines, each with its own extension phone number. These
lines correspond to the LINE keys on your phone. For each line you need to configure a
SIP account.
To configure Lines/SIP Accounts using the phone interface:
1.
Access the SIP Accounts screen (MENU key > Administration menu > SIP
Accounts submenu).
The following screen appears:
2.
To add a new line extension (SIP account), press the Add softkey (otherwise, to
define an existing SIP account, skip to Step 3); the following screen appears:
3.
Choose (using the
and
Navigation keys) the line, and then press the Select
softkey; the SIP Details screen appears:
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4.
Choose the required SIP parameter, and then press the Select softkey to define it:
•
"User ID"
•
"Display Name"
•
"Authentication Name"
•
"Authentication Password"
•
"SIP Proxy Address" - SIP proxy server’s address
5.
After each parameter setting, press the Save softkey to save the setting.
6.
When you have completed all the SIP account settings, press the Back softkey until
you are prompted that the phone is about to reboot itself. The phone reboots
automatically and the warning message 'Phone is rebooting to apply new settings' is
indicated.
21.1.3 420HD IP Phone
Configuration of Lines/SIP Accounts on the 420HD IP phone is identical to that on the
320HD IP phone.
To configure Lines/SIP Accounts using the phone interface:
1.
Access the SIP Accounts screen (MENU key > Administration menu > SIP
Accounts submenu).
The following screen appears:
2.
To add a new line extension (SIP account), press the Add softkey (or skip to Step 3 to
define an existing SIP account).
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3.
Press the navigation control's lower rim to navigate to the added line, and then press
the Select softkey; the SIP Details screen appears.
4.
Choose the required SIP parameter, and then press the Select softkey to define it:
•
"User ID"
•
"Display Name"
•
"Authentication Name"
•
"Authentication Password"
•
"SIP Proxy Address" - SIP proxy server’s address
5.
After each parameter setting, press the Save softkey to save the setting.
6.
When you have completed all the SIP account settings, press the Back softkey until
you are prompted that the phone is about to reboot itself. The phone reboots
automatically and a warning message 'Phone is rebooting to apply new settings' is
indicated.
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21.2
Configuring Lines using Web and Configuration File
Before you can make a call on your phone, you must configure a phone line. Lines can be
configured using the Web or Configuration File, as described below.
To configure lines:
Using the Web:
1.
Access the ‘Line Settings’ page (Configuration tab > Voice Over IP menu >
Line Settings).
Figure 21-1: Voice Web Fields
2.
Configure the Line settings according to the parameters in the table below, and
then click Submit.
Using the Configuration File:
Table 21-11: Line Parameters
Parameter
Description
Note: The value %d refers to the line number.
Line Activate
[voip/line/%d/enabled]
Enables the line.
[0] Disable
[1] Enable
The default for Line-0 is 1 (enabled); for the other lines it is
0 (disabled).
Line User ID
[voip/line/%d/id]
Lines VoIP user’s ID for identification to initiate and accept
calls.
The user’s ID can be up to 30 characters.
Line Display Name
[voip/line/%d/description]
Arbitrary name to intuitively identify the line and that is
displayed to remote parties as your caller ID.
Line Authentication User Name
voip/line/%d/auth_name
User name provided to you from the VoIP service provider.
This is used when sending a response to Unauthorized or
Proxy Authentication Requested (401/407).
The authentication name can be up to 35 characters.
Line Authentication Password
voip/line/%d/auth_password
Password provided to you from the VoIP Service Provider.
This is used when sending a response to Unauthorized or
Proxy Authentication Requested (401/407).
The authentication password can be up to 35 characters.
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Parameter
/voip/line/%d/extension_display
Description
Set the string that is displayed for local extension.
If not set, the local extension display will be the user ID
(self-number).
Note: You can activate DnD per phone line. See Section 22.7 on page 142.
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22. Supplementary Services
Supplementary Services
You can configure various supplementary services supported by your phone such as Call
Waiting, Call Forwarding, Three-way Conferencing, and Message Waiting Indication
(MWI).
22.1
Selecting the Application Server
By default, the phone is set for a generic application server. However, you can select a
specific third-party application server as described below. See Appendix A on page 267 for
more information.
Note: Configuration of specific supplementary services depends on the thirdparty application server used in your organization.
Selecting the Application Server can be done by using the Web or Configuration File, as
described below.
To select the application server:
Using the Web:
1.
Access the ‘Services’ page (Configuration tab > Voice Over IP menu >
Services).
Figure 20-1: Services Page
2.
Select the desired application server according to the table below, and then click
Submit.
Using the Configuration file:
Table 20-1: General Supplementary Services Parameters
Parameter
Description
Type
[voip/services/application_server_type]
Version 2.0.0
Defines the type of the Application server to
which the device is registered.
[Generic] Generic (default)
[Asterisk] Asterisk
[BSFT] BroadSoft
[Coral] Coral
[Metaswitch] Metaswitch
[FreeSWITCH] FreeSWITCH
Note: In addition to the general
supplementary services parameters,
parameters (if any) unique to the selected
Application server become applicable.
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Parameter
Description
/system/current_user_presence_status/enabled
Enables the “presence” feature.
DND SK will be replaced by “status”; phone
will be show and publish the presence state.
22.2
Configuring Call Waiting
Call Waiting can be configured using the Web or Configuration File, as described below.
To configure call waiting:
Using the Web:
1.
Access the ‘Services’ page (Configuration tab > Voice Over IP menu >
Services).
Figure 22-1: Call Waiting Web Fields
2.
Configure the Call Waiting parameters according to the table below, and then
click Submit.
Using the Configuration file:
Table 22-1: Call Waiting Parameters
Parameter
Description
Activate
[voip/services/call_waiting/enabled]
Enables the Call Waiting feature.
[0] Disable
[1] Enable (default)
Call Waiting SIP Reply
[voip/services/call_waiting/sip_reply]
Determines the SIP response that is sent when another
call arrives while a call is in progress:
[RINGING] Ringing - 180 Ringing
[QUEUED] Queued (default) - 182 Queued
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22. Supplementary Services
Configuring Call Forwarding
Call Forwarding can be configured using the Web or Configuration File, as described
below.
To configure call forwarding:
Using the Web:
1.
Access the ‘Services’ page (Configuration tab > Voice Over IP menu >
Services).
Figure 22-2: Call Forwarding Web Fields
2.
Configure the Call Forwarding parameters according to the table below, and then
click Submit.
Using the Configuration file:
Table 22-2: Call Forward Parameters
Parameter
Description
voip/line/%d/call_forward/enabled
Enables the Call Forward feature.
[0] Disable
[1] Enable (default)
Call Forward Type
[voip/line/%d/call_forward/type]
Determines the condition upon which incoming
calls are forwarded to another destination:
[Unconditional] Unconditional - incoming
calls are forwarded independently of the
status of the line.
[Busy] Busy - incoming calls are forwarded
only if the phone is busy.
[No_Reply] No Reply (default) - incoming
calls are forwarded only if the phone does not
answer before a user-defined timeout.
Forward on No Reply Timeout
[voip/line/%d/call_forward/timeout]
If calls are forwarded when the condition is NoReply, then this parameter defines the time (in
seconds) after which incoming calls are
forwarded when this is no reply.
The valid range is 0 to 7200. The default value is
6.
Forward Destination
[voip/line/%d/call_forward/destination]
The destination to which the call is directed when
call forward is activated.
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Parameter
Activate
[voip/line/%d/call_forward/active]
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Description
Activates call forwarding, if it has been enabled
(using the parameter
voip/line/%d/call_forward/enabled).
[0] (default) - Disable
[1] Enable
Note: Call forwarding can also be activated using
the LCD screen (common).
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22. Supplementary Services
Configuring Three-way Conferencing
Three-way conferencing can be configured using the Web or Configuration File, as
described below.
To configure three-way conferencing:
Using the Web:
1.
Access the ‘Services’ page (Configuration tab > Voice Over IP menu >
Services).
Figure 22-3: Three-way Conferencing Web Fields
2.
Configure the Three-way conferencing parameters according to the table below,
and then click Submit.
Using the Configuration file:
Table 22-3: Three-way Conferencing Parameters
Parameter
Mode
[/voip/services/conference/mode]
Remote Conference Media Server
[/voip/services/conference/conf_ms_addr]
Description
Sets the conference mode (when establishing 3Way Conference).
[LOCAL] = phone will establish the conference by
itself.
[REMOTE] = phone will use remote media server
for establishing the conference.
Media server for establishing remote conference.
Relevant only in case the conference mode is
REMOTE.
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22.5
Configuring Automatic Dialing
Automatic Dialing can be configured using the Web or Configuration File, as described
below.
To define the Automatic Dialing parameters:
Using the Web:
1.
Access the ‘Dialing’ page (Configuration tab > Voice Over IP menu > Dialing).
Figure 22-4: Automatic Dialing Web Fields
2.
Configure dialing options according to the parameters in the table below, and
then click Submit.
Using the Configuration File:
Table 22-4: Automatic Dialing Parameters
Parameter
Description
Activate
[voip/dialing/auto_dialing/enabled]
Determines whether automatic dialing is enabled (i.e.,
phone number is automatically dialed when you offhook the phone).
[0] Disable (default)
[1] Enable
Timeout
[voip/dialing/auto_dialing/timeout]
Timeout (in seconds) before automatic dialing occurs
after the phone is off-hooked. When set to 0,
automatic dialing is performed immediately.
The valid range is 0 to 120. The default value is 15.
Destination Phone Number
[voip/dialing/auto_dialing/destination]
The number that is automatically dialed when the
phone is off-hooked.
The valid value can be up to 32 characters.
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22.6
22. Supplementary Services
Configuring Automatic Answer
The Automatic Answer feature is configured using the Configuration File parameters
described in the table below.
Table 22-5: Automatic Answer Parameters
Parameter
Description
voip/auto_answer/enabled
Enables the Automatic Answering feature.
[0] Disable (default)
[1] Enable
When this parameter is enabled and an incoming SIP INVITE
message is received containing information that informs the
IP phone to automatically answer the call, the phone answers
the call immediately or after a timeout, depending on the autoanswer type specified in the INVITE message:
Phone answers after a timeout: The phone automatically
answers the call after a timeout if the INVITE message
includes a SIP Call-Info header with a tag value, "answerafter=" set to a number representing the timeout. During
the timeout interval, the phone rings normally. If the call is
answered or rejected during this interval, then the
automatic answering mechanism is not used. However, if
the phone is left to ring throughout the timeout interval, it
automatically answers the call once this timeout expires.
Phone answers immediately: The phone answers the
call immediately in any of the following cases:
If the SIP Alert-Info header contains the tag value, "ring
answer".
If the SIP Alert-Info header contains the tag value,
"info=alert-autoanswer".
Notes:
If the SIP Call-Info header includes all the above answer
types or any two different types (i.e., “answer-after=”, “ring
answer”, and “alert-autoanswer”), the answer-after= type
takes precedence.
If there is an existing call when an INVITE message for
automatic answer is received, the existing call is
automatically put on hold.
voip/talk_event/enabled
Enables the "talk" event feature.
[0] Disable (default)
[1] Enable
The phone automatically answers an incoming call if it
receives a SIP NOTIFY message with the "talk" event. If a call
is already in progress, the call is put on hold and the incoming
call is answered.
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22.7
Configuring Do Not Disturb
Do not Disturb (DnD) can be configured using the Web or Configuration File, as described
below.
Note: For the 320HD IP Phone, DnD can be activated per phone line.
To configure DnD:
Using the Web:
1.
Access the ‘Services’ page (Configuration tab > Voice Over IP menu >
Services).
Figure 22-5: DnD Web Fields
2.
Select the desired application server according to the table below, and then click
Submit.
Using the Configuration file:
Table 22-6: Do Not Disturb (Services) Parameters
Parameter
Description
Activate
[voip/services/do_not_disturb/enabled]
Enables the DnD feature.
[0] Disable
[1] Enable (default)
Line Number
Defines the extension line for which you want to
activate DnD
voip/line/%d/do_not_disturb/activated
Activates the DnD feature per phone line, if DnD is
enabled using the parameter
voip/services/do_not_disturb/enabled.
[0] - Activate (default)
[1] - Deactivate
Note: DnD can also be activated using the LCD
screen interface (more common).
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22.8
22. Supplementary Services
Configuring Call Pickup
Since the Call Pickup feature is relevant only when Busy Lamp Field (BLF) is activated, the
call pickup parameters appear as BLF related parameters.
To configure Call Pickup :
Using the Web:
1.
Access the ‘Services’ page (Configuration tab > Voice Over IP menu >
Services).
Figure 22-6: Call Pick Up Web Fields
2.
3.
Under the BLF Support group, ensure that the ‘Activate’ field is set to “Enable”.
Configure Call Pickup according to the parameters in the table below, and then
click Submit.
Table 22-7: Call Pickup Parameters
Parameter
Call Pick Up
[/voip/services/call_pickup/enabled]
[voip/services/call_pickup/access_code]
Description
Allows call pickup by pressing on the relevant BLF
key when the state of the remote phone is “ringing”.
[0] Disable
[1] Enable
The phone supports the Call Pick-Up feature,
whereby the user can answer someone else's
telephone call by pressing a user-defined sequence
of phone keys. When the user dials the user-defined
digits plus the other phone number, the incoming
call from the other phone is forwarded to the user's
phone.
The default key sequence is **.
For example, to pick up a call for extension 5000
dial **5000.
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22.9
Configuring Message Waiting Indication
Message Waiting Indication (MWI) can be configured using the Web or Configuration File,
as described below.
To configure MWI:
Using the Web:
1.
Access the ‘Services’ page (Configuration tab > Voice Over IP menu >
Services).
Figure 22-7: MWI Web Fields
2.
Configure the MWI parameters according to the table below, and then click
Submit.
Using the Configuration file:
Table 22-8: Message Waiting Indication Parameters
Parameter
Description
Activate
[voip/services/msg_waiting_ind/enabled]
Enables the MWI feature.
[0] Disable
[1] Enable (default)
Subscribe To MWI
[voip/services/msg_waiting_ind/subscribe]
Determines whether the phone registers to
an MWI server.
[0] Disable (default)
[1] Enable
MWI Server Port
The port number of the MWI server.
The valid range is 1024 to 65535. The
default value is 5060.
[voip/services/msg_waiting_ind/subscribe_port]
MWI Server IP Address or Host Name
[voip/services/msg_waiting_ind/subscribe_address]
MWI Subscribe Expiry Time
[voip/services/msg_waiting_ind/expiraition_timeout]
Voice Mail Number
[voip/services/msg_waiting_ind/voice_mail_number]
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The IP address or host name of the MWI
server.
The default value is 0.0.0.0.
The interval between the MWI Subscribe
messages.
The valid range is 0 to 86400. The default
value is 3600.
Defines the extension number for accessing
your voice mail messages.
The valid value is up to 64 characters.
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Parameter
Description
voip/services/msg_waiting_ind/always_send_port
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If the SIP port is the default port (i.e. 5060),
then remove it from the Request-URI of the
MWI SUBSCRIBE.
[0] Disable
[1] Enable (default)
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22.10 Configuring Busy Lamp Field
Busy Lamp Field (BLF) can be configured using the Web or Configuration File, as
described below.
Note: BLF is only applicable to the 320HD IP Phone.
To configure BLF:
Using the Web:
1.
Access the ‘Services’ page (Configuration tab > Voice Over IP menu >
Services).
Figure 22-8: BLF Support Web Fields
2.
Configure the BLF parameters according to the table below, and then click
Submit.
Using the Configuration file:
Table 22-9: BLF Parameters
Parameter
Description
Activate
[voip/services/busy_lamp_field/enabled]
Enables the BLF feature:
[0] Disable
[1] Enable (default)
BLF Subscription Period
[voip/services/busy_lamp_field/subscripti
on_period]
The interval between BLF and SIP SUBSCRIBE
messages.
The valid range is 0 to 86400. The default value is
3600.
User Resource List
[voip/services/busy_lamp_field/uri]
The user resource list. This must be the username
(not the domain name). For example, if the URI
resource list is [email protected], then only the
value mylist must be entered.
The valid value is up to 64 characters.
Note: This parameter is applicable only if the
parameter
voip/services/application_server_type is set to
BSFT.
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Parameter
Description
Use Registrar as Application Server Address
[voip/services/busy_lamp_field/applicatio
n_server/use_registrar]
Determines whether to use the registrar as the
application server address. When enabled, there is
no need to configure the application server
address or domain name separately.
[0] Disable (default)
[1] Enable
Note: This parameter is applicable only if the
parameter
voip/services/application_server_type is set to
BSFT.
Use Registrar as Application Server Address
[voip/services/busy_lamp_field/applicatio
n_server/addr]
The IP address or host name of the application
server.
The valid value is up to 64 characters. The default
value is 0.0.0.0.
Note: This parameter is applicable only if the
parameter
voip/services/application_server_type is set to
BSFT and
voip/services/busy_lamp_field/application_serv
er/use_registrar is set to 0.
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22.11 Configuring Advice of Charge
Advice of Charge (AOC) can be configured using the Web or Configuration File, as
described below.
To configure AOC:
Using the Web:
1.
Access the ‘Services’ page (Configuration tab > Voice Over IP menu >
Services).
Figure 22-9: AOC Support Web Fields
2.
Configure the AOC parameters according to the table below, and then click
Submit.
Using the Configuration File:
Table 22-10: AOC Parameters
Parameter
Enable
[system/aoc/enabled]
Currency
[system/aoc/currency]
Ratio
[system/aoc/ratio]
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Description
Enables the “advice of charge” feature.
[0] Disable
[1] Enable
Sets the desired currency for AOC display.
The string represents the currency name. (e.g., USD , EUR ,
NIS , etc..)
Sets the conversion ratio from the local currency.
The string represents the ratio between the base currency to the
set currency with a decimal point),( e.g., : 3.8, 4.95 , 1).
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22.12 Enabling Electronic Hook Switch
Electronic Hook Switch (EHS) can be enabled using the Web or Configuration File, as
described below.
To enable EHS:
Using the Web:
1.
Access the ‘Services’ page (Configuration tab > Voice Over IP menu >
Services).
Figure 22-10: EHS Web Fields
2.
Configure the EHS parameters according to the table below, and then click
Submit.
Using the Configuration file:
Table 22-11: EHS Parameters
Parameter
Description
Electronic Hook Switch
[voip/services/electronic_hook_switch/enabled]
Enables the EHS DHSG feature.
[0] Disable (default)
[1] Enable
DHSG is the protocol that uses the special
AUX port.
Supporting headsets can be connected to
the AUX port (in addition to the regular
headset port). This allows the user to
connect and disconnect calls by pressing
the button on the headset.
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23. Volume Levels
Volume Levels
The procedures for configuring volume levels are described below.
23.1
Configuring Automatic Gain Control
Automatic Gain Control can be configured using the Web or Configuration File, as
described below.
To configure the Automatic Gain Control:
Using the Web:
1.
Access the ‘Voice’ page (Configuration tab > Voice Over IP menu > Voice).
Figure 23-1: Auto Gain Control Web Fields
2.
Configure the settings according to the parameters in the table below, and then
click Submit.
Using the Configuration File:
Table 23-1: Automatic Gain Control Parameters
Parameter
Description
Enable Automatic Gain Control
[voip/audio/gain/automatic_gain_control/enabled]
Automatic Gain Control Direction
[voip/audio/gain/automatic_gain_control/direction]
Target Energy
[voip/audio/gain/automatic_gain_control/target_energy]
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Enables the AGC. AGC automatically
adjusts the phone’s voice volume to
compensate for weak or loud signals.
[0] Disable (default)
[1] Enable
Determines whether AGC is located
before the encoder input (CTL_LOCAL)
or after the decoder output
(CTL_REMOTE).
[CTL_REMOTE] For Remote User
(default) - AGC is located after the
Decoder output
[CTL_LOCAL] For Local User - AGC
is located before the Encoder input
The required output energy (in dBm) of
the AGC.
The valid range is -31 to 0 (dB). The
default value is -19.
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Configuring Tone Volume
Tone volume can be configured using the Web or Configuration File, as described below.
To configure tone volume:
Using the Web:
1.
Access the ‘Volume Setting’ page (Configuration tab > Voice Over IP menu >
Volume Settings).
Figure 23-2: Tone Volume Web Field
2.
Configure the tones volume settings according to the parameters in the table
below, and then click Submit.
Using the Configuration file:
Table 23-2: Tone Volume Parameter
Parameter
Tone Volume
[voip/audio/gain/tone_signal_level]
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Description
Call progress tone volume. This volume can be modified
on-the-fly by pressing the phone’s VOLUME key in certain
scenarios.
The valid range is 1 to 31 (-dB). The default value is 10 (10dB).
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23. Volume Levels
Configuring Ringer Volume
The ringer volume can be configured using the Web or Configuration File, as described
below.
To configure the ringer volume:
Using the Web:
1.
Access the ‘Volume Setting page (Configuration tab > Voice Over IP menu >
Volume Settings).
Figure 23-3: Ringer Volume Web Field
2.
Configure the ringer volume settings according to the parameters in the table
below, and then click Submit.
Using the Configuration file:
Table 23-3: Ringer Volume Parameters
Parameter
Description
Ringer Volume
voip/audio/gain/ringer_signal_level
Ringing tone volume. This volume can be modified on-thefly by pressing the phone’s VOLUME key when the phone
is in idle state.
The valid range is [-31] to [31] dB
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23.4
Configuring Speakerphone (Hands Free) Volume
The speakerphone (Hands Free) volume can be configured using the Web or Configuration
File, as described below.
To configure speakerphone (Hands Free) volume:
Using the Web:
1.
Access the ‘Volume Setting’ page (Configuration tab > Voice Over IP menu >
Volume Settings).
Figure 23-4: Hands Free Web Fields
2.
3.
Configure the Volume settings according to the parameters in the table below,
and then click Submit.
Access the ‘Services’ page (Configuration tab > Voice Over IP menu > Volume
Settings).
Figure 23-5: Additional Speaker Volume Web Field
4.
Configure the speakerphone volume according to the parameters in the table
below, and then click Submit.
Table 23-13: Speakerphone (Hands Free) Parameters
Parameter
Description
Hands-free Gain Parameters
Note: Values are in decibels (dB) and represented as follows:
Negative values: use the word "minus" (e.g. =minus9db).
Positive values: use the word "plus" (e.g. =plus9db).
Decimal places: use underscore instead of period (e.g. plus19_5db).
Hands Free Digital Output
[voip/audio/gain/handsfree_digital_output_gain]
Digital output gain (in dB).
The valid range is (-32) to 31 (dB), where 32 is mute.
Hands Free Digital Input
[voip/audio/gain/handsfree_digital_input_gain]
Digital input gain (in dB).
The valid range is (-32) to 31 (dB), where 32 is mute.
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Parameter
Description
Hands Free Analog Output
[voip/audio/gain/handsfree_analog_output_gain]
Analog output gain (in dB).
Valid values: [0db], [minus1_5db],
[minus3db], [minus4_5db], [minus6db],
[minus7_5db], [minus9db],
[minus10_5db], [minus12db],
[minus13_5db], [minus15db],
[minus16_5db], [minus18db],
[minus19_5db], [minus21db],
[minus22_5db], [minus24db],
[minus25_5db], [minus27db],
[minus28_5db], [minus30db],
[minus31_5db], [minus33db],
[minus34_5db], [minus36db],
[minus37_5db], [minus39db],
[minus39db], [minus42db], [minus48db],
[minus54db], [MUTE]
Hands Free Analog Input
[voip/audio/gain/handsfree_analog_input_gain]
Analog input gain (in dB).
Valid values: [0db], [plus1_5db],
[plus3db], [plus4_5db], [plus6db],
[plus7_5db], [plus9db], [plus10_5db],
[plus12db], [plus13_5db], [plus15db],
[plus16_5db], [plus18db], [plus19_5db],
[plus21db], [plus22_5db], [plus24db],
[plus25_5db], [plus27db], [plus28_5db],
[plus30db], [plus31_5db], [plus33db],
[plus34_5db], [plus36db], [plus37_5db],
[plus39db], [plus40_5db], [plus42db],
[plus48db], [plus54db], [MUTE]
Additional Speaker Gain
[voip/audio/gain/additional_speaker_gain]
Additional parameter for speaker gain
configuration.
[0] 0dB
[1] 1dB
[2] 2dB
[3] 3dB
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23.5
Configuring Handset Volume
The handset volume can be configured using the Web or Configuration File, as described
below.
To configure handset volume:
Using the Web:
1.
Access the ‘Volume Setting page (Configuration tab > Voice Over IP menu >
Volume Settings).
Figure 23-6: Handset Volume Settings Page
2.
Configure the handset volume settings according to the parameters in the table
below, and then click Submit.
Using the Configuration File:
Table 23-14: Handset Gain Parameters
Parameter
Description
Handset Gain Parameters
Note: Values are in decibels (dB) and represented as follows:
Negative values: use the word "minus" (e.g. =minus9db).
Positive values: use the word "plus" (e.g. =plus9db).
Decimal places: use underscore instead of period (e.g. plus19_5db).
Handset Digital Output
[voip/audio/gain/handset_digital_output_gain]
Digital output gain (in dB).
The valid range is (-32) to 31 (dB), where 32 is mute.
Handset Digital Input
[voip/audio/gain/handset_digital_input_gain]
Digital input gain (in dB).
The valid range is (-32) to 31 (dB), where 32 is mute.
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Parameter
Description
Handset Analog Output
[voip/audio/gain/handset_analog_output_gain]
Analog output gain (in dB).
Valid values: [0db], [minus1_5db],
[minus3db], [minus4_5db], [minus6db],
[minus7_5db], [minus9db],
[minus10_5db], [minus12db],
[minus13_5db], [minus15db],
[minus16_5db], [minus18db],
[minus19_5db], [minus21db],
[minus22_5db], [minus24db],
[minus25_5db], [minus27db],
[minus28_5db], [minus30db],
[minus31_5db], [minus33db],
[minus34_5db], [minus36db],
[minus37_5db], [minus39db],
[minus39db], [minus42db], [minus48db],
[minus54db], [MUTE]
Handset Analog Input
[voip/audio/gain/handset_analog_input_gain]
Analog input gain (in dB).
Valid values: [0db], [plus1_5db],
[plus3db], [plus4_5db], [plus6db],
[plus7_5db], [plus9db], [plus10_5db],
[plus12db], [plus13_5db], [plus15db],
[plus16_5db], [plus18db], [plus19_5db],
[plus21db], [plus22_5db], [plus24db],
[plus25_5db], [plus27db], [plus28_5db],
[plus30db], [plus31_5db], [plus33db],
[plus34_5db], [plus36db], [plus37_5db],
[plus39db], [plus40_5db], [plus42db],
[plus48db], [plus54db], [MUTE]
Handset Analog Sidetone Gain
[voip/audio/gain/handset_analog_sidetone_gain]
Analog side tone gain (in db).
Valid values: [minus9db], [minus12db]
(default), [minus15db], [minus18db],
[minus21db], [minus24db], [minus27db],
[MUTE]
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23.6
Configuring Headset Volume
The headset volume can be configured using the Web or Configuration File, as described
below.
To configure headset volume:
Using the Web:
1.
Access the ‘Volume Setting page (Configuration tab > Voice Over IP menu >
Volume Settings).
Figure 23-7: Headset Web Fields
2.
Configure the headset volume settings according to the parameters in the table
below, and then click Submit.
Using the Configuration File:
Table 23-4: Headset Gain Parameters
Parameter
Description
Headset Gain Parameters
Note: Values are in decibels (dB) and represented as follows:
Negative values: use the word "minus" (e.g. =minus9db).
Positive values: use the word "plus" (e.g. =plus9db).
Decimal places: use underscore instead of period (e.g. plus19_5db).
Headset Digital Output
[voip/audio/gain/headset_digital_output_gain]
Digital output gain (in dB).
The valid range is (-32) to 31 (dB), where 32 is mute.
Headset Digital Input
[voip/audio/gain/headset_digital_input_gain]
Digital input gain (in dB).
The valid range is (-32) to 31 (dB), where 32 is mute.
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Parameter
Description
Headset Analog Output
[voip/audio/gain/headset_analog_output_gain]
Analog output gain (in dB).
Valid values: [0db], [minus1_5db],
[minus3db], [minus4_5db], [minus6db],
[minus7_5db], [minus9db],
[minus10_5db], [minus12db],
[minus13_5db], [minus15db],
[minus16_5db], [minus18db],
[minus19_5db], [minus21db],
[minus22_5db], [minus24db],
[minus25_5db], [minus27db],
[minus28_5db], [minus30db],
[minus31_5db], [minus33db],
[minus34_5db], [minus36db],
[minus37_5db], [minus39db],
[minus39db], [minus42db], [minus48db],
[minus54db], [MUTE]
Headset Analog Input
[voip/audio/gain/headset_analog_input_gain]
Analog input gain (in dB).
Valid values: [0db], [plus1_5db],
[plus3db], [plus4_5db], [plus6db],
[plus7_5db], [plus9db], [plus10_5db],
[plus12db], [plus13_5db], [plus15db],
[plus16_5db], [plus18db], [plus19_5db],
[plus21db], [plus22_5db], [plus24db],
[plus25_5db], [plus27db], [plus28_5db],
[plus30db], [plus31_5db], [plus33db],
[plus34_5db], [plus36db], [plus37_5db],
[plus39db], [plus40_5db], [plus42db],
[plus48db], [plus54db], [MUTE]
Headset Analog Sidetone Gain
[voip/audio/gain/headset_analog_sidetone_gain]
Analog side tone gain (in db).
Valid values: [minus9db], [minus12db],
[minus15db], [minus18db], [minus21db],
[minus24db], [minus27db], [MUTE]
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Part VI
Advanced Phone
Settings
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24. Locking and Unlocking the IP Phone
Locking and Unlocking the IP Phone
If your phone is deployed with AudioCodes solution for Microsoft Lync Server 2010, you
can lock your phone to prevent anyone from using it (such as making and receiving calls).
Incoming calls can be answered only if you provide the correct unlock password; outgoing
calls can only be made if they are configured as emergency calls (by your administrator).
Notes:
24.1
•
This feature is applicable only when your phone is deployed with
AudioCodes solution for Microsoft Lync Server 2010 and is enabled
by your system administrator. For more information, refer to the
Administrator's Manual.
•
To obtain the unlocking password, contact your system
administrator.
•
When the phone is locked, the emergency number, defined by your
system administrator, can be dialed.
Locking and Unlocking the 310HD IP Phone
The procedures below describe how to lock and then unlock your 310HD IP phone.
To lock the 310HD IP Phone:
Press the star key for a few seconds.
If the phone is locked and you receive an incoming call, you are prompted to enter the
unlock password in order to establish the call, as shown below:
To unlock the 310HD IP Phone:
1.
Press the star key for a few seconds; you are prompted to enter the unlock password:
2.
Enter the password, and then press the ENTER key.
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24.2
Locking and Unlocking the 320HD IP Phone
The procedures below describe how to lock and then unlock your 320HD IP phone.
24.2.1 Single-Line Mode
The procedures below describe how to lock and then unlock your phone in single-line
mode.
To lock your 320HD IP Phone:
Press the star (*) key for a few seconds (about 3 seconds); the phone locks and the
lock icon is displayed on the screen, as shown below:
Lock Icon
To unlock your 320HD IP Phone:
1.
Press the star key until you are prompted for the unlock password:
2.
Enter the password, and then press the OK softkey; the phone is unlocked and the
lock icon no longer appears on the screen.
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24.2.2 Multi-Line Mode
When your phone is configured with multiple lines (extensions), you can perform a lock per
extension or for all extensions, as described in the procedure below.
To lock your 320HD IP Phone:
1.
Press the star key until the Lock screen is displayed:
2.
Choose an extension (or All Extensions to lock all extensions) that you want to lock,
using the
and
Navigation keys, and then press the Lock softkey; the extension
is locked and the lock icon is displayed with the extension number in the idle screen,
as shown below:
Lock Icon
To unlock your 320HD IP Phone:
1.
Press the star key until the Lock screen is displayed:
2.
Choose an extension (or All Extensions to unlock all extensions) that you want to
unlock, using the
and
Navigation keys, and then press the Unlock softkey; you
are prompted to enter the unlock password, as shown below:
3.
Enter the password for unlocking the phone, and then press the OK softkey; the
extension(s) is unlocked and the lock icon no longer appears on the screen.
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24.3
Locking and Unlocking the 420HD IP Phone
The procedures below describe how to lock and then unlock your 420HD IP phone.
24.3.1 Single-Line Mode
The procedures below describe how to lock and then unlock your phone when it's in singleline mode.
To lock your 420HD IP Phone:
Press the star (*) key for approximately 3 seconds; the phone locks and the lock icon
is displayed on the screen:
To unlock your 420HD IP Phone:
1.
Press the star key until you are prompted for the unlock password:
2.
Enter the password, and then press the OK softkey; the phone is unlocked and the
lock icon no longer appears on the screen.
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24.3.2 Two-Line Mode
When your phone is configured with two lines (extensions), you can perform a lock per
extension or for all extensions, as described in the procedure below.
To lock your 420HD IP Phone:
1.
Press the star key until the Lock screen is displayed:
2.
Press the navigation control's upper or lower rim to navigate to an extension that you
want to lock (or All Extensions to lock all extensions) and then press the Lock
softkey; the extension is locked and the lock icon is displayed with the extension
number in the idle screen, as shown below:
To unlock your 420HD IP Phone:
1.
Press the star key until the Lock screen is displayed:
2.
Press the navigation control's upper or lower rim to navigate to an extension that you
want to unlock (or All Extensions to unlock all extensions) and then press the Unlock
softkey; you are prompted to enter the unlock password, as shown below:
3.
Enter the password for unlocking the phone and then press the OK softkey; the
extension(s) is unlocked and the lock icon no longer appears on the screen.
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24.4
Using the Configuration File
The phone’s Lock / Unlock feature can be configured using the configuration file
parameters below:
Table 24-1: Phone Lock/Unlock Parameters
Parameter
Description
/system/lock/[0-0]/enabled
Enables the “phone lock” feature, relevant for
supporting servers only.
/system/server_lock_error_code
Refers to specific error code from the server as the
“lock” operation trigger.
The error code should be 4XX SIP error code.
/system/lock/[0-0]/allow_incoming_calls
Allows incoming calls when the phone is locked.
If allowed, the user will need to enter an unlock
password to answer an incoming call.
If not allowed, the incoming call will be automatically
rejected by the phone.
/system/action_url/lock_status
Sets the action URI to get lock status from phone.
/system/action_url/lock
Sets the action URI to lock the phone.
/system/action_url/unlock
Sets the action URI to unlock the phone.
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25. Phone Directory
Phone Directory
This section describes how to configure the phone directory.
25.1
Corporate Directory
The Corporate Directory can be configured using the Web or Configuration File, as
described below.
25.1.1 LDAP-based Corporate Directory
This section describes how to configure Lightweight Directory Access Protocol (LDAP),
which is an application protocol for accessing and maintaining distributed directory
information services over an IP network. It is fully described under RFC 4510.
To configure LDAP:
Using the Web:
1.
Access the ‘LDAP’ page (Configuration tab > Advanced Applications >
LDAP). If LDAP is set to “Enable”, more configuration parameters appear.
Figure 25-1: LDAP Web Fields
2.
Configure the LDAP settings according to the parameters in the table below, and
then click Submit.
Using the Configuration File:
Table 25-1: LDAP Parameters
Parameter Name
Active
Description
Enables or disable LDAP.
[system/ldap/enabled]
Server Address
[system/ldap/server_address]
Version 2.0.0
Defines the IP address or URL of the LDAP
server.
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Parameter Name
Port
Description
Defines the LDAP service port.
[system/ldap/port]
User Name
[system/ldap/user_name]
Password
Defines the user name used for the LDAP search
request.
Defines the password of the search requester.
[system/ldap/password]
Base
Defines the access point on the LDAP tree.
[system/ldap/base]
Name Filter
[system/ldap/name_filter]
Name Attribute
[system/ldap/name_attrs]
Number Filter
[system/ldap/number_filter]
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Specifies your search pattern for name look ups.
For example:
When you type in the following
field: (&(telephoneNumber=*)(sn=%)), the search
result includes all LDAP records, which have the
‘telephoneNumber’ field set and the ‘(“sn”->surname)’ field starting with the entered prefix.
When you type in the following field:
(|(cn=%)(sn=%)), the search result includes all
LDAP records which have the (“cn”->CommonName) OR (“sn”-->Surname) field
starting with the entered prefix.
When you type in the following field: (!(cn=%)),
the search result includes all LDAP records
which “do not” have the “cn” field starting with the
entered prefix.
Specifies the LDAP name attributes setting,
which can be used to specify the “name”
attributes of each record which is returned in the
LDAP search results.
When you type in the following field, for example,
cn sn displayName”, this requires you to specify
‘cn-->commonName’. This is the Full name of the
user, sn-->Surname, last name or family name
and “displayName” fields for each LDAP record.
Specifies your search pattern for number look
ups.
When you type in the following field, for example,
(|(telephoneNumber=%)(Mobile=%)(ipPhone=%))
, the search result is all LDAP records which have
the “telephoneNumber” OR “Mobile” OR
“ipPhone”field match the number being searched.
When you type in the following field:
(&(telephoneNumber=%)(sn=*)), the search
result is all LDAP records which have the “sn”
field set and the “telephoneNumber” match the
number being searched.
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Parameter Name
Number Attribute
[system/ldap/number_attrs]
Display Name
[system/ldap/display_name]
Max Hits
[system/ldap/max_hits]
Sort Result
[system/ldap/sorting_result]
Description
Specifies the LDAP number attributes setting,
which can be used to specify the “number”
attributes of each record which is returned in the
LDAP search results.
When you type in the following field, for example,
Mobile telephoneNumber ipPhone, you must
specify ‘Mobile’, ‘telephoneNumber’ and
‘ipPhone’ fields for each LDAP record.
Specifies the format in which the “name, e.g.
“Mike Black” of each returned search result is
displayed on the IPPHONE.
When you type in the following field, for
example:%sn, %givenName, the displayed
result returned should be “Black, Mike”.
Specifies the maximum number of entries
expected to be sent by the LDAP server (this
parameter is sent to the LDAP server).
Sorts the search result by display name on the
client side.
system/ldap/predict_text
This parameter appears in the configuration file;
however, it is currently not supported.
Search Timeout
The time out value for LDAP search (this
parameter is sent to the LDAP server).
[system/ldap/search_timeout]
system/ldap/ui/use_right_arrow_active_sea
rch
This parameter appears in the configuration file;
however, it is currently not supported.
system/ldap/lookup_incoming_call
This parameter appears in the configuration file;
however, it is currently not supported.
Call Lookup
Performs an LDAP search during call (search the
display name for a number).
[system/ldap/call_lookup]
Country Code
[system/ldap/country_code]
Area Code
[system/ldap/area_code]
Defines the country code prefix added for number
search.
Defines the area code prefix added for number
search.
system/ldap/minimal_name_search_length
Starts to perform an LDAP search after x
characters are input.
system/ldap/send_queries_while_typing
Sends an LDAP search each time the user
presses a key (all keys with both number and
letters).
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25.1.2 Loading a Text-based Corporate Directory File
The Configuration file can include a link to a user-defined Corporate Directory file, using
the provisioning/corporate_directory_uri parameter. This allows you to upload a
corporate directory to the phone.
The Corporate Directory file includes a list of contacts and their phone numbers. The file
must be a simple text file that can be created using an Excel document and saved as a
CSV file.
The syntax of the corporate directory file must be as follows:
<full name>,<office>,<home>,<mobile>
For example:
John Smith,1234,98765432,574685746
If not all phone numbers are required, the relevant field must be left empty. For example, in
the below directory entry, the home and user-defined numbers are absent:
John Smith,1234,,574685746
To configure the Corporate Directory:
Using the Web:
1.
2.
3.
4.
Prepare the TXT-based file as explained above.
Access the ‘Directory’ page (Configuration tab > Personal Settings menu >
Directory).
Click Browse and select the file to upload.
Click Load Corporate Directory.
Figure 25-2: Corporate Directory Web Field
5.
Configure the Corporate Directory settings according to the parameters in the
table below, and then click Submit.
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25. Phone Directory
Using the Configuration File:
Table 25-2: Provisioning Parameters
Parameter
Description
Note: To add a value to these parameters, enter provisioning/ followed by the parameter name
equals the value (e.g. provisioning/method=dynamic).
provisioning/corporate_directory_uri
Version 2.0.0
The URI for retrieving the corporate directory. The
corporate directory must be included in a separate file that
can be downloaded to the phone during provisioning.
For example:
provisioning/corporate_directory_uri=corporate_dir.tx
t
Notes:
The corporate directory file is downloaded after boot
up and periodically.
If the corporate directory file is new, the phone updates
the information, does not reboot.
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25.2
Local Phone Directory
You can add, edit, or delete directory contacts. A contact’s address can be a telephone
number, IP address, or domain name. You can also download or upload a personal
directory file via the Web interface.
To add a contact to the phone’s directory:
1.
Access the ‘Directory’ page (Configuration tab > Personal Settings menu >
Directory).
Figure 25-3: Add Contact Web Fields
2.
3.
4.
5.
Under the Add Contact group, define the contact:
In the ‘Name’ field, enter the name of the contact.
In the ‘Office’, 'Home' and/or 'Mobile' fields, enter the contact’s telephone
numbers. The contact's number can be defined with an IP address or domain
name (e.g. <number>@<IP address or domain name>).
Click Submit; the contact appears in the Directory list at the bottom of the page.
To edit a contact:
1.
2.
3.
If the contact does not appear in the displayed Directory list, then from the
‘Directory Page’ drop-down list, select the page in the directory that you want
displayed.
In the Directory list, click the number that appears in the ‘No.’ column
corresponding to the contact you want to edit; the contact’s attributes appear in
the Edit Phone group above.
Edit the contact as required, and then click Submit; the contact’s new attributes
are updated in the Directory list.
To delete a contact:
1.
2.
In the Directory list, mark the ‘Select’ check box corresponding to the contact you
want to delete.
Click Delete. (To delete all contacts, click the Delete All button.)
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26. Speed Dial
Speed Dial
Speed Dial can be configured using the Web or Configuration File, as described below.
Note: The phone’s speed dials can also be defined in a simple text-based
editor, placed on a server (e.g., HTTP or FTP/TFTP), and then uploaded
to the phone using a configuration file. For a detailed description, see
Section 26.3.
26.1
Adding a Speed Dial
Adding a Speed Dial can be configured using the Web or Configuration File, as described
below.
To add a speed dial:
Using the Web:
1.
Access the ‘Speed Dial’ page (Configuration tab > Personal Settings menu >
Speed Dial).
Figure 26-1: Speed Dial Web Fields
2.
3.
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In the ‘Number’ field corresponding to the phone’s Speed Dial key (in the ‘Button’
column), enter the speed dial number to which you want to assign the Speed Dial
key.
Click Submit.
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Using the Configuration File:
Table 26-1: Speed Dial Parameters
Parameter Name
provisioning/speed_dial_uri
Description
The URI for retrieving the speed dial list. The speed dial
list must be included in a separate file that can be
downloaded to the phone during provisioning.
For example:
provisioning/speed_dial_uri=speed_dial_list.txt
Notes:
The speed dial file is downloaded after boot up and
periodically.
If the speed dial file is new, the phone reboots.
For creating a Speed Dial file, see Section 26.3.
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26. Speed Dial
Deleting a Speed Dial
To delete speed dials, do one of the following:
26.3
•
Deleting selected speed dial entries: select the ‘Delete’ check boxes
corresponding to the speed dials that you want to delete, and then click Submit.
•
All speed dials: Click Delete All, and then at the prompt, click OK.
•
To clear (unselect) all selected ‘Delete’ check boxes: click Reset.
Saving Speed Dial File Configuration to a File
Once you have configured your Speed Dial in the Web interface, you can save it to a file on
your computer. This may be useful for backing up your configuration or for using the same
configuration by loading it to another phone.
To save a Speed Dial:
1.
2.
Access the ‘Speed Dial’ page (Configuration tab > Personal Settings menu >
Speed Dial).
Click Save Speed Dial.
To load a Speed Dial:
1.
2.
3.
4.
26.4
Access the ‘Speed Dial’ page (Configuration tab > Personal Settings menu >
Speed Dial).
Click Browse…. to select the .cfg file.
Click Open; the selected .cfg filename and path appear on the Web interface
alongside the Browse… button.
Click Load Speed Dial.
Creating a Speed Dial File for Configuration File
The Configuration file can include a link to a user-defined Speed Dial file, using the
provisioning/speed_dial_uri parameter. This allows you to upload speed dial settings to
the phone.
The Speed Dial file must include a list of speed dial configurations. The file must be a
simple text file that can be created using an Excel document and saved as a CSV file.
The syntax of the speed dial file is as follows:
<memory key>,<speed dial phone number>,<type>
where:
memory key denotes the speed dial memory key on the phone.
speed dial phone number denotes the phone number that is automatically dialed,
when the user presses the speed dial key.
type denotes the Speed Dial feature and must be set to “0”.
Below is an example of a Speed Dial file:
1,4418,0
2,4403,0
3,039764432,0
4,4391,0
12,1234,0
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Security Settings
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27. Transport Layer Security and Certificate Management
Transport Layer Security and Certificate
Management
This section describes Transport Layer Security (TLS) and certificate management.
27.1
SIP over TLS Overview
TLS is a cryptographic protocol which provides communication security over the transport
layer (TCP). Generally, the TLS protocol uses private and public keys for authentication.
The authentication is performed by the certification authority (CA). Full protocol
specification is updated in RFC 5246.
Note: You must obtain the relevant CA, Private and Certificate (Public Key)
files from your Certificate Authority (CA).
27.1.1 Obtaining Certificate Authority File
The Certificate Authority (CA) is a body that certifies the ownership of a public key by the
name subject of the certificate.
Figure 27-1: Certificate Web Fields
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27.1.2 Appending Private and Public Key Data
You must edit the Private file that you received from the CA by appending the Private key
and Certificate (Public key) data as shown in the figure below. This action is necessary for
the configuration process to retrieve the relevant data. The IP Phone Private key and
Certificate data is configured by the same parameter (where each parameter specified
below represents a different configuration method (See Section 27.2 on page 183):
Configuration file: ‘/provisioning/certificate/private_key_content’ and
‘provisioning/certificate/private_key_uri’
Web: ‘Private Key and Certificate’
Figure 27-2: Combined Private and Public key Example
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27. Transport Layer Security and Certificate Management
Configuring TLS
This section describes how to configure TLS.
27.2.1 Configuring TLS using the Configuration file
This section describes how to configure TLS using the Configuration file.
27.2.1.1 Configuring TLS using a URI
This section describes how to configure TLS by directing the configuration to a URI location
of the CA Certificate and the Private key files.
Note: Before performing this procedure, ensure that you have uploaded the CA
certificate file and the Private key file to an accessible TFTP, FTP, HTTP,
or HTTPS server.
To configure TLS using a URI:
1.
Open the configuration file using a text editor.
2.
Specify the full URI of the CA Certificate file. For example:
provisioning/certificate/ca_uri= ftp://10.16.2.20/ca_cert.cer
3.
Specify the full URI of the Private key file. For example:
provisioning/certificate/private_key_uri=
https://auth.com/certfiles/private_keys.cert
4.
Save and close the file.
Table 27-1: SIP-over-TLS Parameters
Parameter
Description
SIP Transport Protocol
[voip/signalling/sip/transport_protocol]
Specifies the SIP Transport protocol.
• If using the ‘sip’ prefix, set to ‘TLS’
• If using the ‘sips’ prefix, set to ‘TCP’
TLS Port
[voip/signalling/sip/tls_port]
Defines the local TLS SIP port for SIP messages.
The valid range is 1024 to 65535. The default value
is 5061.
/voip/signalling/sip/enable_sips
If signaling protocol is set to TCP and we want to
activate TLS, this parameter should be enabled. In
this case we will use ‘sips’ prefix instead of “sip:”
Private Key and Certificate
The URI for retrieving the private and public keys
(combined file). The certificate key must be included
in a separate file that can be downloaded to the
phone during provisioning.
For example:
[provisioning/certificate/private_key_uri]
provisioning/certificate/private_key_uri=certificate.cer
Notes:
The certificate file is downloaded only after boot
up, and not periodically.
If the certificate file is new, the phone reboots.
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Parameter
CA Certification
[provisioning/certificate/ca_uri]
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Description
The URI for retrieving the Certificate Authority (CA)
key. The CA must be included in a separate file that
can be downloaded to the phone during
provisioning.
For example:
provisioning/certificate/ca_uri=ca_cert.cer
Notes:
The certificate file is downloaded only after boot
up, and not periodically.
If the certificate file is new, the phone reboots.
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27.2.1.2 Configuring TLS by Loading File Content
This section describes how to configure TLS by loading the CA certificate file and the
Private Key file content to the Configuration file.
Note: Before performing this procedure, ensure that you have uploaded the CA
certificate file and the Private Key file to an accessible TFTP, FTP,
HTTP, or HTTPS server.
To configure TLS by loading CA Certificate file and the Private Key file
content:
1.
Open the configuration file using a text editor.
2.
Specify the ‘private_key_content’ parameter and copy the content of the Private key
and Certificate (Public key) file from the appropriate server. For example:
/provisioning/certificate/private_key_content=
Note: The PRIVATE KEY and CERTIFICATE file data must be appended as
illustrated above.
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3.
Specify the parameter ‘ca_certificate_content’ and then copy the content of the CA
Certificate file: For example:
/provisioning/certificate/ca_certificate_content
4.
Save and close the file.
Table 27-2: SIP over TLS File- Encryption File Content Parameters
Parameter
Description
/provisioning/certificate/private_key_content
The content (as string) of the private key file
and public key (certificate file) (combined file).
You can load the file content here instead of
using the
provisioning/certificate/private_key_uri
parameter
/provisioning/certificate/ca_certificate_content
The content (as string) of the ca_certificate
You can load the file content here instead of
using the provisioning/certificate/ca_uri
parameter
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27.2.2 Manually Configuring TLS using the Web
This section describes how to manually configure TLS for a single telephone using the
Web.
In this procedure, associate the CA Certificate file and the Private key files to the
respective configuration parameter.
Notes:
•
Before performing this procedure, ensure that you have saved the
CA Certificate file and the Private key files in an accessible location.
•
This procedure is generally used for evaluation purposes only.
•
For parameter descriptions, see Table 27-2.
To configure TLS using the Web:
1.
Access the ‘Signaling Protocols’ page (Configuration tab > Voice Over IP menu >
Signaling Protocols > SIP General).
2.
From the ‘SIP Transport Protocol’ drop-down list, select TLS.
3.
In the ‘TLS Port field’, enter “5061”.
Figure 27-3: TLS Port and CA Certification Web Fields
4.
Load the CA certification file (*.cer) using Browse and Load buttons.
5.
Load the combined Private Key and Public Key (*.cer) file using Browse and Load
buttons.
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Figure 27-4: Private and Public Key Web Fields
6.
Click Submit.
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28. Using the Encryption Tool
Using the Encryption Tool
The following describes how to use the Encryption tool.
28.1
Encrypting Configuration Files
This procedure describes how to encrypt the Configuration file. For example, you may wish
to encrypt the configuration file when it is send over an unsecure network.
To encrypt the configuration file:
At the command line prompt, specify the following:
encryption_tool.exe –f <filename>.cfg
where <file name>.cfg specifies the name of the Configuration file that you wish to
encrypt.
Once the Configuration file is encrypted, it receives the suffix ‘.cfx’ (e.g. Conf.cfx). This
is the file that you should specify in the ‘Configuration URL’ and the ‘Dynamic
Configuration URL’ fields when performing automatic provisioning (see Part II
‘Automatic Provisioning’).
28.2
Encrypting Passwords in Configuration File
This procedure below describes how to encrypt IP phone passwords used in the
configuration process, for example, the ‘System’ password and the ‘SIP Authentication’
password.
To encrypt passwords:
1.
At the command line prompt, specify the following:
encryption_tool.exe –s <password_string>
where <password_string> specifies the string of the password that you wish to
encrypt.
Once the password is encrypted, a string is generated with the following syntax:
{"<encrypted_string>"}
For example:
{"0qrNRpSJ6aE="}
2.
Copy the generated string (including the {“ “}) with the syntax specified above to the
relevant parameter in the Configuration file.
For example, if you encrypted the SIP authentication password, the following is
displayed in the relevant line in the configuration file:
voip/line/0/auth_password={"0qrNRpSJ6aE="}
Note: It is recommended to encrypt the System password using this procedure.
If you choose not to do so, then the System password is by default
encrypted using MD5.
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29. Secure Real-time Transport Protocol
Secure Real-time Transport Protocol
Secure Real-time Transport Protocol (SRTP) is a protocol that allows encryption for RTP
data. Since the RTP encryption key is delivered via SIP, this feature is relevant only when
SIP transport is secured. Therefore, when using this feature you also need to use SIP over
TLS.
SRTP can be configured using the Web or Configuration File, as described below.
To configure SRTP:
Using the Web:
1.
Access the ‘Media Streaming’ page (Configuration tab > Voice Over IP menu >
Media Streaming).
Figure 29-1: Enabling SRTP Web Fields
2.
Configure the SRTP settings according to the parameters in the table below, and
then click Submit.
Using the Configuration File:
Table 29-1: SRTP Parameters
Parameter
Description
Enable SRTP Encryption and
Authentication
[voip/media/srtp/enabled]
Enables secured RTP (SRTP).
[0] Disable (default)
[1] Enable
Method
[voip/media/srtp/method]
The SRTP encryption method.
[AES_CM_128_HMAC_SHA1_32] (default)
[AES_CM_128_HMAC_SHA1_80]
[AES_CM_128_ALL_METHODS]
ARIA
Adds the ARIA encryption method to the list of
supported encryption the phone publishes.
[0] Disable (default)
[1] Enable
[voip/media/srtp/aria_support_enabled]
/voip/media/srtp/use_MKI
Defines the Master Key Index from SDP.
/voip/media/srtp/MKI_length
Defines the Maximum length of the Master Key Index.
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30. Configuring 802.1x
Configuring 802.1x
802.1X is an IEEE Standard for port-based Network Access Control (PNAC). It is part of
the IEEE 802.1 group of networking protocols. It provides an authentication mechanism to
devices wishing to attach to a LAN or WLAN.
30.1
Configuring 802.1x using Phone User Interface
The procedures for configuring 802.1x on the 310HD IP Phone and 320HD IP Phone using
the Phone User Interface, are described below.
30.1.1 310HD IP Phone
The procedure below describes how to configure the 802.1x settings using the 310HD IP
Phone.
To configure 802.1x:
1.
Access the Settings screen (MENU key > Administration menu > Network Settings
submenu).
The following screen appears:
2.
Version 2.0.0
Using the
schemes:
and
Navigation keys, choose one of the following IP addressing
•
Disabled – disables the 802.1x feature
•
EAP-MD5 – see Section 30.1.1.1
•
EAP-TLS – see Section 30.1.1.2
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30.1.1.1 EAP-MD5 Mode
This option enables the 802.1x feature in EAP-MD5 mode. In this mode, define the
following:
Identity: Defines the User ID.
MD5 Password: Defines the Password (optional).
To enable the 802.1x feature in EAP-MD5 mode:
1.
2.
3.
Enter the Identity (User ID).
Enter the MD5 Password (optional).
Press the left
Navigation key until you are prompted to save settings, as
shown below:
4.
Select the Yes option (using the
and
ENTER key; the settings are saved.
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30.1.1.2 EAP-TLS Mode
This option enables the 802.1x feature in EAP-TLS mode. In this mode, define the
following:
To enable the 802.1x feature in EAP-TLS mode
1.
2.
3.
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Define the following:
♦
Identity: Enables the User ID.
♦
Root Cert. URL: Enables the full URL of the file which contains the root
certification file.
♦
Trusted Cert URL: Enables the full URL of the file which contains the
trusted certification file.
♦
Local Cert URL: Enables the full URL of the file which contains the local
certification file.
♦
Private key URL: Enables the full URL of the file which contains the private
key file.
♦
Private key Password: (Optional) In case the private key file is encrypted,
input the password to decrypt the file.
Press the left
Navigation key until you are prompted to save settings, as
shown below:
Select the Yes option (using the
and
ENTER key; the settings are saved.
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30.1.2 320HD IP Phone
The procedure below describes how to configure the 802.1x settings using the 320HD IP
Phone.
To configure 802.1x:
1.
Access the Settings screen (MENU key > Administration menu > Network Settings
submenu).
The following screen appears:
2.
Using the
and
Navigation keys, choose 802.1X Settings, and then press the
Select softkey; the following screen appears:
3.
Using the
schemes:
and
Navigation keys, choose one of the following IP addressing
•
Disabled – disables the 802.1x feature
•
EAP-MD5 – see Section 30.1.3.1
•
EAP-TLS - see Section 30.1.3.2
30.1.2.1 EAP-MD5 Mode
The procedure for configuring EAP-MD5 mode for 802.1x is described below.
To configure EAP-MD5 mode for 802.1x:
1.
Using the
and
Navigation keys, choose the EAP-MD5 option, and then press
Select and then Edit; the following screen appears:
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30. Configuring 802.1x
Enter the following information:
•
Identity: User ID
•
Password: MD5 password (optional)
3.
Press the Save softkey; a message appears notifying you that the phone will restart.
4.
Press Apply.
30.1.2.2 EAP-TLS Mode
The procedure for configuring EAP-TLS mode for 802.1x is described below.
To configure EAP-TLS mode for 802.1x:
1.
Using the
and
Navigation keys, choose the EAP-MD5 option, and then press
Select and then Edit; the following screen appears:
2.
Enter the following information:
•
Identity: user ID.
•
Root Certificate URL: full URL of the file which contains root certification file.
•
Trusted Certificate URL: full URL of the file which contains trusted certification
file.
•
Local Certificate URL: full URL of the file which contains local certification file.
•
Private Key URL: full URL of the file which contains the private key file.
•
Private key Password: (option) In case the private key file is encrypted you
should input the password to decrypt the file)
3.
Press the Save softkey; a message appears notifying you that the phone will restart.
4.
Press Apply.
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30.1.3 420HD IP Phone
Configuring 802.1x settings using the 420HD IP Phone is identical to configuring 802.1x
settings using the 320HD IP Phone.
To configure 802.1x:
1.
Access the Settings screen (MENU key > Administration menu > Network Settings
submenu).
2.
Press the navigation control's lower rim to navigate to 802.1X Settings and then press
the Select softkey.
3.
Press the navigation control's lower or upper rim to navigate to and select either:
•
Disabled – disables the 802.1x feature
•
EAP-MD5 – see Section 30.1.3.1
•
EAP-TLS - see Section 30.1.3.2
30.1.3.1 EAP-MD5 Mode
The procedure for configuring EAP-MD5 mode for 802.1x is described below.
To configure EAP-MD5 mode for 802.1x:
1.
Press the navigation control's lower rim to navigate to the EAP-MD5 option and then
press Select and Edit:
2.
Enter the following information:
•
Identity: User ID
•
Password: MD5 password (optional)
3.
Press the Save softkey; a message appears notifying you that the phone will restart.
4.
Press Apply.
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30.1.3.2 EAP-TLS Mode
The procedure for configuring EAP-TLS mode for 802.1x is described below.
To configure EAP-TLS mode for 802.1x:
1.
Press the navigation control's lower rim to navigate to the EAP-TLS option and press
Select and then Edit; this screen appears:
2.
Enter the following information:
•
Identity: user ID.
•
Root Certificate URL: full URL of the file which contains root certification file.
•
Trusted Certificate URL: full URL of the file which contains trusted certification
file.
•
Local Certificate URL: full URL of the file which contains local certification file.
•
Private Key URL: full URL of the file which contains the private key file.
•
Private key Password: (option) In case the private key file is encrypted you
should input the password to decrypt the file)
3.
Press the Save softkey; a message appears notifying you that the phone will restart.
4.
Press Apply.
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30.2
Configuring 802.1x using Web and Configuration File
802.1x can be configured using the Web or Configuration File, as described below.
30.2.1 EAP MD5 Mode
This section describes how to configure 802.1x settings for EAP-MD5.
To configure the phone’s 802.1x settings for EAP-MD5:
Using the Web:
1.
Access the ‘802.1X Settings’ page (Network Connections > 802.1X Settings).
Figure 30-1: EAP MD5 Web Fields
2.
Configure the 802.1x settings according to the parameters in the table below, and
then click Submit.
Using the Configuration File:
Table 30-1: EAP MD5 Parameters
Parameter
Description
EAP Type
[/network/lan/_802_1x/eap_type]
Sets 802.1x Extensible Authentication Protocol mode.
[Disable] = Disables the use of 802.1x
[EAP_MD5]=Authentication is implemented by user
name and password (Password is optional).
[EAP_TLS]= Authentication is implemented by
Certificate, Client Certificate and Client Private Key.
Identity
[/network/lan/_802_1x/md5_identity]
User ID for md5 mode.
MD5 password
[/network/lan/_802_1x/md5_password]
Password for md5 mode. (Leave blank if no
password).
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30.2.2 EAP TLS Mode
This section describes how to configure phone’s 802.1x settings for EAP-TLS.
To configure the phone’s 802.1x settings for EAP-TLS:
Using the Web:
1.
Access the ‘802.1X Settings’ page (Network Settings > 802.1X Settings).
Figure 30-2: EAP TLS Web Fields
2.
Configure the 802.1x settings according to the parameters in the table below, and
then click Submit.
Using the Configuration File:
Table 30-2: EAP TLS Parameters
Parameter
Description
Identity
/network/lan/_802_1x/tls_identity
Identity for TLS mode. (optional)
Root and Intermediate Certificate Filename
[/network/lan/_802_1x/root_ca_url]
URL for the file that contains root and
intermediate Certificate.
Trusted Certificate Filename
[/network/lan/_802_1x/trust_ca_url]
URL for the file that contains a trusted
Certificate.
Local Certificate Filename
[/network/lan/_802_1x/local_ca_url]
URL for the file that contains a local
Certificate.
Private Key Filename
[/network/lan/_802_1x/private_key_url
URL for file that contains the private key.
Private Key Password
[/network/lan/_802_1x/private_key_password]
Optional: If the file that contains the private
key is encrypted, then this is the password
for decryption.
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31. HTTPS Management
HTTPS Management
This section describes HTTPS secure management.
31.1
Web Interface
You can choose to access the IP Phone Web interface using HTTPS for example,
“https://10.16.2.40”.
31.2
Secure Provisioning
You can load firmware and configuration files to the IP phone from an HTTPS server as
specified in the Firmware and Configuration URLs (displayed in the example Automatic
Provisioning screen below).
Figure 30-1: Automatic Provisioning Web Fields
Note: To implement secure provisioning using HTTPS, the HTTPS server on
the far end (from where you are loading the files) must also support
HTTPS.
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Maintenance
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32. Changing User Login Credentials
Changing User Login Credentials
You can change the phone’s login user name and password. This is the login required for
accessing the Web interface and the Administration menu of the phone’s LCD screen.
The default user name and password is "admin" and "1234" respectively.
User Login Credentials can be changed using the Web or Configuration File, as described
below.
To change the login username and password:
Using the Web:
1.
Access the ‘System Authorization’ page (Management tab > Administration
menu > Users).
Figure 32-1: System Authorization Web Fields
2.
3.
4.
5.
In the ‘Username’ field, enter a user name.
In the ‘Password’ field, enter a new password, and then in the ’Confirm Password’
field, re-enter this new password.
Click Submit; a confirmation box appears.
Click OK.
Using the Configuration File:
Table 32-1: Username and Password Parameters
Parameter
Description
Note: To add a value to these parameters, enter system/ followed by the parameter name, equal
sign and then the value (e.g. system/user_name=admin).
Username
[system/user_name]
The phone user name.
The default value is admin.
Note: This parameter is applicable only to the Web and Telnet
interfaces.
Password
[system/password]
The encrypted phone password.
The default value is 1234.
To generate an encrypted password, see Section 28 on page 189.
Note: This parameter is applicable only to the Web and Telnet
interfaces, and LCD display.
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33. Restarting Phones
Restarting Phones
The 310HD, 320HD and 420HD IP Phones can be restarted using the phone user interface
or the Web.
33.1
Restarting Phone using Phone User Interface
The procedures below describe how to restart the phone using the phone user interface.
33.1.1 310HD IP Phone
The procedure below describes how to restart the 310HD IP Phone using the phone user
interface.
To restart the 310HD IP Phone:
1.
Select the Restart submenu (MENU key > Administration menu > Restart):
A confirmation message appears, prompting you to confirm phone reset:
2.
Select the Yes option (using the
ENTER key; the phone resets.
and
Navigation keys), and then press the
Notes:
Version 2.0.0
•
You can remotely reset the phone by sending a SIP NOTIFY
message to the phone with the "check-sync" or "resync" events. This
reset allows for graceful shutdown, whereby the phone resets only
when there are no calls, i.e., when the phone is in idle state.
•
Upon receipt of a "check-sync" or "resync" message, the phone
immediately retrieves the configuration file and reboots if a new
configuration is provided.
•
The phone can be configured to reboot even if no configuration
update is required.
•
The phone supports the optional "reboot" parameter located in the
Event header of the NOTIFY message (e.g. Event: checksync;reboot=true, Event: resync;reboot=false, etc.).
•
When set to "false" or not present, the phone retrieves its new
configuration file and reboots only if the configuration was updated.
•
When set to "true", the phone reboots without trying to get the
configuration file.
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33.1.2 320HD IP Phone
The procedure below describes how to restart the 320HD IP Phone using the phone menu.
To restart the 320HD IP Phone:
1.
Select the Restart submenu (MENU key > Administration menu > Restart
submenu).
A warning message appears, requesting you to confirm:
2.
Press the Yes softkey to confirm phone restart or No if you want to cancel.
33.1.3 420HD IP Phone
The procedure to restart the 420HD IP Phone is identical to that for the 320 HD IP Phone.
To restart the 420HD IP Phone:
1.
Select the Restart submenu (MENU key > Administration menu > Restart
submenu):
A warning message appears requesting you to confirm:
2.
Press the Yes softkey to confirm phone restart or No if you want to cancel.
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33. Restarting Phones
Restarting Phone using Web
You can use the Web interface to restart your phone.
To restart the phone:
1.
Access the ‘Restart System’ page (Management tab > Administration menu >
Restart System).
Figure 33-1: Restart System Web Fields
2.
Click the Restart button; a confirmation box appears prompting you to confirm.
Figure 33-2: Confirmation Box
3.
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Click OK.
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34. Restoring Phone Defaults
Restoring Phone Defaults
Phone default settings can be restored using the phone or Web interface.
34.1
Restoring Factory Defaults using Phone Interface
The procedures for restoring factory defaults using the IP Phone user interface are
described below.
34.1.1 310HD IP Phone
The procedures below describe how to restore the 310HD IP Phone factory defaults using
the phone.
To restore the 310HD IP Phone to default settings:
1.
Select the Restore Defaults submenu (MENU key > Administration menu >
Restore Defaults):
A confirmation message appears, prompting you to confirm factory reset:
2.
Select the Yes option (using the
and
Navigation keys), and then press the
ENTER key; the phone is restored to factory defaults.
Note: You can also restore the phone to default settings by using a hard-coded
sequence of keys on the phone's keypad. To do this:
Version 2.0.0
•
Unplug the power cable from the phone.
•
Press the following keys at the same time MENU and ENTER.
•
While pressing the above keys, plug the power cable into the phone.
Once the power is connected, keep pressing the keys for five
seconds and then release the keys.
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34.1.2 320HD IP Phone
The procedure below describes how to restore the 320HD IP Phone factory defaults using
the phone.
To restore the phone to default settings:
1.
Access the Restore Defaults submenu (MENU key > Administration menu >
Restore Defaults).
A warning message appears, requesting you to confirm:
2.
Press the Yes softkey to confirm reset to defaults or No if you want to cancel.
Notes: You can also restore the phone to default settings by using a hard-coded
sequence of keys on the phone's keypad. To do this:
•
Unplug the power cable from the phone.
•
Press the
together.
•
While pressing the above keys, plug the power cable into the phone.
Once the power is connected, keep pressing the keys for five
seconds and then release keys.
and
Navigation keys and the ENTER key all
34.1.3 420HD IP Phone
The procedure for restoring the 420HD IP Phone factory defaults is identical to that for the
320HD IP Phone.
To restore the phone to default settings:
1.
Access the Restore Defaults submenu (MENU key > Administration menu >
Restore Defaults).
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2.
Press the Select softkey; a warning message appears requesting you to confirm:
3.
Press the Yes softkey to confirm reset to defaults or No if you want to cancel.
Notes: You can also restore the phone to default settings by using a hard-coded
sequence of keys on the phone's keypad. To do this:
34.2
•
Unplug the power cable from the phone.
•
Press the OK and the MENU key together.
•
Plug the power cable into the phone and keep the keys pressed for
five seconds.
Restoring Factory Defaults using the Web
You can restore all your phone’s settings to factory default settings using the Web.
To restore the phone to factory defaults:
1.
Access the ‘Restore Defaults’ page (Management tab > Administration menu >
Restore Defaults).
Figure 34-1: Restore Defaults Web Fields
2.
Click the Submit button; a confirmation box appears prompting you to confirm.
Figure 34-2: Submit Confirmation Box
3.
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Click OK.
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Status and Monitoring
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35. Viewing Network Status
Viewing Network Status
This section describes how to view the network status using the Web interface.
35.1
LAN Status
The following describes how to view LAN status information.
To view LAN status information:
Access the ‘Network Status’ page (Status & Diagnostics tab > System Status menu
> Network Status).
Figure 35-1: LAN Information Web Fields
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35.2
Port Status
The following describes how to view the Port Mode status.
To view Port Mode status:
Access the ‘Network Status’ page (Status & Diagnostics tab > System Status menu
> Network Status).
Figure 35-2: Port Mode Status Web Fields
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35. Viewing Network Status
802.1x Status
The following describes how to view 802.1x status.
To view 802.1x status:
Access the ‘Network Status’ page (Status & Diagnostics tab > System Status menu
> Network Status).
Figure 35-3: 802.1X Status Web Fields
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36. Viewing VoIP Status
Viewing VoIP Status
The following describes how to view VoIP status using the Web interface.
36.1
Phone Status
This section describes how to view the phone status.
To view the phone status:
Access the ‘Network Status’ page (Status & Diagnostics tab > System Status menu
> VoIP Status).
Figure 36-1: Phone Status Web Fields
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36.2
Line Status
This section describes how to view the line status.
To view the line status:
Access the ‘Network Status’ page (Status & Diagnostics tab > System Status menu
> VoIP Status).
Figure 36-2: Line Status Web Fields
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36. Viewing VoIP Status
Current Call Information
The Web interface displays call information of a currently established call.
To view the line information:
Access the ‘Network Status’ page (Status & Diagnostics tab > System Status menu
> VoIP Status).
Figure 36-3: Current Call Information Web Fields
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37. Viewing Call History
Viewing Call History
You can view a list of received calls, missed calls, and dialed numbers. The call duration of
the received and dialed calls is also displayed.
To view call history log:
1.
Access the ‘Call History’ page (Status & Diagnostics tab > History menu > Call
History).
Figure 37-1: Call History Web Fields
2.
From the ‘Type’ drop-down list, select the type of call history (i.e., missed calls,
received calls, and dialed numbers) that you want to view; the table lists the call
history according to the chosen call history type.
3.
You can delete a logged call history entry, by selecting the ‘Delete’ check box
corresponding to the entry that you want to delete, and then clicking the Delete button.
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38. Viewing Phone Version Number
Viewing Phone Version Number
You can view the phone firmware version number via the Web or on the phone itself.
38.1
Viewing Phone Version Number using the Web
You can view the phone’s model name as well as the firmware version currently running on
the phone.
To view the phone’s model and version number:
Access the ‘System Information’ page (Status & Diagnostics tab > System
Information menu > Versions).
Figure 38-1: System Information Web Fields (for example 310HD Model)
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38.2
Viewing Firmware Version using Phone Interface
You can view the phone’s model number and firmware version on the phone.
38.2.1 310HD IP Phone
The procedures below describe how to view the 310HD IP Phone’s model number and
firmware version on the phone.
To view the phone’s firmware version:
1.
From the idle screen, press the MENU key, select the Status menu, and then the
Firmware Version submenu.
The following screen appears:
2.
Press the ENTER key; the phone’s model and firmware version is displayed.
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38.2.2 320HD IP Phone
The procedures below describe how to view the 320HD IP Phone’s model type, firmware
version, and configuration file version.
To view the phone’s firmware version:
Version 2.0.0
1.
Access the Software Versions screen (MENU key > Status menu > Firmware
Version).
2.
The following screen appears, displaying the phone model and the firmware
version currently running on the phone:
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38.2.3 420HD IP Phone
The procedures below describe how to view the 420HD IP Phone’s model type, firmware
version, and configuration file version.
To view the phone’s firmware version:
1.
Access the Firmware Version screen (MENU key > Status menu > Firmware
Version).
2.
The following screen appears, displaying the phone model and the firmware
version currently running on the phone:
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Diagnostics and
Troubleshooting
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A. Server-Specific Configuration
Testing Phone Hardware
The 310HD, 320HD and 420HD IP Phones Built-in Tests (BITs) are described below.
39.1
310HD IP Phone BIT
The 310HD IP Phone provides a built-in test for verifying its operational integrity. These
tests can be performed before deployment and in cases of troubleshooting. These tests are
all performed in one continuous sequence, where each test is followed by another
consecutively, in the order listed below:
Keypad and off-hook/on-hook test
LED test
Handset microphone (recording/speaking) and handset receiver (playing/listening) test
Headset microphone (recording/speaking) and headset headphone/receiver
(playing/listening) test
Speaker microphone (recording/speaking) and speaker receiver (playing/listening) test
MAC address and serial number verification
39.1.1 Test Preparation
Before you can start the built-in test, you need to cable the phone as follows:
1.
Connect the phone’s LAN port to a switch, using a LAN cable.
2.
Ensure that the DHCP server is functioning.
3.
Connect a headset to the phone.
4.
Power on the phone and wait until initialization is complete.
5.
From this stage onwards, follow the procedures described in the subsequent sections
and in consecutive order. In other words, continue with the procedure described in
Section 39.1.2 on page 236.
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39.1.2 Keypad Test
The Keypad test checks the responsiveness and correct functioning of the keys on the
phone’s keypad.
To test the keypad:
1.
Using the keypad, dial the number “0123456789” and then press the star (“*”)
key; the phone’s LCD screen displays the following:
Figure 39-1: Testing Keypad
2.
Press all keys except the keys 0-9, *, and #; for every key pressed, its
corresponding character displayed on the LCD disappears.
Figure 39-2: Testing Keypad – Characters Disappear
.
3.
Continue with the LED test in Section 39.1.3 on page 237.
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39.1.3 LED Test
Upon the successful completion of the Keypad test (in the previous section), the LCD
screen displays the message “All LEDs are on. Press #” and the phone’s single LED lights
up red, as shown in the figure below:
Figure 39-3: Successful Completion of Keypad Test
Continue with Handset test in Section 39.1.4 on page 238.
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39.1.4 Handset Test
Upon successful completion of the LED test (in the previous section), perform the Handset
test. This test verifies the correct functioning of the handset, which includes the following:
Handset’s microphone (transmitter) for speaking
Handset’s receiver (speaker) for listening
To test the handset:
1.
Press the pound (#) key; the LCD screen displays the message ”Handset test. Off
hook to start”.
Figure 39-4: Off hook Handset Test
2.
Off hook the phone; the LCD screen displays the message “Recording”.
Figure 39-5: Recording Test
3.
Speak into the handset microphone; after about five seconds, the LCD screen
displays the message ”Playing”.
Figure 39-6: Playing Test
4.
5.
The voice that was recorded when you spoke into the handset microphone is now
played from the handset receiver.
Continue with the Headset test in Section 39.1.5 on page 239.
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39.1.5 Headset Test
Upon successful completion of the Handset test (in the previous section), perform the
Headset test. This test verifies the correct functioning of the headset (for hands-free
operation), which includes the following:
Headset’s microphone (transmitter) for speaking
Headset’s headphone (receiver) for listening
To test the headset:
1.
After about five seconds after your recorded voice is played in the Handset test
(in the previous section), the LCD screen displays the message, ”Headset test.
Any key to start”.
Figure 39-7: Headset Test
2.
Press any key; the LCD screen displays the ”Recording” message.
Figure 39-8: Headset Recording Test
3.
Speak into the headset’s microphone; after about five seconds, the LCD screen
displays ”Playing” and the voice that was recorded when you spoke into the
headset’s microphone is now played from the headset’s receiver.
Figure 39-9: Headset Playing Test
4.
Version 2.0.0
Continue with the Speaker test in Section 39.1.6 on page 240.
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39.1.6 Speaker Test
Upon successful completion of the Headset test (in the previous section), perform the
Speaker test. This test verifies the correct functioning of the speakers, which includes the
following:
Speaker microphone (transmitter) for speaking (recording)
Speaker receiver for listening (playing)
To test the speaker:
1.
After about five seconds after your recorded voice is played in the Headset test
(in the previous section), the LCD screen displays the message ”Speaker/Mic
test. Any key to start”.
Figure 39-10: Speaker Test
2.
Press any key; the LCD screen displays the ”Recording” message.
Figure 39-11: Speaker Recording Test
3.
Speak into the phone’s speaker microphone; after about five seconds, the LCD
screen displays the message ”Playing”, and the voice that was recorded when
you spoke into the speaker’s microphone is now played from the speaker’s
receiver.
Figure 39-12: Speaker Playing Test
4.
Continue with the MAC Address and Serial Number Verification test in Section
39.1.7 on page 241.
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39.1.7 MAC Address and Serial Number Verification Test
The 310HD, 320HD and 420HD IP Phones MAC Address and Serial Number Verification
Tests are described below.
39.1.7.1 MAC Address and Serial Number Verification Test
The MAC Address and Serial Number Verification test allows you to verify that the 310HD
IP Phone’s MAC address and serial number are correct.
To verify the MAC address and serial number:
1.
After about five seconds after your recorded voice is played in the Speaker test (in the
previous section), the LCD screen displays a MAC address and serial number:
Figure 39-13: MAC Address and Serial Number
2.
Check that the LAN MAC address and serial number are correct.
3.
Press any key to exit the phone’s built-in test
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39.2
Testing 320HD IP Phone BIT
The 320HD IP Phone provides a built-in test for verifying its operational integrity. This test
can be performed before deployment and in cases of troubleshooting. The built-in test is
comprised of numerous tests that are performed in one continuous sequence, where each
test is followed by another consecutively, in the order listed below:
Keypad and off-hook/on-hook test
Green-color LEDs test
Red-color LEDs test
Handset microphone (recording/speaking) and handset receiver (playing/listening) test
Headset microphone (recording/speaking) and headset headphone/receiver
(playing/listening) test
Speaker microphone (recording/speaking) and speaker receiver (playing/listening) test
MAC address and serial number verification
39.2.1 Test Preparation
Before you can start the built-in test, you need to cable the phone as follows:
1.
Connect the phone’s LAN port to a switch, using a LAN cable.
2.
Ensure that the DHCP server is functioning.
3.
Connect a headset to the phone.
4.
Power on the phone and wait until initialization is complete.
5.
From this stage onwards, follow the procedures described in the subsequent sections
and in consecutive order. In other words, continue with the procedure described in
Section 39.1.2 on page 236.
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39.2.2 Keypad and Off-Hook/On-Hook Test
The Keypad and Off-Hook/On-Hook tests checks the responsiveness and correct
functioning of the following:
All the keys on the phone’s keypad
Handset when picked up (off-hook) and placed back on the receiver (on-hook)
To test the keypad and on-hook state:
1.
Using the keypad, dial the number "0123456789" and then press the star ("*") key; the
phone’s LCD screen displays a graphical display of the keypad, where each key has a
corresponding indicator on the LCD screen.
Figure 39-14: Keypad Test
2.
Off-hook the handset; the hook indicator ("H") on the LCD screen turns white on a
black background.
Figure 39-15: Keypad Test – Off-Hook
3.
Version 2.0.0
On-hook the handset; the hook indicator on the LCD screen turns white again.
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4.
Press each key on the phone’s keypad; each key that is pressed turns its
corresponding indicator on the LCD screen to the color black.
Figure 39-16: Keypad Test - Keys
5.
Continue with the Green-color LEDs test in Section 39.1.3 on page 237.
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39.2.3 Green-Color LEDs Test
Upon the successful completion of the Keypad and On-Hook/Off-Hook test (in the previous
section), the LCD screen displays the message "Green LEDs are on". Consequently, all
LEDs light up green (except the VOICE MAIL, HEADSET, SPEAKER, and MUTE LEDs,
which are red only), as shown in the figure below:
Figure 39-17: Green LEDs On
Continue with the Red-Color LEDs test in Section 39.2.4 on page 246.
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39.2.4 Red-Color LEDs Test
Upon the completion of the Green-Color LEDs test (in the previous section), perform the
Red-color LEDs test for verifying that the red light is functioning for the phones LEDs.
To test the LEDs for red-color lighting:
1.
Press any key; all the LEDs turn red, as shown in the figure below:
Figure 39-18: Red LEDs Test
2.
Continue with the Handset test in Section 39.2.5 on page 247.
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39.2.5 Handset Test
Upon successful completion of the Red-color LEDs test (in the previous section), perform
the Handset test. This test verifies the correct functioning of the handset, which includes
the following:
Handset’s microphone (transmitter) for speaking
Handset’s receiver (speaker) for listening
To test the handset:
1.
Press any key; the LCD screen displays the message "Prepare to test handset. Off
hook to start".
Figure 39-19: Handset Test
2.
Off hook the phone; the LCD screen displays the message "start recording".
Figure 39-20: Handset Test - Recording
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3.
Speak into the handset microphone; after about five seconds, the LCD screen
displays the message "start playing".
Figure 39-21: Handset Test- Playing
The voice message that was recorded when you spoke into the handset microphone is
now played from the handset receiver.
4.
Continue with the Headset test in Section 39.1.5 on page 239.
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39.2.6 Headset Test
Upon successful completion of the Handset test (in the previous section), perform the
Headset test. This test verifies the correct functioning of the headset (for hands-free
operation), which includes the following:
Headset’s microphone (transmitter) for speaking
Headset’s headphone (receiver) for listening
To test the headset:
1.
After about five seconds after your recorded voice is played in the Handset test (in the
previous section), the LCD screen displays the message "Prepare to test headset.
Press any key to start".
Figure 39-22: Headset Test
2.
Press any key; the LCD screen displays the message "start recording".
Figure 39-23: Headset Test - Recording
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3.
Speak into the headset’s microphone; after about five seconds, the LCD screen
displays "start playing" and the voice that was recorded when you spoke into the
headset’s microphone is now played from the headset’s receiver.
Figure 39-24: Headset Test - Playing
4.
Continue with the Speaker test in Section 39.1.6 on page 240.
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39.2.7 Speaker Test
Upon successful completion of the Headset test (in the previous section), perform the
Speaker test. This test verifies the correct functioning of the speakers, which includes the
following:
Speaker microphone (transmitter) for speaking (recording)
Speaker receiver for listening (playing)
To test the speaker:
1.
After about five seconds after your recorded voice is played in the Headset test (in the
previous section), the LCD screen displays the message "Speaker/Mic Test. Press
any key".
Figure 39-25: Speaker/Mic Test
2.
Press any key; the LCD screen displays the message "Start recording".
Figure 39-26: Speaker/Mic Test– Recording
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3.
Speak into the phone’s speaker microphone; after about five seconds, the LCD screen
displays the message "playing", and the voice that was recorded when you spoke into
the speaker’s microphone is now played from the speaker’s receiver.
Figure 39-27: Speaker/Mic Test – Playing
4.
Continue with the MAC Address and Serial Number Verification test in Section 39.1.7
on page 241.
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39.2.8 MAC Address and Serial Number Verification Test
The MAC Address and Serial Number Verification test allows you to verify that the phone’s
MAC address and serial number are correct.
To verify the MAC Address and Serial Number:
1.
After about five seconds after your recorded voice is played in the Speaker test (in the
previous section), the LCD screen displays a MAC address and serial number:
Figure 39-28: MAC Address and Serial Number Verification Test
39.3
2.
Check that the LAN MAC address and serial number are correct.
3.
Press any key to exit the phone’s built-in test.
Testing 420HD IP Phone BIT
The BIT for the 420HD IP Phone is identical to that of the 320HD IP Phone (see Section
39.2 on page 242).
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A. Server-Specific Configuration
Recovering Firmware
If the phone is powered off for some reason during the firmware upgrade process, the
phone becomes unusable.
To recover the phone firmware:
1.
Ensure that your DHCP server supports Options 66 (TFTP server address) and 67
(firmware file), and that these are configurable.
2.
Before connecting the phone, verify that the TFTP server is running and the firmware
file for recovery is located in the correct location.
3.
Connect your phone to the IP network, and then connect the phone to the power
outlet;
a.
b.
c.
d.
Version 2.0.0
The phone sends a TFTP request to the IP address indicated in the DHCP Option
66 field to retrieve the firmware file indicated in the DHCP Option 67 field.
The phone, in the DHCP Discover message sends its model name in the DHCP
Option 77 field. The DHCP server, according to the phone model, sets the
appropriate firmware file name in the DHCP Option 67 field sent to the phone
(e.g., 310HD_1.4.1_build_32.img).
The phone then upgrades to the recovery firmware.
After the firmware upgrade process completes, the phone boots up successfully.
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A. Server-Specific Configuration
System Logging (Syslog)
System logging can be configured using the Web or Configuration File, as described
below.
To configure system logging:
Using the Web:
1.
Access the ‘System logging’ page (Status & Diagnostics tab > Diagnostics
menu > Logging).
Figure 41-1: System Logging Web Fields
2.
Configure system logging according to the parameters in the table below, and
then click Submit.
Using the Configuration File:
Table 41-1: Syslog Parameters
Parameter
Description
Note: To add a value to these parameters, enter system/ followed by the parameter name, equal
sign and then the value (e.g. system/syslog/mode=UDP).
Activate
[system/watchdog/enabled]
Version 2.0.0
Enables the system watchdog. Possible values are:
Disable = No Syslog.
• Network = Syslog is sent to the Syslog server.
(Recommended).
• Console = Syslog is sent to the phone console.
(You need to connect a serial cable to view the
logs. This causes delays in phone operation).
• Both = Syslog sends to the Syslog server AND
the console.
Note: It is recommended to leave this parameter at
its default value.
•
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Parameter
Description
Server IP Address or Host Name
[system/syslog/server_address]
The IP address (in dotted-decimal notation) of the
computer you are using to run the Syslog server
(e.g. Wireshark). The Syslog server is an application
designed to collect the logs and error messages
generated by the phone. The default IP address is
0.0.0.0.
Note: This parameter is applicable when Activate is
set to “Network” or “Both”.
Server Port
[system/syslog/server_port]
Defines the UDP port of the Syslog server.
The valid range is 0 to 65,535. The default port is
514.
Note: This parameter is applicable when Activate is
set to “Network” or “Both”.
Note: The following Severity level options are applicable for the fields below:
• None
• Emergency
• Error
• Waiting
• Notice
• Info
• Debug
VoIP Application
Defines multi-layer VoIP application.
SIP Call Control
Defines MTR layer Radvision.
SIP Stack
Defines SIP Stack Radvision.
Control Center
Responsible for Networking and running other
processes.
LCD Display
Defines LCD Display.
CGI
Defines services for the Web server.
Web Server
Defines the phone Web server.
Watchdog
Responsible for keeping other processes running.
802.1X
Defines the security protocol.
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A. Server-Specific Configuration
Phone User Interface Error Messages
The following Phone User Interface error messages may appear on the IP Phone.
Table 42-1: Phone User Interface Error Messages
Message
Description
LAN Link failure
The LAN link is disconnected. Registration failure
Received error or no response from the SIP proxy Notes:
Version 2.0.0
•
With both errors, the “ringer” LED remains red until the error is fixed.
•
While the error message is displayed, the user can`t dial or initiate a
call.
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A. Server-Specific Configuration
Media Debug Information
Recording parameters allows you to debug the voice activity of the phone, using the Web
or Configuration File, as described below.
To define media debug information:
Using the Web:
1.
Access the ‘Recording’ page (Status & Diagnostics tab > Diagnostics menu >
Recording).
Figure 43-1: Recording Web Fields
2.
Configure Media Debug Recording according to the parameters in the table
below, and then click Submit.
Using the Configuration File:
Table 43-1: Packet Recording Parameters
Parameter
Description
Enable DSP Recording
[voip/packet_recording/enabled]
Activates the packet recording
mechanism.
[0] Disable (default)
[1] Enable
Remote IP Address or Host Name
[voip/packet_recording/remote_ip]
The IP address (in dotted-decimal
notation) of the remote computer to
which the recorded packets are sent.
The recorded packets should be
captured by a network sniffer (such as
Wireshark).
The default value is 0.0.0.0.
Remote Port
[voip/packet_recording/remote_port]
Defines the UDP port of the remote
computer to which the recorded
packets are sent.
The valid range is 1024 to 65535. The
default value is 50000.
Enable RTP Recording
[voip/packet_recording/rtp_recording/enabled]
Activates the DSP RTP recording.
[0] Disable (default)
[1] Enable
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Parameter
Description
Enable EC Debug Recording
[voip/packet_recording/ec_debug_recording/enabled
]
Activates the Echo Canceller Debug
recording.
[0] Disable (default)
[1] Enable
Enable Network Recording
[voip/packet_recording/network_recording/enabled]
Activates the DSP network (TDM Out)
recording.
[0] Disable (default)
[1] Enable
Enable TDM Recording
[voip/packet_recording/tdm_recording/enabled]
Activates the DSP TDM (TDM In)
recording.
[0] Disable (default)
[1] Enable
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A. Server-Specific Configuration
Port Mirroring
Port mirroring can be configured using the Web or Configuration File, as described below.
To configure port mirroring:
Using the Web:
1.
Access the ‘Recording’ page (Configuration tab > Network Connections menu
> Network Settings).
Figure 44-1: Port Mirroring
2.
Configure port mirroring according to the parameters in the table below, and then
click Submit.
Using the Configuration File:
Table 44-1: Port Mirroring Parameters
Parameter
Activate
[network/pc_port_mirroring/enabled]
Version 2.0.0
Description
Enables port mirroring.
[0] Disable (default) - The LAN/PC Network
interfaces operate in SWITCH mode.
[1] Enable - The LAN/PC Network interfaces
operate in HUB mode. The network traffic on the
LAN port is reflected in the PC port.
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Part XI
Appendices
Reader’s Notes
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A
A. Server-Specific Configuration
Server-Specific Configuration
Note: This appendix is only applicable to the 320HD IP Phone.
A.1
Lync (SPS only)
Lync can be configured using the Configuration File, as described below.
Table A-1: Busy Lamp Field Parameter
Parameter
Description
/voip/services/busy_lamp_field/color_settings
Version 2.0.0
267
Relevant for presence feature.
Valid values:
[OPTION_1] = BLF refers the presence state
of remote parties.
[OPTION_2] = BLF refers the call state of
remote parties (default).
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A.2
BroadWorks
The configuration of the BLF feature is unique when the selected application server is
BroadWorks™.
To configure BLF for BroadWorks:
1.
Access the Services page (Configuration tab > Voice Over IP menu > Services).
The figure’s callouts below correspond to the steps in this procedure:
Figure A-1: BLF Configuration for BroadWorks Web Fields
2.
From the ‘Application Server Type’ drop-down list, select BSF.
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A. Server-Specific Configuration
In the BLF Support group, do the following:
a.
b.
c.
d.
From the ‘Activate’ drop-down list (voip/services/busy_lamp_field/enabled
parameter), select Enable.
In the ‘BLF Subscription Period’ field
(voip/services/busy_lamp_field/subscription_period parameter), enter the interval
between BLF and SIP SUBSCRIBE messages.
In the ‘User Resource List’ field (voip/services/busy_lamp_field/uri parameter),
enter the resource list URI to which the phone can subscribe to in order to get the
BLF information from the application server.
Define the Application Server Address:
♦
Same address as SIP Registrar: From the ‘Use Registrar as Application
Server Address’
(voip/services/busy_lamp_field/application_server/use_registrar parameter)
drop-down list, select Enable. The Registrar’s address is used as the
application server’s address, defined by the parameter
voip/signalling/sip/sip_registrar/addr (refer to Section 17.2 on page 93).
♦
Unique address: From the ‘Use Registrar as Application Server Address’
drop-down list, select "Disable", and then in the "Application Server Address
or Domain Name" field
(voip/services/busy_lamp_field/application_server/addr parameter), enter the
IP address or domain name of the application server.
4.
Click Submit.
5.
Define speed dial keys with the BLF feature (see Section 26 on page 175).
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A.3
Configuring BLF for Asterisk, Coral and Metaswitch
The configuration of the BLF feature is unique when the selected application server is
Asterisk, Coral, or Metaswitch.
To configure BLF for Asterisk application server:
1.
Access the Services page (Configuration tab > Voice Over IP menu > Services).
The figure’s callouts below correspond to the steps in this procedure:
Figure A-2: BLF Configuration for Application Server Type - Asterisk
2.
3.
From the ‘Application Server Type’ drop-down list, select one of the following:
•
Asterisk
•
Coral
•
Metaswitch
In the BLF Support group, configure the following:
a.
b.
From the ‘Activate’ drop-down list (voip/services/busy_lamp_field/enabled
parameter), select Enable.
(Optional) In the ‘BLF Subscription Period’ field
(voip/services/busy_lamp_field/subscription_period parameter), enter the interval
between BLF and SIP SUBSCRIBE messages.
Note: The application server’s address is the same as the SIP Registrar
address defined by the parameter voip/signalling/sip/sip_registrar/addr
(see Section 17.2 on page 93).
4.
Click Submit.
5.
Define speed dial keys with the BLF feature (see Section 26 on page 175).
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B
B. Typical Setup Scenario Example
Typical Setup Scenario Example
In the sample scenario below, the IT manager plans to deploy out-of-the-box IP phones in
the office. AudioCodes’ configuration tool is used to prepare a configuration file per phone
(based on MAC addresses). The configuration files are downloaded from the office’s HTTP
server. DHCP is used to point the phone to the HTTP server.
To prepare the template configuration file:
1.
2.
Connect an out-of-the box IP Phone to a regular LAN and wait for it to boot up.
Check the phone’s IP address using the User Interface (Menu > Status >
Network Status).
3. Connect to the phone’s embedded web server using http://<phone IP address>.
4. Make sure the phone is running the latest firmware version (shown on the home
page). If not, obtain a new firmware file and download it using
Management > Manual Update > Firmware Upgrade.
5. If the phone’s configuration has been modified in the past, restore the default
configuration using Management > Administration > Restore Defaults.
6. Access Management > Manual Update > Configuration File and save a
copy of the original configuration file.
7. Make the required configuration changes (e.g. VoIP parameters) via Web.
8. Make sure the IP Phone functions properly in your network.
9. Access Management > Manual Update > Configuration file and save the
modified configuration file.
10. Compare the two configuration files and create a file containing only the modified
lines.
To create MAC-specific files using AudioCodes’ configuration tool:
1.
2.
3.
4.
Edit the template configuration file and replace each parameter value that should
be different per user (e.g. the line ID) with a tag (e.g. [id]).
Prepare a *.csv file with a list of the MAC addresses of the phones you plan to
deploy and the parameters values each user.
Run the configuration too to generate the configurations files (one per phone).
Save the configuration files on the HTTP server.
To prepare the DHCP server:
•
Configure OPTION 160 on the DHCP with the URL of the configuration files on
the HTTP server. Use the string <MAC> to indicate the phone’s MAC address, for
example: http://192.168.2.1/:310HD_<MAC>_conf.cfg.
To verify the setup:
1.
2.
3.
4.
Version 2.0.0
Connect one of the IP Phones for deployment to the network.
Follow the status displayed on the LCD. Make sure the phone received an IP
address and is upgrading the configuration.
The phone reboots with the new configuration.
Make sure all functionality works well (e.g. can make VoIP calls).
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C
C. SIP Support (RFC, Headers)
SIP Support (RFC, Headers)
The following is a list of supported SIP RFCs and Methods that you can use to create for
the IP Phone.
Table C-1: Supported IETF RFC’s
RFC Number
RFC Title
RFC 2327
SDP
RFC 2617
HTTP Authentication: Basic and Digest Access
Authentication
RFC 2782
A DNS RR for specifying the location of services
RFC 2833
Telephone event
RFC 3261
SIP
RFC 3262
Reliability of Provisional Responses in SIP
RFC 3263
Locating SIP Servers
RFC 3264
Offer/Answer Model
RFC 3265
(SIP)-Specific Event Notification
RFC 3310
Hypertext Transfer Protocol (HTTP) Digest
Authentication Using Authentication and Key
Agreement (AKA)
RFC 3311 (Partially Supported)
UPDATE Method
RFC 3326 (Partially Supported)
Reason header
RFC 3389
RTP Payload for Comfort Noise
RFC 3515
Refer Method
RFC 3605
RTCP attribute in SDP
RFC 3611
RTP Control Protocol Extended Reports (RTCP XR)
RFC 3665
SIP Basic Call Flow Examples
RFC 3711
The Secure Real-time Transport Protocol (SRTP)
RFC 3725
Third Party Call Control
RFC 3842
MWI
RFC 3891
"Replaces" Header
RFC 3892 (Sections 2.1-2.3 and 3 are
supported)
The SIP Referred-By Mechanism
RFC 3960 (Partially Supported)
Early Media and Ringing Tone Generation in SIP
(partial compliance)
RFC 3966
The tel URI for Telephone Numbers
RFC 4028 (Partially Supported)
Session Timers in the Session Initiation Protocol
RFC 4240
Basic Network Media Services with SIP - NetAnn
draft-ietf-sip-privacy-04.txt (Partially
Supported)
SIP Extensions for Network-Asserted Caller Identity
using Remote-Party-ID header
draft-ietf-sipping-cc-transfer-05
Call Transfer
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RFC Number
RFC Title
draft-ietf-sipping-realtimefax-01
SIP Support for Real-time Fax: Call Flow Examples
draft-choudhuri-sip-info-digit-00
SIP INFO method for DTMF digit transport and
collection
draft-mahy-sipping-signaled-digits-01
Signaled Telephony Events in the Session Initiation
Protocol
Note: The following SIP features are not supported:
Administrator's Manual
•
Preconditions (RFC 3312)
•
SDP - Simple Capability Declaration (RFC 3407)
•
S/MIME
•
Outbound, Managing Client-Initiated Connections (RFC 5626)
•
SNMP SIP MIB (RFC 4780)
•
SIP Compression – RFC 5049 (SigComp)
•
ICE (RFC 5245)
•
Connected Identity (RFC 4474)
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C.1
C. SIP Support (RFC, Headers)
SIP Compliance Tables
The SIP device complies with RFC 3261, as shown in the following subsections.
C.1.1
SIP Methods
The device supports the following SIP Methods:
Table C-2: Supported SIP Methods
Method
Supported
Comments
INVITE
Yes
ACK
Yes
BYE
Yes
CANCEL
Yes
REGISTER
Yes
Send only
REFER
Yes
Inside and outside of a dialog
NOTIFY
Yes
INFO
Yes
OPTIONS
Yes
PRACK
Yes
PUBLISH
Yes
SUBSCRIBE
Yes
Version 2.0.0
Send only
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C.1.2
SIP Headers
The device supports the following SIP Headers:
Table C-3: Supported SIP Headers
Header Field
Supported
Accept
Yes
Alert-Info
Yes
Allow
Yes
Authorization
Yes
Call-ID
Yes
Call-Info
Yes
Contact
Yes
Content-Length
Yes
Content-Type
Yes
Cseq
Yes
Date
Yes
Diversion
Yes
Encryption
No
Expires
Yes
Fax
Yes
From
Yes
History-Info
Yes
Join
Yes
Max-Forwards
Yes
MIN-SE
Yes
P-Asserted-Identity
Yes
P-Preferred-Identity
Yes
Proxy- Authenticate
Yes
Proxy- Authorization
Yes
Prack
Yes
Record- Route
Yes
Refer-To
Yes
Referred-By
Yes
Replaces
Yes
Remote-Party-ID
Yes
Retry-After
Yes
Route
Yes
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C. SIP Support (RFC, Headers)
Header Field
Supported
Session-Expires
Yes
Supported
Yes
Timestamp
Yes
To
Yes
Unsupported
Yes
User- Agent
Yes
Via
Yes
Voicemail
Yes
Warning
Yes
WWW- Authenticate
Yes
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D
D. IP Phone Specifications
IP Phone Specifications
The 310HD, 320HD and 420HD IP Phones specifications are listed in the table below.
Table D-1: 310HD, 320HD and 420HD IP Phone Specifications
Feature
Details
VoIP Signaling Protocols
SIP: RFC 3261, RFC 2327 (SDP)
Data Protocols
IPv4, TCP, UDP, ICMP, ARP, DNS and DNS SRV for SIP Signaling
SIP over TLS (SIPS)
802.1p/Q for Traffic Priority and QoS
VLAN Discovery Mechanism (CDP, LLDP)
ToS (Type of Service) field, indicating desired QoS DHCP Client
NTP Client
Media Processing
Voice Coders: G.711, G.723.1, G.729A/B, G.722, RTA.
Acoustic Echo Cancelation: G.168-2004 compliant, 64-msec tail
length
Adaptive Jitter Buffer 300 msec
Voice Activity Detection
Comfort Noise Generation
Packet Lost Concealment
RTP/RTCP Packetization (RFC 3550, RFC 3551), SRTP (RFC
3711)
DTMF Relay (RFC 2833)
Telephony Features
Call Hold / Un-Hold
Call Transfer
3-Way Conferencing (with local mixing)
Redial
Caller ID Notification
Call Waiting Indication
Message Waiting Indication (including MWI LED)
Local and Corporate Directories
Automatic On-hook Dialing
Automatic Answering (Alert-Info header and "talk" event)
CWRR (Call Waiting Reminder Ring)
Secondary Dial Tone
Call Logs: Missed/Received Calls and Dialed Numbers
Speed Dial (Including BLF support for Asterisk and BroadWorksTM)
Dial Plan
URL Dialing
Advice of Charge (AOC)
Call Forward (Unconditional, Busy, No Answer)
Redundant SIP Proxy Mechanism
Remote Conferencing (RFC 4240)
Multiple Lines for 320HD (currently supports up to 4 lines)
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Feature
Details
Configuration/
Management
TR-069 for remote configuration and management
LCD Display User Interface Language Support (Various Languages)
Web-based Management (HTTP/HTTPS)
Auto-Provisioning (via TFTP, FTP, HTTP, and HTTPS) for firmware
and configuration file upgrade
DHCP options (66, 67, and 160) for Auto-provisioning
DHCP options (12, 60, and 77) for device information
Private labeling mechanism
Configuration file encryption (entire file and individual parameters)
Debugging Tools
Syslog mechanism
DSP recording
Port mirroring
VoIP Status Web page
310HD Hardware
LCD screen: 2 * 16 characters
Connectors interfaces:
2 x RJ-45 ports (10/100BaseT Ethernet) for WAN and LAN
RJ-9 port (jack) for Headset
RJ-9 port (jack) for Handset
Mounting:
Wall mounting
Adjustable angle tilt for desktop mounting
Power:
DC jack adapter 12V
Power supply AC 100 ~ 240V
PoE: IEEE802.3af (optional)
Keys:
10 x speed dial keys
VOICE MAIL message hotkey
4-way navigation keys with ENTER Key
MENU
DIRECTORY
REDIAL
HOLD
MUTE
TRANSFER
CONFERENCE
FORWARD
DND
VOLUME control key
HEADSET
SPEAKER
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D. IP Phone Specifications
Feature
Details
320HD Hardware
LCD screen: Graphic LCD (132*64) mono
Connectors interfaces:
2 x RJ-45 ports (10/100BaseT Ethernet) for WAN and LAN
RJ-9 port (jack) for Headset
RJ-9 port (jack) for Handset
Mounting:
Wall mounting
Adjustable angle tilt for desktop mounting
Power:
DC jack adapter 12V
Power supply AC 100 ~ 240V
PoE: IEEE802.3af (optional)
Keys:
12 x speed dial keys
Line 1 ~ 2 (LED)
4 x softkeys
VOICE MAIL message hotkey with LED indicator
4-way navigation keys with ENTER Key
MENU
REDIAL
HOLD
MUTE (LED)
TRANSFER
VOLUME control key
HEADSET
SPEAKER
420HD Hardware
LCD screen: Graphic LCD (132*64) mono
Connectors interfaces:
2 x RJ-45 ports (10/100BaseT Ethernet) for WAN and LAN
RJ-9 port (jack) for Headset
RJ-9 port (jack) for Handset
Mounting:
Wall mounting
Adjustable angle tilt for desktop mounting
Power:
DC jack adapter 12V
Power supply AC 100 ~ 240V
PoE: IEEE802.3af (optional)
Keys:
4 x softkeys
VOICE MAIL message hotkey
4-way navigation keys with ENTER Key
MENU
REDIAL
HOLD
MUTE
TRANSFER
VOLUME control key
HEADSET
SPEAKER
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Feature
Headset Compatibility
Details
GN.com 2100 – both single and dual
Jabra BIZ 2400 – both single and dual
GN.com 9350
Plantronics H261N
Plantronics H251N
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D. IP Phone Specifications
Reader's Notes
Version 2.0.0
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AudioCodes Series of High Definition IP Phones
HD VoIP
Administrator’s Manual
310HD / 320HD / 420HD IP Phones
Version 2.0.0
www.audiocodes.com
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