400HD Series IP Phone Administrator's Manual Ver. 2.2.16 (M20)

400 HD Series IP Phon e Administrator's Manual Ver. 2.2.16 (M20) - Generic SIP

Administrator's Manual

AudioCodes High-Definition IP Phones Series

405/405HD, 420HD, 430HD, 440HD IP

Phones

Version 2.2.16

Contents 
400HD Series IP Phones 
 
- ii - 
 
Table of Contents 
400HD Series IP Phone Administrator's Manual Ver. 2.2.16 (M20) - Generic SIP ..................1 
Notice ............................................................................................................................... xii 
WEEE EU Directive ........................................................................................................................ xii 
Customer Support ........................................................................................................................ xii 
Stay in the Loop with AudioCodes ................................................................................................ xii 
Abbreviations and Conventions .................................................................................................... xii 
Related Documentation ............................................................................................................... xiii 
Document Revision Record .......................................................................................................... xiii 
Documentation Feedback ............................................................................................................ xiv 
1  Introduction .................................................................................................................1 
Part I ...................................................................................................................................2 
Configuration Tools ............................................................................................................ 2 
2  IP Phone User Interface ................................................................................................3 
2.1  Accessing the Administration Menu ..................................................................................... 3 
2.2  Changing Display Language .................................................................................................. 4 
3  Web Interface ...............................................................................................................5 
3.1  Accessing Web Interface ...................................................................................................... 5 
3.2  Getting Started with the Web .............................................................................................. 6 
3.3  Configuring the Web Interface's Port ................................................................................... 7 
3.4  Configuring User Login Credentials ...................................................................................... 7 
4  Configuration File .........................................................................................................8 
4.1  Introduction .......................................................................................................................... 8 
4.2  File Syntax............................................................................................................................. 8 
4.3  Linking Multiple Files ............................................................................................................ 9 
4.4  Downloading the Configuration File from the Phone ........................................................... 9 
4.5  Creating Configuration Files using VoIProvision Utility ......................................................... 9 
4.5.1  Configuration File Format ................................................................................................. 10 
4.5.2  Global Configuration File .................................................................................................. 10 
4.5.3  VoIProvision Utility Overview ........................................................................................... 10 
4.5.4  CSV File ............................................................................................................................ 10 
4.5.5  Template File ................................................................................................................... 11 
4.5.6  Generated Configuration Files .......................................................................................... 11 
4.5.7  Starting the VoIProvision Utility........................................................................................ 11 
4.5.8  Usage ............................................................................................................................... 11 
4.6  Using the Encryption Tool .................................................................................................. 12 
4.6.1  Encrypting Configuration Files .......................................................................................... 12 

Contents 
400HD Series IP Phones 
 
- iii - 
 
6.2  Encrypting Passwords in the Configuration File ................................................................ 12 
Device Manager Pro ................................................................................................... 13 
Part II ................................................................................................................................ 14 
Automatic Provisioning .................................................................................................... 14 
Introduction ............................................................................................................... 15 
Updating the Configuration File Manually .................................................................. 16 
Setting up Network for Auto Provisioning ................................................................... 17 
Obtaining Firmware and Configuration Files ............................................................... 18 
1  Provisioning Hunt Order ..................................................................................................... 18 
2  Dynamic URL Provisioning .................................................................................................. 18 
2.1  Provisioning using DHCP Option 160 ................................................................................ 21 
2.2  Technician's Digit Key Code .............................................................................................. 21 
2.3  Provisioning using DHCP Option 66/67 ............................................................................. 22 
2.4  Provisioning using DHCP Option 43 .................................................................................. 23 
2.5  Provisioning using the User-Class Option.......................................................................... 24 
2.6  SIP SUBSCRIBE and NOTIFY Messages............................................................................... 32 
2.7  Hardcoded Domain Name for Provisioning Server ............................................................ 34 
2.8  Cached Address of Last Provisioning Server Used ............................................................. 34 
2.9  Redirect Server ................................................................................................................. 35 
3  Static URL Provisioning ....................................................................................................... 36 
Part III ............................................................................................................................... 38 
Quick Setup ...................................................................................................................... 38 
Quick Setup ................................................................................................................ 39 
Part IV ............................................................................................................................... 40 
Networking ...................................................................................................................... 40 
Introduction ............................................................................................................... 41 
Configuring Date and Time Manually .......................................................................... 42 
1  Configuring Daylight Saving Time ....................................................................................... 43 
2  Configuring the NTP Server ................................................................................................ 46 
3  Configuring NTP Server via DHCP ....................................................................................... 48 
Configuring IP Network Settings ................................................................................. 49 
1  Configuring Static IP Address .............................................................................................. 49 
1.1  Configuring Static IP Address on the Phone ...................................................................... 49 
1.2  Configuring IP Network Settings ....................................................................................... 50 
2  Configuring Partial DHCP .................................................................................................... 52 

Contents 
400HD Series IP Phones 
 
- iv - 
 
Configuring LAN and PC Port Settings ......................................................................... 53 
Configuring VLAN Settings .......................................................................................... 54 
1  Configuring Manual or Automatic VLAN Assignment ......................................................... 56 
1.1  Configuring Manual VLAN Assignment to the Phone ........................................................ 56 
1.2  Configuring Automatic VLAN Assignment to the Phone .................................................... 56 
1.3  Configuring VLAN via DHCP Provisioning Path .................................................................. 56 
Part V ................................................................................................................................ 57 
VoIP Settings .................................................................................................................... 57 
Configuring SIP Settings .............................................................................................. 58 
1  Configuring General SIP Settings ........................................................................................ 58 
2  Configuring Proxy and Registration .................................................................................... 62 
2.1  Configuring Proxy Redundancy ......................................................................................... 64 
2.2  Device Registration Failover/Failback ............................................................................... 67 
2.2.1  Failover ............................................................................................................ 67 
2.2.2  Failback ............................................................................................................ 68 
2.3  Preventing Unregistering after Changing Settings and Reloading ..................................... 68 
3  Configuring a Line ............................................................................................................... 69 
4  Configuring Shared Call Appearance .................................................................................. 71 
5  Configuring SIP Timers ........................................................................................................ 72 
6  Configuring SIP QoS ............................................................................................................ 74 
7  Configuring SIP Reject Code ............................................................................................... 74 
Configuring Dialing ..................................................................................................... 75 
1  Configuring Voice Dialing through VocaNOM ..................................................................... 75 
2  Configuring General Dialing Parameters ............................................................................ 77 
3  Configuring Auto Redial ...................................................................................................... 79 
4  Configuring Dial Tones ........................................................................................................ 80 
5  Configuring DTMF ............................................................................................................... 82 
6  Configuring Digit Maps and Dial Plans ................................................................................ 83 
7  Configuring Headset LED to Stay On ................................................................................... 85 
8  Configuring Default Audio Device ....................................................................................... 86 
Configuring Ring Tones ............................................................................................... 87 
1  Configuring Distinctive Ring Tones ..................................................................................... 87 
1.1  Example of Configuring a Distinctive Ring ......................................................................... 88 
2  Configuring CPT Regional Settings ...................................................................................... 89 
3  Uploading Ring Tones ......................................................................................................... 91 
4  Configuring Beeps to Headsets when a Call Comes in to a Call Center ............................... 92 
5  Configuring the Phone to play Fast Busy Tone if Automatically Disconnected on Remote 
Side  93 

Contents 
400HD Series IP Phones 
 
- v - 
 
6  Configuring the Beep (Ring) to Play via an Answering Device ............................................ 94 
Configuring Media Settings ......................................................................................... 95 
1  Configuring Media Streaming ............................................................................................. 95 
2  Configuring RTP Port Range and Payload Type ................................................................... 96 
3  Configuring RTP QoS ........................................................................................................... 97 
4  Configuring RTP/SRTP Capability Negotiation .................................................................... 97 
5  Configuring Codecs ............................................................................................................. 98 
6  Configuring OPUS Management ....................................................................................... 100 
Configuring Voice Settings ........................................................................................ 101 
1  Configuring Gain Control .................................................................................................. 101 
2  Configuring Jitter Buffer ................................................................................................... 101 
3  Configuring Silence Compression ..................................................................................... 102 
4  Configuring Noise Reduction ............................................................................................ 103 
5  Configuring Echo Cancellation .......................................................................................... 104 
Configuring Extension Lines ...................................................................................... 105 
1  On the Phone .................................................................................................................... 105 
2  Using the Web Interface and Configuration File ............................................................... 106 
Configuring Supplementary Services ......................................................................... 108 
1  Selecting the Application Server ....................................................................................... 108 
2  Configuring Call Waiting ................................................................................................... 109 
3  Configuring Call Forwarding ............................................................................................. 110 
4  Configuring a Conference ................................................................................................. 111 
5  Allowing the Initiator to Drop out of a Conference .......................................................... 111 
6  Configuring Automatic Dialing .......................................................................................... 112 
7  Configuring Automatic Answer ......................................................................................... 113 
8  Configuring Do Not Disturb (DnD) .................................................................................... 115 
9  Configuring Call Pick Up .................................................................................................... 116 
10 Configuring Message Waiting Indication .......................................................................... 117 
11 Configuring Busy Lamp Field............................................................................................. 118 
12 Configuring Advice of Charge ........................................................................................... 119 
13 Configuring a Tone to Alert to Long Hold ......................................................................... 119 
14 Disabling the HOLD Key .................................................................................................... 120 
15 Configuring Onhook Disconnect when Held ..................................................................... 120 
16 Configuring Ringing on the Default Audio Device ............................................................. 121 
17 Allowing an Incoming Call when the Phone is Locked ...................................................... 122 
18 Allowing Call Center Agents to Record Welcome Greetings ............................................. 122 
19 Enabling the Electronic Hook Switch ................................................................................ 123 

Contents 
400HD Series IP Phones 
 
- vi - 
 
20 Disabling the Hard Mute Key on the Phone ...................................................................... 124 
21 Configuring Attended and Semi-Attended Call Transfer ................................................... 124 
22 Configuring Blind Transfer ................................................................................................ 125 
23 Creating a Speed Dial File for Configuration File .............................................................. 125 
Configuring Volume Levels ....................................................................................... 126 
1  Configuring Gain Control .................................................................................................. 126 
2  Configuring Tone Volume ................................................................................................. 128 
3  Configuring Ringer Volume ............................................................................................... 128 
4  Configuring Speaker Volume ............................................................................................ 129 
5  Configuring Handset Volume ............................................................................................ 132 
6  Configuring Headset Volume ............................................................................................ 134 
Part VI ............................................................................................................................. 136 
Advanced Phone Settings ................................................................................................ 136 
Configuring the Phone Directory ............................................................................... 137 
1  Configuring the Corporate Directory ................................................................................ 137 
1.1  Configuring the LDAP-based Corporate Directory ........................................................... 137 
1.2  Loading a Text-based Corporate Directory File ............................................................... 140 
2  Modifying the Local Phone Directory ............................................................................... 141 
Configuring Keys ....................................................................................................... 142 
1  Configuring Function Keys ................................................................................................ 142 
1.1  430HD Phone ................................................................................................................. 142 
1.2  440HD Phone ................................................................................................................. 144 
1.3  Configuring Additional Function Keys ............................................................................. 145 
2  Configuring Programmable Keys ...................................................................................... 145 
2.1  430HD and 440HD Phones ............................................................................................. 145 
2.1.1  Configuring Multiple Extension Lines ............................................................. 146 
2.1.2  Configuring a Key Event ................................................................................. 148 
3  Configuring Speed Dials .................................................................................................... 149 
3.1  420HD and 405/405HD Phone Models ........................................................................... 149 
3.2  Deleting Speed Dials ....................................................................................................... 150 
3.3  Saving Configured Speed Dials........................................................................................ 150 
3.4  Creating a Speed Dial File for the Configuration File ....................................................... 151 
4  Configuring Softkeys ......................................................................................................... 152 
4.1  Configuring Programmable Softkeys (PSK)...................................................................... 155 
5  Configuring Navigation Control Button Positions ............................................................. 156 
5.1  Saving Configured Keys .................................................................................................. 157 
5.2  Loading Saved Keys to Phones ........................................................................................ 157 

Contents 
400HD Series IP Phones 
 
- vii - 
 
Disabling Hard Keys and Softkeys ............................................................................. 158 
Configuring Paging .................................................................................................... 160 
1  Configuring a Key for Paging using the Web Interface ..................................................... 160 
1.1  Configuring Barge-in ....................................................................................................... 161 
2  Configuring Paging Using the Configuration File .............................................................. 162 
Configuring Feature Key Synchronization.................................................................. 163 
Part VII ............................................................................................................................ 164 
Security ........................................................................................................................... 164 
Implementing X.509 Authentication ......................................................................... 165 
1  Factory-Set Certificates and AudioCodes Trusted Root CA ............................................... 166 
2  User-Generated Certificates ............................................................................................. 166 
3  External Trusted Root CAs ................................................................................................ 167 
Loading a Certificate ................................................................................................. 169 
1  Loading the Trusted Root CA Certificate to the Phone ..................................................... 169 
1.1  Loading Trusted Root CA Certificate Using Configuration File ......................................... 169 
2  Loading the Client Certificate to the Phone ...................................................................... 170 
2.1  Loading the Client Certificate to the Phone using the Configuration File ........................ 170 
2.2  Enabling Server-side Authentication (Mutual Authentication) ........................................ 171 
3  Generating a Certificate Signing Request ......................................................................... 171 
4  Using Previously Loaded Certificates ................................................................................ 172 
Configuring SIP TLS ................................................................................................... 173 
1  Configuring TLS ................................................................................................................. 173 
1.1  Configuring SIP TLS using the Web Interface .................................................................. 174 
Configuring 802.1x .................................................................................................... 175 
1  Configuring 802.1x in the Phone Screen ........................................................................... 175 
1.1  Configuring EAP-MD5 Mode ........................................................................................... 176 
1.2  Configuring EAP-TLS Mode ............................................................................................. 176 
2  Configuring 802.1x Using Web and Configuration File ..................................................... 177 
2.1  Configuring EAP MD5 Mode ........................................................................................... 177 
2.2  Configuring EAP TLS Mode ............................................................................................. 178 
Configuring SRTP ...................................................................................................... 179 
Configuring HTTP/S................................................................................................... 181 
Logging into a Remote HTTP/S Server from the Phone .............................................. 182 
Securing the Web Interface using HTTP/S ................................................................. 183 
1  Provisioning ...................................................................................................................... 184 

Contents 
400HD Series IP Phones 
 
- viii - 
 
MAC-Based Authentication....................................................................................... 185 
Configuring HELD ...................................................................................................... 186 
Part VIII ........................................................................................................................... 187 
Maintenance ................................................................................................................... 187 
Changing Administrator Login Credentials ................................................................ 188 
Restarting Phones..................................................................................................... 189 
1  Restarting from the Phone ............................................................................................... 189 
2  Restarting the Phone using the Web Interface ................................................................. 189 
Restoring Phone Defaults ......................................................................................... 190 
1  Restoring Factory Defaults from the Phone's Screen ....................................................... 190 
2  Restoring Factory Defaults using the Web Interface ........................................................ 191 
Part IX ............................................................................................................................. 192 
Status and Monitoring..................................................................................................... 192 
Determining Network Status .................................................................................... 193 
1  Determining LAN Status ................................................................................................... 193 
2  Determining Port Status ................................................................................................... 193 
3  Determining 802.1x Status ............................................................................................... 193 
Determining VoIP Status ........................................................................................... 194 
1  Determining Phone Status ................................................................................................ 194 
2  Determining Line Status ................................................................................................... 194 
3  Determining Memory Status ............................................................................................ 195 
4  Viewing Current Call Information ..................................................................................... 196 
Viewing Call History .................................................................................................. 197 
Accessing System Information .................................................................................. 198 
1  Accessing Phone Firmware Version .................................................................................. 198 
1.1  Accessing Firmware Version using the Web Interface ..................................................... 198 
1.2  Accessing Firmware Version from the Phone's Screen .................................................... 198 
2  Viewing Phone Firmware Release Information ................................................................ 199 
2.1  Viewing Firmware Release Information in the Web Interface ......................................... 199 
2.2  Viewing Firmware Release Information on the Phone .................................................... 199 
Monitoring Quality of Experience ............................................................................. 200 
1  Configuring Remote Voice Quality Monitoring ................................................................. 200 
1.1  Configuring RTCP Extended Report................................................................................. 201 
1.2  Configuring Voice Quality Monitoring ............................................................................ 202 

Contents 
400HD Series IP Phones 
 
- ix - 
 
Part X .............................................................................................................................. 203 
Diagnostics and Troubleshooting..................................................................................... 203 
47  Diagnosing Phone Hardware ..................................................................................... 204 
47.1  Testing Keypad and Hook ................................................................................................. 205 
47.2  Testing Handset ................................................................................................................ 205 
47.3  Testing the Headset .......................................................................................................... 205 
47.4  Testing Hands Free ........................................................................................................... 205 
48  Recovering Firmware ................................................................................................ 206 
49  Configuring System Logging (Syslog) ......................................................................... 207 
49.1  Analyzing and Debugging Traffic using Regular Syslog ..................................................... 207 
49.2  Analyzing and Debugging Traffic using 'Lightweight Syslog' ............................................. 209 
50  Viewing Error Messages Displayed in the Phone Screen............................................ 210 
51  Debugging using Packet Recording Parameters ......................................................... 211 
52  Creating a Crash Dump File ....................................................................................... 213 
53  Configuring Port Mirroring........................................................................................ 214 
54  Enabling Tracing ....................................................................................................... 215 
Part XI ............................................................................................................................. 217 
Appendices ..................................................................................................................... 217 
A  Configuring Phones in Server-Specific Deployments ................................................. 218 
A.1  BroadSoft's BroadWorks................................................................................................... 218 
A.1.1  Configuring BLF .............................................................................................................. 218 
A.1.2  Configuring Call Forwarding ........................................................................................... 220 
A.1.2.1  From the Phone ............................................................................................. 220 
A.1.2.2  From BroadSoft's BroadWorks ....................................................................... 220 
A.1.3  Configuring DnD ............................................................................................................. 221 
A.1.4  Configuring FKS .............................................................................................................. 222 
A.1.5  Using SIP Authentication for Xsi Access .......................................................................... 222 
A.1.6  Configuring Phones to Connect to the Xsi Interface using HTTP/S Authentication .......... 223 
A.1.7  Configuring Shared Call Appearance ............................................................................... 225 
A.1.8  Setting up a Remote Conference .................................................................................... 228 
A.1.9  Loading the Corporate Directory to the Phone ............................................................... 229 
A.1.10  Adding a Contact to the Corporate Directory.................................................................. 230 
A.1.11  Deleting a Contact from the Corporate Directory ........................................................... 230 
A.1.12  Disabling Handset Mode ................................................................................................ 230 
A.1.13  Displaying a Message in Agents' Phone Screens ............................................................. 231 
A.1.14  Changing Phone Screen Backlight Timeout ..................................................................... 231 
A.2  Asterisk, Coral and Metaswitch ........................................................................................ 232 

Contents 
400HD Series IP Phones 
 
- x - 
 
A.2.1  Configuring BLF .............................................................................................................. 232 
A.3  Genesys SIP Server for Contact Centers ........................................................................... 233 
A.3.1  Using DHCP to Auto Provision Phones ............................................................................ 233 
A.3.2  Verifying Firmware Version ............................................................................................ 233 
A.3.3  Accessing a Phone's Web Interface ................................................................................ 233 
A.3.4  Configuring Dual Registration to Ensure SIP Business Continuity for Agents ................... 234 
A.3.5  Disabling the Web Interface ........................................................................................... 237 
A.3.6  Forcing a Reboot on Provisioning ................................................................................... 238 
A.3.7  Provisioning using TFTP / FTP / HTTP / HTTPS in DHCP Options 66/67 ............................ 238 
A.3.8  Enabling Agents to Sign in with Phone Numbers ............................................................ 238 
A.3.9  Locking Agents' Phones' Alphabetical Keys ..................................................................... 239 
A.3.10  Playing a Beep on an Incoming Call ................................................................................ 239 
A.3.11  Enabling Proactive Mute ................................................................................................ 240 
A.3.12  Configuring Automatic Answer ....................................................................................... 240 
A.3.13  Regulating the 'Logged out' Message ............................................................................. 241 
A.3.14  3PCC (Third Party Call Control) ....................................................................................... 241 
A.3.14.1  Enabling 3PCC Calls ........................................................................................ 242 
A.3.15  Disabling Handset Mode ................................................................................................ 243 
A.3.16  Changing Phone Screen Backlight Timeout ..................................................................... 243 
A.3.17  Displaying a Message on Agents' Phones........................................................................ 244 
A.3.18  Configuring a Redundant (Backup) Genesys Server ........................................................ 244 
A.3.18.1  Configuring Retransmission Timer T1 Using the Configuration File................. 245 
A.4  Genband: KBS Softswitch Solution ................................................................................... 245 
A.4.1  Configuring Shared Line Appearance .............................................................................. 246 
A.4.2  Configuring Call Pickup ................................................................................................... 248 
A.4.3  Setting up a Remote Conference .................................................................................... 251 
A.4.4  Configuring Parameters Mandatory for Phones in a Genband Environment ................... 252 
B  Configuring Automatic Call Distribution (ACD) .......................................................... 253 
B.1  Softkey Display and Command Menu Options ................................................................. 257 
C  Recovering AudioCodes' IP Phone............................................................................. 258 
C.1  Identifying that the Phone is in Recovery Mode ............................................................... 258 
C.2  Verifying that the Phone is in Recovery Mode ................................................................. 259 
C.3  Recovering the Phone....................................................................................................... 260 
C.4  Verifying that the Phone is Downloading the Image File .................................................. 262 
C.4.1  Verifying Using Wireshark .............................................................................................. 262 
C.4.2  Verifying Using tftpd64................................................................................................... 264 
C.4.3  Verifying on the Phone ................................................................................................... 264 
D  Deploying AudioCodes IP Phones - Use Case ............................................................. 266 
D.1  Preparing Configuration (cfg) Files for the Enterprise Customer ...................................... 266 
D.1.1  Saving the Phone's Default Configuration to File ............................................................ 266 

Contents

400HD Series IP Phones

- xi -

D.1.2 Preparing a global.cfg Configuration File ........................................................................ 267

D.1.3 Generating MAC-specific <private>.cfg Configuration Files............................................. 267

D.1.3.1 Preparing a csv File ........................................................................................ 267

D.1.3.2 Preparing a Template File .............................................................................. 268

D.1.3.3 Using AudioCodes' VoIProvision Tool ............................................................. 269

D.2 Preparing the DHCP Server to Automatically Provision Phones ....................................... 270

D.3 Making Sure Phones are Correctly Provisioned ................................................................ 270

E Supported SIP RFCs and Headers .............................................................................. 271

E.1 SIP Compliance Tables ...................................................................................................... 272

E.1.1 SIP Methods ................................................................................................................... 272

E.1.2 SIP Headers .................................................................................................................... 273

F Specifications ........................................................................................................... 274

G RTCP-XR Parameters ................................................................................................. 277

H Example SIP - PUBLISH Message ............................................................................... 279

Notices

400HD Series IP Phones

- xii -

Notice

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 can be downloaded from

https://www.audiocodes.com/library/technical-documents.

This document is subject to change without notice.

Date Published: March-18-2024

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 services are provided by AudioCodes or by an authorized

AudioCodes Service Partner. For more information on how to buy technical support for AudioCodes

products and for contact information, please visit our website at

https://www.audiocodes.com/services-support/maintenance-and-support.

Stay in the Loop with AudioCodes

Abbreviations and Conventions

Each abbreviation, unless widely used, is spelled out in full when first used.

Notices

400HD Series IP Phones

- xiii -

Related Documentation

Document Name

440HD IP Phone User’s Manual

440HD IP Phone Quick Guide

430HD IP Phone User’s Manual

430HD IP Phone Quick Guide

420HD IP Phone User’s Manual

420HD IP Phone Quick Guide

405 and 405HD IP Phone User’s Manual

405 and 405HD IP Phone Quick Guide

Device Manager Pro Administrator’s Manual

One Voice Operations Center (OVOC) IOM Manual

OVOC User’s Manual

Document Revision Record

LTRT

Description

11947

Version 2.2.2.

11948

Version 2.2.4. 405. CC features: Supervisor Listen, Select Ring Audio Device, Disable Hands-Free

Mode, Greeting Recording, BroadSoft-based ACD Hoteling, SHA2, Blind Transfer, Drop From Local

Conference, Factory-Set Certificates and AudioCodes Trusted Root CA, Factory-Installed Certificates

Status Displayed, Send DTMF via SIP and via RTP Together, HTTP/S Provisioning, CDP Enhanced,

Restoring Phone Settings to Defaults; Slovak, Czech, Turkish.

11949

Version 2.2.8 – preliminary. Proxy and Registrar parameter values, Headset LED, ring-tone, RTP

Port Range (media_port), Codec Type updated, Media Streaming – Codecs updated, DnD Activate,

Function Keys pgs, Firmware Release Information, RTCP-XR, Disabling Handset Mode, Displaying a

Message in Agents’ screens, hide ACW softkey, 3PCC restored.

11950

Version 2.2.8 – official. Multiple lines. Dual registration (Genesys), 3PCC. Voip/talk_event.

Unregister_on_voip_reload. Lightweight Syslog. Recovering phone.

11951

Version 2.2.12. 3DES. Multiple Lines. System/syslog/mode. Locking A-B keys – applicability. User-

Class. Distinctive Ring Tone. 405HD. 430HD with high resolution screen.

11953

Version 2.2.14. Call Log sync. Technician’s Digit Key Code. XSI. EMS > OVOC server. Removed

Redundant Outbound proxy; new note recommending DNS queries. OPUS payload.

Auto_answer_use_180. Paging. Barge-in. voip/services/group_paging/codec.

Voip/services/group_paging/end_income_paging_timeout. System Logging. BroadWorks Xsi over

HTTP/S.

11955

SIP SUBSCRIBE and NOTIFY Messages. USB Headsets. VocaNOM. Disable HOLD key. Disable mute

key. Disabling Hard Keys and Softkeys. SRTP. Backlight timeout. USB headsets.

11956

PAI on Replay. Automatic Answer: new Note. Tone after long hold. Client Certificate: modified

descriptions. Server-side Authentication: modified descriptions. Plantronics link.

Notices

400HD Series IP Phones

- xiv -

LTRT

Description

11959

Preventing Unregistering. DTMF level. Voip/media/srtp/NegotiationMode. SRTCP. Device Manager

Pro. USB Headset. Recovery Mode notification. DIGICert. Genband support. Remove the ‘lifetime’

parameter. OpenSSL. ‘provisioning/period/type’ description improved.

11970

personal_settings/menu/callautoanswering/enabled. Configuring parameters mandatory for

Genband.

11971

voip/ring_via_answer_device/enabled

11972

Beep (Ring) via Answering Device. Network/lan/_802_1x/eap_type.

Network/lan/_802_1x/eap_identity. Network/lan/_802_1x/eap_identity.

Voip/services/sla/barging/enable. Voip/sla/group/0/description. Voip/sla/group/0/enabled.

Voip/sla/group/1/description. Voip/sla/group/1/enabled. Combined ACD/Hoteling.

Voip/services/enhanced_ACD/enabled.

11974

RTP/SRTP Capability Negotiation

11975

Removed a server

11976

Fix to UN & PWD parameters

09966

Added SIP instance

09969

Added Headset restriction and fixed document formatting

09970

Changed EAP TLS Parameter

09973

Added Configuration File Parameter

09976

Typo in Primary_Fallback

Documentation Feedback

AudioCodes continually strives to produce high quality documentation. If you have any comments

(suggestions or errors) regarding this document, please fill out the Documentation Feedback form

on our Website at https://online.audiocodes.com/documentation-feedback.

1. Introduction

400HD Series IP Phones

- 1 -

1 Introduction

This manual is intended for the system administrator responsible for setting up and configuring the

420HD, 430HD, 440HD, 405 and 405HD 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 screen - easy-to-use, menu-driven 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.

◼ Device Manager Pro (see the Device Manager Pro Administrator's Manual).

For a detailed description on hardware installation and for operating the phone's call features, see

the User's Manual.

• Boldened parameters enclosed in square brackets [ ] in the tables in this document

indicate configuration file parameters.

• Web interface parameters are displayed in regular font above their web counterparts.

Part I Configura tion Tools

Part I

Configuration Tools

2. IP Phone User Interface

400HD Series IP Phones

- 3 -

2 IP Phone User Interface

The 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.

Screenshots of the 440HD are shown in this document. Screenshots of the 420HD and

405/405HD are only shown if they significantly differ.

2.1 Accessing the Administration Menu

Here's how to access the Administration menu on the phone.

• The phone is password protected. The default password is 1234. To change the login

password, use the phone's Web interface or Configuration file.

• After entering the password, the access session is applied to all the submenus.

• To change the Administration screen's login password, use the phone's Web interface

or use the configuration file.

To access the Administration screen:

1. Press the MENU key on the phone and navigate down to Administration.

Alternatively, after pressing the MENU key you can press an item's number to navigate to

the item, for example, press 6 to navigate to Administration.

2. Press Select; you're prompted for a password.

3. Enter the administration password (Default: 1234) and then press the OK softkey.

2. IP Phone User Interface

400HD Series IP Phones

- 4 -

2.2 Changing Display Language

Here's how to change the language in the phone screen. Language can be configured using the Web

interface or configuration file.

To choose a language using the Web interface:

1. Open the Language page (Configuration tab > Personal Settings menu > Language).

Figure 1: Language

2. Select the language according to the parameter in the table below, and then click Submit; the

phone reboots and changes the screen language.

To choose a language using the configuration file:

◼ Use the table below as reference.

Table 1: Language Display Parameters

Parameter

Description

Phone Display Language

[personal_settings/language]

Determines the screen language. See the Release

Notes for the list of languages supported.

[personal_settings/lcd_contrast]

Determines the screen contrast. Configure to a

level that is comfortable for the user.

Range: 0-35 (430HD / 440HD) and

0-30 (420HD). The default depends on the

hardware revision.

[personal_settings/blf_lcd_contrast]

Applies only to 440HD phone. Determines the

sidecar BLFs contrast. Configure to a level

comfortable for the user.

Range: 0-39. Default: 16.

3. Web Interface

400HD Series IP Phones

- 5 -

3 Web Interface

This section describes the phone's Web interface. You can use the Web interface to configure the

device.

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.

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 2: Web Interface Login

The administrator's default login user name and password are admin and 1234

respectively.

4. Alternatively, if your DHCP and DNS servers are synchronized, you can access the phone Web

browser by using the following method:

http://<Phone Model>-<MAC Address>.<Domain Name>

E.g., http://440hd-001122334455.corp.YourCompany.com

5. Enter the User name and Password, and then click OK.

3. Web Interface

400HD Series IP Phones

- 6 -

3.2 Getting Started with the Web

The areas of the Web interface are shown below:

Figure 3: 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.

Toolbar

3. Web Interface

400HD Series IP Phones

- 7 -

3.3 Configuring the Web Interface's Port

Here's how to assign a port number to the Web interface, using the configuration file.

To configure the Web interface port using the configuration file:

◼ Use the table below as reference.

Table 2: Port Parameters

Parameter

Description

[system/http_server_port]

Assigns a port number to the Web interface.

The HTTP server by default uses port number 80. Range: 0-

65535.

[system/https_server_port]

Assigns a port number to the Web interface.

The HTTPS server by default uses port number 443. Range: 0-

65535.

3.4 Configuring User Login Credentials

Here's how to configure the phone user's name and password.

To configure the phone user's name and password using the Web interface:

1. Open the User Account page (Management menu > Administration > Users):

Figure 4: User Account

2. Configure using the table below as reference, and click Submit.

To configure using the configuration file:

◼ Use the table below as reference.

Table 3: User Name and Password Parameters

Parameter

Description

Username

[system/web_user_name]

The user account’s username. Default: user.

Applies only to the Web interface.

Password

[system/web_user_password]

The encrypted phone password. Default: 1234.

Applies only to the Web interface, and phone screen.

4. Configuration File

400HD Series IP Phones

- 8 -

4 Configuration File

This section describes the configuration file and the parameters you can configure in it.

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. The subsections below describe

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:

Make sure the configuration file conforms to these 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=440HD

voip/line/0/enabled=1

voip/line/0/id=1234

voip/line/0/description=440HD

voip/line/0/auth_name=1234

voip/line/0/auth_password=4321

4. Configuration File

400HD Series IP Phones

- 9 -

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=440HD

include 440HD_<MAC>_voip.cfg

include vlan_conf.cfg

include network_conf.cfg

include provisioning_conf.cfg

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=440HD

include http://10.10.10.10/440HD_<MAC>_voip.cfg

include https://remote-pc/vlan_conf.cfg

include tftp://10.10.10.10/440HD_<MAC>_network.cfg

include ftp://remote-pc/provisining_conf.cfg

If no URL is provided in the Configuration file, the files are retrieved according to the

provisioning information (e.g., DHCP Option 160 as well as Option 66/67).

4.4 Downloading the Configuration File from the Phone

For more information, see Maintenance.

4.5 Creating Configuration Files using VoIProvision Utility

When installing AudioCodes' phones, the network administrator typically configures each installed

phone automatically. Using DHCP options or other methods, the phone can be instructed to

download a configuration file. This file is typically unique to each phone, based on the MAC address.

This MAC-specific configuration file is generated with 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. As an IP phone vendor, AudioCodes is required to

provide a standalone provisioning tool that will enable the provisioning of its phones in such

environments.

AudioCodes provides a tool that assists in the automatic generation of configuration files. These files

can be generated for the initial configuration of the phones and then later regenerated for

subsequent configuration updates as required.

4. Configuration File

400HD Series IP Phones

- 10 -

4.5.1 Configuration File Format

The detailed format of the phones' configuration files are described in the appendix. The following

is an output example of an automatically generated MAC-specific file:

system/type=440HD

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

4.5.2 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 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.

4.5.3 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 phone.

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 phone and the template file maps these tagged records to a

configuration file format, which can be read by the phone.

4.5.4 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 phone.

The csv file is usually exported from the customer's IP-PBX or some other database and typically

contains the list of phones (e.g., MAC, extension ID, user name and password of each phone).

Table 4: 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. Configuration File

400HD Series IP Phones

- 11 -

4.5.5 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=440HD

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

4.5.6 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.

4.5.7 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

4.5.8 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,Ethan).

◼ There is no restriction on the format of the tags (e.g., [tag] or @tag@).

◼ 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 to generate

a separate .cfg file per csv line record (e.g., [mac].cfg).

4. Configuration File

400HD Series IP Phones

- 12 -

4.6 Using the Encryption Tool

AudioCodes' phones use the Triple Data Encryption Standard (3DES) algorithm for encryption. Here's

how to use the encryption tool.

4.6.1 Encrypting Configuration Files

Here's 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').

4.6.2 Encrypting Passwords in the Configuration File

Here's how to encrypt 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="}

It's recommended to encrypt the 'System' password using this procedure. If you choose

not to, the 'System' password is by default encrypted using MD5.

5. Device Manager Pro

400HD Series IP Phones

- 13 -

5 Device Manager Pro

Network administrators can provision an enterprise's phones from the server of the One Voice

Operations Center (OVOC) module, Device Manager Pro.

◼ Device Manager Pro and OVOC share the same server location.

◼ For more information on using Device Manager Pro to provision phones, see the

Device Manager Pro Administrator's Manual.

To configure provisioning phones from the OVOC server using the configuration file:

1. Open the Configuration File page in the Web interface (Management tab > Manual Update >

Configuration File).

2. Configure the OVOC server parameters using the table below as reference.

Table 5: OVOC Server Parameters

Parameter

Description

[ems_server/keep_alive_period]

The OVOC server sends a keep alive message at a configured

interval to verify that its link with the network is operating. If

no reply is received, the link is determined to be down or not

working.

Default: 60 minutes

[ems_server/provisioning/url]

Defines the URL of the OVOC server, for example,

http://10.1.8.23:8081

[ems_server/user_name]

Defines the username of the administrator who'll use the

OVOC server for provisioning, for example, John Smith.

[ems_server/user_password]

Defines the password (encrypted) of the network

administrator who'll provision the phones from the OVOC

server, for example: {"Y6QYmP53BDkoTvuIFjEBuQ=="}

AudioCodes’ Device Manager Pro also supports headset management capability in addition

to phone management capability. IGS-1071The management interface displays (for

example) in the Devices Status page:

◼ Column 'USB Headset Type' displaying a headset connected to a phone’s USB port

◼ Column 'IPP Model' displaying the USB icon

For more information on the Device Manager Pro, see the Device Manager Pro

Administrator's Manual.

Part II Automatic Prov isioning

Part II

Automatic Provisioning

400HD Series IP Phone Administrator's Manual Ver. 2.2.16 (M20) - Generic SIP 6. Introduction

- 15 -

6 Introduction

By default, the phone is ready for out-of-the-box deployment using its automatic provisioning

capabilities.

The 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.

◼ In the DHCP Discover message, the phone publishes its model name in Option fields

60 and 77 (e.g., 440HD). 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 is powered off for some reason during the firmware upgrade process, the

phone will be unusable and the recovery process must be performed.

◼ 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).

Automatic mass provisioning of phones using DHCP can alternatively be performed from

the OVOC’s IP Phones Manager Pro module. For more information, see the Device Manager

Pro Administrator's Manual.

7. Updating the Configuration File Manually

400HD Series IP Phones

- 16 -

7 Updating the Configuration File Manually

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. Open the Configuration File page (Management tab > Manual Update menu > Configuration

File). The current configuration file settings are displayed in the text pane.

Figure 5: Configuration File (430HD)

2. Click the Loading new Configuration File button; the following page appears:

Figure 6: Load New Configuration File

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.

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.

When creating a new configuration file, make sure the system/type parameter in the

configuration file is set to the correct phone model (e.g., 430HD).

8. Setting up Network for Auto Provisioning

400HD Series IP Phones

- 17 -

8 Setting up Network for Auto Provisioning

The phone supports dynamic VLAN discovery, dynamic IP addressing (DHCP), and NTP (as client).

For manual configuration of Network Settings, see Section 13.

9. Obtaining Firmware and Configuration Files

400HD Series IP Phones

- 18 -

9 Obtaining Firmware and Configuration Files

The Web interface allows you to:

◼ Automatically update firmware and configuration files

◼ Manually update firmware and configuration files

9.1 Provisioning Hunt Order

The phone always attempts to use the first provisioning method listed below (DHCP Option 160). If

it cannot use this method, it attempts to use the second method listed below, and so on, until it

reaches a successful provisioning method. This is called the provisioning 'hunt order'. The 'hunt

order' is:

1. DHCP Option 160 (see Section 9.2.1)

2. DHCP Options 66-67 (see Section 9.2.2)

3. DHCP Options 43 (see Section 9.2.4)

4. SIP SUBSCRIBE and NOTIFY Messages (see Section 9.2.6)

5. Static and Globally Accessible Domain (see Section 9.2.7)

6. Cached Addresses of the Last Provisioning Server Used on Reboots (see Section 9.2.8)

7. AudioCodes Redirect server (see Section 9.2.9)

9.2 Dynamic URL Provisioning

Dynamic Host Configuration Protocol (DHCP) can be used to automatically provision the phone. The

DHCP feature can be configured using the Web interface or configuration file.

To configure DHCP using the Web interface:

1. Open the Automatic Update page (Management tab > Automatic Update menu > Automatic

Provisioning).

Figure 7: Automatic Provisioning – Dynamic URL

2. Configure the parameters using the table below as reference, and then click Submit.

To configure DHCP using the configuration file:

◼ Use the table below as reference.

9. Obtaining Firmware and Configuration Files

400HD Series IP Phones

- 19 -

Table 6: DHCP Automatic Provisioning Parameters

Parameter

Description

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 option 160

as well as option 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 as well as 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 is440HD.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;

<MODEL>_<MAC>_conf.cfg - retrieved firmware file is

different_firmware_name.img and the configuration file name is <Model

type>_<MAC address>_conf.cfg. For example,

440HD_001122334455_conf.cfg

• https://192.168.2.1/;<MODEL>_<MAC>_conf.cfg - if the model is

440HD, the retrieved firmware file is 440HD.img and the

configuration file name is 440HD_<MAC Address>_conf.cfg.

For example, 440HD_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 440HD.img image file. For

example:

◼ Option 66: 192.168.2.1 or myTFTPServer

◼ Option 67: 440HD_2.2.2.img

Note:

◼ 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.

9. Obtaining Firmware and Configuration Files

400HD Series IP Phones

- 20 -

Parameter

Description

Random Provisioning Time

[provisioning/random_pro

visioning_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:

◼ [every5minutes] Minimum definable time. Sets the interval at every five

minutes.

◼ [every15minutes] Sets the interval at every five minutes.

◼ [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] Irrespective of what value is defined, the phone always checks

on powerup, but if powerup is defined, the phone will check only on

powerup.

Every (Check Period =

Hourly)

[provisioning/period/hourl

y/hours_interval]

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.

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/week

ly/day]

The day in the week for automatically checking for new firmware and

configuration files.

◼ [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/week

ly/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.

9. Obtaining Firmware and Configuration Files

400HD Series IP Phones

- 21 -

9.2.1 Provisioning using DHCP Option 160

Phones can get a provisioning URL from DHCP Option 160, 66/67 or 43. Option 160 has the highest

priority, following by Option 66/67, and then Option 43. DHCP Option 160 can be configured using

the Web interface.

To configure DHCP Option 160 using the Web interface:

1. Open the Automatic Update page (Management tab > Automatic Update menu > Automatic

Provisioning).

Figure 8: Automatic Provisioning - DHCP Option 160

2. From the 'Provisioning Method' dropdown, select DHCP Option (Dynamic URL).

3. In the 'DHCP Option Value' field, enter 160.

4. Configure the remaining parameters, and then click Submit.

5. After reboot, confirm that the firmware and configuration files have been updated.

9.2.2 Technician's Digit Key Code

Technicians installing phones at customer sites do not need to connect laptops to phones to

provision them. After connecting phones to the network, technicians can enter a specific digit key

code which changes the phones' provisioning URL to the server's URL. If the code that the technician

enters matches, the phones are automatically provisioned from that server.

The feature requires software customization.

9. Obtaining Firmware and Configuration Files

400HD Series IP Phones

- 22 -

9.2.3 Provisioning using DHCP Option 66/67

Phones can get a provisioning URL from DHCP Option 66/67. Option 160 has the highest priority,

following by Option 66/67, and then Option 43. The table below shows the behaviors for Option

66/67.

Table 7: Auto Provisioning via DHCP Option 66/67

Option 66

Option 67

Result

Comment

Doesn’t exist or empty

Any

No URL from Option 66/67

When Option 66

doesn’t exist, or

it’s empty, the

phone cannot

get a URL from

Option 66/67.

Server address exists but there

is no protocol header such as

TFTP, FTP, HTTP, HTTPS.

File names do not exist.

Example:

Audiocodes.com

192.168.0.11

Non-existent

Firmware URL:

Tftp://audiocodes.com/<hardw

are type>.img

Configuration file url:

Tftp://audiocodes.com/.<mac>

cfg

When protocol

is not specified,

tftp is added as

the default

protocol.

Contains

names.

Example:

abc.img;efg.cfg

Firmware URL:

Tftp://audiocodes.com/abc.img

Configuration file URL:

Tftp://audiocodes.com/efg.cfg

Server address exists

File names do not exist.

Example:

http://Audiocodes.com

http://192.168.0.11

Non-existent

Firmware URL:

http://audiocodes.com/<hardw

are type>.img

Configuration file URL:

http://audiocodes.com/.<mac>

cfg

Contains

names.

Example:

abc.img;efg.cfg

Firmware URL:

http://audiocodes.com/abc.img

Configuration file URL:

http://audiocodes.com/efg.cfg

Server address exists.

File names exist.

Example:

http://Audiocodes.com/abc.ima

ge;efg.cfg

Any

Firmware URL:

http://audiocodes.com/abc.img

Configuration file URL:

http://audiocodes.com/efg.cfg

If any file name

exists in Option

66, the names

in Option 67 are

ignored.

To operate with DHCP Options 66 and 67:

◼ Configure DHCP Options 66 and 67 in the DHCP server, instead of configuring Option 160. See

the DHCP server related documentation for detailed information.

9. Obtaining Firmware and Configuration Files

400HD Series IP Phones

- 23 -

9.2.4 Provisioning using DHCP Option 43

Phones can get a provisioning URL from DHCP Option 43. Option 160 has the highest priority,

following by Option 66/67, and then Option 43.

To operate with DHCP Options 43:

◼ Configure DHCP Options 43 in the DHCP server. Use the example in the figure below as

reference.

Figure 9: Provisioning using DHCP Option 43 in the DHCP Server

• 01 is the sub option

• 27 is the length (in HEX) of the provisioning path string that you configured

• The remainder is the provisioning path, in ASCII code.

Example: tftp://192.168.202.103/test.img;test.cfg

9. Obtaining Firmware and Configuration Files

400HD Series IP Phones

- 24 -

9.2.5 Provisioning using the User-Class Option

Provision using the User-Class Option if vendor phones other than those of AudioCodes are deployed

in the same enterprise as AudioCodes' phones and a DHCP Option cohabitation issue consequently

occurs.

Here's how to configure provisioning of AudioCodes phones using the User-Class Option when other

vendor phones in the enterprise point to the same DHCP server and use one of the standard DHCP

Options described in the previous sections.

To configure provisioning of AudioCodes phones using the User-Class Option:

1. Determine the DHCP server hosting the phones.

2. Determine if DHCP Options are assigned to IPv4 or IPv6 addresses.

• The examples below show DHCP server acrtplab-ad.audiocodes.local

• The examples below show IPv4 addresses

Figure 10: DHCP Options Assigned to IPv4 Addresses

3. Define a separate User Class for each AudioCodes phone model deployed (420HD, 430HD and

440HD phone models): Right-click the IPv4 server icon and from the popup menu, select

Define User Classes…

Figure 11: Defining User Classes

9. Obtaining Firmware and Configuration Files

400HD Series IP Phones

- 25 -

The DHCP User Classes screen opens.

Figure 12: DHCP User Classes

4. Click the Add… button.

Figure 13: New Class

5. In the New Class screen, enter Display name and Description as shown in the figure above,

and then in the ASCII field, enter the User Class Phone Type (see the Packet Bytes window in

Wireshark below for an example of the 420HD phone, and see the table below for the other

AudioCodes phone models) to be sent from the phone during DHCP Discover via Option 77

(supported by DHCP Server 2008). Do this for each AudioCodes phone model so that a User

Class entry for each model deployed will exist when completed.

Figure 14: Packet Bytes Window

9. Obtaining Firmware and Configuration Files

400HD Series IP Phones

- 26 -

6. Make sure one DHCP User Class entry exists for each AudioCodes phone model deployed in

the enterprise.

Figure 15: DHCP User Classes

Table 8: DHCP User Class Entry for Each AudioCodes Phone Model Deployed

Display Name

Description

ASCII

420HD

AudioCodes 420HD IP Phone

420HD

430HD

AudioCodes 430HD IP Phone

430HD

440HD

AudioCodes 440HD IP Phone

440HD

405

AudioCodes 405 IP Phone

405

7. Configure Scope Option 160. This is not a standard Scope Option, so it needs to be created. To

create it on the server, select the IP version (IPv4) and select Set Predefined Options…

Figure 16: Set Predefined Options

9. Obtaining Firmware and Configuration Files

400HD Series IP Phones

- 27 -

8. From the 'Option class' dropdown, select DHCP Standard Options, and then click the Add…

button.

Figure 17: Predefined Options and Values

9. Add the AudioCodes 160 Option as shown below, and then click OK.

Figure 18: Option Type – Add AudioCodes 160 Option

9. Obtaining Firmware and Configuration Files

400HD Series IP Phones

- 28 -

10. Add the OVOC server location using HTTP. In the figure below, it's http://<OVOC server IP

address>/firmwarefiles;ipp/dhcpoption160.cfg. See the Device Manager Pro Administrator’s

Manual for more information.

Figure 19: Predefined Options and Values – Add OVOC Server Location

Make sure you defined in the enterprise's DHCP server http://<OVOC server IP

address>/firmwarefiles;ipp/dhcpoption160.cfg for DHCP Option 160.

11. Decide if the DHCP Scope Option needs to be assigned to phones in a specific VLAN (Scope),

or to the entire server (acrtplab-ad.audiocodes.local) for IPv4 addresses.

VLAN Scope

12. Assign to a specific VLAN (Scope of IP addresses such as the Scope below 172.17.0.0, or to

multiple Scopes, to be performed separately on each Scope).

a. If selecting a VLAN, expand the 'Scope Leased' folder, select 'Scope Options', and then

select Configure Options from the popup menu.

9. Obtaining Firmware and Configuration Files

400HD Series IP Phones

- 29 -

Figure 20: 'Scope Leased' Folder - Configure Options

-OR-

b. Select the collapsed folder 'Scope Leased' and in the main screen, right-click 'Scope

Options' and select Configure Options…

Figure 21: Configure Options 1

-OR-

c. Server Option

9. Obtaining Firmware and Configuration Files

400HD Series IP Phones

- 30 -

13. If assigning to the entire server (acrtplab-ad.audiocodes.local), select the 'Server Options'

folder under server IPv4, right-click 'Server Options' and select Configure Options...

Figure 22: Configure Options 2

14. In the Server Options page (or Scope Options page) that opens, select the Advanced tab,

make sure DHCP Standard Options remains selected, and select 420HD User Class for the first

phone model to be defined. Scroll through the Available Options (all are cleared) and select

only 160 AudioCodes 160 Option.

The figure below shows the Server Options page. The Scope Options page is identical. Note

that the String value you defined for Scope Option 160 is automatically populated so

it's unnecessary to change it. Note also that if additional DHCP Options are required (such as

DNS or time server) that are different from the Servers Options for the rest of the Scopes on

the server, they can also be selected, but this is typically not needed.

Figure 23: Server Options

9. Obtaining Firmware and Configuration Files

400HD Series IP Phones

- 31 -

15. Click Apply and then follow the same procedure to add the 430HD and 440HD user classes.

After adding them, click the OK button.

You've successfully created three separate Scope Options that will only allow AudioCodes phones to

connect to the Device Manager Pro when they boot up and will prevent other vendor phones from

receiving the OVOC as their provisioning server.

Figure 24: Three Scope Options Created

9. Obtaining Firmware and Configuration Files

400HD Series IP Phones

- 32 -

9.2.6 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.

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-13c4-

50029-386d4398-66dc40c-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-386d4398-

6ad00ca2-7ca3606e

Expires: 0

Event: ua-profile;profile-

type="application";model="440HD";version="2.2.2"

Max-Forwards: 70

Supported: replaces,100rel

Accept: application/url

Contact: <sip:[email protected]:5060>

User-Agent: AUDC-IPPhone/2.2.2

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-386d4398-

6ad00ca2-7ca3606e

Contact: <sip:10.13.2.37:5060>

To: <sip:224.0.1.75:5060>

From: <sip:[email protected]:5060>;tag=87a5a8-25020d0a-13c4-

50029-386d4398-66dc40c-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-386d4398-

6ad00ca2-7ca3606e

9. Obtaining Firmware and Configuration Files

400HD Series IP Phones

- 33 -

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-13c4-

50029-386d4398-66dc40c-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;profile-

type="application";model="440HD";version="2.2.2"

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-13c4-

50029-386d4398-66dc40c-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-386d4398-

6ad00ca2-7ca3606e

Expires: 0

Event: ua-profile;profile-

type="application";model="440HD";version="2.2.2"

Max-Forwards: 70

Supported: replaces,100rel

Accept: application/url

Contact: <sip:[email protected]:5060>

User-Agent: AUDC-IPPhone/2.2.2

Content-Length: 0

With the above method, the phone uses its built-in auto-provisioning mechanism while the

provisioning information is retrieved through the NOTIFY message.

9. Obtaining Firmware and Configuration Files

400HD Series IP Phones

- 34 -

9.2.7 Hardcoded Domain Name for Provisioning Server

If no higher-priority provisioning method applied, the phone automatically searches in the DNS

server for the domain named "ProvisioningServer". After the DNS server gives the domain IP address,

the phone contacts the provisioning server. The phone tries to retrieve firmware and configuration

files using URL tftp://ProvisioningServer/<Phone Model Name>/

For example:

◼ The phone tries to obtain the following firmware file: tftp://ProvisioningServer/440HD.img

where 440 is optional; if omitted, the phone will try to retrieve the firmware file according to

its model name.

◼ The phone tries to obtain the following configuration file: tftp://ProvisioningServer/<MAC

address>.cfg

where MAC address is optional; if omitted, the phone will try to retrieve the configuration

file according to its MAC address.

(e.g., tftp://ProvisioningServer/440HD/001122334455.cfg)

The network administrator must configure a DNS entry called "ProvisioningServer" on the DNS server

and set it to the TFTP server IP address.

If Generic Domain Name is used, the automatic provisioning mechanism periodically tries

to retrieve new firmware/configuration from Provisioning Server domain name.

9.2.8 Cached Address of Last Provisioning Server Used

These are the addresses of the last provisioning servers used, stored in cache memory.

When the device starts up and connects to the provisioning server, it can pull firmware, configuration

and private label files from the provisioning server using the cached address of the last provisioning

server used.

After the phone creates a successful connection with a provisioning server, this server's address is

cached by the phone. The next time the phone is rebooted, if it doesn't receive provisioning details,

the device performs provisioning using the cached IP address.

9. Obtaining Firmware and Configuration Files

400HD Series IP Phones

- 35 -

9.2.9 Redirect Server

You can use the AudioCodes Redirect server to direct you to the appropriate Provisioning server URL

to download the relevant configuration and firmware files.

After the phone is powered up and network connectivity is established, it automatically requests

provisioning information. If it doesn't get these files according to the regular provisioning hunt order

methods, it sends an HTTPS request to the AudioCodes HTTPS Redirect server. The server responds

to the phone with an HTTPS Redirect response containing the URL of the provisioning server where

the firmware and configuration files are located. After the phone successfully connects to the

provisioning server URL, the Automatic Update mechanism commences.

◼ The MAC addresses of the phones and the provisioning server's URL are pre-

configured on the HTTPS Redirect server. For more information, contact AudioCodes

support.

◼ The default URL of the Redirect server is:

provisioning/redirect_server_url=https://redirect.audiocodes.com

◼ This address can be reconfigured if required.

Figure 25: Redirect Server Configuration Process

1. Device sends HTTPS request to AudioCodes HTTPS Redirect server.

2. Redirect server sends HTTPS response with redirect URL of the provisioning server.

3. Phone sends request to redirected URL (i.e., provisioning server).

For security, communication between the phone and the HTTPS Redirect server is encrypted (HTTPS)

and uses the pre-installed AudioCodes factory-set certificate to authenticate itself with the HTTPS

Redirect server and to verify authenticity of the latter. If the redirect URL (where the configuration

file is stored) also uses the HTTPS protocol, the phone can use a regular certificate or the AudioCodes

factory-set certificate to authenticate itself and to validate the server’s certificate if a trusted root

certificate (regular) is configured.

The phone repeats the redirect process whenever it undergoes a reset to factory defaults.

9. Obtaining Firmware and Configuration Files

400HD Series IP Phones

- 36 -

9.3 Static URL Provisioning

Here's how to configure the phone using the Static URL method.

To configure static provisioning information using the Web interface:

1. Open the Automatic Provisioning page (Management tab > Automatic Update menu >

Automatic Provisioning).

Figure 26: Automatic Provisioning – Static URL

2. Configure the parameters using the table below as reference, and then click Submit.

To configure static provisioning information using the configuration file:

◼ Use the table below as reference.

Table 9: Static URL Automatic Provisioning Parameters

Parameter

Description

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

Option 160 and Options 66/67 for provisioning

◼ [Static] Static URL - Using Static URL for provisioning

Firmware URL

[provisioning/firmware/url]

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 is440HD.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.

9. Obtaining Firmware and Configuration Files

400HD Series IP Phones

- 37 -

Parameter

Description

Configuration URL

[provisioning/configuration/url]

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://192.168.2.1/440HD_<MAC>_conf.cfg - retrieved

configuration file name is 440HD_<MAC Address>_conf.cfg, for

example, 440HD_001122334455_conf.cfg

Note: This parameter is applicable only when 'method' is configured to

Static.

Part III Quick Setup

Part III

Quick Setup

10. Quick Setup

400HD Series IP Phones

- 39 -

10 Quick Setup

The Web interface provides a Quick Setup page that lets you configure basic features to quickly set

up your phone to operational level.

For Quick Setup parameters descriptions, see Part IV (Networking) and Part V (VoIP

Features).

To quickly set up your phone:

1. Open the Quick Setup page (Configuration tab >Quick Setup menu >Quick Setup).

Figure 27: Quick Setup

2. For descriptions of parameters on this page:

• See Section 13.1.2 for descriptions of parameters under the LAN Setup group

• See Section 16.2 for descriptions of parameters under the SIP Proxy and Registrar group

• See Section 21 for descriptions of parameters under the Line Settings group

Part IV Networking

Part IV

Networking

11. Introduction

400HD Series IP Phones

- 41 -

11 Introduction

Here's how to configure network settings manually, if required.

By default, the network settings are set for automatic provisioning. However, if you need

to change them, you can do so manually, as described in this section.

12. Configuring Date and Time Manually

400HD Series IP Phones

- 42 -

12 Configuring Date and Time Manually

By default, date and time settings are automatically provisioned via the enterprise DHCP

server when the phone is connected to the Internet and to the power supply, but you can

manually change them if required. This section shows how.

The phone automatically retrieves date and time from a Network Time Protocol (NTP) server when

connected to the internet. To configure the NTP server for automatic provisioning of date and time,

see Section 12.2. NTP is a protocol for distributing Coordinated Universal Time (UTC) by

synchronizing the clocks of computer systems over packet-switched, variable-latency data networks.

Date/time can also be manually configured in the Web interface.

To manually configure date and time using the Web interface:

1. Open the Date and Time page (Configuration tab > Advanced Applications menu > Date and

Time).

Figure 28: Date and Time

2. Configure the 'Set System Time' parameter.

3. Set the 'Time Display Format' to 24 Hours or 12 Hours

4. Set the 'Date Display Format' to European or American. See the table below.

5. Click Submit.

To configure date and time using the configuration file:

◼ Use the table below as reference.

Table 10: Date Display Format

Parameter

Description

Date Display Format

[system/ntp/date_display_format]

Select either:

◼ EUROPEAN (default)

◼ AMERICAN

The European date format is DDMMYYYY.

The American format is MMDDYYYY.

12. Configuring Date and Time Manually

400HD Series IP Phones

- 43 -

12.1 Configuring Daylight Saving Time

You can configure Daylight Saving Time using the Web interface or configuration file.

To configure Daylight Saving Time using the Web interface:

1. Open the Date and Time page (Configuration tab > Advanced Applications menu > Date and

Time) as described above.

2. Set the 'Active' parameter to Enable; the page shown below opens.

Figure 29: NTP & Time Settings

3. Set the 'Obtain Time Zone from DHCP' parameter to Disable; the Daylight Saving Time page

shown below opens:

Figure 30: Daylight Saving Time

4. Configure the settings using the table below as reference.

12. Configuring Date and Time Manually

400HD Series IP Phones

- 44 -

To configure Daylight Saving Time using the configuration file:

◼ Use the table below as reference.

Table 11: Daylight Saving Time Parameters

Parameter

Description

Active

[system/daylight_saving/activate]

Determines whether the phone automatically detects

the Daylight Saving Time for the selected Time Zone.

◼ [DISABLE] Disable (default)

◼ [ENABLE] Enable

Start Time

[system/daylight_saving/start_date]

This subsection defines the starting day for the

daylight saving offset.

◼ [month] - defines specific month in year

◼ [day] - defines specific day in month

◼ [hour] - defines specific hour in day

◼ [minute] - defines specific minute in hour

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=30

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.

End Time

[system/daylight_saving/end_date]

This subsection 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=22

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.

12. Configuring Date and Time Manually

400HD Series IP Phones

- 45 -

Parameter

Description

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.

[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]

12. Configuring Date and Time Manually

400HD Series IP Phones

- 46 -

12.2 Configuring the NTP Server

The Network Time Protocol (NTP) server can be configured using the Web interface or configuration

file. When activated, date and time are automatically obtained from the NTP server.

To configure the NTP server using the Web interface:

1. Open the Date and Time page (Configuration tab > Advanced Applications menu > Date and

Time).

2. Configure the parameters using the table below as reference, and then click Submit.

Figure 31: NTP & Time Settings

12. Configuring Date and Time Manually

400HD Series IP Phones

- 47 -

To configure the NTP server using the configuration file:

◼ Use the table below as reference.

Table 12: NTP Server Parameters

Parameter

Description

Active

[system/ntp/enabled]

Enables the NTP server from which the phone automatically

retrieves the date and time.

◼ [0] Disable

◼ [1] Enable – obtains the time information automatically

from a configured NTP server (default)

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.

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 phone screen.

◼ [24Hour] (default)

◼ [12Hour]

12. Configuring Date and Time Manually

400HD Series IP Phones

- 48 -

12.3 Configuring NTP Server via DHCP

If the 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).

If both options (4 and 42) are received, priority is given to Option 42.

◼ Time Zone – (Option 2)

The 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. This is the updated

system time.

To manually configure NTP / GMT offset via DHCP using Web interface:

1. Open the Date and Time page (Configuration tab > Advanced Applications menu > Date and

Time).

2. From the 'Obtain Time Zone From DHCP' drop-down list, select Disable.

Figure 32: NTP and Time Settings

3. Configure the NTP and Time Settings according to the parameters in the table below, and

then click Submit.

These values will have no affect if TimeZone is set to be obtained from DHCP. If Time Zone

and NTP server are manually set, the phone acts as described above but the values are

obtained from the configuration file and not from DHCP.

Table 13: NTP Server and GMT Parameters

Parameter

Description

[system/ntp/gmt_offset]

Default: 00:00

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

13. Configuring IP Network Settings

400HD Series IP Phones

- 49 -

13 Configuring IP Network Settings

The following section shows 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 screen

◼ Web interface and configuration file

13.1.1 Configuring Static IP Address on the Phone

Here's how to configure Static IP Address on the phone on the phone. The 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 and select the LAN Connection Type option (MENU key > Administration > Network

Settings).

2. Navigate to and select Static IP or Automatic IP (DHCP) IP addressing scheme.

3. If you select Static IP, continue with the next step (Step 4); else skip to Step 5.

4. 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, navigate to and choose the parameter, and then press

the Edit softkey, for example, navigate to IP Address and press the Edit softkey:

13. Configuring IP Network Settings

400HD Series IP Phones

- 50 -

c. 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.

d. Press the Save and then Apply softkey.

5. Press the Save softkey.

13.1.2 Configuring IP Network Settings

Here's how to configure IP Network settings using the Web interface or configuration file.

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 interface:

1. Open the LAN Settings page (Configuration tab > Network Connections menu > Network

Settings).

Figure 33: Network Settings

2. Configure the LAN parameters using the table below as reference, and then click Submit.

13. Configuring IP Network Settings

400HD Series IP Phones

- 51 -

To define the phone's LAN settings using the configuration file:

◼ Use the table below as reference.

Table 14: Network Settings Parameters

Parameter

Description

IP Type

[network/lan_type]

Defines the IP addressing method:

◼ [STATIC] Static IP (default)- Phone's IP address is defined manually

◼ [DHCP] Automatic IP DHCP - Phone's IP address is acquired

automatically from a DHCP server

network/lan/fixed_ip

This subsection defines the relevant parameters 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_na

me]

The domain name.

Domain Name Server (DNS)

Primary DNS

[network/lan/fixed_ip/primary_dn

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.

13. Configuring IP Network Settings

400HD Series IP Phones

- 52 -

13.2 Configuring Partial DHCP

Partial DHCP can be configured with the following parameters:

Table 15: Partial DHCP Parameters

Parameter

Description

Partial DHCP

network/lan/dhcp

This subsection defines the parameters to configure if 'lan_type' is

configured to DHCP.

Domain Name - Manual

[network/lan/dhcp/domain_name/en

abled]

Enables setting the domain name manually.

◼ [0] Disable (default)

◼ [1] Enable

Note: If enabled, network/lan/fixed_ip/domain_name must be set.

IP Address - Manual

[network/lan/dhcp/ip_address/enable

Enables setting the IP address manually.

◼ [0] Disable

◼ [1] Enable (default)

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

◼ [1] Enable (default)

Note: If enabled, network/lan/fixed_ip/netmask must be set.

[network/lan/dhcp/gateway/enabled]

Enables setting the default gateway manually.

◼ [0] Disable

◼ [1] Enable (default)

Note: If enabled, network/lan/fixed_ip/gateway must be set.

Primary DNS - Manual

[network/lan/dhcp/primary_dns/enab

led]

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/ena

bled

Enables setting the secondary DNS manually.

◼ [0] Disable (default)

◼ [1] Enable

Note:If enabled, network/lan/fixed_ip/secondary_dns must be set.

DHCP-Related Parameters

network/lan/dhcp/ntp/server_list/ena

bled

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/en

abled

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)

14. Configuring LAN and PC Port Settings

400HD Series IP Phones

- 53 -

14 Configuring LAN and PC Port Settings

Port settings can be configured using the Web interface or configuration file.

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 configure the phone's port settings using the Web interface:

1. Open the Network Settings page (Configuration tab > Network Connections menu > Network

Settings).

Figure 34: Network Settings - Port Mode

2. Configure the parameters using the table below as reference, and then click Submit.

To define the phone's port settings using the configuration file:

◼ Use the table below as reference.

Table 16: Port Settings

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

[FULL _ 1Gbps] = 1 Gbit/s port + full duplex

PC Port Mode

[network/pc/port_mode]

Sets the computer 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

[DISABLE] = Disables the PC port mode

15. Configuring VLAN Settings

400HD Series IP Phones

- 54 -

15 Configuring VLAN Settings

You can configure VLAN settings using the Web interface or configuration file.

To configure the phone's VLAN settings using the Web interface:

1. In the Web interface access the Network Settings page (Configuration tab > Network

Connections menu > Network Settings).

Figure 35: Network Settings - VLAN Settings

2. Configure the settings using the table below as reference, and then click Submit.

15. Configuring VLAN Settings

400HD Series IP Phones

- 55 -

To configure the phone's VLAN settings using the configuration file:

◼ Use the table below as reference.

Table 17: VLAN Settings

Parameter

Description

VLAN Discovery Mode

[network/lan/vlan/mode]

Determines how VLAN is assigned to your phone, i.e., manually or

automatically, and if automatically, according to which protocol.

◼ Disable [Disable]

◼ Manual Configuration of VLAN [Manual] - If selected, the screen

extends to also display 'VLAN ID' and 'VLAN Priority' (see these

settings below) for static configuration of VLAN ID and priority. See

Section 15.1 below for a detailed explanation.

◼ Automatic Configuration of VLAN (CDP) [CDP] - VLAN discovery

mechanism based on Cisco Discovery Protocol (CDP). See Section

15.1 below for a detailed explanation.

◼ Automatic Configuration of VLAN (LLDP) [LLDP] - VLAN discovery

mechanism based on LLDP. See Section 15.1 below for a detailed

explanation.

◼ Automatic Configuration of VLAN (CDP+LLDP) [CDP_LLDP] (default) -

VLAN discovery mechanism based on LLDP and CDP. LLDP is higher

priority. See below for a detailed explanation.

Period

[network/lan/vlan/period]

The time period, in seconds, between discovery messages when

configured to CDP, LLDP or CDP+LLDP. The default value is 30.

VLAN ID

[network/lan/vlan/id]

The VLAN ID. The valid range is 0 to 4094. The default 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 is 0.

PC Port VLAN Activate

[network/lan/vlan/pc_port_

tagging/enable]

Default = Disable [0]. Change to Enable (1) for the traffic from the PC to

the network to be VLAN-tagged.

Note: Changing the parameter settings requires a hard reset of the

phone (power on/off).

Vlan Switch

/network/lan/vlan/vlan_switch/

enable

Enable or disable vlan switching on the fly

◼ [0] Disable (default)

◼ [1] Enable

15. Configuring VLAN Settings

400HD Series IP Phones

- 56 -

15.1 Configuring Manual or Automatic VLAN Assignment

You can configure the VLAN to be assigned manually or automatically to your phone. This section

shows when to configure what, and why.

15.1.1 Configuring Manual VLAN Assignment to the Phone

Configure manual assignment of the VLAN to set up two separate VLANs in your enterprise, one for

voice (your phone) and the other for data (your pc). Security considerations may require this. If you

configure manual assignment, the switch in your enterprise will assign the VLAN to your phone. See

Sections 13.1.1 and 13.1.2 for details.

15.1.2 Configuring Automatic VLAN Assignment to the Phone

Configure automatic assignment of VLAN if you do not need to separate voice from data, i.e., if there

are no security considerations requiring it. In this case, configure either:

◼ Automatic Configuration of VLAN (CDP) [CDP]

◼ Automatic Configuration of VLAN (LLDP) [LLDP] -OR-

◼ Automatic Configuration of VLAN (CDP+LLDP) [CDP_LLDP]

What you select depends on whether the switch deployed in your enterprise supports Cisco-

proprietary Cisco Discovery Protocol (CDP), or LLDP (Link Layer Discovery Protocol) which is a vendor-

neutral protocol used by devices in an IEEE 802 LAN to advertise their identity, capabilities, and

neighbors. Not all switches support CDP. If You're unsure, select CDP+LLDP. LLDP includes enhanced

LLDP for Media Endpoint Devices, i.e., LLDP-MED, to specifically address voice applications.

15.1.3 Configuring VLAN via DHCP Provisioning Path

VLAN can be configured using (1) Link Layer Discovery Protocol (LLDP) (2) Cisco Discovery Protocol

(CDP) and (3) manually.

If (1) is unsuccessful, (2) is attempted.

If (2) is unsuccessful, (3) is attempted.

If (3) is unsuccessful, (4) is attempted.

The capability provides an alternative VLAN configuration option. vi93303

Part V VoIP Settings

Part V

VoIP Settings

16. Configuring SIP Settings

400HD Series IP Phones

- 58 -

16 Configuring SIP Settings

You can configure the following SIP settings using the Web interface or configuration file:

◼ General

◼ Proxy and Registration

◼ SIP Timers

◼ SIP QoS

16.1 Configuring General SIP Settings

The phone's General SIP settings can be configured using the Web interface or configuration file.

To configure general SIP settings using the Web interface:

1. Open the Signaling Protocols page (Configuration tab > Voice Over IP menu > Signaling

Protocols).

Figure 36: Signaling Protocols- SIP General

2. Configure the SIP General parameters using Table 18 below as reference, and then click

Submit.

To configure General SIP parameters using the configuration file:

◼ Use the table below as reference.

Table 18: 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.

16. Configuring SIP Settings

400HD Series IP Phones

- 59 -

Parameter

Description

[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.

Note:

◼ 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)

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.

16. Configuring SIP Settings

400HD Series IP Phones

- 60 -

Parameter

Description

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

[voip/signalling/sip/block_callerid_on_outgoing_calls]

Can be configured only if the BroadSoft

BroadWorks application server is used.

When enabled, the outgoing INVITE

message is sent with an anonymous From

header and P-Asserted-Identityheader.

◼ [0] Disable (default)

◼ [1] Enable

For example:

◼ FROMheader contains anonymous URI:

From: "Anonymous"

sip:[email protected]

◼ P-Asserted-Identityheader:

P-Asserted-Identity: "1001"

111555100[email protected]

Incoming Anonymous Call Blocking

[voip/signalling/sip/anonymous_calls_blocking]

Can be configured only if the BroadSoft

BroadWorks application server is used.

When enabled, incoming INVITE messages

with anonymous From header are rejected.

◼ [0] Disable (default)

◼ [1] Enable

For example:

From:"Anonymous"<sip:anonymous@anon

ymous.invalid>

The phone responds with a SIP 403

"Forbidden" response.

[voip/signalling/sip/auth_retries]

Defines the number of times authenticated

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/enabl

ed]

Sets the Display Name in the 'To' and 'From'

fields of the SIP REGISTER message.

◼ [0] Disable (default)

◼ [1] Enable

16. Configuring SIP Settings

400HD Series IP Phones

- 61 -

Parameter

Description

[voip/signalling/sip/semi_transfer_with_no_cancel/enabled]

Determines whether semi-attendant

transfer is performed without sending the

SIP CANCEL message to the remote side.

◼ [0] Disable (default)

◼ [1] Enable

Note:

◼ 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/draft-ietf-

sipping-cc-transfer-03.

◼ Existing / current behavior is retained for

backward compatibility (disabled by

default)

[voip/signalling/sip/PAI_On_Replay/enable]

Enables the P-Asserted Identity header to

be added to “18x” and “200” responses.

◼ [0] (Default) PAI header is not added to

“18x” and “200” responses

◼ [1] PAI header is added to “18x” and

“200” responses

/voip/signalling/sip/add_sip_instance/enable

Allows Universally Unique Identifier (UUID)

URN to be constructed and used in the

"+sip.instance" Contact header field

parameter for outbound behavior.

◼ [0] Disable (default)

◼ [1] Enable

16. Configuring SIP Settings

400HD Series IP Phones

- 62 -

16.2 Configuring Proxy and Registration

Here's how to configure Proxy and Registration settings using the Web interface or configuration

file.

To configure Proxy and Registration using the Web interface:

1. Open the Signaling Protocol page (Configuration tab > Voice Over IP menu > Signaling

Protocols).

Figure 37: SIP Proxy and Registrar

2. Configure Proxy and Registration parameters using the table below as reference, and then

click Submit.

To configure Proxy and Registration using the configuration file:

◼ Use the table below as reference.

Table 19: 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.

Default: 0.0.0.0

Proxy Port

[voip/signalling/sip/proxy_port]

The UDP or TCP port of the SIP proxy server.

Range: 1024 to 65535. Default: 5060.

16. Configuring SIP Settings

400HD Series IP Phones

- 63 -

Parameter

Description

Enable Registrar Keep Alive

[voip/signalling/sip/registrar_ka/enabled]

Determines whether to use the registration keep-alive

mechanism based on SIP OPTION messages.

◼ [0] Disable (default)

◼ [1] Enable

Note:

◼ 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 keep-alive

messages are re-sent periodically.

Registrar Keep Alive Period

[voip/signalling/sip/registrar_ka/timeout]

Defines the registration keep-alive time interval (in

seconds) between Keep-Alive messages.

Range: 40 to 65536. Default: 60.

Maximum Number of Authentication Retries

[voip/signalling/sip/proxy_timeout]

The SIP proxy server registration timeout (in seconds).

Range: 0 to 86400. Default: 3600.

Use SIP Proxy IP and Port for Registration

[voip/signalling/sip/use_proxy_ip_port_for_registr

ar]

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

Use SIP Registrar

[voip/signalling/sip/sip_registrar/enabled]

Determines whether the phone registers to a separate

SIP Registrar server.

◼ [0] Disable (default)

◼ [1] Enable

Registrar IP Address or Host Name

[voip/signalling/sip/sip_registrar/addr]

[Only displayed if the 'Use SIP Registrar' parameter is

enabled.]

The IP address or host name of the Registrar server.

Default: 0.0.0.0

Registrar Port

[voip/signalling/sip/sip_registrar/port]

[Only displayed if the 'Use SIP Registrar' parameter is

enabled.] The UDP or TCP port of the Registrar server.

Range: 1024 to 65535. Default: 5060.

Registration Failed Expires

[voip/signalling/sip/registration_failed_timeout]

If registration fails, this parameter determines the

interval between the register messages periodically

sent until successful registration. Range: 1 to 86400.

Default: 300.

Use SIP Outbound Proxy

[voip/signalling/sip/sip_outbound_proxy/enabled]

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

Outbound Proxy IP Address or Host Name

[voip/signalling/sip/sip_outbound_proxy/addr]

[Only displayed if the 'Use SIP Outbound Proxy'

parameter is enabled.] 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. Default: 0.0.0.0.

16. Configuring SIP Settings

400HD Series IP Phones

- 64 -

Parameter

Description

Outbound Proxy Port

[voip/signalling/sip/sip_outbound_proxy/port]

[Only displayed if the 'Use SIP Outbound Proxy'

parameter is enabled.] The port on which the

outbound proxy listens.

Range: 1024 to 65535. Default: 5060.

[voip/signalling/sip/register_before_expires_perce

nt]

Allows administrators to configure the registration

expired time. The registration expired time is that time

that lapses before the refresh registration message is

sent. Default: 15%. Non-percentage values are 5-85.

These represent the time that must lapse before the

new registration message is sent, for example, 15%

means that if the expiration time is 100 seconds, the

registration refresh message will be sent after 85% of

the registration expiring timeout. In releases before

version 2.2.12, it was 33%.

Redundant Proxy Mode

[voip/signalling/sip/redundant_proxy/mode]

See the next section.

It’s recommended to use DNS queries to complete FQDN for a redundant outbound proxy.

16.2.1 Configuring Proxy Redundancy

The Redundant Proxy feature allows the configuration of a backup SIP proxy server to increase QoS

stability. After the 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 interface or Configuration file.

To configure Proxy Redundancy using the Web interface:

1. Open the Signaling Protocol page (Configuration tab > Voice Over IP menu > Signaling

Protocols).

16. Configuring SIP Settings

400HD Series IP Phones

- 65 -

Figure 38: Proxy Redundancy

2. Configure the Proxy Redundancy parameters using the table below as reference, and then

click Submit.

To configure Proxy Redundancy using the configuration file:

◼ Use the table below as reference.

Table 20: SIP Proxy Server Redundancy Parameters

Parameter

Description

[voip/signaling/sip/redundant_proxy/enabled

]

Mandatory for the phone to operate in redundancy.

Commands the phone to operate with the other

voip/signalling/sip/redundant_proxy parameters.

Redundant Proxy Mode

[voip/signalling/sip/redundant_proxy/mode]

Mandatory for the phone to operate in redundancy. Defines

the two proxies' mode of operation: Primary-Fallback or

Simultaneous. Defines a backup SIP proxy server to increase

QoS stability. Enable the parameter if you want to operate

with a proxy server that will serve as a backup if the first

goes down.

◼ Disable = (Default) Phone doesn't use redundant proxy.

If set to Disable in the Web interface, it will set the

previously described ini file parameter

redundant_proxy/enabled as well.

◼ Primary_Fallback = Phone registered to redundant proxy

if the primary proxy does not respond to SIP signaling

messages.

◼ Simultaneous = Applies only in some environments. If

selected, dual registration is performed; the phone

registers simultaneously to both servers.

Redundant Proxy Address

[voip/signalling/sip/redundant_proxy/addres

[Only displayed if the 'Redundant Proxy Mode' parameter is

enabled (Primary-Fallback)]

Defines the IP address of the backup proxy server. Default:

0.0.0.0

Redundant Proxy Keep Alive Period

voip/signalling/sip/redundant_proxy/keepaliv

e_period

[Only displayed if the 'Redundant Proxy Mode' parameter is

enabled (Primary-Fallback)]

Defines how often a keep alive message is sent by the

phone to the proxy server.

Range: 0 to 300.

Default: Every 60 seconds.

16. Configuring SIP Settings

400HD Series IP Phones

- 66 -

Parameter

Description

Redundant Outbound Proxy Port

voip/signalling/sip/redundant_proxy/port

[Only displayed if the 'Redundant Proxy Mode' parameter is

enabled (Primary-Fallback)]

Defines the UDP or TCP port of the backup redundant proxy

server. If occupied by other enterprise devices, you can

configure another.

Range: 1024 to 65535. Default = 5060.

Switch back to Primary SIP proxy when

available

[voip/signalling/sip/redundant_proxy/symme

tric_mode]

[Only displayed if the 'Redundant Proxy Mode' parameter is

enabled (Primary-Fallback)]

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.

◼ [0] = Asymmetric (default). In this 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. Therefore, the phone

assigns the primary proxy a higher priority for

registration. If asymmetric mode is configured and the

primary server goes down, an attempt will be made to

revert to the primary server.

◼ [1] = Symmetric. In this 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 whom it has registered does not respond.

Therefore, the phone assigns both proxies the same

priority for registration. If symmetric mode is configured

and the primary server goes down, you'll operate with

the redundant proxy without ever reverting to the

primary unless the redundant proxy also goes down.

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

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.

16. Configuring SIP Settings

400HD Series IP Phones

- 67 -

16.2.2 Device Registration Failover/Failback

16.2.2.1 Failover

This feature enables a secondary server to take over the functions of the primary server on the

enterprise network, if SIP communication between the SIP access device and the primary proxy

server is blocked or delayed or the primary server isn't available.

No phone functionality is lost when the secondary server takes over.

• For failover to function, the Proxy DNS server must be configured with a list of the

names of the proxies, in order and priority, i.e, SRV record. Before the phone tries to

these). The DNS server send a prioritized list. The phone sends a registration request

to the first SIP server; if it isn’t responsive in n time retries (i.e.,

'outgoing_request_no_response_timeout' parameter), it goes to the second, etc.,

until it gets a response.

• SIP Proxy/Outbound Proxy must be configured as the host name.

To configure failover using the configuration file:

◼ Use the table below as reference.

Table 21: Device Registration Failover Parameters

Parameter

Description

SIP Transport Protocol

[voip/signalling/sip/transport_protocol]

Either:

◼ UDP (default)

◼ TCP

◼ TLS encryption

In the SIP protocol, Name Authority

Pointers (NAPTRs) are used to map servers

and user addresses. Combined with Service

Records (SRVs), they enable determining

the service types available for a name, the

name to use for an SRV lookup, and the port

and 'A' DNS records to use to find the IP for

the service.

[voip/signalling/sip/outgoing_request_no_response_timeout

_ms]

This is the timeout, in milliseconds, that

lapses until the phone failovers to the

secondary proxy. Default: 32000

Outbound Proxy IP Address or Host Name

Configure this parameter as an SRV host

name.

Outbound Proxy Port

[voip/signalling/sip/sip_outbound_proxy/port]

Configure a value of 65535 for this

parameter. Configure the parameter when

you're using an Outbound Proxy.

Either configure this parameter or the

parameter 'Proxy Port'.

Proxy IP Address or Host Name

Configure this parameter as an SRV host

name.

16. Configuring SIP Settings

400HD Series IP Phones

- 68 -

Parameter

Description

Proxy Port

[voip/signalling/sip/proxy_port]

Configure a value of 65535 for this

parameter. Configure the parameter when

you're using a regular Proxy server.

Either configure this parameter or the

parameter 'Outbound Proxy Port'.

Registrar Port

[voip/signalling/sip/sip_registrar/port]

Configure this parameter when you're using

a regular Proxy server.

16.2.2.2 Failback

To configure failback using the configuration file:

◼ Use the table below as reference.

Table 22: Device Registration Failback Parameter

Parameter

Description

[voip/signalling/sip/failback_retry_timeout]

[Only applies to BroadSoft]. Applies only if

you're operating with the DNS mode of

failover, i.e., with a DNS server.

◼ [0] Disable (default) – it'll never try to

access back to the first one.

◼ [n] Time, in seconds, that must lapse

before failback is performed.

16.2.3 Preventing Unregistering after Changing Settings and Reloading

An unregistration message is by default sent after making a change to the VoIP application

configuration and reloading.

The VoIP application by default sends a SIP Registration message with Expires:0 (unregister).

The network administrator can change the default and prevent unregistering.

To prevent unregistering:

◼ Use the table below as reference.

Table 23: Preventing Unregistering

Parameter

Description

[voip/signalling/sip/unregister_on_voip_reload]

Either:

◼ [0] (Default) SIP Registration message with

Expires:0 (unregister) is sent.

◼ [1] SIP Registration message with Expires:0

(unregister) is not sent.

16. Configuring SIP Settings

400HD Series IP Phones

- 69 -

16.3 Configuring a Line

Here's how to configure a line.

The Web interface page of the 440HD / 430HD phone are shown here. The Web pages of

the 420HD / 405/405HD phones are identical, except:

• on the 420HD/405/405HD phones, 'Line Mode' is not supported

• on the 420HD/405/405HD phones, 'Label' is supported

• on the 420HD/405/405HD phones, two lines can be configured; on the 430HD/440HD

phones, six lines can be configured.

To configure a line using the Web interface:

1. Open the Line Settings page (Configuration > Voice Over IP > Line Settings):

Figure 39: Line Settings

2. Configure Line Mode using the table below as reference, and then click Submit.

16. Configuring SIP Settings

400HD Series IP Phones

- 70 -

To configure line mode using the configuration file:

◼ Use the table below as reference.

Table 24: Line Settings

Parameter

Description

Line Display Name

[voip/line/0/description]

Defines the SIP User ID which is sent in “INVITE” packets to the

called party in the “From” field, and should appear to the called

party as “Caller ID”. Default: 400HD

Line Activate

[voip/line/0/enabled]

◼ Activates or deactivates the line. See also Section. 25.2.1.1.

[0] = Disabled (this is the default for the second line and higher in

the configuration file).

◼ [1] = Enabled (this is the default for the first line voip/line/0/ in

the configuration file).

Line User ID

[voip/line/0/id]

Defines the SIP User ID provided by the SIP server which the phone

attempts to associate itself with during the registration process. This

is also the default ID sent in the “INVITE” if the Line Display Name

above is left blank.

Default: 0

Line Authentication User Name

[voip/line/0/auth_name]

Defines the SIP username credential used in the registration process

when attempting to associate with the above Line ID.

Default: 0

Line Authentication Password

[voip/line/0/auth_password]

Defines the SIP password associated with the above Line ID

identifier during the registration process.

Default: 0

Line Mode

[voip/line/0-5/line_mode]

Applies only to the 430HD and 440HD phones.

Determines the line mode: PRIVATE (default) or SHARED. PRIVATE

line - only presented with private call appearances. SHARED line –

lets users share an extension number and manage a call as a group.

When an employee places a call on a SHARED line on hold, the call

can be resumed from any other employee sharing the line. A

SHARED line is a line that is only presented with shared call

appearances. Icons displayed in the phone's screen indicate if lines

are configured in a Shared Call Appearance group, or as private

lines.

A hollow icon indicates a phone configured in an SCA group.

A solid icon indicates a phone configured as private.

Line Label

[voip/line/0/extension_display]

Supported only on the 430HD/440HD phones. Defines the label

displayed in the phone screen. To meet requirements for users with

the 405/405HD and 420HD phones, use the parameter

‘voip/line/0/description’ (see Table 24 for more information).

16. Configuring SIP Settings

400HD Series IP Phones

- 71 -

16.4 Configuring Shared Call Appearance

Figure 40: Shared Call Appearance

Parameter

Description

[voip/line/0/shared_call_appearance

/call_info_expiration_timeout]

Default: 3600

[voip/line/0/shared_call_appearance

/call_info_subscription_failed_timeo

ut]

Default: 60

[voip/line/0/shared_call_appearance

/line_seize_expiration_timeout]

Default: 15

[voip/line/0/shared_call_appearance

/speed_dial_delay]

Default: 2

[voip/line/0/shared_call_appearance

/waiting_to_line_seize_tone]

Default: SILENCE

16. Configuring SIP Settings

400HD Series IP Phones

- 72 -

16.5 Configuring SIP Timers

SIP Timers can be configured using the Web interface or configuration file.

To configure SIP timer settings using the Web interface:

1. Open the Signaling Protocol page (Configuration tab > Voice Over IP menu > Signaling

Protocols).

Figure 41: Signaling Protocols - SIP Timers

2. Configure the SIP Timers parameters using the table below as reference, and then click

Submit.

To configure SIP timer settings using the configuration file:

◼ Use the table below as reference.

Table 25: 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.

Note also:

If dual registration / redundant Genesys server is configured

and the configuration file parameter

'voip/signalling/sip/redundant_proxy/dual_reg/t1' is then

configured, its value will override 'Retransmission Timer T1'.

See also Section A.3.18 and Section A.3.18.1.

16. Configuring SIP Settings

400HD Series IP Phones

- 73 -

Parameter

Description

[voip/signalling/sip/redundant_proxy/dual_r

eg/t1]

Only relevant if dual registration / redundancy server is

configured. Allows quicker retransmission of SIP messages

than the Web interface parameter 'Transmission Timer T1'

and overrides it if configured.

◼ Default: 20 milliseconds. Range: 20-200.

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

[voip/signalling/sip/sip_invite_timer]

The SIP INVITE timer according to RFC 3261.

◼ The valid range is 0 to 65535. The default value is 32000.

Session-Expires

[voip/signalling/sip/session_timer]

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.

Min-SE

[voip/signalling/sip/min_session_interval]

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.

[voip/signalling/sip/unregister_on_voip_relo

ad]

If the VoIP application needs to be reloaded, the application

by default sends a SIP Registration message with Expires:0,

which means unregister.

By setting this parameter to 1, the application will not send

the unregistration message when its reloaded.

16. Configuring SIP Settings

400HD Series IP Phones

- 74 -

16.6 Configuring SIP QoS

SIP Quality of Service (QoS) can be configured using the Web interface or configuration file.

To configure SIP QoS using the Web interface:

1. Open the Signaling Protocol page (Configuration tab > Voice Over IP menu > Signaling

Protocols).

Figure 42: Quality of Service

2. Configure the SIP QoS parameters using the table below as reference, and then click Submit.

To configure SIP QoS using the configuration file:

◼ Use the table below as reference.

Table 26: 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.

16.7 Configuring SIP Reject Code

Reject Code can be configured using the Web interface or configuration file.

To configure Reject Code using the Web Interface:

1. Open the Services page (Configuration tab > Voice Over IP menu > Services > General

Parameters).

Figure 43: General Parameters - Reject Code

2. Configure 'Reject Code' using the table below as reference, and then click Submit.

To configure Reject Code using the configuration file:

◼ Use the table below as reference.

Table 27: Reject Code Parameter

Parameter

Description

Reject Code

[voip/services/reject_code]

Configures the reject code that the phone sends when the Reject

softkey is pressed or while DND is activated.

Valid values are:

[CODE_603]

[CODE_486]

17. Configuring Dialing

400HD Series IP Phones

- 75 -

17 Configuring Dialing

Dialing parameters can be configured using the Web interface or configuration file.

17.1 Configuring Voice Dialing through VocaNOM

AudioCodes' phones are directly integrated with AudioCodes' VocaNOM service to allow voice

dialing. The network administrator can configure voice dialing capabilities on phones, allowing users

to voice dial any other user in the same corporate directory.

To configure voice dialing using the Web interface:

1. Open the Services page (Configuration tab > Voice Over IP menu > Services) and scroll down

until the VocaNOM section.

Figure 44: Web Interface - VocaNOM

2. Configure the parameters using the table below as reference, and then click Submit.

17. Configuring Dialing

400HD Series IP Phones

- 76 -

To enable the service using the configuration file:

◼ Use the table below as reference.

Table 28: Voice-Dialing Parameter Descriptions

Parameter

Description

VocaNOM Number

[voip/services/vocanom/number]

Defines the number that the phone dials to access the

VocaNOM server, either directly, or indirectly, via the Skype for

Business server. Example: 7777

VocaNOM Label

[voip/services/vocanom/label]

Defines the name that will be displayed in phone screens after

users press their configured VocaNOM key to voice-dial

another party using the VocaNOM service. Default: VocaNOM

Use VocaNOM server directly

[voip/services/vocanom_server/enabled]

Can be enabled or disabled. The user's experience remains the

same whether enabled (direct voice dialing) or disabled

(indirect voice dialing). Direct or indirect voice dialing occurs in

the background, so user experience is unaffected. When

enabled (direct voice dialing), the call is forwarded directly to

the server. When disabled (indirect voice dialing), the call is

forwarded via the Skype for Business server. The VocaNOM

server can be on premises or in the cloud.

◼ [0] Access to the VocaNOM server is indirect via the Skype

for Business server [default]

◼ [1] Access to the VocaNOM server is direct

Server IP Address

[voip/services/vocanom_server/ip_addres

Only displayed in the Web interface if the previous parameter

(above) is enabled. Defines the VocaNOM server's IP address.

The server can be either in the AWS cloud (Amazon Web

Services) or on premises. Default: 0.0.0.0

[voip/services/vocanom_server/port]

Defines the port number on the VocaNOM server. Its value

must match Transport Mode.

◼ 5060 [for UDP, TCP]

◼ 5061 [default] [for TLS]

[voip/services/vocanom/transport_mode]

Defines the Transport Mode for sending SIP messages.

◼ TLS [Default]

◼ UDP

◼ TCP

17. Configuring Dialing

400HD Series IP Phones

- 77 -

17.2 Configuring General Dialing Parameters

General dialing parameters can be configured using the Web interface or configuration file.

To configure dialing using the Web interface:

1. Open the Dialing page (Configuration tab > Voice Over IP menu > Dialing).

Figure 45: Web Interface Dialing

2. Configure the parameters using the table below as reference, and then click Submit.

To configure dialing using the configuration file:

◼ Use the table below as reference.

Table 29: 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). Range is 0 to 10.

Default = 5.

Interdigit Short Timeout

[voip/dialing/interdigit_short_timeout]

Shorter than 'Dialing Timeout' (see above). Default: 3 seconds.

Implemented as 0S for the Dial Map. If a user wants to make an

international call by dialing 00 and wants to dial the

secretary/operator by dialing 0, the user can do both by adding

0S to the Dial Map. For example, if the digit map string= *xx|[2-

9]11|0S|[2-9]xxxxxxxxx|1xxx[2-9]xxxxxx, it has 0S in it. When

the user dials 0, 0 will match 0S and will therefore start the

'Interdigit Short Timeout' timer. After this timeout, 0 is dialed

out. User can dial 00 or 0123 within the 'Interdigit Short

Timeout'. After the 'Dialing Timeout', the string is dialed out.

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

Range is 3 to 32. Default = 32.

Enable Dialing Complete Key

[voip/dialing/dial_complete_key/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.

17. Configuring Dialing

400HD Series IP Phones

- 78 -

Parameter

Description

Dialing Complete Key

[voip/dialing/dial_complete_key/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.

Range: 1 to 300. Default = 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. To initiate a call, users 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).

17. Configuring Dialing

400HD Series IP Phones

- 79 -

17.3 Configuring Auto Redial

The administrator is responsible for enabling/disabling the auto-redial feature. If enabled and a

called party is unavailable because they're busy (for example), the caller's phone's SCREEN prompts

Extension Busy. Activate auto redial on busy?

If the caller then activates auto-redial by pressing Yes, the busy extension is automatically redialed

every n seconds.

The administrator is also responsible for configuring this frequency.

To configure auto redial using the Web interface:

1. In the Dialing page, scroll down until the Automatic Redial On Busy section (VoIP > Dialing >

Automatic Redial On Busy).

Figure 46: Automatic Redial On Busy

2. Configure the parameters using the table below as reference, and then click Submit.

To configure dialing using the configuration file:

◼ Use the table below as reference.

Table 30: Automatic Redial On Busy Parameters

Parameter

Description

Activate

[voip/dialing/automatic_redial_on_busy/e

nabled]

Allows the administrator disable/enable the feature.

◼ [0]=Disabled

◼ [1]=Enabled

Default: 0

Timeout

[voip/dialing/automatic_redial_on_busy/r

etry_timer]

Visible only if the feature is enabled. Range: 3-120. Default: 30.

If the feature is activated and the timer lapses, an outgoing call

to the busy destination is established. If the feature is

activated, a countdown screen is displayed: Dialing <ext>

within <x>s (Line <n>) The screen shows the timer, the remote

extension and the line number.

17. Configuring Dialing

400HD Series IP Phones

- 80 -

17.4 Configuring Dial Tones

Dial Tones settings can be configured using the Web interface or configuration file.

To configure Dial Tones using the Web interface:

1. Open the Dialing page (Configuration tab > Voice Over IP menu > Dialing).

Figure 47: Dialing Page - Tones

2. Configure the tones parameters using the table below as reference, and then click Submit.

3. Open the Services page (Configuration tab > Voice Over IP menu > Services) and scroll down

to the General Parameters.

Figure 48: Services Page - Tones

4. Configure the tones parameters using the table below as reference.

17. Configuring Dialing

400HD Series IP Phones

- 81 -

To configure Dial Tones using the configuration file:

◼ Use the table below as reference.

Table 31: 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.

Range:1 to 300. Default: 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.

Range:1 to 300. Default: 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.

Range:1 to 300. Default: 120.

Secondary Dial Tone

[voip/dialing/secondary_dial_tone/enabled]

◼ Enables the secondary dial tone.

◼ [0] Disable (default) - Phone doesn't use secondary

dial tone.

◼ [1] Enable - Phone plays secondary dial tone if the

secondary dial tone key is pressed (first digit). For

example, when pressing 9 to get an external dial

tone, a different dial tone (not configurable) is

played as the second dial tone.

Secondary Dial Tone Key

[voip/dialing/secondary_dial_tone/key_sequence]

Defines the secondary dial tone is played if this is the

first key pressed.

◼ Range: 0 to 9. Default: 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_duration

]

Defines the duration for which a stutter tone is played

when you have unheard messages.

◼ Range:1000 to 60000.

◼ Default: 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)

17. Configuring Dialing

400HD Series IP Phones

- 82 -

17.5 Configuring DTMF

Dual-Tone Multi-Frequency (DTMF) signaling can be configured using the Web interface or

configuration file.

To configure DTMF using the Web interface:

1. Open the Dialing page (Configuration tab > Voice Over IP menu > Dialing).

Figure 49: DTMF Transport Mode

2. You can choose one of three available DTMF transport methods:

3. The Web interface allows selecting Inband, RFC 2833, or Via SIP as the DTMF transport type.

4. Configure the parameter using the table below as reference, and then click Submit.

To configure DTMF using the configuration file:

◼ Use the table below as reference.

Table 32: DTMF Transport Mode

Parameter

Description

DTMF Transport Mode

[voip/media/out_of_band_dtmf]

DTMF transport mode.

◼ [INBAND] Inband

◼ [RFC2833] RFC 2833 (default)

◼ [VIA_SIP] Via SIP

[voip/media/dtmf_via_sip_force_flag]

Must be set to 1 to enable Via SIP as DTMF transport type.

/voip/audio/gain/dtmf_rtp_event_signal_

level

Allows the network administrator to control the DTMF tones

level.

◼ 0 db (Minimum)

◼ 31 db (Maximum)

◼ The Web interface parameter 'DTMF Transport Mode' and the cfg file parameter

'voip/media/out_of_band_dtmf' are related; changing one changes the other.

◼ If the cfg file parameter 'voip/media/dtmf_via_sip_force_flag' is enabled, a SIP

message is sent in addition to the RTP message. If it is disabled, only one message is

sent, according to the selected DTMF transport type.

17. Configuring Dialing

400HD Series IP Phones

- 83 -

17.6 Configuring Digit Maps and Dial Plans

Digit maps and Dial plans can be configured using the Web interface or configuration file.

To configure digit map and dial plan using the Web interface:

1. Open the Dialing page (Configuration tab > Voice Over IP menu > Dialing).

Figure 50: Digit Map and Dial Plan

2. Configure the parameters using the table below as reference, and then click Submit.

To configure digit map and dial plan using the configuration file:

◼ Use the table below as reference.

Invalid Tokens will be ignored by the application.

Table 33: Digit Map and Dial Plan Parameters

Parameter

Description

Digit Map

[voip/signalling/sip/digit_map]

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[2-

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

• 9011xxx.: International rule with prefix

• 1[2-9]xxxxxxxxx: LD rule without prefix

17. Configuring Dialing

400HD Series IP Phones

- 84 -

Parameter

Description

• 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.

Dial Plan

voip/signalling/sip/number_rules

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.

17. Configuring Dialing

400HD Series IP Phones

- 85 -

17.7 Configuring Headset LED to Stay On

IT administrators can configure the headset LED to stay on when the phone is on standby and when

it is in conversation mode.

Headset must be configured as the default audio device for the feature to function (see

Section 17.8).

To configure the headset LED to stay on:

◼ Use the table below as reference.

Table 34: Headset LED Parameter

Parameter

Description

[voip/highlight_audio_device]

Allows the headset LED to stay on when the phone is on standby and when

it is in conversation mode.

Functions only when headset is configured as the default audio device.

Configure either:

◼ [NONE] (Default) Headset LED illuminates only when the phone is in

conversation mode.

◼ [HEADSET] = Headset LED illuminates when the phone is on standby and

when it is in conversation mode

17. Configuring Dialing

400HD Series IP Phones

- 86 -

17.8 Configuring Default Audio Device

The default audio device (speaker or headset) can be configured using the Web interface or

configuration file.

To configure default audio device using the Web interface:

1. Open the Dialing page (Configuration tab > Voice Over IP menu > Dialing) and scroll down to

Default Audio Device.

Figure 51: Default Audio Device

2. Configure the parameter using the table below as reference and then click Submit.

To configure default Audio Device using the configuration file:

◼ Use the table below as reference.

Table 35: Audio Device Parameter

Parameter

Description

Audio Device

[voip/answer_device]

Sets the default audio device to answer or initiate a new call when no

explicit audio device is set.

For example:

◼ When pressing the Answer softkey.

◼ 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]

18. Configuring Ring Tones

400HD Series IP Phones

- 87 -

18 Configuring Ring Tones

Here's how to configure and upload ring tones to the phone.

18.1 Configuring Distinctive Ring Tones

You can configure a phone to ring in a distinct tone per caller, thus facilitating caller recognition and

saving others from unnecessary disruptions to their activities if the phone is shared.

To configure a distinctive ring tone using the Web interface:

1. Open the Tones page (Configuration tab > Personal Settings menu > Tones) and scroll down

until 'Distinctive Ringing Setting'.

Figure 52: Distinctive Ringing

2. Configure using the table below as a reference.

To configure a distinctive ring using the configuration file:

◼ Use the table below as a reference.

Table 36: Distinctive Ringing Parameters

Parameter

Description

Ringing Tone Name

[voip/distinctive_ringing/0-4/ringtone]

A name to assign to a distinctive ring tone. The default ring

tone names are Ring01 – Ring11. (Optionally, you can select

and manually upload a customized ring tone – see Section

18.3). If you do not enter a name, the phone assigns the

tone's filename (without the .wav file extension) as the

name of the tone.

Ringing Tone Types

[voip/distinctive_ringing/0-4/type]

This is the 'Alert-Info' header's content in the INVITE

message. It should be configured in the SIP proxy or

application server.

Used to distinguish between different calls.

18. Configuring Ring Tones

400HD Series IP Phones

- 88 -

18.1.1 Example of Configuring a Distinctive Ring

Here's how to configure a ring tone whose name is Ring05 to ring when the Alert-Info Header

received in the INVITE message contains external_call1.

Example using the configuration file:

◼ Configure parameter 'voip/distinctive_ringing/4/ringtone' to Ring05

◼ Configure parameter 'voip/distinctive_ringing/4/type' to externall_call1

Example using the Web interface:

Figure 53: Distinctive Ringing

The Alert-Info header must contain external_call1, as shown below. This is the INVITE the phone

receives from the proxy / application server.

Figure 54: Example of the Alert-Info Header

The phone will play ring tone 5 irrespective of the selected line ring tone.

18. Configuring Ring Tones

400HD Series IP Phones

- 89 -

18.2 Configuring CPT Regional Settings

It's important to match your phone's Call Progress Tones (CPT) to the country in which your phone

is located. Here's how to configure it.

To configure your region using the Web interface:

1. Open the Tones page (Configuration tab > Personal Settings > Tones).

Figure 55: Tones - Regional Settings

2. From the 'Current Location' drop-down list, select the country in which your phone is located.

Use the table below as reference.

3. Click Submit.

To configure regional location using the configuration file:

◼ Use the table below as reference.

Table 37: Regional Parameters

Parameter

Description

Current Location

[voip/regional_settings/selected_country]

Defines the country in which your phone is located. The

behavior and parameters of analog telephones lines vary

between countries. CPTs are country-specific. The phone

automatically selects the correct regional settings according

to this parameter. Supported countries are:

◼ [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 (Default)

◼ [India] India

[voip/regional_settings/use_config_file_valu

es]

Enables the user-defined CPT. When this parameter is

enabled, the 'selected_country' parameter is not relevant

and the CPT values below can be determined by the user.

◼ [0] - Disable (default)

◼ [1] - Enable

18. Configuring Ring Tones

400HD Series IP Phones

- 90 -

Parameter

Description

Call Progress Tones (CPT)

Note: Up to 10 CPTs can be configured (voip/regional_settings/call_progress_tones/0…9).

[voip/regional_settings/call_progress_tones

/%d/enabled]

Enables the specific CPT.

◼ [0] - Disable

◼ [1] - Enable

[voip/regional_settings/call_progress_tones

/%d/name]

Defines the name of the CPT.

[voip/regional_settings/call_progress_tones

/%d/cadence]

Defines the cadence type of the tone.

◼ [0] - Continuous signal

◼ [1] - Cadence signal

◼ [2] - Burst signal

[voip/regional_settings/call_progress_tones

/%d/frequency_a]

Defines the low frequency (in Hz) of the tone.

Range:300 - 1980 Hz, in steps of 1 Hz.

Unused frequencies must be set to zero.

[voip/regional_settings/call_progress_tones

/%d/frequency_b]

Defines the high frequency (in Hz) of the tone.

Range:300 - 3000 Hz, in steps of 1 Hz.

Unused frequencies must be set to zero.

[voip/regional_settings/call_progress_tones

/%d/frequency_a_level]

Output level of the low frequency tone (in -dBm) in Call

Progress generation. Range: 0 - 63, where 63 is mute.

[voip/regional_settings/call_progress_tones

/%d/frequency_b_level]

Output level of the low frequency tone (in -dBm) in Call

Progress generation.Range:0 - 63, where 63 is mute.

[voip/regional_settings/call_progress_tones

/2/name]

Default: busy_tone The calling party hears a busy tone if the

called party's line is busy. The busy tone complies with

international telcom standards in traditional non-VOIP

telephony systems.

[voip/regional_settings/call_progress_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. In units of 10

msec.Range:0 - 10000.

[voip/regional_settings/call_progress_tones

/%d/tone_off_0]

tone_off_0 to tone_on_3.

If the signal is Cadence, then this value represents the off

duration, in units of 10 msec. If 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.Range:0 - 10000.

18. Configuring Ring Tones

400HD Series IP Phones

- 91 -

18.3 Uploading Ring Tones

New Ring Tones can be uploaded using the Web interface or configuration file.

• 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 PCM 16-kHz audio sample rate and 16-bit audio sample

size, Intel PCM encoding).

• For the phone to use an uploaded ring tone, select it on the phone (see the phone's

User's Manual).

To upload a ring tone using the Web interface:

1. Open the Tones page (Configuration tab > Personal Settings menu > Tones).

Figure 56: Upload Ringing Tone

2. In the 'Ringing Tone Name' field, enter the name of the ring tone file 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.

3. 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.

4. Click Submit; the file is loaded to the phone and displayed in the Ring Tone Name list.

To delete Ring Tones using the Web interface:

• Select the 'Delete' check boxes corresponding to the ring tone that you want to delete,

and then click Submit.

18. Configuring Ring Tones

400HD Series IP Phones

- 92 -

To upload Ring Tones using the configuration file:

• Use Table 38 below as reference.

Table 38: Ring Tone Parameters

Parameter

Description

Ringing Tone Name File Location

[provisioning/ring_tone_uri]

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

Note:

◼ 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.

◼ For the feature to function, the file must first be

compressed to zip / tgz format. The phone won't accept

a simple .wav file format.

[personal_settings/lines/0/ring_tone] -

[personal_settings/lines/3/ring_tone]

Lets administrators set a ring tone for each line extension

(up to four line extensions). Administrators can choose any

one of the eleven ring tones available: Ring01 - Ring11

18.4 Configuring Beeps to Headsets when a Call Comes in to a

Call Center

You can configure a beep instead of ringing to be played to agents' headsets when a call comes in to

a Call Center. The beep is heard even if 'Auto answer' is configured to 0.

To play beeps to headsets instead of ringing:

◼ Use the table below as reference.

Table 39: Configuring Beeps to be Played to Headsets when Calls Come in

Parameter

Description

[voip/beep_to_ringing_device/enabled]

Enables/disables beeping the device.

◼ [0] Disable (default)

◼ [1] Enable

[voip/beep_to_ringing_device/number_of_beeps]

If the feature is enabled, the number of beeps

must be configured. Default: 3.

18. Configuring Ring Tones

400HD Series IP Phones

- 93 -

18.5 Configuring the Phone to play Fast Busy Tone if

Automatically Disconnected on Remote Side

You can configure the phone to play a fast busy tone if it is automatically disconnected on the remote

side. You can also configure for how long this fast busy tone is played. When the phone plays the

tone, it also displays a 'Disconnected' message for the same length of time.

To configure this feature:

◼ Use the table below as reference.

Table 40: Configuring the Phone to Play a Fast Busy Tone when Automatically Disconnected on Remote Side

Parameter

Description

[enable_remote_disconnect_warningTone]

Allows you to enable or disable playing a fast

busy tone if the phone is automatically

disconnected on the remote side.

◼ [0] (default) If the phone accepts an

incoming call and the remote side

automatically ends it (disconnects), the

phone does not play any tone and no

message is displayed.

◼ [1] If the phone accepts an incoming call

and the remote side automatically ends

(disconnects) it, the phone plays a fast busy

tone and displays a Disconnected message

(see the parameter description below).

[voip/dialing/automatic_disconnect_delay_timer]

Defines for how long the fast busy tone is

played and for how long the 'Disconnected'

message is displayed if the warningTone

parameter above is enabled and the phone is

automatically disconnected on the remote side.

Default: 3000 ms.

18. Configuring Ring Tones

400HD Series IP Phones

- 94 -

18.6 Configuring the Beep (Ring) to Play via an Answering Device

Network administrators can configure whether or not the beep (ring) plays via an answering device

where the answering device is the speaker, headset and handset.

To configure this feature:

◼ Use the table below as reference.

Table 41: Configuring the Beep (Ring) to Play via an Answering Device

Parameter

Description

voip/ring_via_answer_device/enabled

Defines whether or not the beep (ring) plays via

an answering device. The answering device is

the speaker, headset and handset.

◼ [0] The beep doesn't play via an answering

device (default)

◼ [1] The beep plays via an answering device

19. Configuring Media Settings

400HD Series IP Phones

- 95 -

19 Configuring Media Settings

Here's how to configure media settings.

19.1 Configuring Media Streaming

The Configuring Media Streaming feature can be configured using the configuration file. Configure

the parameters using the table below as reference.

Table 42: 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_conne

ction_detection]

If enabled an active call will be automatically disconnected if

no RTP packet is received within pre-defined time.

◼ [0] Disable (default)

◼ [1] Enable

◼ [voip/media/broken_conne

ction_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. Default: 10.

19. Configuring Media Settings

400HD Series IP Phones

- 96 -

19.2 Configuring RTP Port Range and Payload Type

RTP Port Range and Payload Type can be configured using the Web interface or configuration file.

To configure RTP Port Range and Payload Type using the Web interface:

1. Open the Media Streaming page (Configuration tab > Voice Over IP menu > Media Streaming).

Figure 57: Media Streaming

2. Configure 'RTP Port Range' and 'Payload Type'parameters using the table below as reference,

and then click Submit.

To configure RTP Port Range and Payload Type using the configuration file:

◼ Use the table below as reference.

Table 43: 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.

Range: 96 - 127. Default: 101.

RTP Port Range

[voip/media/media_port]

Defines the base port for the range of Real Time Protocol (RTP) voice

transport ports which the enterprise IT administrator must open on

the network's firewall.

Default: 4000. Valid possible ports (if the default is selected as base

port): 4000-4126. If, for example, 5000 is selected as the base port,

the valid possible ports will be 5000-5126.

19. Configuring Media Settings

400HD Series IP Phones

- 97 -

19.3 Configuring RTP QoS

RTP QoS can be configured using the Web interface or configuration file.

To configure RTP QoS using the Web interface:

1. Open the Media Streaming page (Configuration tab > Voice Over IP menu > Media Streaming).

Figure 58: Quality of Service

2. Configure the QoS parameter using the table below as reference, and then click Submit.

To configure RTP QoS using the configuration file:

◼ Use the table below as reference.

Table 44: RTP QoS Parameter

Parameter

Description

Type of Service (ToS)

[voip/media/media_tos]

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). Default: 184.

19.4 Configuring RTP/SRTP Capability Negotiation

Support for RTP/SRTP capability negotiation according to a subset of RFC 5939, in compliance with

Genesys / Avaya environment requirements. The phone sends an SDP offer with RTP and SRTP

capabilities according to configuration parameter

To configure RTP/SRTP Capability Negotiation using the configuration file:

◼ Use the table below as reference.

Table 45: RTP/SRTP Capability Negotiation Parameter

Parameter

Description

[voip/media/srtp/NegotiationMode]

Configurable values are:

◼ Basic (default) [RTP/SRTP negotiation according to the

document IMTC Best Practices for SIP Security. This mode is

supported by Broadsoft, Microsoft and many other vendors]

◼ RFC 5939 [RTP/SRTP capability negotiation "a=tcap", "a=acap"

and “a=pcfg” attributes as described in RFC 5939]

19. Configuring Media Settings

400HD Series IP Phones

- 98 -

19.5 Configuring Codecs

Codecs can be configured using the Web interface or configuration file.

To define the Codecs using the Web interface:

1. Open the Media Streaming page (Configuration tab > Voice Over IP menu > Media Streaming).

Figure 59: Media Streaming - Codecs

2. Configure the parameters using the table below as reference, and then click Submit.

To define the Codecs using the configuration file:

◼ Use the table below as reference.

Table 46: Codec Parameters

Parameter

Description

[voip/codec/codec_info/%d/enabl

ed]

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. 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

19. Configuring Media Settings

400HD Series IP Phones

- 99 -

Parameter

Description

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 (default)

◼ [PCMA] G.711 A-Law

◼ [PCMU] G.711 Mu-Law

◼ [G729] G.729

◼ OPUS

For example, voip/codec/codec_info/0/name=G722.

Note:

◼ If OPUS is selected from the list of codecs on a phone that doesn’t

support the OPUS codec, a warning pops up: 'The hardware

doesn't support the OPUS codec' (see the figure below).

◼ If a call starts with the OPUS codec and the user makes a

conference call, the phone switches from the OPUS codec to G.711

for the conference call.

◼ To enable OPUS management for enhanced voice quality, see

Section 19.6.

[voip/media/opus_payload]

Allows the network administrator to configure the OPUS dynamic

payload type. Default: 111.

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.

OPUS Jitter Buffer Minimum Delay

[voip/codec/opus/jitter_buffer/mi

n_delay]

The initial and minimum delay of the OPUS adaptive Jitter Buffer

mechanism, which compensates for network impairments. The value

should be set according to the expected average jitter in the network

(in milliseconds).

◼ Range: 30-500. Default: 30.

[system/activation_keys/amr_code

Activation key (string) required to unlock AMR coder (relevant for

supporting firmware only).

19. Configuring Media Settings

400HD Series IP Phones

- 100 -

Figure 60: Media Streaming - Codecs

19.6 Configuring OPUS Management

OPUS management for enhanced voice quality allows the OPUS audio codec's configuration to be

changed on the fly when impairments are detected in the network. The OPUS functions at a lower

channel bit rate and consumes less bandwidth, delivering better voice quality despite the poor

network conditions. OPUS management can be configured using configuration file.

To configure OPUS management using the configuration file:

◼ Use the table below as reference.

Table 47: OPUS Management Parameter

Parameter

Description

[voip/voice_quality/mode]

Enables OPUS management for enhanced voice quality.

◼ [DISABLE] Disable (default)

◼ [ENABLE_RTCP_FEEDBACK] Enable

20. Configuring Voice Settings

400HD Series IP Phones

- 101 -

20 Configuring Voice Settings

Here's how to configure voice settings.

20.1 Configuring Gain Control

See Section 23.1 for detailed information.

20.2 Configuring Jitter Buffer

Jitter Buffer can be configured using the Web interface or configuration file.

To define Jitter Buffer using the Web interface:

1. Open the Voice page (Configuration tab > Voice Over IP menu > Voice) and then scroll down

to Jitter Buffer.

Figure 61: Voice – Jitter Buffer

2. Configure the Jitter Buffer parameters using the table below as reference, and then click

Submit.

To define Jitter Buffer using the configuration file:

◼ Use the table below as reference.

Table 48: 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).

◼ Range:10 to 150. Default: 10.

Optimization Factor

[voip/audio/jitter_buffer/optimization_factor]

The adaptation rate of the jitter buffer mechanism.

Higher values cause the jitter buffer to respond faster

to increased network jitter.

◼ Range: 0 to 13. Default: 10.

20. Configuring Voice Settings

400HD Series IP Phones

- 102 -

20.3 Configuring Silence Compression

The Silence Compression feature can be configured using the Web interface or configuration file.

To configure Silence Compression using the Web interface:

1. Open the Voice page (Configuration tab > Voice Over IP menu > Voice) and then scroll down

to Silence Compression.

Figure 62: Voice - Silence Compression

2. Configure the parameter using the table below as reference, and then click Submit.

To configure Silence Compression using the configuration file:

◼ Use the table below as reference.

Table 49: 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

20. Configuring Voice Settings

400HD Series IP Phones

- 103 -

20.4 Configuring Noise Reduction

The Noise Reduction feature can be configured using the Web interface.

It's strongly advisable not to change the default values without consulting AudioCodes.

To configure Noise Reduction using the Web interface:

1. Open the Voice page (Configuration tab > Voice Over IP menu > Voice) and then scroll down

to Noise Reduction.

Figure 63: Voice - Noise Reduction

2. Configure the parameters referring to the table below, and then click Submit.

Table 50: Noise Reduction Parameters

Parameter

Description

Enable Noise Reduction

Efficiently suppresses stationary background noise

(e.g., office noise) and enhances speech by improving

SNR (Signal to Noise Ratio) and emphasizing the

important features of the desired speech signal. On the

phones, an additive noise is captured due to

environmental settings, such as office noise, air

conditioning, etc.

◼ Disable

◼ Enable (default)

Enable Noise Reduction Suppression

Enables NRS.

◼ Disable

◼ Enable (default)

Enable Noise Reduction Post Gain

Enables the noise reduction post gain. The post gain is

additional attenuation of the signal which is performed

after the noise reduction operation. It is needed if the

noise is highly non-stationary or of low SNR. Note that

the post gain attenuates also the speech signal.

◼ Disable (default)

◼ Enable

Stationary Level:(0-3)

Configures the noise reduction sensitivity to stationary

noise.

0 = sensitive to stationary noise only.

3 = sensitive to highly non-stationary noise.

Default: 2

Maximal Attenuation:(0-15)

Defines the maximal noise attenuation during noise-

only periods. Default: 2.

20. Configuring Voice Settings

400HD Series IP Phones

- 104 -

20.5 Configuring Echo Cancellation

• It is strongly advisable to leave the echo cancellation parameters at their defaults and

not to configure different values.

• Contact your AudioCodes representative if you encounter an echo cancellation

related issue.

You can view the following echo cancellation related parameters in the configuration file

(Management tab > Manual Update > Configuration File).

◼ voip/audio/echo_cancellation/enabled

◼ voip/audio/echo_cancellation/extended_nlp/enabled

◼ voip/audio/echo_cancellation/handset/HPF_mode

◼ voip/audio/echo_cancellation/handsfree/HPF_mode

◼ voip/audio/echo_cancellation/headset/HPF_mode

◼ voip/audio/echo_cancellation/nlp/max_delay

◼ voip/audio/echo_cancellation/nlp/mode

21. Configuring Extension Lines

400HD Series IP Phones

- 105 -

21 Configuring Extension Lines

21.1 On the Phone

Before you can make a call, you must configure an extension line (SIP account). Here's how to

configure an extension line on the phone.

• 440HD / 430HD phone screens are shown here.

• 420HD / 405 / 405HD phone screens are virtually identical.

To configure an extension line on the phone:

1. Open the SIP Accounts screen (MENU key > Administration > SIP Accounts).

2. To add a new line extension (SIP account), press the Add softkey or skip to Step 3 to define an

existing line extension (SIP account).

3. 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:

• Line Mode (applies only to the 440HD/ 430HD phones)

• User ID

• Display Name

• Authentication Name

• Authentication Password

• SIP Proxy Address

5. After each parameter setting, press the Save softkey to save the setting.

6. After completing configuring SIP account settings, press the Back softkey until you're

prompted that the phone is rebooting. The screen indicates a warning message: 'Phone is

rebooting to apply new settings'.

21. Configuring Extension Lines

400HD Series IP Phones

- 106 -

21.2 Using the Web Interface and Configuration File

Here's how to configure an extension line (SIP account) using the Web interface and configuration

file.

The Web interface page of the 440HD / 430HD phone are shown here. The Web pages of

the 420HD / 405/405HD phones are identical, except that on the 420HD/405/405HD

phones:

• 'Line Label' is displayed instead of 'Line Mode'

• Two lines can be configured instead of six as on the 430HD/440HD phones

To configure an extension line (SIP account) using the Web interface:

1. Open the Line Settings page (Configuration tab > Voice Over IP menu > Line Settings).

Figure 64: Line Settings

2. Configure the Line Settings using the table below as reference, and then click Submit.

21. Configuring Extension Lines

400HD Series IP Phones

- 107 -

To configure an extension line (SIP account) using the configuration file:

◼ Use the table below as reference. %d refers to the line number.

Table 51: Line Parameters

Parameter

Description

Line Activate

[voip/line/%d/enabled]

Activates or deactivates the line. See also Section. 25.2.1.1.

◼ [0] = Disabled (this is the default for the second line and

higher in the configuration file)

◼ [1] = Enabled (this is the default for the first line voip/line/0/

in the configuration file).

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.

Line Label

[voip/line/%d/extension_display]

[Only applies to the 420HD and 405/405HD phones]

Set the string that will be displayed in the phone screen for local

extension. If not set, the local extension displayed will be the

user ID (self-number).

Line Mode

[voip/line/%d/line_mode]

[Only applies to 430HD/440HD phones]

Determines the line mode: PRIVATE (default) or SHARED.

SHARED allows enterprise employees to share an extension

number and manage a call as a group. When an employee places

a call on a SHARED line on hold, the call can be resumed from

any other employee sharing the line.

You can activate DnD per phone line (see Section 22.8).

22. Configuring Supplementary Services

400HD Series IP Phones

- 108 -

22 Configuring 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.

Configuration of specific supplementary services depends on the third-party application

server used in your organization.

To select the application server using the Web interface:

1. Open the Services page (Configuration tab > Voice Over IP menu > Services).

Figure 65: Services

2. Select the application server and then click Submit. Use the table below as reference.

To select the application server using the configuration file:

◼ Use the table below as reference.

Table 52: General Supplementary Services Parameters

Parameter

Description

Type

[voip/services/application_server_type]

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

◼ [GENBAND] Kandy Business Solutions softswitch solution.

Note:

◼ Parameters unique to the selected application server become

applicable in addition to this page's parameters.

◼ For more information about Genband’s KBS environment, see

the Kandy Business Solutions Feature Description Guide.

Presence

[system/current_user_presence_status/

enabled]

Only displayed if the application server selected [FreeSWITCH]

supports it. Enables the presence feature. The DND softkey on

the phone is replaced by Status; the phone shows and publishes

the presence status.

Feature Key Synchronization

Only displayed if BSFT is selected.

22. Configuring Supplementary Services

400HD Series IP Phones

- 109 -

22.2 Configuring Call Waiting

Call Waiting can be configured using the Web interface or configuration file.

To configure call waiting using the Web interface:

1. Open the Services page (Configuration tab > Voice Over IP menu > Services).

Figure 66: Services - Call Waiting

2. Configure the Call Waiting parameters using the table below as reference, and then click

Submit.

To configure call waiting using the configuration file:

◼ Use the table below as reference.

Table 53: 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

Generate Tone

[/voip/services/call_waiting/generate_tone

/enabled]

Determines whether the phone plays a call waiting tone:

◼ [0] The phone doesn’t play a call waiting tone.

◼ [1] The phone plays a call waiting tone (default).

22. Configuring Supplementary Services

400HD Series IP Phones

- 110 -

22.3 Configuring Call Forwarding

Call Forwarding can be configured using the Web interface, configuration file, or phone screen. In a

BroadSoft environment, Call Forwarding can be configured in the BroadSoft BroadWorks application

server (see under Appendix A for detailed information).

To configure call forwarding using the Web interface:

1. Open the Services page (Configuration tab > Voice Over IP menu > Services).

Figure 67: Services - Call Forward

2. Configure the parameters using the table below as reference, and click Submit.

To configure call forwarding using the configuration file:

◼ Use the table below as reference.

Table 54: Call Forward Parameters

Parameter

Description

Enable

[voip/line/%d/call_forward/enabled]

Enables the Call Forward feature.

◼ [0] Disable

◼ [1] Enable (default)

Activate

[voip/line/%d/call_forward/active]

Activates call forwarding, if it was enabled with the

parameter above.

◼ [0] (default) - Disable

◼ [1] Enable

Note: Call Forwarding is typically activated on the phone

screen (see the User's Manual).

Call Forward Type

[voip/line/%d/call_forward/type]

Determines the condition on 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 No-Reply, then

this parameter defines the time (in seconds) after which

incoming calls are forwarded when this is no reply. Range:0

- 7200. Default: 6.

Forward Destination

[voip/line/%d/call_forward/destination]

The destination to which the call is directed when call

forward is activated.

To configure call forwarding using the phone's screen:

◼ See the User's Manual for detailed information.

22. Configuring Supplementary Services

400HD Series IP Phones

- 111 -

22.4 Configuring a Conference

Three-way conferencing can be configured using the Web interface or configuration file.

To configure three-way conferencing using the Web interface:

1. Open the Services page (Configuration tab > Voice Over IP menu > Services).

Figure 68: Services - Conference

2. Configure the conferencing parameters using the table below as reference, and then click

Submit.

To configure three-way conferencing using the configuration file:

◼ Use the table below as reference.

Table 55: Conference Parameters

Parameter

Description

Mode

[/voip/services/conference/mode]

Sets the conference mode (when establishing 3-Way

Conference).

[LOCAL] = phone will establish the conference by itself.

[REMOTE] = phone will use remote media server to

establish the conference.

Remote Conference Media Server

[/voip/services/conference/conf_ms_addr]

Relevant only if 'Mode'(above) is REMOTE.

Defines the media server for establishing remote

conference.

For more information on this feature, see RFC 4579, Session Initiation Protocol (SIP) - Call Control -

Conferencing for User Agents.

22.5 Allowing the Initiator to Drop out of a Conference

The phone can be configured to allow the initiator of a 3-way conference to drop out the conference

when they on-hook the phone.

To configure the capability using the configuration file:

◼ Use the table below as reference.

Table 56: Allowing a Conference Initiator to Drop Out when On-Hooking

Parameter

Description

[voip/drop_from_3wc_when_on_hook]

Allows the initiator of a 3-way conference to drop out

of the conference when they on-hook the phone.

◼ [0] = The initiator won't drop out of the conference

when they on-hook the phone.

◼ [1] = The initiator will drop out of the conference

when they on-hook the phone.

22. Configuring Supplementary Services

400HD Series IP Phones

- 112 -

22.6 Configuring Automatic Dialing

Automatic Dialing can be configured using the Web interface or configuration file.

To define Automatic Dialing using the Web interface:

1. Open the Dialing page (Configuration tab > Voice Over IP menu > Dialing).

Figure 69: Dialing - Automatic Dialing

2. Configure dialing options according to the parameters in the table below, and then click

Submit.

To define Automatic Dialing using the configuration file:

◼ Use the table below as reference.

Table 57: 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 off-hook

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.

22. Configuring Supplementary Services

400HD Series IP Phones

- 113 -

22.7 Configuring Automatic Answer

The Automatic Answer feature is configured using the configuration file. Use the table below as

reference.

Table 58: 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 phone to automatically answer the call, the

phone answers the call immediately or after a timeout,

depending on the auto-answer 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, answer-after= 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.

Note:

◼ 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.

[voip/advanced_auto_answer/timeout]

The timeout before the call is answered (in seconds).

Range: 0 – 60 (seconds)

◼ [5] = 5 seconds (default)

◼ [0] = immediately

22. Configuring Supplementary Services

400HD Series IP Phones

- 114 -

Parameter

Description

[voip/advanced_auto_answer/type]

◼ SIP_Header (default) = identical to the parameter

'voip/auto_answer/enabled' described above

◼ Manual = the phone automatically answers incoming

calls according to the timeout configured in the

'voip/advanced_auto_answer/timeout' parameter

[voip/auto_answer_use_180/enabled]

◼ Determines whether or not the phone will ring before

auto answer, or if auto answer will occur immediately,

before the phone rings.

◼ [0] Disable (default) = No ringing occurs before auto

answer; auto answer occurs immediately

◼ [1] Enable – Ringing occurs before auto answer

• The configuration file parameter ‘voip/auto_answer/enabled’ must be set to 1 to

support the following:

▪ voip/advanced_auto_answer/timeout

▪ voip/advanced_auto_answer/type

▪ voip/auto_answer_use_180/enabled

IGS-2246

22. Configuring Supplementary Services

400HD Series IP Phones

- 115 -

22.8 Configuring Do Not Disturb (DnD)

The Do not Disturb (DnD) feature can be configured using the Web interface or configuration file. It

can also be configured in BroadSoft's BroadWorks (see under Appendix A.1.3).

To configure DnD using the Web interface:

1. Open the Services page (Configuration tab > Voice Over IP menu > Services).

Figure 70: Services - DnD

2. Configure using the table below as reference, and then click Submit.

To configure DnD using the configuration file:

◼ Use the table below as reference.

Table 59: Do Not Disturb Parameters

Parameter

Description

Enable

[voip/services/do_not_disturb/enabled]

Enables the DnD feature.

◼ [0] Disable

◼ [1] Enable (default)

Activate

[voip/line/%d/do_not_disturb/activated]

Activates the DnD feature per phone line, if the

parameter 'voip/services/do_not_disturb/enabled' is

enabled.

◼ [0] – Deactivate (default)

◼ [1] - Activate

Three DnD configurations are possible in phones' idle

screens:

1. If DnD is disabled, no notification will be displayed

2. If DnD is enabled for one line extension, one

notification is displayed

3. If DnD is configured for two line extensions, two

notifications are displayed.

Note: DnD can also be activated in the phone's screen

(more common).

To configure DnD on the phone:

◼ See the phone's User's Manual for detailed information.

22. Configuring Supplementary Services

400HD Series IP Phones

- 116 -

22.9 Configuring Call Pick Up

Call Pick Up only applies to the 430HD and 440HD phones.

Since the Call Pick Up feature is relevant only when Busy Lamp Field (BLF) is activated, the call pickup

parameters appear as BLF related parameters.

To configure Call Pick Up using the Web interface:

1. Open the Services page (Configuration tab > Voice Over IP menu > Services).

Figure 71: Services - BLF Support - Call Pick Up

2. Under the BLF Support section, set the 'Activate' field to Enable.

3. Configure Call Pick Up according to the parameters in the table below, and then click Submit.

Table 60: Call Pick Up Parameters

Parameter

Description

Call Pick Up

[voip/services/call_pickup/enabled]

Allows call pickup by pressing on the relevant BLF key

when the remote phone's state is 'ringing'.

◼ [0] Disable

◼ [1] Enable

Access code

[voip/services/call_pickup/access_code]

Allows a user to answer another's call by pressing a user-

defined sequence of phone keys. The default sequence is

**. The user dials the sequence + the other user's phone

number; the incoming call from the other phone is

forwarded to the user's phone.

For example, to pick up a call for extension 5000, dial

**5000.

22. Configuring Supplementary Services

400HD Series IP Phones

- 117 -

22.10 Configuring Message Waiting Indication

The Message Waiting Indication (MWI) feature can be configured using the Web interface or

configuration file.

To configure MWI using the Web interface:

1. Open the Services page (Configuration tab > Voice Over IP > Services).

Figure 72: Services - MWI

2. Configure using the table below as reference, and then click Submit.

To configure MWI using the configuration file:

◼ Use the table below as reference.

Table 61: MWI Parameters

Parameter

Description

Voice Mail Number

[voip/services/msg_waiting_ind/voice_mail_number]

Defines the extension number for accessing your

voice mail messages.

◼ The valid value is up to 64 characters.

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 IP Address or Host Name

[voip/services/msg_waiting_ind/subscribe_address]

The IP address or host name of the MWI server.

Default: 0.0.0.0

MWI Server Port

[voip/services/msg_waiting_ind/subscribe_port]

The port number of the MWI server.

Range: 1024-65535. Default: 5060.

MWI Subscribe Expiry Time

[voip/services/msg_waiting_ind/expiraition_timeout]

The interval between the MWI Subscribe

messages.

Range:0-86400. Default: 3600

[voip/services/msg_waiting_ind/always_send_port]

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)

22. Configuring Supplementary Services

400HD Series IP Phones

- 118 -

22.11 Configuring Busy Lamp Field

The Busy Lamp Field (BLF) feature can be configured using the Web interface or configuration file.

BLF only applies to the 430HD and 440HD phones.

To configure BLF using the Web interface:

1. Open the Services page (Configuration tab > Voice Over IP menu > Services).

Figure 73: Services - BLF

2. Configure using the table below as reference, and then click Submit.

To configure BLF using the configuration file:

◼ Use the table below as reference.

Table 62: BLF Parameters

Parameter

Description

Activate

[voip/services/busy_lamp_field/enabled]

Enables the BLF feature:

◼ [0] Disable

◼ [1] Enable (default)

Call Pick Up

See Section 22.9.

Access code

See Section 22.9.

BLF Subscription Period

[voip/services/busy_lamp_field/subscription_per

iod]

The interval between BLF and SIP SUBSCRIBE messages.

Range: 0 to 86400. Default: 3600.

User Resource List

[voip/services/busy_lamp_field/uri]

[Only displayed if 'Type' is set to BSFT]. 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.

Use Registrar as Application Server Address

[voip/services/busy_lamp_field/application_serv

er/use_registrar]

[Only displayed if 'Type' is set to BSFT]. 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

Use Registrar as Application Server Address

[voip/services/busy_lamp_field/application_serv

er/addr]

[Only displayed if 'Type' is set to BSFT and the

'voip/services/busy_lamp_field/application_server/use_

registrar' parameter is set to 0.]. Defines the IP address

or host name of the application server. The valid value is

up to 64 characters. Default: 0.0.0.0

22. Configuring Supplementary Services

400HD Series IP Phones

- 119 -

22.12 Configuring Advice of Charge

The Advice of Charge (AOC) feature can be configured using the Web interface or configuration file.

The feature permits an accurate estimate of the size of the bill which will eventually be charged to

be displayed.

To configure AOC using the Web interface:

1. Open the Services page (Configuration tab > Voice Over IP menu > Services) and then scroll

down to AOC Support.

Figure 74: Services – AOC Support

2. Configure the parameters using the table below as reference, and then click Submit.

To configure AOC using the configuration file:

◼ Use the table below as reference.

Table 63: AOC Parameters

Parameter

Description

Enable

[system/aoc/enabled]

Enables the 'advice of charge' feature.

◼ [0] Disable

◼ [1] Enable

Currency

[system/aoc/currency]

Sets the required currency for AOC display. The string represents the

currency name (e.g., USD, EUR, NIS, etc.)

Ratio

[system/aoc/ratio]

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.

22.13 Configuring a Tone to Alert to Long Hold

Network administrators can configure an audible indication to be played after a call has been on

hold for a long time. After a call has been on hold for a long time (the time is configurable), a

reminder tone will be played every 10 seconds until the call is taken off hold. To configure the feature

using the configuration file, use the table below as reference.

Table 64: Reminder Tone after Long Hold

Parameter

Description

[voip/lhcwrr_enabled]

Enables the feature.

◼ [1] (Default) Enabled. After the length of time configured for

configuration file parameter voip/lhcwrr_wait_time lapses (see the

parameter below), a reminder tone (beep) is played every 10 seconds

until the call is taken off hold.

◼ [0] Disabled. No reminder tone (beep) is played, regardless of how long

the call is on hold.

[voip/lhcwrr_wait_time]

Defines the length of time that must lapse before a reminder tone (beep) is

played. The tone will then be played every 10 seconds until the call is taken

off hold. Default: 120 seconds.

22. Configuring Supplementary Services

400HD Series IP Phones

- 120 -

22.14 Disabling the HOLD Key

The network administrator can disable hard keys and softkeys on phones using the configuration

file. The feature is motivated by the requirement on the part of some enterprises to control settings

remotely to comply with company policy.

Here's how to disable the HOLD key on the phone's keypad.

To disable the HOLD key using the configuration file:

◼ Use the table below as reference.

Table 65: Disabling the HOLD Key

Parameter

Description

[system/disable_hold_button]

Disables the HOLD key on the phone's keypad. The HOLD key

is used to place a call on hold, or to resume a held call.

Besides the HOLD key, other hard keys that can be disabled include speaker, headset, voicemail,

REDIAL, CONTACTS, MENU, TRANSFER, VOL and mute.

For more information, see Section 26.

22.15 Configuring Onhook Disconnect when Held

Only applies to AudioCodes' SIP-100 and SIP-200 devices.

Here's how to configure whether the device automatically goes idle (i.e., on-hook) when the last

remaining call is disconnected.

To configure onhook disconnect when held, using the configuration file:

◼ Use the table below as reference.

Table 66: Onhook Disconnect when Held

Parameter

Description

[voip/onhook_disconnect_when_held/enabl

ed]

Choose either:

◼ [0] Disable (default) - If the receiver is placed on-hook

after a call is put on hold, the call is put on speaker

rather; it isn't disconnected.

◼ [1] Enable - If the receiver is placed on-hook after a call is

put on hold, the call is disconnected.

22. Configuring Supplementary Services

400HD Series IP Phones

- 121 -

22.16 Configuring Ringing on the Default Audio Device

Here's how to configure whether ringing is heard on the default audio device.

To configure whether ringing is heard on the default audio device, using the configuration file:

◼ Use the table below as reference.

Table 67: Configuring Ringing on the Default Audio Device

Parameter

Description

[voip/called_ringing_device]

Determines which device rings when a call comes in.

Configure either:

◼ SPEAKER (default)

◼ HEADSET

◼ BOTH

◼ NO_RING

In call centers in which agents typically work close to one

another wearing headsets, the administrator can configure

HEADSET to prevent incoming calls ringing on speakers which

creates a noisy environment.

The parameter is bundled with the two parameters below

('hands_free_mode' and 'supervisor_listen_in') to maximize

flexibility for call center administrators, who can - for

example - disable ringing on the speaker yet enable hands-

free mode (talking and listening using the speaker).

[voip/hands_free_mode/enabled]

When disabled [0]:

◼ hands-free mode becomes unavailable

◼ pressing the speaker key does not have any effect

◼ when answering a call, the headset is the default audio

Configure:

◼ [1] = Enabled (default)

◼ [0] = Disabled

[voip/services/supervisor_listen_in/enabl

ed]

If enabled, a call center supervisor can pick up an agent's

handset and listen in on the conversation that the agent is

conducting on headphones with the customer, without the

customer at the other end sensing that the supervisor is

listening in (because the supervisor is in effect muted).

Configure:

◼ [1] = Enabled

◼ [0] = Disabled (default)

22. Configuring Supplementary Services

400HD Series IP Phones

- 122 -

22.17 Allowing an Incoming Call when the Phone is Locked

Here's how to allow or not allow an incoming call when the phone is locked.

To allow an incoming call when the phone is locked, using the configuration file:

◼ Use the table below as reference.

Table 68: Allowing an Incoming Call when the Phone is Locked

Parameter

Description

[system/lock/2-5/allow_incoming_calls]

Allows incoming calls when the phone is locked (default).

◼ 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/lock/2-5/enabled]

Enables the phone's lock feature. Relevant for supporting

servers only.

22.18 Allowing Call Center Agents to Record Welcome Greetings

Here's how to let Call Center agents record welcome greetings.

To let Call Center agents record welcome greetings, using the configuration file:

◼ Use the table below as reference.

Table 69: Letting Call Center Agents Record Welcome Greetings

Parameter

Description

[voip/services/greeting/beep/enabled]

Enables a beep to be heard by the agent when the recorded

welcome greeting finishes playing.

◼ [1] = Enabled (default)

◼ [0] = Disabled

[voip/services/greeting/enabled]

Lets agents in a call center record directly on their phones

personal voice greetings which play automatically when

callers call in, to welcome callers to the service they’re

seeking.

Configure:

◼ [1] = Enabled

◼ [0] = Disabled (default)

22. Configuring Supplementary Services

400HD Series IP Phones

- 123 -

22.19 Enabling the Electronic Hook Switch

The phone supports the Electronic Hook Switch (EHS) DHSG feature. Calls can be answered and

volume level can be changed with EHS-capable headsets. The feature is supported on the following

headsets:

◼ Jabra® PRO 920

◼ Jabra® PRO 9450

The headset's base unit connects to the phone's headphone port. The Audio connector connects to

the headphone's port. The management connector connects to the Auxiliary port using a DHSG cable

which can be ordered from AudioCodes.

The feature can be enabled using the Web interface or configuration file. The feature allows users

to handle calls, i.e., answer calls and change volume level, with EHS-capable wireless headsets at a

distance from the phone.

To enable the EHS using the Web interface:

1. Open the Services page (Configuration tab > Voice Over IP menu > Services) and scroll down

to the General Parameters section.

Figure 75: VoIP- Services – General Parameters

2. Configure the 'Electronic Hook Switch' parameter using the table below as reference, and

then click Submit.

To enable EHS using the configuration file:

• Configure the EHS parameter using the table below as reference, and then click Submit.

Table 70: EHS Parameter

Parameter

Description

Electronic Hook Switch

[voip/services/electronic_hook_switch/enabled

]

Enables the EHS DHSG-standard feature.

◼ [0] Disable (default)

◼ [1] Enable

DHSG (Drahtlose Hör-Sprechgarnitur) is the protocol used

to convert a wireless headset's internal control signals to

a commonly supported standard, and which uses the

special AUX port.

Supported wireless 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.

The base unit of the headset connects to the phone's headset port, i.e., to the same port that all

headsets' base units connect to. The Audio connector must be connected to the headphones port.

The management connector must be connected to the Auxiliary port using a DHSG-standard cable

which can be ordered from AudioCodes.

22. Configuring Supplementary Services

400HD Series IP Phones

- 124 -

22.20 Disabling the Hard Mute Key on the Phone

The network administrator can disable the mute hard key on phones using the configuration file. The

feature is motivated by the requirement on the part of some enterprises to control the key remotely

to comply with company policy.

To disable the hard mute key on the phone using the configuration file:

◼ Use the table below as reference.

Table 71: Disabling the Hard Mute Key on the Phone

Parameter

Description

[voip/block_mute_key]

Allows network administrators to configure

enabling or disabling the hard mute key on the

phone.

◼ [0] (default) Allows the hard mute key on

the phone to function regularly.

◼ [1] Disables the hard mute key on the

phone.

In addition to the mute key, hard keys that can be disabled include speaker, headset, voicemail,

REDIAL, CONTACTS, MENU, TRANSFER, HOLD, and VOL. See also Section 26 for information about

disabling these keys.

vi102498

22.21 Configuring Attended and Semi-Attended Call Transfer

Here's how to configure a softkey with attended and semi-attended call transfer functionality, using

the configuration file.

To configure a softkey with attended / semi-attended call transfer functionality:

◼ Use the table below as reference:

Table 72: Configuring a Softkey with Attended and Semi-Attended Call Transfer Functionality

Parameter

Description

[personal_settings/soft_keys/ongoing_call/0/key_f

unction]

Default: Hold (all phone models except 405HD).

Change it to TRANSFER to configure the softkey with

attended / semi-attended call transfer functionality.

22. Configuring Supplementary Services

400HD Series IP Phones

- 125 -

22.22 Configuring Blind Transfer

You can configure a softkey with blind transfer functionality using the configuration file.

To configure a softkey with blind transfer functionality:

◼ Use the table below as reference:

Table 73: Configuring a Softkey with Blind Transfer Functionality

Parameter

Description

[personal_settings/soft_keys/ongoing_call/0/key_f

unction]

If you configure BLIND_TRANSFER, a softkey with blind

transfer functionality will be displayed in the phone

screen:

◼ On the 405/405HD phone (exclusively), the softkey

displayed will be BXfer. The 405/405HD phone

does not feature a hard TRANSFER key like the

other phones. This softkey enables that

functionality.

◼ On the other phones (besides the 405/405HD), the

softkey displayed is HOLD.

22.23 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:

where:

<memory key> is the speed dial memory key on the phone.

<speed dial phone number> is 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

23. Configuring Volume Levels

400HD Series IP Phones

- 126 -

23 Configuring Volume Levels

Here's how to configure volume levels.

23.1 Configuring Gain Control

Automatic Gain Control can be configured using the Web interface or configuration file.

It's strongly advisable not to change the default values.

To configure Automatic Gain Control using the Web interface:

1. Open the Voice page (Configuration tab > Voice Over IP menu > Voice).

Figure 76: Voice - Gain Control

2. Configure the settings according to the parameters in the table below, and then click Submit.

To configure Automatic Gain Control using the configuration file:

◼ Use the table below as reference.

Table 74: Automatic Gain Control Parameters

Parameter

Description

Max Gain

The index of the maximal gain.

Default: 15 dB. The index runs from 0 (0 dB)

to 18 (18 dB).

Fast Adaptation Gain Slope

The rate of changes in Automatic Gain Control

gain during Fast Adaptation Mode. Default:

19 (10dB/sec). 0.25-70.00 10dB/sec.

Fast Adaptation Duration (0 - 65535)

The duration of the Fast Adaptation Mode, in

msec. Default: 1500 msec.

23. Configuring Volume Levels

400HD Series IP Phones

- 127 -

Parameter

Description

Slow Adaptation Gain Slope

The rate of changes in Automatic Gain Control

gain during Slow Adaptation Mode. Default:

19 (10dB/sec). 0.25-70.00 10dB/sec.

Enable Automatic Gain Control

[voip/audio/gain/automatic_gain_control/enabled]

Enables the AGC. AGC automatically adjusts

the phone's voice volume to compensate for

weak or loud signals.

◼ [0] Disable (default)

◼ [1] Enable

Automatic Gain Control Direction

[voip/audio/gain/automatic_gain_control/direction]

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

Target Energy

[voip/audio/gain/automatic_gain_control/target_energy]

The required output energy (in dBm) of the

AGC.

The valid range is -31 to 0 (dB). The default

value is -19.

[voip/audio/gain/automatic_gain_control/NB/handset_tar

get_energy]

Default = -16

[voip/audio/gain/automatic_gain_control/NB/headset_tar

get_energy]

The valid range is -31 to 0

Default = -16

[voip/audio/gain/automatic_gain_control/WB/handset_tar

get_energy]

Default = -16

[voip/audio/gain/automatic_gain_control/WB/headset_tar

get_energy]

The valid range is -31 to 0

Default = -16

[voip/audio/gain/automatic_gain_control/fast_adap_gain_

slope]

Default = 3_50

Valid values:

[REBOOT], [0_25], [0_50], [0_75], [1_00],

[1_25], [1_50], [1_75], [2_00], [2_50], [3_00],

[3_50], [4_00], [4_50], [5_00], [5_50], [6_00],

[7_00], [8_00], [9_00], [10_00], [11_00],

[12_00], [13_00], [14_00], [15_00], [20_00],

[25_00], [30_00], [35_00], [40_00], [50_00],

[70_00]

[voip/audio/gain/automatic_gain_control/max_gain]

Default = 15

The valid range is 0 to 18

23. Configuring Volume Levels

400HD Series IP Phones

- 128 -

23.2 Configuring Tone Volume

Tone volume can be configured using the configuration file.

It's strongly advisable not to change the default values.

To configure tone volume using the configuration file:

◼ Use the table below as reference.

Table 75: Tone Volume Parameter

Parameter

Description

Tone Volume

[voip/audio/gain/tone_signal_level]

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).

23.3 Configuring Ringer Volume

The ringer volume can be configured using the configuration file.

It's strongly advisable not to change the default values.

To configure the ringer volume using the configuration file:

◼ Use the table below as reference.

Table 76: Ringer Volume Parameters

Parameter

Description

Ringer Volume

voip/audio/gain/ringer_signal_level

Ringing tone volume. This volume can be modified on-the-fly by

pressing the phone's VOLUME key when the phone is in idle

state.

The valid range is [-31] to [31] dB

Important! When plugging in a USB headset, the ringer will not be heard on incoming call

23. Configuring Volume Levels

400HD Series IP Phones

- 129 -

23.4 Configuring Speaker Volume

The speaker volume can be configured using the configuration file.

It's strongly advisable not to change the default values.

To configure speaker volume using the configuration file:

◼ Use the table below as reference.

Table 77: Speaker Parameters

Parameter

Description

Hands-free Gain Parameters

Note: Values are in decibels (dB)

◼ Decimal places: Use underscore instead of period (e.g., plus19_5db).

[voip/audio/gain/NB/handsfree_digital_input_gain]

Digital input gain (in dB) – Narrow Band. Default =

The valid range is (-32) to 31 (dB), where -32 is

mute.

[voip/audio/gain/NB/handsfree_digital_output_gain]

Digital output gain (in dB) – Narrow Band. Default =

The valid range is (-32) to 31 (dB), where -32 is

mute.

[voip/audio/gain/WB/handsfree_digital_input_gain]

Digital input gain (in dB) – Wide Band. Default = 6.

The valid range is (-32) to 31 (dB), where -32 is

mute.

[voip/audio/gain/WB/handsfree_digital_output_gain]

Digital output gain (in dB) – Narrow Band. Default =

The valid range is (-32) to 31 (dB), where -32 is

mute.

[voip/audio/gain/NB/handsfree_analog_output_gain]

Analog output gain (in dB) – Narrow Band.

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]

23. Configuring Volume Levels

400HD Series IP Phones

- 130 -

Parameter

Description

[voip/audio/gain/WB/handsfree_analog_output_gain]

Analog output gain (in dB) – Wide Band.

Valid values:

[0db] (default), [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]

[voip/audio/gain/NB/handsfree_analog_input_gain]

Analog input gain (in dB) – Narrow Band

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] (default), [plus48db], [plus54db],

[MUTE]

[voip/audio/gain/WB/handsfree_analog_input_gain]

Analog input gain (in dB) – Wide Band.

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] (default), [plus48db], [plus54db],

[MUTE]

[voip/audio/gain/NB/additional_speaker_gain]

Additional parameter for speaker gain

configuration, for Narrow Band.

Valid values:

◼ [0] 0dB

◼ [1] 1dB

◼ [2] 2dB

◼ [3] 3Db (default)

23. Configuring Volume Levels

400HD Series IP Phones

- 131 -

Parameter

Description

[voip/audio/gain/WB/additional_speaker_gain]

Additional parameter for speaker gain

configuration, for Wide Band.

Valid values:

◼ [0] 0dB

◼ [1] 1dB

◼ [2] 2dB

◼ [3] 3dB (default)

23. Configuring Volume Levels

400HD Series IP Phones

- 132 -

23.5 Configuring Handset Volume

The handset volume can be configured using the configuration file.

It's strongly advisable not to change the default values.

To configure handset volume using the configuration file:

◼ Use the table below as reference.

Table 78: Handset Gain Parameters

Parameter

Description

Handset Gain Parameters

Note: Values are in decibels (dB)

[voip/audio/gain/NB/handset_digital_output_gain]

Digital output gain (in dB) – Narrow Band.

Default = 2.

The valid range is (-32) to 31 (dB), where -32 is

mute.

[voip/audio/gain/NB/handset_digital_input_gain]

Digital input gain (in dB) – Narrow Band. Default

= 0.

The valid range is (-32) to 31 (dB), where -32 is

mute.

[voip/audio/gain/WB/handset_digital_input_gain]

Digital input gain (in dB) – Wide Band. Default =

The valid range is (-32) to 31 (dB), where -32 is

mute.

[voip/audio/gain/WB/handset_digital_output_gain]

Digital output gain (in dB) – Wide Band. Default

= 2.

The valid range is (-32) to 31 (dB), where -32 is

mute.

[voip/audio/gain/NB/handset_analog_output_gain]

Analog output gain (in dB), for Narrow Band.

Valid values:

[0DB] (default), [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]

23. Configuring Volume Levels

400HD Series IP Phones

- 133 -

Parameter

Description

[voip/audio/gain/WB/handset_analog_output_gain]

Analog output gain (in dB), for Wide Band.

Valid values:

[0DB] (default), [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]

[voip/audio/gain/NB/handset_analog_input_gain]

Analog input gain (in dB), for Narrow Band.

Default: PLUS30DB

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]

[voip/audio/gain/WB/handset_analog_input_gain]

Analog input gain (in dB), for Wide Band.

Default: PLUS30DB

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]

[voip/audio/gain/handset_analog_sidetone_gain]

Analog side tone gain (in db).

Valid values: [minus9db], [MINUS21DB]

(default), [minus15db], [minus18db],

[minus21db], [minus24db], [minus27db],

[MUTE]

23. Configuring Volume Levels

400HD Series IP Phones

- 134 -

23.6 Configuring Headset Volume

Headset volume can be configured using the configuration file.

It's strongly advisable not to change the default values.

To configure headset volume using the configuration file:

◼ Use the table below as reference.

Table 79: Headset Gain Parameters

Parameter

Description

Headset Gain Parameters

Note: Values are in decibels (dB)

◼ Decimal places: Use underscore instead of period (e.g., plus19_5db).

[voip/audio/gain/NB/headset_digital_output_gain]

Digital output gain (in dB) – Narrow Band. Default = 4.

The valid range is (-32) to 31 (dB), where -32 is mute.

[voip/audio/gain/NB/headset_digital_input_gain]

Digital input gain (in dB) – Narrow Band. Default = 0.

The valid range is (-32) to 31 (dB), where -32 is mute.

[voip/audio/gain/WB/headset_digital_output_gain

]

Digital output gain (in dB) – Wide Band. Default = -4.

The valid range is (-32) to 31 (dB), where -32 is mute.

[voip/audio/gain/WB/headset_digital_input_gain]

Digital input gain (in dB) – Wide Band. Default = 0.

The valid range is (-32) to 31 (dB), where -32 is mute.

[voip/audio/gain/NB/headset_analog_output_gain

]

Analog output gain (in dB), for Narrow Band.

Valid values: [0DB](default), [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]

[voip/audio/gain/WB/headset_analog_output_gai

As above, but for Wide Band.

[voip/audio/gain/NB/headset_analog_input_gain]

Analog input gain (in dB).

Valid values: [PLUS31_5DB](default), [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]

23. Configuring Volume Levels

400HD Series IP Phones

- 135 -

Parameter

Description

[voip/audio/gain/WB/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], [PLUS31_5DB](default),

[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]

[voip/audio/gain/headset_analog_sidetone_gain]

Analog side tone gain (in db).

Valid values: [minus9db], [MINUS12DB](default),

[minus15db], [minus18db], [minus21db], [minus24db],

[minus27db], [MUTE]

Part VI Advanced Phone Settings

Part VI

Advanced Phone Settings

24. Configuring the Phone Directory

400HD Series IP Phones

- 137 -

24 Configuring the Phone Directory

Here's how to configure the phone directory.

24.1 Configuring the Corporate Directory

The Corporate Directory can be configured using the Web interface or configuration file.

24.1.1 Configuring the LDAP-based Corporate Directory

Here's 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 interface:

1. Open the LDAP page (Configuration tab > Advanced Applications > LDAP). If LDAP is set to

Enable, extended configuration parameters are displayed.

Figure 77: LDAP

2. Configure the LDAP settings according to the parameters in the table below, and then click

Submit.

24. Configuring the Phone Directory

400HD Series IP Phones

- 138 -

To configure LDAP using the configuration file:

◼ Use the table below as reference.

Table 80: LDAP Parameters

Parameter Name

Description

Active

[system/ldap/enabled]

Enables or disable LDAP.

Server Address

[system/ldap/server_address]

Defines the IP address or URL of the LDAP server.

Port

[system/ldap/port]

Defines the LDAP service port.

User Name

[system/ldap/user_name]

Defines the user name used for the LDAP search request.

Password

[system/ldap/password]

Defines the password of the search requester.

Base

[system/ldap/base]

Defines the access point on the LDAP tree.

Name Filter

[system/ldap/name_filter]

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.

Name Attribute

[system/ldap/name_attrs]

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.

24. Configuring the Phone Directory

400HD Series IP Phones

- 139 -

Parameter Name

Description

Number Filter

[system/ldap/number_filter]

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.

Number Attribute

[system/ldap/number_attrs]

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.

Display Name

[system/ldap/display_name]

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”.

Max Hits

[system/ldap/max_hits]

Specifies the maximum number of entries expected to be

sent by the LDAP server (this parameter is sent to the LDAP

server).

Sort Result

[system/ldap/sorting_result]

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

[system/ldap/search_timeout]

The time out value for LDAP search (this parameter is sent to

the LDAP server).

system/ldap/ui/use_right_arrow_active_se

arch

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

[system/ldap/call_lookup]

Performs an LDAP search during call (search the display name

for a number).

Country Code

[system/ldap/country_code]

Defines the country code prefix added for number search.

Area Code

[system/ldap/area_code]

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).

24. Configuring the Phone Directory

400HD Series IP Phones

- 140 -

24.1.2 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.

Three types of corporate directory files are supported: txt, cfg, and xml

The corporate directory file includes a list of contacts and their phone numbers.

The syntax of the corporate directory file must be as follows:

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

directory entry below, the home and user-defined numbers are absent:

John Smith,1234,,574685746

To configure the Corporate Directory using the Web interface:

1. Prepare the file as explained above.

2. Open the Directory page (Configuration tab > Personal Settings menu > Directory).

3. Click Browse and select the file to upload.

4. Click Load Corporate Directory.

Figure 78: Corporate Directory

5. Configure the Corporate Directory settings according to the parameters in the table below,

and then click Submit.

To configure the Corporate Directory using the configuration file:

◼ Use the table below as reference.

Table 81: Provisioning Parameters

Parameter

Description

provisioning/corporate_directory_uri

The URI for retrieving the corporate directory. The corporate

directory must be included in a separate file to be loaded to the

phone during provisioning.

For example:

provisioning/corporate_directory_uri=corporate_dir.txt

Note:

◼ The corporate directory file is loaded after boot up and after

that, periodically.

◼ If the corporate directory file is new, the phone updates the

information and does not reboot.

24. Configuring the Phone Directory

400HD Series IP Phones

- 141 -

24.2 Modifying the 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 through the

Web interface.

To add a contact to the phone's directory:

1. Open the Directory page (Configuration tab > Personal Settings menu > Directory).

Figure 79: Directory - Add Contact

2. Under the Add Contact group, define the contact:

3. In the 'Name' field, enter the name of the contact.

4. 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>).

5. Click Submit; the contact appears in the Directory list at the bottom of the page.

To edit a contact:

1. 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.

2. 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.

3. 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. In the Directory list, mark the 'Select' check box corresponding to the contact you want to

delete.

2. Click Delete. (To delete all contacts, click the Delete All button).

25. Configuring Keys

400HD Series IP Phones

- 142 -

25 Configuring Keys

Here's how to configure keys.

The following keys can be configured:

◼ Function Keys (see Section 25.1) – applies to the 430HD and 440HD phones

◼ Programmable Keys (see Section 25.2) - applies to the 430HD and 440HD phones

◼ Speed Dials (see Section 25.3) – applies to the 420HD and 405/405HD phones

◼ Softkeys (see Section 25.4) – applies to the 420HD and 405/405HD phones

◼ Navigation Keys (see Section 25.5) – applies to all phones

25.1 Configuring Function Keys

Here's how to configure Function Keys.

Applies only to the 430HD and 440HD phone models.

25.1.1 430HD Phone

Up to 12 Function Keys can be configured as Speed Dials+BLFs and for Multicast Paging (for

configuring Multicast Paging, see Section 26 Disabling Hard Keys and Softkeys, on page 158). On the

phone, the 12 Function Keys are located on the right side. The label panes, next to the BLFs, are

covered. You can use the Web interface or configuration file to configure Function Keys.

To configure a Function Key as a Speed Dial using the Web interface:

1. In the Web interface, open the Function Keys page (Configuration tab > Personal Settings >

Function Keys).

Figure 80: 430HD Phone Personal Settings - Function Keys

25. Configuring Keys

400HD Series IP Phones

- 143 -

2. Configure a Function Key using the table below as reference.

Table 82: 430HD Phone Function Keys Parameters

Parameter Name

Description

Key

[personal_settings/functional_key/0-11/key_label]

Used to define a label for a key.

Type

[personal_settings/functional_key/0-11/type]

Choose either:

SPEED_DIAL (default) -or-

SPEED_DIAL_BLF

The speed-dial feature helps users quickly dial

numbers that are often used or hard to

remember.

PAGING

When the Paging feature is enabled, you can

define Paging Groups (see Section 26 Disabling

Hard Keys and Softkeys, on page 158).

Number

[personal_settings/functional_key/0-

11/speed_dial_number]

The telephone number which the speed dial

dials. The speed-dial feature helps users quickly

dial numbers that are frequently used or that are

hard to remember.

25. Configuring Keys

400HD Series IP Phones

- 144 -

25.1.2 440HD Phone

A total of 33 Function Keys can be configured on the 440HD phone. Of these, you can configure up

to 12 Function Keys as Speed Dials+BLFs. When more than 12 Function Keys are configured, these

keys can only be assigned as regular Speed Dials or for Multicast Paging (see Section 26 Disabling

Hard Keys and Softkeys, on page 158). You can scroll to additional pages to define these keys.

To configure Function Keys as Speed Dials+BLFs in the Web interface:

1. Open the Function Keys page (Configuration tab > Personal Settings menu > Function Keys).

Figure 81: 440HD Phone Personal Settings - Function Keys

2. Configure Function Keys using Table 83 as reference.

Table 83: 440HD Phone Function Keys Parameters

Parameter Name

Description

Key

[personal_settings/functional_key/n-n/key_label]

Used to define a label for a key.

Type

[personal_settings/functional_key/n-n/type]

Choose either SPEED_DIAL (default) -or-

SPEED_DIAL_BLF. The speed-dial feature helps

users quickly dial numbers that are often used or

hard to remember.

PAGING

When the Paging feature is enabled, you can

define Paging Groups (see Section 26 Disabling

Hard Keys and Softkeys, on page 158).

Number

[personal_settings/functional_key/n-

n/speed_dial_number]

The telephone number which the speed dial dials.

The speed-dial feature helps users quickly dial

numbers that are frequently used or that are

hard to remember.

The first Web page shows Function Keys 1-12 (0-11 in the configuration file). The second

Web page shows 18-27 (17-27 in the configuration file). The third Web page shows 28-38

(27-37 in the configuration file).

25. Configuring Keys

400HD Series IP Phones

- 145 -

25.1.3 Configuring Additional Function Keys

Applies only to the 440HD phone model.

You can configure more than 12 Function Keys using the Web interface or the configuration file.

Note the following when configuring more than 12 Function Keys:

◼ A total of 33 Speed Dials+BLFs can be configured. The 33 SDs+BLFs are configured on pages 1,

2 and 3 of the phone's BLF sidecar. Users define 12 SDs+BLFs and then when defining the

13th, the 12th SD+BLF shows Next and the name in the 12th moves to the 13th.

◼ In all other environments, pages 2 and 3 can be configured as SDs only, without BLF

functionality. Only page 1 allows SD+BLF.

◼ In the Web interface, the 'Page' dropdown lets you navigate to page 2 and 3.

25.2 Configuring Programmable Keys

Applies to the 430HD and 440HD phone models.

25.2.1 430HD and 440HD Phones

Six programmable keys are located adjacent to the screen. There are three on each side. They're

labeled 1-6 in the Web interface and 12-17 in the configuration file.

You can configure them as

◼ Extension lines (SIP accounts) (see the next section)

◼ Speed Dials / Speed Dials+BLFs (see Section 25.2.1.1)

◼ Key Events such as MISSED_CALLS (see Section 25.2.1.2)

To access the Programmable Keys in the Web interface:

1. Open the Programmable Keys page (Configuration tab > Personal Settings menu >

Programmable Keys).

Figure 82: Programmable Keys (430HD/440HD Phones)

25. Configuring Keys

400HD Series IP Phones

- 146 -

2. Use the table below as reference when configuring the 430HD/440HD phones.

Table 84: Programmable Keys Parameters (430HD/440HD Phones)

Parameter Name

Description

Line Key

Select 1-6. Network administrators can configure these six Programmable Keys;

see the next parameter 'Key Type'. The keys are located on either side of the

screen, three on either side. The uppermost left is 1; the lowermost right is 6.

Users are disabled from configuring them as Line Keys (SIP Accounts). The first

two are by default set to lines. The other four are by default empty.

Key Type

The six Programmable Keys can be configured as:

◼ SIP Account (Extension Line)

◼ SPEED DIAL

◼ SPEED DIAL+BLF

◼ Key Event

Key Label

Enter a Label for the key reflecting the event, for example, enter Dialed in the

'Key Label' field if you're programming a Dialed Calls Key Event. The label will

appear in the screen.

Paging Group Name

The name of the Paging Group. All phone extensions assigned to this group are

paged when a message is multicast from an extension in the group.

Paging Multicast

Address

The IP address via which multicast messages traverse. When a user announces or

receives a multicast message, the message traverses this IP address. You should

assign a different IP address for each group. This enables the multicasting of

paged messages to different groups simultaneously.

Paging Multicast Port

The port via which multicast messages traverse. When a user announces or

receives a multicast message, the message traverses this port. This port number

does not need to be unique for each Paging Group.

25.2.1.1 Configuring Multiple Extension Lines

The administrator can configure a phone's six Programmable Keys as extension lines (SIP accounts).

To configure multiple extension lines:

1. Open the Line Settings page (Configuration > Voice Over IP > Line settings).

Figure 83: Line Settings

2. Configure Line Settings for each Line Number. See Table 51 for detailed information about the

Line Settings. You can use the figure above as a reference example.

3. Open the Programmable Keys page (Configuration > Personal Settings > Programmable Keys).

25. Configuring Keys

400HD Series IP Phones

- 147 -

Figure 84: Programmable Keys

4. Configure a Programmable Key for each Line Key you configured in Step 2.

5. Associate each Line Key (i.e., Programmable Key) with a SIP Account (i.e., Line Number). You

can alternatively perform the association using the configuration file. Use the table below as

reference:

Table 85: Associating Line Keys with SIP Accounts using the configuration file

Parameter Name

Description

[voip/line_key/0/line]

Default: 0. Indicates that this is the first Programmable Key, located in

the upper left corner of the screen.

Leave the setting at its default.

[voip/line_key/1/line]

Default: 0. Indicates that the second Programmable Key is assigned to

line 0 by default.

Change it to 1 to assign the Programmable Key to line 1.

[voip/line_key/2-5/line]

Defaults: 2-5. Indicates the third through to the sixth Programmable

Key.

[personal_settings/functional_key/

12-17/key_label]

These are the labels displayed in the screen next to Programmable

Keys 0-5.

Define the string of characters you want displayed (name, extension,

etc.).

[personal_settings/functional_key/

12-17/type]

Default: SIP_ACCOUNT. Can be set to perform other functions such as

Speed Dial or Key Event.

Do not change these for keys you want to assign to SIP lines.

25. Configuring Keys

400HD Series IP Phones

- 148 -

25.2.1.2 Configuring a Key Event

You can set as Key Events the six keys flanking the screen, as well as sidecar key 13 and up.

To configure a Key Event:

1. Enable the key_event parameter:

• Set personal_settings/functional_key/12/type to KEY_EVENT

(Default = SIP_ACCOUNT)

• Set personal_settings/functional_key/13-17/type to KEY_EVENT

(Default = empty)

• personal_settings/functional_key/12/type = the uppermost left key flanking the

screen

• personal_settings/functional_key/13-17/type = the other five keys flanking the screen

2. Set voip/line_key/0/key_event to the event you require - use the table below as reference.

Table 86: Configuring a Key Event

Parameter Name

Description

[voip/line_key/0-5/key_event]

Defines an event to allocate to each of the six keys flanking the

screen.

Only becomes operative if the type parameter is set to KEY_EVENT as

shown above.

Default: MISSED_CALLS.

You can change the default to:

◼ DIALED_CALLS

◼ RECEIVED_CALLS

◼ FORWARD_ALL

◼ DND

25. Configuring Keys

400HD Series IP Phones

- 149 -

25.3 Configuring Speed Dials

This section applies to the 420HD and 405/405HD phone models.

25.3.1 420HD and 405/405HD Phone Models

Up to nine dialpad keys can be configured as Speed Dials. You can configure them in the Web

interface or using the configuration file.

To configure a dialpad key as a Speed Dial in the Web interface:

1. In the Web interface, open the Function Keys page (Configuration tab > Personal Settings >

Function Keys).

Figure 85: Personal Settings – Speed Dials (420HD and 405/405HD Phones)

2. In the 'Number' field, enter the phone number to which to assign the Speed Dial.

Keys 1-9 correspond to the 1-9 keys on the device's dial pad.

3. After configuring a Speed Dial, click Submit for the setting to take effect on the phone.

4. Alternatively, you can use the Save Function Keys button to save the configuration in a file,

which you can then browse to and load to the phone using the Load Function Keys button.

5. You can select one or more configured Speed Dials under the 'Delete' column, and then click

the Delete All button.

25. Configuring Keys

400HD Series IP Phones

- 150 -

To configure a key as a Speed Dial using the configuration file:

◼ Open the Configuration File page (Management tab > Manual Update > Configuration File)

and locate the Function Key parameters. Use the table below as reference when configuring

the parameters.

Table 87: Speed Dials Parameters

Parameter Name

Description

[personal_settings/functional_key/0-

8/speed_dial_extension]

Used to define a label for a Speed Dial.

[personal_settings/functional_key/0/speed_dial_number]

The telephone number which the speed dial

dials. The speed-dial feature helps users

quickly dial numbers that are frequently used

or that are hard to remember.

[personal_settings/functional_key/0/type=SPEED_DIAL]

The feature helps users quickly dial numbers

that are often used or that are hard to

remember.

25.3.2 Deleting Speed Dials

You can delete configured Speed Dials.

To delete Speed Dials:

1. Check the 'Delete' box corresponding to the Speed Dial that you want to delete, and then click

Submit

-or-

2. Click the Delete All button, and then at the prompt, click OK

To clear (unselect) all selected 'Delete' check boxes:

◼ Click the Reset button.

25.3.3 Saving Configured Speed Dials

After configuring Speed Dials in the Web interface, you can save the configuration in a cfg file on

your computer and then load it to other phones.

To save Speed Dials in a cfg file:

1. Open the Function Keys page (Configuration tab > Personal Settings menu > Function Keys).

2. Click Save Function Keys; the configuration is saved in a .cfg file.

25. Configuring Keys

400HD Series IP Phones

- 151 -

25.3.4 Creating a Speed Dial File for the Configuration File

The configuration file parameter 'provisioning/speed_dial_uri' can be configured to point to a user-

defined Speed Dial file, for speed dial settings to be uploaded to the phone when the cfg file is

uploaded.

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:

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

25. Configuring Keys

400HD Series IP Phones

- 152 -

25.4 Configuring Softkeys

• The section applies to the 420HD and 405/405HD phone models.

• When Genesys’ ACD is enabled, the Soft Keys item in the Keys Configuration menu is

not displayed.

This section describes how to configure softkeys. Four softkeys, located below the screen, can be

configured. Their functionality is context sensitive according to the phone's state. Here's how to

configure softkeys that are activated when the phone is in idle state and when it is in call state.

Following are the four default (preconfigured) softkeys (0-3), when the phone is in idle state and

when it is in call state.

Table 88: Default Softkeys

Key

Idle State

Call State

Directory

Hold

Missed

Conf

Forward

New Call

Do Not Disturb (Status)

End

When more than four softkeys are configured, users can scroll to additional pages, where on each

configured page, the fourth key becomes the More softkey.

◼ Up to 20 (0-19) softkey functions can be configured for when the phone is in idle call state.

◼ Up to 20 (0-19) softkey functions can be configured for when the phone is in call state.

◼ Up to 12 (0-11) programmable softkey (PSKs) functions can be configured to either a call state

softkey or an idle state softkey.

To program the softkeys in the Web interface:

1. Open the Programmable Keys page (Configuration tab > Personal Settings menu >

Programmable Keys).

Figure 86: Softkeys (420HD and 405/405HD Phone)

25. Configuring Keys

400HD Series IP Phones

- 153 -

2. Configure the softkey parameters using the table below as reference.

Table 89: Softkeys Parameters (420HD/405/405HD Phone)

Parameter Name

Description

Key 1-4

From any of the 1-4 softkeys dropdown menus, you can

select one of the following:

◼ None

◼ Missed Calls

◼ Received Calls

◼ Dialed Calls

◼ All Calls

◼ Directory

◼ DnD All

◼ Forward All

◼ SPEED DIAL

◼ SPEED DIAL+BLF

◼ SIP Account

To configure softkeys using configuration file

◼ Use the table below as reference:

Table 90: SoftKey Parameters

Parameter Name

Description

Softkey

[personal_settings/soft_key/n-n] (Idle Call State)

[personal_settings/soft_keys/ongoing_call/n-n]

(Ongoing call state)

Indicates the softkey number.

Where <n-n> defines the softkey number.

Possible values: 0-19

Function (Idle call state):

[personal_settings/soft_key/n-n/key_function]

Select one of the following key function types for

the Idle call state:

◼ Missed_calls

◼ Received_calls

◼ Dialed_calls

◼ All_calls

◼ Directory

◼ Dnd_all

◼ Forward_all

◼ Speed dial

◼ Speed dial + blf

◼ PSK

Function (Ongoing call state):

[personal_settings/soft_keys/ongoing_call/n-

n/key_function]

Select one of the following key function types for

the Ongoing call state (configuration file only):

◼ Rec_call

◼ Transfer

◼ Blind_transfer

◼ Hold

◼ Conf

◼ New_call

◼ End

◼ PSK

25. Configuring Keys

400HD Series IP Phones

- 154 -

Parameter Name

Description

Number

personal_settings/soft_key/n-

n/speed_dial_number=<number>

Where <number> is the telephone number which the

speed dial dials.

Define the telephone number which the speed

dial dials. The speed-dial feature helps users

quickly dial numbers that are frequently used or

that are hard to remember.

[personal_settings/soft_keys/n-n/psk_index] (Idle call

state)

[personal_settings/soft_keys/ongoing_call/n-

n/psk_index] (Ongoing call state)

Where n-n is the softkey number.

Defines the PSK index. This index identifies the

PSK which is assigned to the softkey.

You can define up to 11 PSKs (0-11). Default=12

(non-valid value)

Note that there are separate index number series

for the Idle and Ongoing call states. However,

each index number represents unique

functionality. For example, if you configure

psk_index 1 to activate an intercom (an Idle state

function), you cannot use the same index

(psk_index 1) to connect to a Voicemail server

(Ongoing Call state function).

personal_settings/soft_keys/psk/n-

n/is_dial_required=0 (Idle and Ongoing call states)

Where n-n is the PSK index number.

Determines whether a personal dialing code is

required for the PSK. When this parameter is

enabled, the user is prompted on the phone to

enter a personal code to activate this event. For

example, to connect to a Voicemail server.

This parameter is only applicable when

'Programmable SK' is set as the key_function.

[personal_settings/soft_keys/psk/n-n/PSKlabel] (Idle

and Ongoing Call states)

Where n-n is the PSK index number.

Defines the PSK label which is displayed on the

phone's screen for the configured PSK.

This parameter is only applicable when 'PSK' is set

as the key_function.

[personal_settings/soft_keys/psk/n-n/prefix] (Idle and

Ongoing Call states)

Where n-n is the PSK index number.

Defines the prefix which sends a SIP INVITE to the

softswitch to activate this feature (event). For

example *70.

This parameter is only applicable when

'Programmable SK' is set as the key_function.

Up to 128 characters (any characters).

25. Configuring Keys

400HD Series IP Phones

- 155 -

25.4.1 Configuring Programmable Softkeys (PSK)

You can configure a programmable key function and assign it to a softkey (Programmable Softkey-

PSK) for either idle state or call state. The PSK can be used for performing actions such as connecting

to a Voicemail (Ongoing Call state) server, returning the details of the last call (Idle state), connecting

to the Conference server (Idle state) and activate an intercom (Idle state). When these softkeys are

configured with such functionality, and the user presses these softkeys, the Enterprise's server

(softswitch or application server) is instructed to perform these actions. The instructions to the

softswitch or application server are applied using a prefix in the SIP INVITE message. An additional

feature enables the user to enter a personal code before the softkey functionality can be activated.

For example, the user wishes to activate their Voice Mail to hear messages whenever the softkey

configured for this feature is pressed. In this case, the user dials a prefix, for example *70, and then

is prompted to enter a personal code to access their voice mail i.e not configured on the phone, only

entered e.g., '1234'. Once this code is entered, the user is connected to the Enterprise's Voice Mail

server and can listen to their messages.

The following example shows the configuration of softkey 0 for connecting to a Voicemail server.

Note that in this example, psk index-1 is assigned to function key-0.

personal_settings/soft_key/0/key_function=PSK

personal_settings/soft_key/0/psk_index=1

personal_settings/soft_keys/psk/1/is_dial_required=1

personal_settings/soft_keys/psk/1/label=Voicemail

personal_settings/soft_keys/psk/1/prefix=*70

• You can configure the PSK to perform any action that is supported by your

enterprise’s softswitch or application server. AudioCodes provides the ability to

configure a calling prefix and a dialing code and to include these in the SIP INVITE.

• The PSK can only be configured using the configuration file.

25. Configuring Keys

400HD Series IP Phones

- 156 -

25.5 Configuring Navigation Control Button Positions

This section applies to all phone models.

Each of the four positions of the navigation control button on the phone, i.e., Up, Down, Left, and

Right, as well as its OK button, can be configured to perform a one of five functions:

◼ None (default)

◼ Missed Calls

◼ Received Calls

◼ Dialed Calls

◼ All Calls

◼ Directory

For example, from the 'Navigation Control Up' key dropdown in the Navigation Keys section of the

page (see the figure below), select Missed Calls, and then click the Submit button; the user will be

able to press the upper rim of the navigation control button on their phone to display the Missed

Calls on the phone screen.

The OK button can also be configured to perform one of these five functions. A Delete option

adjacent to each Navigation Control key and adjacent to 'Ok Button' can be selected to remove the

function from the key/button.

To configure navigation control button positions in the Web interface:

1. Open the Programmable Keys page (Configuration tab > Personal Settings menu >

Programmable Keys) and locate the Navigation Keys section.

Figure 87: Navigation Keys

2. From the 'Navigation Control Up' key dropdown, select (for example) Missed Calls, and then

click the Submit button; the user will be able to press the upper rim of the navigation control

button on their phone to display the Missed Calls in the phone screen.

3. [Optionally] Select the Delete option next to a key to remove the function.

25. Configuring Keys

400HD Series IP Phones

- 157 -

25.5.1 Saving Configured Keys

This section applies to all phone models.

After configuring softkeys, the configuration can be saved in a cfg file on a computer and then loaded

to other phones.

To save the configured softkeys in a cfg file:

1. Open the Programmable Keys page (Configuration tab > Personal Settings menu >

Programmable Keys) and locate the Load and Save section.

Figure 88: Load and Save

2. Click Save Programmable Keys; the configured keys are saved in a cfg file.

25.5.2 Loading Saved Keys to Phones

This section applies to all phone models.

After configuring softkeys, the configuration can be saved in a cfg file on a computer and then loaded

to other phones.

To load the saved keys to other phones:

1. Open the Programmable Keys page (Configuration tab > Personal Settings menu >

Programmable Keys).

2. Click Browse…. to select the cfg file.

3. Click Open; the selected cfg filename and path appear on the Web interface alongside the

Browse… button.

4. Click Load Programmable Key; the file is uploaded to the phone.

26. Disabling Hard Keys and Softkeys

400HD Series IP Phones

- 158 -

26 Disabling Hard Keys and Softkeys

The network administrator can disable hard keys and softkeys on phones using the configuration

file. The feature is motivated by the requirement on the part of some enterprises to control the

setting remotely to comply with company policy.

Hard keys that can be disabled include speaker, headset, voicemail, REDIAL, CONTACTS, MENU,

TRANSFER, HOLD, VOL and mute.

Example 1: To disable the phone's REDIAL hard key, the configuration file parameter

personal_settings/key/redial/enabled can be set to 0.

Example 2: To disable the option to restart the phone, the configuration file parameter

personal_settings/menu/restart/enabled can be set to 0.

Configuration file parameters that can be configured to disable hard keys and softkeys also include

these in the table below.

Table 91: Parameters that can be Configured to Disable Hard Keys / Softkeys

Configuration File Parameters

personal_settings/soft_keys/display_idle_screen_keys_when_dialing/enabled

personal_settings/key/speaker_device/enabled

personal_settings/key/headset_device/enabled

personal_settings/key/voice_mail/enabled

personal_settings/key/redial/enabled

personal_settings/key/contacts/enabled

personal_settings/key/menu/enabled

personal_settings/key/hold/enabled

personal_settings/key/volume/enabled

personal_settings/key/mute/enabled

personal_settings/menu/call_log/enabled

personal_settings/menu/directory/enabled

personal_settings/menu/keys_configuration/enabled

personal_settings/menu/keys_configuration/speed_dial_keys/enabled

personal_settings/menu/keys_configuration/soft_keys/enabled

personal_settings/menu/keys_configuration/navigation_keys/enabled

personal_settings/menu/settings/enabled

personal_settings/menu/language/enabled

personal_settings/menu/ring_tone/enabled

personal_settings/menu/callwaiting/enabled

personal_settings/menu/date_and_time/enabled

personal_settings/menu/lcd_contrast/enabled

personal_settings/menu/backlight_timeout/enabled

26. Disabling Hard Keys and Softkeys

400HD Series IP Phones

- 159 -

Configuration File Parameters

personal_settings/menu/answer_device/enabled

personal_settings/menu/restart/enabled

personal_settings/menu/status/enabled

personal_settings/menu/administration/enabled

personal_settings/menu/languages/english/enabled

personal_settings/menu/languages/spanish/enabled

personal_settings/menu/languages/russian/enabled

personal_settings/menu/languages/portuguese/enabled

personal_settings/menu/languages/portuguesebrazilian/enabled

personal_settings/menu/languages/german/enabled

personal_settings/menu/languages/ukrainian/enabled

personal_settings/menu/languages/french/enabled

personal_settings/menu/languages/frenchcanadian/enabled

personal_settings/menu/languages/italian/enabled

personal_settings/menu/languages/hebrew/enabled

personal_settings/menu/languages/polish/enabled

personal_settings/menu/languages/korean/enabled

personal_settings/menu/languages/finnish/enabled

personal_settings/menu/languages/chinese/enabled

personal_settings/menu/languages/chinesetraditional/enabled

personal_settings/menu/languages/turkish/enabled

personal_settings/menu/languages/japanese/enabled

personal_settings/menu/languages/slovak/enabled

personal_settings/menu/languages/czech/enabled

personal_settings/menu/callautoanswering/enabled

personal_settings/speed_dial_programming/enabled

personal_settings/new_call_screen/call_log_soft_key/enabled

personal_settings/new_call_screen/directory_soft_key/enabled

personal_settings/soft_keys/incoming_call/sk_reject/enable

personal_settings/key/transfer/enabled

personal_settings/key/reject/enabled

27. Configuring Paging

400HD Series IP Phones

- 160 -

27 Configuring Paging

Live announcements can be made (paged) from a phone to a group of phones to notify a team (for

example) that a meeting is about to commence. The paged announcement is multicast via a

designated group IP address, in real time, on all idle phones in the group, without requiring listeners

to pick up their receivers. The name of the group is displayed on phone screens when the paging call

comes in.

A key for paging a group can be configured using the Web interface, configuration file or on the

phone itself (see the User's Manual).

27.1 Configuring a Key for Paging using the Web Interface

A key can be configured for paging using the Web Interface.

To configure a key for paging using the Web interface:

1. Open the Services page (Configuration > Voice Over IP > Services) and enable the paging

feature.

Figure 89: Services - Enable Paging

2. In the Paging page, from the 'Enabled' dropdown, select Enable.

3. Configure the Multicast Paging Group; open the Function Keys page (Configuration menu >

Voice Over IP > Personal Settings > Function Keys).

4. Choose the Function Key for paging and then from 'Type List' select Paging.

5. Configure the Paging Group to assign to the phone, according to the configuration parameters

described in the table below.

6. Repeat the above procedure for each Paging Group you want to configure (assigning a

separate Function key for each additional Paging Group).

Table 92: Paging Function Key Parameters

Parameter

Description

Paging Group Name

The name of the Paging Group. All phone extensions assigned to this group are

paged whenever one of the extensions in the group multicasts a message.

Paging Multicast

Address

The IP address via which multicast messages traverse. When a user announces or

receives a multicast message, the message traverses this IP address. You should

assign a different IP address for each group. This enables the multicasting of

paged messages to different groups simultaneously.

Paging Multicast Port

The port via which multicast messages traverse. When a user announces or

receives a multicast message, the message traverses this port. This port number

does not need to be unique for each Paging Group.

27. Configuring Paging

400HD Series IP Phones

- 161 -

27.1.1 Configuring Barge-in

When barge-in is disabled (default), users who're in regular calls when a paging call comes in are

prompted in their phone screens to accept or reject the paging call. If they accept, the regular call is

put on hold and the paging call is heard. If they reject, the regular call is continued and the paging

call goes unheard.

The prompt to accept or reject a paged call is only relevant to users who're in regular calls.

If they're not in regular calls, the prompt is displayed irrespective of whether barge-in is

disabled or enabled.

When barge-in is enabled, paging calls interrupt (barge in on) regular calls in progress, without

prompting users with an option to accept or reject the paging call.

To enable barge-in using the Web interface:

1. Open the Services page (Configuration > Voice Over IP > Services) and under Paging, select

Enable; the 'Barge-in' dropdown is displayed.

2. From the 'Barge-in' dropdown, select Enable.

Figure 90: Enable Barge-in

To enable barge-in using the configuration file:

◼ Use the table below as reference.

Table 93: Barge-in Parameters

Parameter

Description

[voip/services/allow_barge_in/enabled]

Enables paging to interrupt (i.e., barge into) regular calls

currently in progress.

◼ [0] Disabled (default)

◼ [1] Enabled

See the phone's User's Manual for examples.

27. Configuring Paging

400HD Series IP Phones

- 162 -

27.2 Configuring Paging Using the Configuration File

Here's how to configure the paging feature using the configuration file.

To configure paging using the configuration file:

◼ Use the table below as reference.

Table 94: Configuration File Paging Parameters

Parameter

Description

[voip/services/group_paging/enabled]

Enables group paging.

◼ [0] Disabled (default)

◼ [1] Enabled

[voip/services/group_paging/group/0-

11/activated]

Defines the group to page to.

Default: Group 0

[voip/services/group_paging/group/0-

11/multicast_addr]

Defines the multicast address for group 0-11 to page to. Default:

224.0.1.0

[voip/services/group_paging/group/0-

11/name]

Defines the paging group name to display in the screen.

[voip/services/group_paging/group/0-

11/port]

Defines the multicast port for group 0 to page to.

Default: 8888

[voip/services/group_paging/codec]

The codec of the paging RTP. Since the phones have many DSP

versions and different DSPs support different codecs, the codec

of the paging call can be configured.

Available options are:

◼ PCMU, PCMA

◼ G729

◼ G722

◼ G726

◼ RTA

◼ RTA_WB

◼ G722_8000

Note: Phones that are in the same paging group must use the

same codec.

[voip/services/group_paging/end_incom

e_paging_timeout]

Defines the timeout that begins after the phone detects that it is

not receiving RTP. The phone ends the incoming paging call

when the timeout expires.

Default: 800 milliseconds. Optionally, you can configure 800~

milliseconds to 100000 milliseconds.

28. Configuring Feature Key Synchronization

400HD Series IP Phones

- 163 -

28 Configuring Feature Key Synchronization

Here's how to configure Feature Key synchronization.

To configure Feature Key synchronization using the configuration file:

◼ Use the table below as reference.

Table 95: Feature Key Synchronization Parameters

Parameter

Description

[system/feature_key_synchronization/enabl

ed]

Disables/enables Feature Key synchronization.

[system/feature_key_synchronization/forwa

rd/0-3/destination]

Forward destination. The number of the telephone to which

the call is made.

[system/feature_key_synchronization/status

/0-3/fks_status]

The status of the Feature Key synchronization.

Select:

◼ [FKS_NONE] (Default)

◼ FKS_DND]

◼ [FKS_CFA]

◼ [FKS_CFB] -or-

◼ [FKS_CFNA]

Part VII Se curity

Part VII

Security

29. Implementing X.509 Authentication

400HD Series IP Phones

- 165 -

29 Implementing X.509 Authentication

X.509 certificates can be used to authenticate a connection with a remote server or HTTP/S client

browser. The certificates may be implemented in one of or a combination of the following SSL

handshake negotiation scenarios:

◼ The phone is a client who needs to authenticate the remote server e.g., provisioning server to

which it is attempting to connect.

In this case, the phone needs to load the certificate and Trusted CA used by the remote

server.

◼ The remote server needs to authenticate the incoming connection request from the phone

client.

In this case, the remote server needs to load the certificate and Trusted CA used by the

phone.

◼ The phone is a server who needs to authenticate an incoming connection request from a

remote HTTP client browser.

In this case, the phone needs to load the certificate and Trusted CA used by the remote HTTP

client browser.

◼ The remote HTTP client browser needs to authenticate the phone to which it is attempting to

connect.

In this case, the remote HTTP client browser needs to load the certificate and Trusted CA used

by the phone.

The following types of certificates can be used to authenticate the connections described in the

above scenarios:

◼ Factory-set Certificates (see Section 29.1):

Certificates that are loaded to the AudioCodes IP Phone using an AudioCodes certificate and

AudioCodes Trusted Root CA.

◼ User-Generated Certificates (see Section 29.2):

Certificates that are generated by the user that may use the AudioCodes Trusted Root CA or

an external CA.

29. Implementing X.509 Authentication

400HD Series IP Phones

- 166 -

29.1 Factory-Set Certificates and AudioCodes Trusted Root CA

AudioCodes IP phones are loaded with factory-set preinstalled certificate files: private key file,

certificate file and a Trusted Root CA file that is signed by AudioCodes (including DIGICert).

• The Web interface provides visual indication that factory certificates are installed:

▪ The System Information page displays 'MAC Address' and 'Device Certificate'

parameters.

▪ The values for the 'Device Certificate' parameter can be Installed, Self-Signed, or

Not Installed.

• The phone's screen visually indicates that factory certificates are installed.

▪ The Release Information menu (MENU button > Status) displays the 'Device

Certificate' parameter.

▪ The values of the 'Device Certificate' parameter can be Installed, Self-Signed, or

Not Installed.

Whenever the phone authenticates with a remote server, it can be authenticated using these

certificate files. Each phone receives a uniquely generated private key certificate file based on its

MAC address.

• If the remote server is configured to authenticate the client and AudioCodes factory-

set certificates are used for authentication, then the AudioCodes Certificate and

AudioCodes Trusted Root CA must be downloaded to the remote server. These files

can be downloaded from the AudioCodes Website. For more information, contact

your local AudioCodes sales representative.

• If you use the AudioCodes Redirect server to obtain firmware and configuration files,

then the factory-set certificates are used to authenticate the connection with this

server.

29.2 User-Generated Certificates

If an organizational certificate Infrastructure (PKI) is used, you may wish to instead use certificates

provided by your security administrator. You can define up to five additional user-generated

certificates, which can be configured to secure different types of connections and paired with

external Trusted Root CAs. The following remote server connection types can be configured with

user-generated certificates:

◼ 802.1x RADIUS server

◼ SIP TLS server

◼ HTTP/S Provisioning server

When user-generated certificates are loaded to the device to authenticate a specific connection

type, then this certificate is used to secure the connection with the assigned connection type. For

example, if you load Certificate A for connecting to an HTTPS Provisioning server, then whenever

there is an attempt by the phone to connect to a Provisioning server, then the connection is

authenticated using Certificate A.

• You can load one certificate for each connection type.

• If you do not load a certificate to support a specific connection type, then the factory-

set certificate is used to authenticate the connection. For example if you load user-

generated certificates to support Automatic Updates (Provisioning server) and SIP TLS

server connections, and there is an attempt by the phone to connect to a RADIUS

server, then this connection is authenticated using the AudioCodes factory-set

installed certificate.

• You can use the AudioCodes Trusted Root CA with a user-generated certificate.

• You can use the same certificate for different server connection types.

29. Implementing X.509 Authentication

400HD Series IP Phones

- 167 -

29.3 External Trusted Root CAs

The Certificate Authority is a body that certifies ownership of a certificate by the name subject of

the certificate.

Figure 91: Certificate

You can define up to five external Trusted Root CAs, which may be configured to secure different

types of connections and paired with the loaded user-generated certificates (see Section 29.2).

If you do not load any Trusted Root CAs to the phone, then when there is an attempt to

connect to a remote server or an attempt by a browser to open the Web interface using

HTTPS, the AudioCodes Trusted Root CA is used to authenticate the connection.

29. Implementing X.509 Authentication

400HD Series IP Phones

- 168 -

Supported CAs are:

◼ Comodo_AAA_Certificate_Services.cer

◼ Comodo_AddTrust_External_CA_Root.cer

◼ Cybetrust_Baltimore_CyberTrust_Root.cer

◼ Cybetrust_GlobalSign_Root_CA.cer

◼ Cybetrust_GTE_CyberTrust_Global_Root.cer

◼ DigiCert_SHA2_High_Assurance_Server_CA

◼ DigiCertGlobalRootCA

◼ DigiCertHighAssuranceEVRootCA

◼ DigiCertSHA2SecureServerCA

◼ DST_Root_CA_X3

◼ Entrust_Entrust.net_Certification_Authority_2048.cer

◼ Entrust_Entrust.net_Secure_Server_Certification_Authority.cer

◼ Equifax_Equifax_Secure_Certificate_Authority.cer

◼ g3cert

◼ GeoTrust_GeoTrust_Global_CA.cer

◼ GeoTrust_Global_CA

◼ GeoTrustEVRSACA2018

◼ Go_Daddy_Go_Daddy_Class_2_Certification_Authority.cer

◼ Go_Daddy_Root_Certificate_Authority-G2

◼ Go_Daddy_Starfield_Class_2_Certification_Authority.cer

◼ Go_Daddy_www.valicert.com_1.cer

◼ Go_Daddy_www.valicert.com_2.cer

◼ Go_Daddy_www.valicert.com_3.cer

◼ GoDaddy8x8pilot

◼ isrgrootx1.pem

◼ letsencryptauthorityx3

◼ VeriSign_Class_2_Public_Primary_Certification_Authority.cer

◼ VeriSign_Class_3_Public_Primary_Certification_Authority.cer

◼ VeriSign_Class_3_Public_Primary_Certification_Authority_G1.cer

◼ VeriSign_Class_3_Public_Primary_Certification_Authority_G2.cer

◼ VeriSign_Class_3_Public_Primary_Certification_Authority_G3.cer

◼ VeriSign_Thawte_Premium_Server_CA.cer

◼ VeriSign_Thawte_Server_CA.cer

30. Loading a Certificate

400HD Series IP Phones

- 169 -

30 Loading a Certificate

Here's how to:

◼ Load the Trusted Root CA Certificate to the Phone (see below).

◼ Load the Client Certificate to the Phone (see Section 30.2).

◼ Generate a Certificate Signing Request (CSR) (see Section 30.3).

30.1 Loading the Trusted Root CA Certificate to the Phone

Here's how to load the Trusted Root CA certificate to the phone.

To load the trusted root CA certificate to the phone:

1. Open the Root CA Certificate page (Configuration tab > Security menu > Root CA Certificate).

Figure 92: Root CA Certificate

2. Click Browse to navigate to the certificate file, and then click the Load button to upload it to

the phone.

You can load a maximum of five certificates to the phone. Click the Del button to delete a load

if necessary. Click the Display button to display the certificate if you wish to view it.

30.1.1 Loading Trusted Root CA Certificate Using Configuration File

Here's how to load a Trusted Root CA certificate using the configuration file.

Using this method, Trusted Root CA certificates files are loaded to the phone when it is

powered up.

To load a Trusted Root CA certificate file using the configuration file:

◼ Use the table below as reference.

Table 96: Root CA Certificate Parameters

Parameter

Description

Root CA 1

[security/ca_certificate/0/uri]

The first root CA certificate loaded to the phone.

Root CA 2

[security/ca_certificate/1/uri]

The second root CA certificate loaded to the phone.

Root CA 3

[security/ca_certificate/2/uri]

The third root CA certificate loaded to the phone.

Root CA 4

[security/ca_certificate/3/uri]

The fourth root CA certificate loaded to the phone.

Root CA 5

[security/ca_certificate/4/uri]

The fifth root CA certificate loaded to the phone.

30. Loading a Certificate

400HD Series IP Phones

- 170 -

30.2 Loading the Client Certificate to the Phone

The section shows how to load the Client Certificate to the phone.

To load the Client Certificate to the phone:

1. Open the Client Certificate page (Configuration tab > Security menu > Client Certificate).

Figure 93: Client Certificate

30.2.1 Loading the Client Certificate to the Phone using the Configuration File

Here's how to load a Client Certificate file to the phone using the configuration file.

Using this method, client certificates files are loaded to the phone when it is powered up.

To load a client certificate file using the configuration file:

◼ Use the table below as reference.

Table 97: Client Certificate Parameters

Parameter

Description

Certificate

[security/sip_certificate_uri]

Downloads from this URI to the phone a Client Certificate

for SIP TLS (SIP calls with Transport Layer Security).

Private Key

[security/sip_private_key_uri]

Downloads from this URI to the phone a Client Private Key

for SIP TLS (SIP calls with Transport Layer Security).

Certificate

[security/ieee802_1x_certificate_uri]

Downloads from this URI to the phone a Client Certificate

for 802.1X Authentication.

Private Key

[security/ieee802_1x_private_key_uri]

Downloads from this URI to the phone a Client Private Key

for 802.1X authentication.

Certificate

[security/autoupdate_certificate_uri]

Downloads from this URI to the phone an external

certificate that is used to secure the connection with the

automatic provisioning server. Identical to downloading

the Automatic Updates certificate from the Client

Certificates page in the phone’s Web interface.

Private Key

[security/autoupdate_private_key_uri]

Downloads from this URI to the phone a private key that

is used to secure the connection with the automatic

provisioning server. Identical to downloading the

Automatic Updates certificate from the Client Certificates

page in the phone’s Web interface.

30. Loading a Certificate

400HD Series IP Phones

- 171 -

30.2.2 Enabling Server-side Authentication (Mutual Authentication)

You can enable server-side authentication of a connection with the RADIUS and Provisioning server.

OpenSSL 1.0.2p is supported. This open source version supports SHA2 algorithms.

Table 98: Server-side Authentication

Parameter

Description

Verify RADIUS remote server certificate

[security/ieee802_1x/verify_server_certificate]

Configures the phone to verify received server certificates

over a secure EAP-TLS connection.

Verify Provisioning server certificate

[security/provisioning/verify_server_certificate

]

Configures the phone to verify received server certificates

over a secure HTTPS connection with a provisioning

server.

30.3 Generating a Certificate Signing Request

The section shows how to generate a certificate signing request (CSR) to send to the Certificate

Authority (CA) for the CA to sign the Client Certificate.

To generate a CSR:

1. Open the Certificate Signing Request page (Configuration tab > Security menu > Certificate

Signing Request).

Figure 94: Certificate Signing Request

2. Press the Generate Certificate Signing Request button; the phone creates a CSR file.

3. Press the Save Certificate Signing Request button and download the CSR file to your PC.

4. Send the CSR file to the Certificate Authority to sign the Client Certificate.

5. You can load the Client Certificate to the phone for 802.1X Authentication or SIP TLS.

30. Loading a Certificate

400HD Series IP Phones

- 172 -

30.4 Using Previously Loaded Certificates

If you have upgraded to this version and your phones have pre-installed certificates, then the CA

configuration parameter values from previous versions are translated to version 2.2.2 parameter

values as described below.

It is highly recommended to change the CA configuration by using the methods described

in the above sections.

◼ Certificate file settings for versions prior to version 2.2.2:

security/ca_certificate_uri= xxx

security/certificate_uri=zzz

security/private_key_uri=yyy

where:

xxx is the uri of the root CA

zzz is the uri of the certificate file

yyy is the uri of the private key file

◼ Certificate file settings translated to version 2.2.2:

security/autoupdate_certificate_uri=zzz

security/autoupdate_private_key_uri= yyy

security/ca_certificate/0/uri=xxx

security/ca_certificate/1/uri=

security/ca_certificate/2/uri=

security/ca_certificate/3/uri=

security/ca_certificate/4/uri=

security/ieee802_1x_certificate_uri=zzz

security/ieee802_1x_private_key_uri= yyy

security/sip_certificate_uri= zzz

security/sip_private_key_uri= yyy

The CA configuration parameters prior to version 2.2.2 are no longer used on the phone.

31. Configuring SIP TLS

400HD Series IP Phones

- 173 -

31 Configuring SIP TLS

Here's how to manage Transport Layer Security (TLS) and certificates. TLS is a cryptographic protocol

which provides communication security over the transport layer (TCP). TLS is used to secure the

phone's SIP signaling connections, Web interface, and Telnet server over TCP/IP. Typically, TLS

protocol uses Private and Public keys for authentication. A Certification Authority (CA) performs

authentication. Full protocol specification is updated in RFC 5246.

Before you can connect to a TLS server, you need to make sure the same certificate and

Trusted Root CA are loaded to the phone and to the TLS server.

31.1 Configuring TLS

Here's how to configure TLS for the SIP connection between the phone and a TLS server.

To configure TLS using the configuration file:

◼ Use the table below as reference.

Table 99: 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.

Range:1024 - 65535. Default: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:”

31. Configuring SIP TLS

400HD Series IP Phones

- 174 -

31.1.1 Configuring SIP TLS using the Web Interface

Here's how to configure SIP TLS using the Web interface.

This procedure is typically used for evaluation purposes only.

To configure TLS using the Web interface:

1. Open the Signaling Protocols page (Configuration tab > Voice Over IP menu > Signaling

Protocols) and scroll down to the SIP General screen section.

2. From the 'SIP Transport Protocol' drop-down list, select TLS.

3. In the 'TLS Port' field, enter 5061.

4. Configure the other parameters using Table 18 as reference.

Figure 95: Signaling Protocols - SIP General

5. Click Submit.

32. Configuring 802.1x

400HD Series IP Phones

- 175 -

32 Configuring 802.1x

802.1X is an IEEE Standard for Port-based Network Access Control (PNAC). It's part of the IEEE 802.1

group of networking protocols. It provides an authentication mechanism for devices wishing to

connect to a LAN or WLAN.

The employee's PC negotiates 802.1X. Messages are sent transparent to the enterprise switch. The

phone is uninvolved in the negotiation; however, if an employee's PC is disconnected, their phone

notifies the switch. If an employee's PC is disconnected from the phone, a PROXY-EAP-LOGOFF

mechanism lets the phone immediately log off the port from the authentication server to prevent

anyone else from connecting to it.

The phone performs like this:

◼ Phone and PC connected to phone's PC port successfully perform 802.1X authentication. The

authentication server records the phone and PC as authorized.

◼ If the PC is disconnected from the phone's PC port, the phone sends an EAPoL-Logoff message

for the PC. The authentication server then records the PC as unauthorized.

◼ If the PC reconnects to the phone's PC port, the authentication server requests the PC to

perform 802.1X authentication again.

Before you can connect to a 802.1x server, you need to make sure the same certificate and

Trusted Root CA are loaded to the phone and to the 802.1x.

32.1 Configuring 802.1x in the Phone Screen

Here's how to configure 802.1x in the phone screen.

To configure 802.1x in the phone screen:

1. On the phone, open the 802.1x Settings screen (MENU key > Administration > Network

Settings > 802.1xSettings).

2. Navigate to and select either:

• Disabled – disables the 802.1x feature

• EAP-MD5 – see Section 32.1.1

• EAP-TLS - see Section 32.1.2

32. Configuring 802.1x

400HD Series IP Phones

- 176 -

32.1.1 Configuring EAP-MD5 Mode

Here's how to configure EAP-MD5 mode for 802.1x using the phone's screen.

To configure EAP-MD5 mode for 802.1x using the phone's screen:

1. 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.

32.1.2 Configuring EAP-TLS Mode

Here's how to configure EAP-TLS mode for 802.1x using the phone's screen.

To configure EAP-TLS mode for 802.1x using the phone's screen:

1. Navigate to the EAP-TLS option and press Select

2. Press the Save softkey; a message appears notifying you that the phone will restart.

3. Press Apply.

To configure EAP TLS using the Configuration File:

◼ Open the Configuration File page (Management tab > Manual Update > Configuration File)

and configure the parameters using the table below as reference.

Table 100: EAP TLS Parameters

Parameter

Description

EAP Type

[network/lan/_802_1x/eap_type]

Sets 802.1X EAP mode.

[Disable] = Disables the use of 802.1X

[EAP_TLS]= Authentication is implemented by

Certificate, Client Certificate, and Client Private Key.

[network/lan/_802_1x/tls_identity]

This parameter is optional; it’s sufficient to configure

the previous parameter

‘network/lan/_802_1x/eap_type’.

If end users are using the EAP_TLS method for

example, they can exercise the option to change the

value of ‘network/lan/_802_1x/tls_identity’ from

the default <HWType_MAC> to <user@domin>

(more common), or any other string up to 31

characters.

32. Configuring 802.1x

400HD Series IP Phones

- 177 -

32.2 Configuring 802.1x Using Web and Configuration File

Here's how to configure 802.1x using the Web interface or configuration file.

32.2.1 Configuring EAP MD5 Mode

Here's how to configure 802.1x settings for EAP-MD5 using the Web interface.

To configure 802.1x settings for EAP-MD5 using the Web interface:

1. Open the 802.1X Settings page (Configuration > Network Connections > 802.1X Settings) and

select MD5 as 'EAP Type'.

Figure 96: Web Interface –801.1X Settings - EAP-MD5

2. Configure the 802.1x settings according to the parameters in the table below, and then click

Submit.

To configure 802.1x settings for EAP-MD5 using the configuration file:

◼ Use the table below as reference.

Table 101: 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).

32. Configuring 802.1x

400HD Series IP Phones

- 178 -

32.2.2 Configuring EAP TLS Mode

Here's how to configure the phone's 802.1x settings for EAP-TLS using the Web interface or

configuration file.

To configure 802.1x settings for EAP-TLS using the Web interface:

1. Open the 802.1X Settings page (Configuration > Network Settings > 802.1X Settings) and

select TLS as 'EAP Type'.

Figure 97: Web Interface –801.1X Settings - EAP-TLS

2. Click Submit.

33. Configuring SRTP

400HD Series IP Phones

- 179 -

33 Configuring SRTP

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,

so when using this feature you also need to use SIP over TLS.

SRTP can be configured using the Web interface or configuration file.

To configure SRTP using the Web interface:

1. Open the Media Streaming page (Configuration tab > Voice Over IP menu > Media Streaming)

and scroll down to SRTP.

Figure 98: SRTP

• From the drop-down list select an encryption level:

 DO NOT SUPPORT ENCRYPTION [Default] [SRTP is disabled]

 SUPPORT ENCRYPTION [Negotiation]

 REQUIRE ENCRYPTION [SRTP is enabled; both sides must support encryption]

2. Click Submit.

To configure SRTP using the configuration file:

◼ Use the table below as reference.

Table 102: SRTP Parameters

Parameter

Description

SRTP Encryption and Authentication

[voip/media/srtp/mode]

◼ Three encryption levels are supported:

◼ DoNotSupportEncryption [Default] [SRTP is disabled]

◼ SupportEncryption [Negotiation]

◼ RequireEncryption [SRTP is enabled; both sides must

support encryption]

◼ Note regarding backward compatibility:

In Version 2.2.16, this configuration file parameter

replaced the legacy voip/media/srtp/enabled

configuration file parameter.

◼ The configuration file parameter

voip/media/srtp/enabled=1 in previous releases is

compatible with the configuration file parameter

voip/media/srtp/mode=REQUIRE_ENCRYPTION in this

release

[voip/media/srtp/negotiation/mode]

◼ If voip/media/srtp/mode=SUPPORT_ENCRYPTION, two

SRTP negotiation modes are supported:

◼ Basic (default) [RTP/SRTP negotiation according to the

document IMTC Best Practices for SIP Security. This

mode is supported by Broadsoft, Microsoft and many

other vendors]

◼ RFC5939 [RTP/SRTP capability negotiation using the

attributes "a=tcap", "a=acap" and "a=pcfg" as

described in RFC 5939]

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]

33. Configuring SRTP

400HD Series IP Phones

- 180 -

Parameter

Description

/voip/media/srtp/use_MKI

Defines the usage of the SRTP Master Key Index.

◼ [0] = MKI is not used (default)

◼ [1] = MKI is used

/voip/media/srtp/MKI_length

Defines the maximum length of the SRTP Master Key

Index. Range: 1 - 4. Default: 1.

/voip/media/srtp/use_lifetime

Allows the removal of the ‘lifetime’ parameter from the

SRTP Crypto line in SDP. According to RFC 4568, an

optional ‘lifetime’ parameter such as "2^31" must be

added to the a=crypto line. This parameter allows the

removal of the lifetime in all phone crypto lines in SDP.

Configurable parameter values are:

◼ [0] = the lifetime is removed

◼ [1] = the lifetime is retained (default)

/voip/media/srtp/RTCP_encrypt_enabled

Default: 1. If set to 0, UnencryptedSRTCP will present at

the end of the “a=crypto” line in the SDP offer, for

example:

a=crypto:1 AES_CM_128_HMAC_SHA1_32

inline:rcO4NFj0PcKk3Pbo7IVhVqpCpQI3MWytScjRL1IS|2^

31 UNENCRYPTED_SRTCP

/voip/media/srtp/RTP_encrypt_enabled

Default: 1. If set to 0, UnencryptedSRTP will present at

the end of the “a=crypto” line in the SDP offer, for

example:

a=crypto:1 AES_CM_128_HMAC_SHA1_32

inline:rcO4NFj0PcKk3Pbo7IVhVqpCpQI3MWytScjRL1IS|2^

31 UNENCRYPTED_SRTP

/voip/media/srtp/RTP_auth_enabled

Default: 1. If set to 0, UnauthenticatedSRTP will present at

the end of the “a=crypto” line in the SDP offer, for

example:

a=crypto:1 AES_CM_128_HMAC_SHA1_32

inline:TDejshzv6Y04By7Add2KuZaJ9YrvteWSENcpBMZ4|2

^31 UNAUTHENTICATED_SRTP

34. Configuring HTTP/S

400HD Series IP Phones

- 181 -

34 Configuring HTTP/S

HTTP/S login authentication can be configured to secure the connection between the phones and a

provisioning server, such as the BroadWorks Device Management Provisioning server. Once the

connection is secure, software and/or configuration files can be downloaded to the phone.

HTTP/S authentication is supporting using the following methods (configured on the remote server):

◼ Basic – (RFC 2617) username and password are sent in plain text over plain HTTP over the

network.

◼ Digest – a hash function is applied to the password before sending it over the network,

therefore it is more secure as usernames and passwords are encrypted

• The enterprise requires an HTTP/S server to support this feature.

• The authentication method is configured on the remote side e.g., Provisioning server.

35. Logging into a Remote HTTP/S Server from the Phone

400HD Series IP Phones

- 182 -

35 Logging into a Remote HTTP/S Server from the

Phone

During automatic provisioning, the phone can optionally prompt the user to enter the login

credentials (username and password) of the provisioning server.

The prompt occurs during the server's authentication process, when it is recognized that an HTTP

username and/or password has not been specified, or that these credentials are incorrect.

If so, and if the prompt feature is enabled, the 'Prov. Credentials' screen pops up, prompting the user

to enter or reenter these login credentials.

To configure HTTP/S login authentication in the configuration file:

◼ Use the table below as reference:

Table 103: HTTP/S Login Authentication

Parameter

Description

[provisioning/configuration/http_auth/ui_interactio

n_enabled]

Enables the user to be prompted to enter the HTTP

username and password on the phone during the

automatic provisioning process whenever the login

credentials to the provisioning server have not been

entered or are incorrect.

◼ [0] = (default) The phone's Settings menu's Prov.

Credentials option is not available on the phone

and therefore the user cannot interactively enter

the HTTP password and username. In this case,

you must enter values for the HTTP username

and password in the configuration file, as

specified below.

◼ [1] = The user can be prompted to enter the

HTTP username and password interactively.

Whenever this value is configured and the phone

attempts to connect to a remote server, then the

user is prompted to enter or reenter these

credentials.

In addition, the user can manually go the Settings

menu option Prov. Credentials to enter their

username and password.

When this value is enabled, then it is highly

recommended to remove the HTTP password and

username entries from the configuration file:

◼ provisioning/configuration/http_auth/password

◼ [provisioning/configuration/http_auth/user_nam

[provisioning/configuration/http_auth/user_name]

Defines a username required by the HTTP/S server

for logging in with authentication.

[provisioning/configuration/http_auth/password]

Defines a password required by the HTTP/S server

for logging in with authentication.

36. Securing the Web Interface using HTTP/S

400HD Series IP Phones

- 183 -

36 Securing the Web Interface using HTTP/S

Here's how to secure management of the phone with HTTP/S authentication, using the Web

interface.

To authenticate the connection between the phone and a remote HTTP/S browser, make

sure the same certificate and Trusted Root CA are loaded to the phone and to the remote

HTTP/S browser. For more information, see Section 30.

To secure management of the phone with HTTP/S authentication, using the Web interface:

1. In your Web browser, enter the URL, for example, https://10.13.2.5; the following screen

opens:

Figure 99: Securing Web Interface Management with HTTP/S

2. Click the Continue to this website link; the Windows Security dialog opens.

3. Enter your 'User Name' and 'Password' (Default = admin and 1234); the home page of phone's

Web interface opens. You're now managing the phone using the Web interface, secured by

HTTP/S.

36. Securing the Web Interface using HTTP/S

400HD Series IP Phones

- 184 -

36.1 Provisioning

Your phones are automatically provisioned by your enterprise's DHCP server when you initially

connect them to the IP network and to the power supply.

You can then configure periodic automatic provisioning by DHCP server, in the Web interface.

To configure periodic automatic provisioning by DHCP server:

1. In the Web interface, open the Automatic Provisioning screen (Management tab > Automatic

Update > Automatic Provisioning).

Figure 100: Automatic Provisioning

2. Configure the firmware file (img) and configuration file (cfg) to be pulled from your

enterprise's HTTP server by the DHCP server, in the 'Dynamic Firmware URL' parameter and

'Dynamic Configuration URL' parameter.

3. Configure the provisioning period.

To implement secure provisioning using HTTP/S, the HTTP/S server on the far end (from

where you are loading the files) must also support HTTP/S.

37. MAC-Based Authentication

400HD Series IP Phones

- 185 -

37 MAC-Based Authentication

Here's how to configure MAC-based authentication.

To configure MAC-based authentication:

◼ Use the table below as reference:

Table 37-1: Authentication

Parameter

Description

[provisioning/configuration/mac_address_in_header

]

Enables MAC-based authentication.

◼ [0] = (default) Don't insert the phone's MAC

address in the header.

◼ [1] = Insert the phone's MAC address in the

header.

38. Configuring HELD

400HD Series IP Phones

- 186 -

38 Configuring HELD

HELD (HTTP-Enabled Location Delivery) is a protocol that enables devices to request

information about their location to a Location Information Service or LIS. Devices that support

the HELD protocol are able to tightly integrate with ERS via the HELD service. The HELD

Service is available in ERS Enterprise SIP Accounts and needs special activation and

configuration. HELD-compliant hard-phones and softphones send their network information

to the ERS in an XML request. ERS responds by determining the phone’s location based on the

pre-provisioned network map, sends this information to the phone in the form of locationURI

or civic address. At call-time, the phone sends a SIP invite containing the locationURI, enabling

ERS to retrieve the phone’s location and route the call to the proper PSAP.

Table 2: HELD Configuration

Parameter

Description

/location/HELD/server_url

Specify the HELD Server URL.

/location/HELD/request_location_type

Either LocationURI, Civic, LocationURI_and_Civic.

/location/HELD/nai.enable

Network Access Identifier (NAI)

Boolean, 1 is default

/location/HELD/Identity

Set the vendor-specific element to include in a location request

message.

CompanyID is default.

/location/HELD/Identity_value

Set the value for the vendor-specific element to include in a

location request message.

/security/HELD_certificate_url

Certificate URL to use for server secure connection

/security/HELD_private_key_url

Private key URL to use for server secure connection

Table 3: HELD Status

Parameter

Description

/status/diagnostics/lldp/chassis/chassisId

Chassis ID of the switch that the device is connected to.

MAC address: 12-digit hexadecimal number represented

by colon- hexadecimal notation. Interface Name: String

format.

Example: <ChassisID>AgcEiFqSa oqA</ChassisID>

/status/diagnostics/lldp/chassis/portId

ID of the switch port that the device is connected to.

MAC address: 12-digit hexadecimal number represented

by colon- hexadecimal notation. Interface Name: String

format.

Example: <PortID>BAkFR2kyLzA vMTc=</PortID>

/status/diagnostics/lldp/chassis/chassisIdType

ERS supports the following Chassis ID subtypes:

◼ “Switch Hostname” / (6)

◼ “Switch IP” /(5)

◼ “Switch MAC Address”/(4)

/status/diagnostics/lldp/chassis/portIdType

ERS supports the following Port ID subtypes:

◼ “Port Name” (5)

◼ “Port MAC Address” /(3)

Part VIII Maintenance

Part VIII

Maintenance

39. Changing Administrator Login Credentials

400HD Series IP Phones

- 188 -

39 Changing Administrator Login Credentials

You can change the administrator phone's login user name and password. This is the login required

to access the Web interface and the Administration menu on the phone. The default administrator

user name and password is admin and 1234 respectively.

Administrator Login Credentials can be changed using the Web interface or configuration file.

To change the administrator's login username and password using the Web interface:

1. Open the Users page (Management tab > Administration > Users).

Figure 101: Users – Administrator Account

2. In the 'Username' field, enter a username.

3. In the 'Password' field, enter a new password, and then in the 'Confirm Password' field, re-

enter this new password.

4. Click Submit; a confirmation prompt appears.

5. Click OK.

To change the login username and password using the configuration file:

◼ Use the table below as reference.

Table 4: Username and Password Parameters

Parameter

Description

Username

[system/user_name]

The Administrator account username. The default value is admin.

Note: This parameter is applicable only to the Web interface.

Password

[system/password]

The phone password is by default encrypted.

The default value is 1234.

To regenerate an encrypted password, see Section 4.6.2.

Note: This parameter applies to the Web interface as well as to the phone's

screen.

40. Restarting Phones

400HD Series IP Phones

- 189 -

40 Restarting Phones

The phone can be restarted from the phone or using the Web interface.

40.1 Restarting from the Phone

Here's how to restart the phone from the phone.

To restart the phone from the phone:

1. On the phone, select the Restart option. Either:

• MENU key > Administration > Restart)

-or-

• MENU key > Settings > Restart

Here's the Administration screen's Restart option:

A warning message appears requesting you to confirm:

2. Press the Yes softkey to confirm the restart or No to cancel.

40.2 Restarting the Phone using the Web Interface

You can use the Web interface to restart your phone.

To restart the phone:

1. Open the Restart System page (Management > Administration > Restart System).

Figure 102: Restart System

2. Click the Restart button; a confirmation box appears prompting you to confirm.

Figure 103: Confirmation Box

3. Click OK.

41. Restoring Phone Defaults

400HD Series IP Phones

- 190 -

41 Restoring Phone Defaults

Phone default settings can be restored from the phone's screen or using the Web interface.

41.1 Restoring Factory Defaults from the Phone's Screen

Here's how to restore factory defaults from the phone's screen.

To restore the phone to default settings:

1. On the phone, open the Restore Defaults screen (MENU key > Administration > Restore

Defaults).

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 to cancel.

You can restore the phone's settings to their defaults without needing access to the

'Administration' menu or (2) administrator access to the Web interface.

To restore the phone's settings to their defaults if necessary:

1. Long-press the OK and MENU keys simultaneously and while pressed, unplug the

power cable.

2. Plug the power cable back into the phone and continue to press the OK + MENU keys

for +-5 seconds as the boot process starts after connecting the power supply.

3. Release the OK + MENU keys; the phone' settings are restored to their defaults.

41. Restoring Phone Defaults

400HD Series IP Phones

- 191 -

41.2 Restoring Factory Defaults using the Web Interface

You can restore your phone's settings to factory defaults using the Web interface.

To restore the phone to factory defaults:

1. Open the Restore Defaults page (Management > Administration > Restore Defaults).

Figure 104: Restore Defaults

2. Click Submit.

Figure 105: Confirmation to Restore Defaults

3. Click OK.

Part IX Sta tus and Monitoring

Part IX

Status and Monitoring

42. Determining Network Status

400HD Series IP Phones

- 193 -

42 Determining Network Status

Here's how to determine network status using the Web interface.

42.1 Determining LAN Status

Here's how to determine LAN status information.

To determine LAN status information:

◼ Open the Network Status page (Status & Diagnostics tab > System Status menu > Network

Status).

Figure 106: LAN Information

42.2 Determining Port Status

Here's how to determine Port Mode status using the Web interface.

To determine Port Mode status:

◼ Open the Network Status page (Status & Diagnostics tab > System Status menu > Network

Status).

Figure 107: Port Mode Status

42.3 Determining 802.1x Status

Here's how to determine 802.1x status.

To determine 802.1x status:

◼ Open the Network Status page (Status & Diagnostics tab > System Status menu > Network

Status).

Figure 108: 802.1X Status

43. Determining VoIP Status

400HD Series IP Phones

- 194 -

43 Determining VoIP Status

Here's how to determine VoIP status using the Web interface.

43.1 Determining Phone Status

Here's how to determine phone status.

To determine phone status:

◼ Open the VoIP Status page (Status & Diagnostics tab > System Status menu > VoIP Status).

Figure 109: VoIP Status - Phone Status

43.2 Determining Line Status

Here's how to determine line status.

To determine line status:

◼ Open the Network Status page (Status & Diagnostics tab > System Status menu > VoIP Status).

Figure 110: Line Status

43. Determining VoIP Status

400HD Series IP Phones

- 195 -

43.3 Determining Memory Status

Here's how to determine the device's memory status in real time, using the three Linux commands

that are most frequently used to obtain data related to a device's memory state.

To determine memory status:

1. Open the Memory Status page (Status & Diagnostics > System Status > Memory Status).

Figure 111: Memory Status

2. From the dropdown list select a Linux command from the three available:

• meminfo

• ps

• top

Use the table below as reference.

Table 5: Memory Status – Linux Commands

Linux Command

Description

Meminfo

Provides you a snapshot of memory usage on the device.

Provides you a snapshot of the current processes running on the device's CPU.

top

Provides you an ongoing look at processor activity in real time. Displays a

listing of the most CPU-intensive tasks on the system.

3. Click Refresh; the information requested is displayed.

43. Determining VoIP Status

400HD Series IP Phones

- 196 -

Figure 112: Memory Status – Linux meminfo Command – Displayed Information

4. Click Save (as text file) (optional); the information provided by the Linux command is saved to

a txt file. You can use the file to make sure all data is stored correctly in memory and to

diagnose possible issues such as voice quality, jitter, or memory leakage.

5. Click Maximize (optional); the information pane is maximized for an optimal viewing

experience.

43.4 Viewing Current Call Information

The Web interface displays call information of a currently established call.

To view current call information:

◼ Open the Network Status page (Status & Diagnostics tab > System Status menu > VoIP Status):

Figure 113: Web Interface –Line 1 Call Information

44. Viewing Call History

400HD Series IP Phones

- 197 -

44 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:

1. Open the Call History page (Status & Diagnostics tab > History menu > Call History).

Figure 114: Call History

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.

45. Accessing System Information

400HD Series IP Phones

- 198 -

45 Accessing System Information

45.1 Accessing Phone Firmware Version

You can determine the phone firmware version using the Web interface or from the phone screen.

45.1.1 Accessing Firmware Version using the Web Interface

You can determine the firmware version currently running on the phone.

To view the phone's firmware version:

◼ Open the System Information page (Status & Diagnostics tab > System Information menu >

General).

Figure 115: System Information–Firmware Version

45.1.2 Accessing Firmware Version from the Phone's Screen

Here's how to determine firmware version on the phone.

To determine the phone's firmware version from the screen:

◼ On the phone, open the Firmware Version screen (MENU key > Status > Firmware Version):

45. Accessing System Information

400HD Series IP Phones

- 199 -

45.2 Viewing Phone Firmware Release Information

You can view phone firmware release information in the Web interface or in the phone screen.

45.2.1 Viewing Firmware Release Information in the Web Interface

To view firmware release information in the Web interface:

◼ Open the Release Information page (Status & Diagnostics tab > System Information menu >

Release Information).

Figure 116: System Information – Release Information

45.2.2 Viewing Firmware Release Information on the Phone

Here's how to view firmware release information on the phone.

To view firmware release information on the phone:

1. On the phone, open the Release Information screen (MENU key > Status > Release

Information).

2. Release information includes:

• BL Version

• Build Time

• DSP FW Version

• Hardware Type

• Log

• Software Version

• Software Type

46. Monitoring Quality of Experience

400HD Series IP Phones

- 200 -

46 Monitoring Quality of Experience

You can configure the phone to send Quality of Experience reports to a QoE collecting server, such

as the AudioCodes SEM server. This mechanism is implemented using RTCP-XR (RTCP Extended

Reports). These extended reports include voice quality data events, such as Jitter Buffer, Packet Loss,

Delay and Burst, which are collected by the phone during the VoIP session.

When the SIP PUBLISH feature is enabled, upon the termination of the VoIP session e.g., call

disconnect or Hold states, values are calculated for each voice quality data event and sent to the

QoE server in a SIP PUBLISH message.

RTCP XR information publishing is implemented on the phone according to RFC 6035.

RTP Control Protocol Extended Reports (RTCP XR) is a VoIP management control that defines a set

of metrics containing information for assessing VoIP call quality and for diagnosing problems. RTCP

XR (RFC 3611) extends the RTCP reports defined in RFC 3550 by providing additional VoIP metrics for

Quality of Experience.

46.1 Configuring Remote Voice Quality Monitoring

To report voice quality events from the phone to a Quality of Experience Server (QoE):

◼ Configure the phone to retrieve RTCP XP events on voice quality data (see Section 46.1.1) -

and-

◼ Configure the phone to send SIP PUBLISH messages to the QoE server, including the RTCP XP

events described above and the SIP call messages (see Section 46.1.2).

46. Monitoring Quality of Experience

400HD Series IP Phones

- 201 -

46.1.1 Configuring RTCP Extended Report

Here's how to configure RTCP-XR (Extended Report for RTP Control Protocol) working mode. You

must enable the phone to retrieve RTCP-XR events using one of the methods described in the table

below (this feature is by default disabled).

To configure RTCP_XR working mode using the Web interface:

1. Open the Media Streaming page (Configuration tab > Voice Over IP menu > Media Streaming),

and then scroll down to the RTCP-XR section.

Figure 117: Media Streaming - RTCP-XR

2. Configure the 'RTCP-XR Voice Quality Statistics Mode' parameter, using the following table

below as reference.

To configure RTCP_XR working mode using the configuration file:

Table 6: RTCP_XR Parameters

Parameter

Description

RTCP-XR Voice Quality Statistics Mode

[voip/rtcp_xr/vq_statistics/mode]

Sets RTCP_XR working mode.

Select either:

◼ [DISABLE] (default). In this state, no RTCP events are

retrieved from the phone and the SIP PUBLISH is not sent,

regardless of the state of parameter

'qoe_publish_enabled' (see below).

◼ [EVENTS_ONLY]. In this state, RTCP-XR events with voice

quality parameter calculations are sent internally on the

phone every five seconds. Each calculation is made on

the basis of these RFC 3611 parameters: BT=7, block

length = 8SSRC of source, loss rate, discard rate, burst

density, gap density, burst duration, gap duration, round

trip delay, end system delay, signal level, noise level,

Gmin, R factor, ext. R factor, MOS-LQ, MOS-CQ, RX

config, JB nominal, JB maximum and JB abs max. The

phone sends the summarized RTCP-XR events to the

Skype for Business server / OVOC server via SIP SERVICE

messages (in Genesis-SIP, SIP PUBLISH messages are

used).

◼ [REMOTE_AND_EVENTS]. In this state, the phone sends

RTCP-XR events to the remote calling party (i.e., party A

sends these events to party B) every five seconds during

the VoIP session. The phone sends the summarized

RTCP-XR events to the Skype for Business server / OVOC

server via SIP SERVICE messages (in Genesis-SIP, SIP

PUBLISH messages are used).

46. Monitoring Quality of Experience

400HD Series IP Phones

- 202 -

46.1.2 Configuring Voice Quality Monitoring

You can set up the phone to report SIP PUBLISH messages to a remote QoE server.

To configure voice quality monitoring using the configuration file:

◼ Use the table below as reference.

Table 7: Voice Quality Monitoring Parameters

Name

Role

[/voip/qoe/qoe_publish_enabled]

Determines whether or not to send PUBLISH messages (Default-0).

[/voip/qoe/qoe_server_address]

Sets the QoE server address/hostname to which PUBLISH

messages will be sent (Default-0.0.0.0).

[/voip/qoe/qoe_server_port]

Sets the port to which the PUBLISH messages will be sent (Default-

5060).

For a full listing of RTCP XR parameters that may be sent to the QoE server, see Appendix G.

For example SIP PUBLISH messages, see Appendix H.

Part X Diagnos tics and Troubleshooting

Part X

Diagnostics and

Troubleshooting

47. Diagnosing Phone Hardware

400HD Series IP Phones

- 204 -

47 Diagnosing Phone Hardware

Here's how to test the phone's hardware functionality.

To diagnose phone hardware:

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. On the keypad, key in the number 0123456789 and then press the star key (*); the diagnostic

tests are displayed on the phone screen.

Figure 118: Diagnostic Tests Displayed on the Phone

6. Run all tests consecutively (MENU option 0 – All Tests) -OR- run a single test (MENU options

1-10).

Option

Test

Description

All Tests

1-10 (listed below), performed in one continuous sequence, where each test is followed

by the next in the order listed.

Product Info

Displays MAC address, SN (serial number), VERSION, IP address, manufacturing

company (AudioCodes) and phone model. Verify that product information is correct.

Main Screen

& Backlight

Tests the backlight in the main screen. Press OK successively to test correct

functionality.

BLF screen &

Backlight

(Only applicable to 440HD model) Tests the backlight in the BLF. Press OK successively

to verify All White, All Black, Backlight On and Backlight Off. Observe screen

notifications.

Keypad and

Hook

Tests off-hook/on-hook functionality. Verify correct functionality. For more details, see

Section 47.1.

LED

Tests the LEDs. Press OK successively; screen displays 'Green LEDs are on' followed by

'Red LEDs are on' followed by 'Blue LEDs are on' followed by 'All LEDs are on'. Verify

correct functionality.

Handset

Tests handset microphone for recording/speaking and handset receiver (speaker) for

playing/listening. Verify correct functionality. For more details, see Section 47.2.

Headset

Tests headset microphone (recording/speaking) and headset headphone/receiver

(playing/listening). Verify correct functionality. For more details, see Section 47.3

below.

Hands Free

Tests speaker microphone (recording/speaking) and speaker receiver

(playing/listening).Verify correct functionality. For more details, see Section 47.4 below.

USB

(Applies only to 430HDand 440HD models) Plug in a USB device; 'USB device was

detected' should be displayed in the screen. Remove it; 'USB device was removed'

should be displayed. Verify correct functionality.

AUX

Used for debugging.

47. Diagnosing Phone Hardware

400HD Series IP Phones

- 205 -

47.1 Testing Keypad and Hook

To test keypad and hook:

1. On the phone, select MENU option 4, Keypad and Hook; the screen displays a graphic

representation of the physical keypad (screens differ slightly across phone models).

Figure 119: Keypad and Hook Test– On-Hook (e.g., 440HD)

2. Off-hook the handset; the background of the hook indicator ('H') turns black.

Figure 120: Keypad Test – Off-Hook (e.g., 440HD)

3. On-hook the handset; the background of the hook indicator ('H') reverts to white.

4. Press each key on the keypad; its representation in the screen turns black.

47.2 Testing Handset

To test the handset:

1. Select MENU option 6, Handset; the screen indicates 'Off-hook phone'.

2. Off-hook the phone; the screen indicates 'Start recording…'Speak into the handset until the

screen displays 'Start playing…' and you hear your speech playing back.

47.3 Testing the Headset

To test the headset:

1. Select MENU option 7, Headset; the screen indicates 'Press any key'.

2. Press any key; the screen indicates 'Start recording…' Speak until the screen displays 'Start

playing…' and you hear your speech playing back.

47.4 Testing Hands Free

To test hands free:

1. Select MENU option 7, Hands Free; the phone screen indicates 'Press any key'.

2. Press any key; the phone screen indicates 'Start recording…' Speak into the phone speaker for

a few seconds until the phone screen displays 'Start playing…' and you hear your speech

playing back through the speaker.

48. Recovering Firmware

400HD Series IP Phones

- 206 -

48 Recovering Firmware

Here's how to recover the phone's firmware. See also Appendix C for more detailed information. If

the phone is powered off for some reason during the firmware upgrade process, the phone becomes

unusable. The recovery process is also available when the phone is connected to a VLAN.

To recover the phone firmware:

1. Make sure 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. 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.

b. 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., 440HD_2.2.2.img).

c. The phone then upgrades to the recovery firmware.

d. After the firmware upgrade process completes, the phone boots up successfully.

49. Configuring System Logging (Syslog)

400HD Series IP Phones

- 207 -

49 Configuring System Logging (Syslog)

System logging (Syslog) can be configured using the Web interface or configuration file.

Two Syslog options are available:

◼ Regular Syslog (see the next section)

◼ Lightweight Syslog (see Section 49.2).

49.1 Analyzing and Debugging Traffic using Regular Syslog

The System Logging (Syslog) feature is used for traffic analysis and debugging. The feature can be

configured using the Web interface or configuration file.

To configure system logging using the Web interface:

1. Open the System Logging page (Status & Diagnostics > Diagnostics > Logging).

Figure 121: Web Interface –System Logging

2. Configure using the table below as reference.

To configure system logging using the configuration file:

◼ Use the table below as reference.

Table 8: Syslog Parameters

Parameter

Description

Activate

[system/syslog/mode]

Enables Syslog. 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 to

the console.

◼ Note: It is recommended to leave this parameter at its

default value.

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

49. Configuring System Logging (Syslog)

400HD Series IP Phones

- 208 -

Parameter

Description

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

◼ Warning

◼ Notice

◼ Info

◼ Debug

VoIP Application

[system/syslog/component/voip_application]

Defines multi-layer VoIP application.

SIP Call Control

[system/syslog/component/sip_call_control]

Defines MTR layer Radvision.

SIP Stack

[system/syslog/component/sip_stack]

Defines SIP Stack Radvision.

Control Center

[system/syslog/component/control_center]

Responsible for Networking and running other processes.

LCD Display

[system/syslog/component/lcd_display]

Defines the phone screen display.

Web

[system/syslog/component/web_server]

Defines the phone Web server.

Watchdog

[system/syslog/component/watchdog]

Responsible for keeping other processes running.

802.1X

[system/syslog/component/ieee802_1x]

Defines the security protocol.

Kernel

Defines the core of the phone (430HD and 440HD only).

DSP

[system/syslog/component/dsp]

Defines the voice engine of the phone (430HD and 440HD

only).

XSI

[system/syslog/component/xsi]

Defines logging for the xsi feature.

Default: DEBUG

Infra

[system/syslog/component/infra]

Defines logging for code infrastructure.

Default: DEBUG

49. Configuring System Logging (Syslog)

400HD Series IP Phones

- 209 -

49.2 Analyzing and Debugging Traffic using 'Lightweight Syslog'

A 'Lightweight Syslog' logging mechanism allows you to perform phone logging without affecting

phone performance.

To enable the Lightweight Syslog:

◼ In the Web interface, open the phone's System Logging page (Status & Diagnostics tab >

Diagnostics > Logging), set the 'Activate' parameter to Network and provide a valid IP address

and server port. Do not set any of the options (keep all as None).

50. Viewing Error Messages Displayed in the Phone Screen

400HD Series IP Phones

- 210 -

50 Viewing Error Messages Displayed in the Phone

Screen

The table below shows the error messages that may be viewed on the phone.

Table 9: Error Messages Displayed in the Phone Screen

Message

Description

LAN Link failure

The LAN link is disconnected.

Registration failure

Received error or no response from the SIP proxy

• 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.

51. Debugging using Packet Recording Parameters

400HD Series IP Phones

- 211 -

51 Debugging using Packet Recording Parameters

Packet recording parameters allow you to debug voice activity on the phone, using the Web interface

or configuration file.

To debug using the Web interface:

1. Open the Recording page (Status & Diagnostics tab > Diagnostics menu > Recording).

Figure 122: Recording

2. Configure the packet recording parameters using the table below as reference, and then click

Submit.

51. Debugging using Packet Recording Parameters

400HD Series IP Phones

- 212 -

To debug using the configuration file:

◼ Use the table below as reference.

Table 10: Recording Parameters

Parameter

Description

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 DSP Recording

[voip/packet_recording/enabled]

Activates the packet recording mechanism.

◼ [0] Disable (default)

◼ [1] Enable

Enable RTP Recording

[voip/packet_recording/rtp_recording/enabled]

Only displayed if the parameter 'Enable

DSP Recording' is enabled.

Activates the DSP RTP recording.

◼ [0] Disable (default)

◼ [1] Enable

Enable EC Debug Recording

[voip/packet_recording/ec_debug_recording/enabled]

Activates the Echo Canceller Debug

recording.

◼ [0] Disable (default)

◼ [1] Enable

Enable Noise Reduction Debug Recording

Traffic on the network stops when the

MUTE key is activated.

◼ [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

[voip/packet_recording/cng_debug_recording/enabled]

Default=0

52. Creating a Crash Dump File

400HD Series IP Phones

- 213 -

52 Creating a Crash Dump File

Here's how to create a crash dump file using the Web interface. Crash dump copies historical

processes to a file. The file is created at the time the problem occurs. You must send it to AudioCodes

Customer Technical Support for troubleshooting.

To download a crash dump file using the Web interface:

1. Open the Crash Dump page (Status & Diagnostics tab > Diagnostics menu > Crash Dump).

Figure 123: Crash Dump

Table 11: Crash Dump Parameters

Parameter

Description

Total tombstones

The number of crashes on the phone.

Last tombstone

The date and time of the last crash (the exact time of the

crash).

2. Click Download to save the crash dump file on your computer.

53. Configuring Port Mirroring

400HD Series IP Phones

- 214 -

53 Configuring Port Mirroring

Traffic on the phone's LAN port can be duplicated on its PC port to record calls, analyze traffic, and

troubleshoot issues.

This port mirroring feature can be configured using the Web interface or configuration file.

To configure port mirroring using the Web interface:

1. Open the Recording page (Configuration tab > Network Connections menu > Network

Settings).

Figure 124: Web Interface –Port Mirroring

2. Configure port mirroring according to the parameters in the table below, and then click

Submit.

To configure port mirroring using the configuration file:

◼ Use the table below as reference.

Table 12: Port Mirroring Parameters

Parameter

Description

Activate

[network/pc_port_mirroring/enabled]

Enables port mirroring.

◼ [0] Disable (default) - LAN/PC network interfaces operate

in SWITCH mode.

◼ [1] Enable - LAN/PC network interfaces operate in HUB

mode. The network traffic on the LAN port is reflected in

the PC port.

54. Enabling Tracing

400HD Series IP Phones

- 215 -

54 Enabling Tracing

Here's how to set up the phone to store trace messages. You can then send the saved logged trace

to AudioCodes Customer Technical Support for effective troubleshooting and diagnosis.

• During regular phone operation it's advisable to disable debug tracing for improved

performance.

• It's advisable to enable debug tracing only if bugs are encountered and AudioCodes

asks you to provide tracing logs.

To enable tracing:

1. In the Web interface, open the Tracing System Key Behavior page (Status & Diagnostics tab >

Diagnostics > Tracing).

Figure 125: Tracing System Key Behavior

2. Set the 'Max File Size' field to 1024 (see the table below).

3. Set the 'Trace Level' field to Debug to activate tracing to debug level, and then click Submit

(see the table below).

4. Click Clean log, and then power up the phone.

5. To save the log file to your computer, click Save log.

6. Send the saved logged trace to AudioCodes Customer Technical Support for troubleshooting.

54. Enabling Tracing

400HD Series IP Phones

- 216 -

To configure tracing using the configuration file:

◼ Use the table below as reference.

Table 13: Tracing Parameters

Parameter

Description

[system/trace/console_printf/enabled]

Enables / disables console print.

◼ [0] Disable (default) – Disables console print.

◼ [1] Enable - Enables console print.

Trace Level

[system/trace/level]

During regular phone operation it's advisable to

disable debug tracing for improved performance. It's

advisable to enable debug tracing only if bugs are

encountered and AudioCodes asks you to provide

tracing logs, in which case set this parameter to

Debug.

Valid values are:

◼ None (default)

◼ Emergency

◼ Error

◼ Warning

◼ Notice

◼ Info

◼ Debug

Max File Size

[system/trace/max_file_size]

Set to 1024

Default: 200

- 217 -

Part XI Appendices

Part XI

Appendices

A. Configuring Phones in Server-Specific Deployments

400HD Series IP Phones

- 218 -

A Configuring Phones in Server-Specific

Deployments

This appendix shows how to configure phones in server-specific deployments.

A.1 BroadSoft's BroadWorks

Features supported in a BroadSoft environment are:

Table 14: Features Supported in a BroadSoft Environment

Feature

405/405HD

420HD

430HD

440HD

BSFT DMS for provisioning

√

BLF Support

√

Call FWD

√

Call Transfer

√

Call Park

√

Call hold

√

Dial Plan

√

Caller ID

√

Message Waiting Indication

√

Local 3-Way Conference

√

DND

√

Feature key Sync

√

Network Conference

√

Shared Call Appearance

√

Broadworks Phone book support

√

Voice Message Support

√

DNS SRV Lookup for Redundancy and register failover

√

ACD (Automatic Call Distribution)

√

A.1.1 Configuring BLF

Configuration of the BLF feature is unique when the selected application server is BroadSoft's

BroadWorks application platform.

This section only applies to 430HD and 440HD phones.

To configure BLF in a BroadSoft environment:

1. Open the Services page (Configuration tab > Voice Over IP menu > Services).

A. Configuring Phones in Server-Specific Deployments

400HD Series IP Phones

- 219 -

Figure 126: BLF Configuration in a BroadSoft Environment

2. From the 'Application Server Type' drop-down list, select BSFT.

3. Under BLF Support do this:

a. From the 'Activate' drop-down (voip/services/busy_lamp_field/enabled parameter),

select Enable.

b. In the 'BLF Subscription Period' field (voip/services/busy_lamp_field/subscription_period

parameter), enter the interval between BLF and SIP SUBSCRIBE messages.

c. 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.

d. 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 (see Section 16.2).

 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/addrparameter), 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 25).

A. Configuring Phones in Server-Specific Deployments

400HD Series IP Phones

- 220 -

A.1.2 Configuring Call Forwarding

This section is only applicable to 430HD and 440HD phones.

A.1.2.1 From the Phone

Call Forwarding can be configured in a BroadSoft environment using the phone's screen.

Before configuring Call Forwarding in the phone's screen, make sure 'Feature Key

Synchronization' in the phone's Web interface is disabled (see Section A.1.4).

To configure call forwarding on the phone:

◼ See the phone's User's Manual for detailed information.

A.1.2.2 From BroadSoft's BroadWorks

Call Forwarding can also be configured from the BroadSoft BroadWorks application platform. Call

Forwarding status can also be retrieved from it.

To enable BroadWorks-managed Call Forwarding, you must first enable 'Feature Key

Synchronization' using the Web interface (see Appendix A.1.4).

The figure below shows Call Forwarding configuration and status in BroadSoft's BroadWorks.

Figure 127: Configuring Call Forwarding using BroadSoft's BroadWorks

The 'Forward No Reply' timeout can be configured as 'number of rings' rather than as

'seconds' if the BroadSoft Feature Key is enabled (by configuring the cfg file parameter

'voip/line/0/call_forward/timeout_mode' to RINGS_COUNT instead of to the default

SECONDS). For example, if the BroadSoft Feature Key is enabled and

'voip/line/0/call_forward/timeout_mode' is configured to RINGS_COUNT, the phone will

by default ring 2r (2 rings) before the call is forwarded. The setting can be changed

according to user preference to 4r (4 rings), for example. The feature allows compliance

with BroadSoft's Feature Key Synchronization method. 97562

For detailed information, see related BroadSoft documentation.

A. Configuring Phones in Server-Specific Deployments

400HD Series IP Phones

- 221 -

A.1.3 Configuring DnD

The DnD feature can be configured in the phone's screen or in the Web interface.

Before configuring DnD in the phone's screen, make sure 'Feature Key Synchronization' in

the phone's Web interface is disabled (see Section A.1.4).

The DnD feature can also be configured using the BroadSoft BroadWorks application server. The

figures below show the DnD configuration and status screens in BroadWorks.

Figure 128: Configuring DnD in BroadSoft's BroadWorks - Status

Figure 129: Configuring DnD in BroadSoft's BroadWorks

For detailed information, see related BroadSoft documentation.

A. Configuring Phones in Server-Specific Deployments

400HD Series IP Phones

- 222 -

A.1.4 Configuring FKS

Enabling Feature Key Synchronization synchronizes the DnD and Call Forward functionalities with

the BroadSoft BroadWorks server. After activating the feature, the DnD and Call Forward

functionalities are performed by BroadWorks rather than the phone. For more information on DnD

functionality, see Section 22.8.

To enable Feature Key Synchronization:

1. In the Web interface, open the Services screen (Configuration > Voice over IP > Services).

Figure 130: Services

2. From the 'Type' dropdown under Application Server, select the BSFT (BroadSoft) option; the

'Feature Key Synchronization' dropdown is displayed.

Figure 131: Services - Feature Key Synchronization

3. From the 'Feature Key Synchronization' dropdown, select Enable and then Submit.

A.1.5 Using SIP Authentication for Xsi Access

BroadSoft environment users can enter their BroadWorks user credentials for Xtended Services

Interface (Xsi) access. The phones use SIP authentication data to authenticate Xsi access. The phones

send the BroadWorks user ID to the Xtended Services Platform (Xsp) to identify the user, along with

the SIP authentication user name and password to authenticate access to the Xsi.

A. Configuring Phones in Server-Specific Deployments

400HD Series IP Phones

- 223 -

A.1.6 Configuring Phones to Connect to the Xsi Interface using HTTP/S

Authentication

BroadSoft environment users can enter their BroadWorks user credentials for Xsi access using

HTTP/S authentication. The phone supports three Xsi services:

◼ Call Center list

◼ Users can be assigned up to three call centers that will be displayed on the right side of the

user's phone screen.

◼ Configured on programmable keys 4-6, three call centers Dept. B, Dept. C and Dept. A are

displayed on the screen above. The network administrator can enable | disable the feature

using configuration file parameter xsi/callcenter/update which by default is enabled.

◼ The feature allows enterprise front desk personnel to indicate their availability status

(available or unavailable), in each call center, to the BroadWorks server. The server then

efficiently distributes incoming calls to front desk personnel, saving callers from the

inconvenience of unanswered referrals or disconnections.

◼ Contact Synchronization

◼ Contact directories are pulled directly from the BroadWorks server. Case-insensitive Abc

name search is performed instantly. Supported directories are Group Directory, Enterprise

Directory, Group Common, Enterprise Common and Personal Directory. The network

administrator can enable | disable the feature using configuration file parameter

xsi/contact/enable which by default is enabled. The feature cannot coexist with contacts

saved locally on the phone.

◼ Call Log Synchronization

Call Logs are pulled directly from the BroadWorks server. The phone displays the following

Call Logs: All Calls, Missed Calls, Received Calls and Dialed Calls. The network administrator

can enable | disable the feature using configuration file parameter xsi/calllog/enable which

by default is enabled.

A. Configuring Phones in Server-Specific Deployments

400HD Series IP Phones

- 224 -

To connect phones to BroadWorks over HTTP/S using the configuration file:

◼ Use the table below as reference.

Table 15: Connecting Phones to BroadWorks over HTTP/S – Configuration File Parameters

Parameter

Description

[xsi/callcenter/update]

◼ [1] The phone gets the call center service from

the BroadWorks server

◼ [0] The phone does not get the call center service

from the BroadWorks server (default).

[xsi/contact/enable]

◼ [1] Contacts and contact information are saved

on the BroadWorks server

◼ [0] Contacts and contact information are saved

locally on the phone (default)

[xsi/host]

Defines the phones host, BroadWorks server. The

phones receive three web services from the server.

1. Contacts list

2. Call Log (history of calls dialed, received, missed,

etc.)

3. Call Center

[xsi/http_port]

Defines the BroadSoft BroadWorks online server

port number. Default: 80. Must be configured for the

phones to receive web services from the BroadSoft

BroadWorks online server over HTTP.

[xsi/http_security]

Defines the authentication method phones will use

opposite the BroadSoft BroadWorks online server.

Can be HTTP or HTTPS. Must be configured for the

phones to receive web services over HTTP/S from

the BroadSoft BroadWorks online server.

[xsi/update_time]

Every 20 minutes (default) phones update their call

center status. Programmable keys on the phone

indicate call centers' statuses.

◼ Key illuminated red = Phone unavailable

◼ Key illuminated green = Phone available

Phone users can press a key to change the phone's

status in that call center. If a user presses a key

illuminated red, the key turns green and the phone's

status in that call center changes from unavailable to

available. If a key is illuminated green and the user

presses it, the phone's status in that call center

changes from available to unavailable.

The BroadWorks server updates phone status every

20 minutes by default.

[xsi/user_id]

Defines the user ID required for the phone to access

the BroadSoft BroadWorks online server.

[xsi/password]

Defines the password required for the phone to

access the BroadSoft BroadWorks online server.

[xsi/authentication_with_SIP]

Allows for XSI authentication using SIP credentials.

1 = Enable

0 = (Default) Disable

A. Configuring Phones in Server-Specific Deployments

400HD Series IP Phones

- 225 -

A.1.7 Configuring Shared Call Appearance

The SCA feature enables multiple phones to be associated in an SCA group so that calls can be made

or received on any phone in the group.

• The 440HD phone fully supports the Shared Call Appearance (SCA) feature.

• The 405 / 405HD / 420HD / 430HD phones support the feature as participant only.

Figure 132: Shared Call Appearance with Multiple Call Appearance

To configure Shared Calls Appearance:

1. In the BroadSoft server, assign Shared Call Appearance to the user: Under the 'User' level,

select the Call Control option, and then click the Shared Call Appearance tab.

Figure 133: BroadSoft Server - Assigning Shared Calls Appearance to a User

2. On the Shared Call Appearance page shown in the figure above, click Add to configure an

'Identity/Device Profile Type', and then use the table below as reference to configure the

parameters.

Speed Dials

for quick

transfer from

main line

A. Configuring Phones in Server-Specific Deployments

400HD Series IP Phones

- 226 -

Table 16: BroadSoft Server - Shared Call Appearance – Identity/Device Profile Type

Parameter

Description

Alert all appearances for Click-to-Dial calls

See BroadSoft's documentation for detailed

information.

Select this option when you want your Click-To-Dial

calls to ring all phones that have your line

appearance.

Clear the option if you prefer your line to ring your

phone only.

Alert all appearances for Group Paging calls

See BroadSoft's documentation for detailed

information.

Allow call retrieve from another location

See BroadSoft's documentation for detailed

information.

Select this option when you use a feature access

code to automatically retrieve a call that was

answered at another Shared Call Appearance of your

number.

Multiple Call Arrangement

See BroadSoft's documentation for detailed

information.

Select On to allow multiple calls using your phone

number / ID to be dialed or answered

simultaneously across all Shared Call Appearances of

your number.

Select Allow bridging between locations when you

want to use a feature access code to bridge a 3-way

conference call automatically for any call that has

been answered at another Shared Call Appearance

of your number.

Select Enable Call Park notification for the phone to

alert the user visually and audially when a parked

call is received.

Bridge Warning tone

See BroadSoft's documentation for detailed

information.

Select the type of Bridge Warning tone treatment

you prefer when you bridge and join a call using a

feature access code.

Select None to apply no tone alert treatment upon

your entry to the call.

Select Barge-in only to provide a single tone alert.

Select Barge-in and repeat every 30 seconds to

provide a tone alert at that interval.

3. Click the Edit link adjacent to the selected Line/Port to modify a specific phone that has a

Shared Call Appearance of your line.

4. Click OK.

A. Configuring Phones in Server-Specific Deployments

400HD Series IP Phones

- 227 -

Figure 134: BroadSoft Server – Shared Call Appearance Add

Table 17: BroadSoft Server - Shared Call Appearance Add

Parameter

Description

Identity/Device Profile Name

See BroadSoft's documentation for detailed information.

From the dropdown, select the Identity/Device Profile Type you

configured previously.

Line/Port

Enter the required SIP register address-of-record, for example

(shown in the figure above):

242111409[email protected]

Enable this location

Select this option to enable this user station.

Allow Origination from this location

Select this option to allow calls to be made from this user

station.

Allow Termination to this location

Select this option to allow calls to be received at this user

station.

5. In the Web interface, access the Line Settings page (Voice Over IP > Line Settings) and from

the 'Line 1 Mode' dropdown, select Shared (line).

Figure 135: Line Settings - Shared Line

Icons displayed on the phone's screen indicate if line keys are configured in a Shared Call

Appearance group, or as private lines. A hollow icon indicates a phone configured in an

SCA group. A solid icon indicates a phone configured as private.

A. Configuring Phones in Server-Specific Deployments

400HD Series IP Phones

- 228 -

A.1.8 Setting up a Remote Conference

Here's how to set up BroadSoft’s remote conference feature. More than three participants can be

added to a remote conference call. A 'local' conference only supports a maximum of three. The

feature must be enabled on BroadSoft's BroadWorks server for it to function.

To set up the remote conference feature using the Web interface:

1. In the Web interface, open the Services page (Configuration > Voice Over IP > Services).

Figure 136: Services - Conference

2. From the 'Type' dropdown under 'Application Server', select BSFT. The feature must be

enabled on BroadSoft's BroadWorks server for it to be supported.

3. From the 'Mode' dropdown under 'Conference', select Remote.

4. In the 'Remote Conference Media Server' field, enter the address of the server hosting the

remote conference. In the example shown in the figure above, the server's address is

conference@iop1broadworks.net

5. Click Submit; the user can now establish a remote conference from the phone (see the

phone's User's Manual for detailed information).

To set up the remote conference feature using the configuration file:

◼ Use the table below as reference.

Table 18: Remote Conference Parameters

Parameter

Description

[voip/services/application_server_type]

Set to BSFT.

[voip/services/conference/conf_ms_addr]

Set the address of the server hosting the remote

conference. Example:

conference@i[email protected]

roadworks.net

[voip/services/conference/mode]

Set the mode to REMOTE.

A. Configuring Phones in Server-Specific Deployments

400HD Series IP Phones

- 229 -

A.1.9 Loading the Corporate Directory to the Phone

Here's how to load the Corporate Directory to the phone. The Corporate Directory can be loaded

either

◼ manually, by loading a (configurable) txt, cfg, or xml file listing all employees and their details,

to the phone, via the Directory page in the Web interface

◼ automatically, by placing the (configurable) txt, cfg, or xml file on the BroadSoft BroadWorks

server, and then when the phone is connected to the network, the phone pulls and

automatically uploads the file from the server.

To load the Corporate Directory manually, via the Web interface:

1. In the Web interface, open the Directory page (Configuration > Personal Settings > Directory).

Figure 137: Directory

2. In the Corporate Direction section of the screen, click Browse and navigate to and select the

txt, cfg, or xml file on your PC.

3. Click the adjacent Load Corporate Directory button; the file is loaded onto the phone.

4. Verify the upload: On the phone, press the MENU key, navigate down to select the Directory

option, and then select Corporate directory; the directory opens displaying the contacts listed

in the corporate directory.

To load the Corporate Directory automatically, via the BroadSoft server:

◼ Place the txt, cfg, or xml file on the BroadSoft BroadWorks server; when the phone is

connected to the network, it pulls and automatically loads the file to the phone.

A. Configuring Phones in Server-Specific Deployments

400HD Series IP Phones

- 230 -

A.1.10 Adding a Contact to the Corporate Directory

Here's how to add a contact to the Corporate Directory.

To add a contact to the Corporate Directory:

◼ In the Directory page shown in the figure above, enter the contact's Name, Office, Home and

Mobile number fields, and then click Submit.

A.1.11 Deleting a Contact from the Corporate Directory

Here's how to delete a contact from the Corporate Directory.

To delete a contact to the Corporate Directory:

1. Scroll down in the Directory page (shown in the figure above), and then select the contact to

delete.

Figure 138: Directory – Delete a Contact

2. Click Delete; the contact is deleted from the Corporate Directory. You can select multiple

contacts and then click Delete All.

A.1.12 Disabling Handset Mode

Administrators can disable handset mode using the configuration file parameter

'voip/handset_mode/enabled' whose default is enabled. Some call centers don't want agents to

work with any device other than headsets. In this case, their administrators can change the

parameter default to disabled.

Some call centers don't even connnect the handsets to the phones. In this case, even

though the handsets are not physically connected to the phones, administrators should

disable the new parameter.

To disable handset mode:

◼ Use the table below as reference.

Table 19: BroadSoft Server - Shared Call Appearance Add

Parameter

Description

[voip/handset_mode/enabled]

When disabled [0], the handset becomes unavailable.

Configure either:

◼ [1] = Enabled (default)

◼ [0] = Disabled

A. Configuring Phones in Server-Specific Deployments

400HD Series IP Phones

- 231 -

A.1.13 Displaying a Message in Agents' Phone Screens

Call center administrators can use a configuration file parameter to define a message that will be

displayed in agents' phone screens, for example: 'Reminder: Your calls might be recorded'. Agents

will then see this message, together with the date (in month/day format), displayed in their screens

when their phones are in idle state.

To display a message in Agents' phone screens:

◼ Use the table below as reference.

Table 20: Displaying a Message in Agents' Phone Screens

Parameter

Description

[system/display/message_on_screen]

Defines a message that will be displayed in agents'

phone screens together with the date (in month/day

format)

A.1.14 Changing Phone Screen Backlight Timeout

Call center administrators can use a configuration file parameter to change phone screens’ backlight

timeout. A phone screen's backlight timeout can also be changed on the phone by the user, unless

the call center administrator has disabled the possibility with the configuration file parameter

personal_settings/menu/backlight_timeout/enabled (see Section 26 for more information about

disabling phone hard keys and softkeys).

To change the backlight timeout:

◼ Use the table below as reference.

Table 21: Backlight Timeout

Parameter

Description

[system/lcd/backlight/timeout]

Range: 0-6

0 = Always On

1 = 10 seconds (default)

2 = 20 seconds

3 = 30 seconds

4 = 40 seconds

5 = 50 seconds

6 = 60 seconds

A. Configuring Phones in Server-Specific Deployments

400HD Series IP Phones

- 232 -

A.2 Asterisk, Coral and Metaswitch

A.2.1 Configuring BLF

Configuration of the BLF feature is unique when the selected application server is Asterisk, Coral, or

Metaswitch.

This section only applies to the 430HD and 440HD phones.

To configure BLF for Asterisk application server:

1. Open the Services page (Configuration tab > Voice Over IP menu > Services).

Figure 139: BLF Configuration for Application Server Type - Asterisk

2. From the 'Application Server Type' drop-down list, select either:

• Asterisk

• Coral -or-

• Metaswitch

3. In the BLF Support group, configure the following:

• 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.

The application server's address is the same as the SIP Registrar address defined by

parameter voip/signalling/sip/sip_registrar/addr (see Section 16.2).

4. Click Submit.

5. Define speed dial keys with the BLF feature (see Section 25).

A. Configuring Phones in Server-Specific Deployments

400HD Series IP Phones

- 233 -

A.3 Genesys SIP Server for Contact Centers

This section shows system administrators how to quickly set up AudioCodes' IP phones to operate

with a Genesys SIP Server in a Genesys contact center.

A.3.1 Using DHCP to Auto Provision Phones

After connecting the LAN ports of your phones to the IP network and then connecting the phones to

the power supply, the phones will by default send a request to the Genesys contact center's network

server which will then automatically allocate an IP address and send configuration information to

each phone.

Make sure that the DHCP (Dynamic Host Configuration Protocol) options in your contact center's

DHCP server are correctly configured (see Section 9.2).

A.3.2 Verifying Firmware Version

After automatic provisioning, make sure the phone's firmware version is correct.

To verify firmware version:

◼ Navigate to the Firmware Status screen on the phone (MENU key > Status > Firmware Status).

A.3.3 Accessing a Phone's Web Interface

Use a standard Web browser such as Microsoft Internet Explorer to access any phone's Web

interface. Use the phone's IP address as the URL.

To obtain the phone's IP address:

◼ Open the Network Status screen on the phone screen (MENU key > Status > Network Status)

and navigate down to IP Address:

To access the phone's Web interface:

1. Open the Web browser and in the URL address field enter the phone's IP address (for

example, http://10.22.13.118 or https://10.22.13.118):

The Web login window opens.

The default User Name and Password are admin and 1234 respectively.

2. Alternatively, if your DHCP and DNS servers are synchronized, you can access the phone Web

browser using the following method:

http://<Phone Model>-<MAC Address>.<Domain Name>

E.g., http://440hd-001122334455.corp.YourCompany.com

3. Enter the User Name and Password, and then click OK.

A. Configuring Phones in Server-Specific Deployments

400HD Series IP Phones

- 234 -

A.3.4 Configuring Dual Registration to Ensure SIP Business Continuity for

Agents

Here's how to configure dual registration for Genesys SIP Business Continuity, using the Web

interface or configuration file.

AudioCodes' 420HD phone supports dual registration for integrating into Genesys' SIP Business

Continuity architecture.

SIP Business Continuity provides the ability for a group of agents to continue offering critical business

functions to customers in the event of a loss of all Genesys components running at a particular site.

The SIP Business Continuity architecture uses a synchronized, two-site deployment, where Genesys

switch and server components are mirrored at each site in an active-active configuration, so that any

agent can log in to either switch, at any time.

In a standalone SIP Server configuration with Business Continuity mode activated, AudioCodes

420HD phone will register on two sites simultaneously (i.e., register on both peer SIP Servers at the

same time).

To configure using the Web Interface:

1. In the Web interface, access the SIP Proxy and Registrar section in the Signaling Protocol

screen (Configuration menu > Voice Over IP > Signaling Protocol):

Figure 140: Signaling Protocol – SIP Proxy and Registrar

If you choose to use the BroadSoft-based Automatic Call Distributor (ACD) method in a SIP

Business Continuity deployment, the 'voip/signalling/sip/redundant_proxy/mode' cfg file

parameter and the 'Redundant Proxy Mode' Web interface parameter cannot be set to

Simultaneous.

A. Configuring Phones in Server-Specific Deployments

400HD Series IP Phones

- 235 -

Figure 141: Signaling Protocol – SIP Proxy and Registrar – Secondary Proxy

2. Configure the parameters using table below as reference.

To configure using configuration file:

◼ Use the table below as reference.

Table 22: SIP Proxy and Registrar Parameters

Parameter

Description

Use SIP Proxy

[voip/signalling/sip/use_proxy]

Determines whether to use a SIP Proxy server.

Configure [1] Enable.

◼ [0] Disable (default)

◼ [1] Enable

Proxy IP Address or Host Name

[voip/signalling/sip/proxy_address]

Enter the IP address or host name (for example,

audiocodes.com) of the SIP proxy server. Default:

0.0.0.0

Proxy Port

[voip/signalling/sip/proxy_port]

The UDP or TCP port of the SIP proxy server. Range:

1024 to 65535. Default: 5060.

Enable Registrar Keep Alive

[voip/signalling/sip/registrar_ka/enabled]

Determines whether to use the registration keep-

alive mechanism based on SIP OPTION messages.

◼ [0] Disable (default)

◼ [1] Enable

Note:

◼ 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 keep-alive

messages are re-sent periodically.

Registrar Keep Alive Period

[voip/signalling/sip/registrar_ka/timeout]

Defines the registration keep-alive time interval (in

seconds) between Keep-Alive messages.

Range: 40 to 65536. Default: 60.

A. Configuring Phones in Server-Specific Deployments

400HD Series IP Phones

- 236 -

Parameter

Description

Maximum Number of Authentication Retries

[voip/signalling/sip/proxy_timeout]

The SIP proxy server registration timeout (in

seconds).

Range: 0 to 86400. Default: 3600.

Registration Failed Expires

[voip/signalling/sip/registration_failed_timeout]

If registration fails, this parameter determines the

interval between the register messages periodically

sent until successful registration.

Range: 1 to 86400. Default: 300.

Use SIP Registrar

[voip/signalling/sip/sip_registrar/enabled]

Determines whether the phone registers to a

separate SIP Registrar server.

◼ [0] Disable (default)

◼ [1] Enable

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.

Default: 0.0.0.0.

Registrar Port

[voip/signalling/sip/sip_registrar/port]

The UDP or TCP port of the Registrar server.

Range: 1024 to 65535. Default: 5060.

Use SIP Outbound Proxy

[voip/signalling/sip/sip_outbound_proxy/enabled]

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

Outbound Proxy IP Address or Host Name

[voip/signalling/sip/sip_outbound_proxy/addr]

The IP address of the outbound proxy (for example,

audiocodes.com; i.e., the same as that configured for

the 'Proxy IP Address or Host Name' parameter

above). If this parameter is set, all outgoing

messages (including Registration messages) are sent

to this Proxy according to the Stack behavior.

Default: 0.0.0.0

Outbound Proxy Port

[voip/signalling/sip/sip_outbound_proxy/port]

The port on which the outbound proxy listens.

Range: 1024 to 65535. Default: 5060.

A. Configuring Phones in Server-Specific Deployments

400HD Series IP Phones

- 237 -

Parameter

Description

Redundant Proxy Mode

[voip/signalling/sip/redundant_proxy/mode]

The call center's network administrator can select

either

◼ Disable -OR-

◼ Primary Fallback -OR-

◼ Simultaneous

For the dual-registration feature, select

Simultaneous; two proxies are registered

simultaneously so that at least one should be up and

running at any time, preventing the call center from

going down.

Note that when using the BroadSoft ACD in a SIP

Business Continuity deployment, this parameter

cannot be set to Simultaneous.

Secondary Proxy Address

[voip/signalling/sip/secondary_proxy/address]

Displayed only when Simultaneous is selected for

'Redundant Proxy Mode' (see previous parameter).

Define the IP address of the secondary proxy that

will be up simultaneously with the primary.

Secondary Proxy Port

[voip/signalling/sip/secondary_proxy/port]

Displayed only when Simultaneous is selected for

'Redundant Proxy Mode' (see the parameter before

the previous). Define the port of the secondary proxy

that will be up simultaneously with the primary.

A.3.5 Disabling the Web Interface

This feature lets the call center's network administrator block Web interface access to agents

employed in the call center.

To disable access using configuration file:

◼ Use the table below as reference.

Table 23: Disabling the Web Interface

Parameter

Description

[system/web/enabled]

Determines whether or not to enable access to

the phone's Web interface.

◼ [0] Disable (access prohibited)

◼ [1] Enable (default) (access enabled)

This can avoid a potential scenario in which agents

deliberately or by accident disrupt the operation

of the call center.

A. Configuring Phones in Server-Specific Deployments

400HD Series IP Phones

- 238 -

A.3.6 Forcing a Reboot on Provisioning

This feature lets the call center's network administrator configure a forced reboot on agents' phones

after provisioning.

To force a reboot on provisioning using configuration file:

◼ Use the table below as reference.

Table 24: Forcing a Reboot on Provisioning

Parameter

Description

[voip/services/notify/check_sync/force_reboot_enable

Determines whether or not to force a reboot on

provisioning.

◼ [0] Disable (default)

◼ [1] Enable

A.3.7 Provisioning using TFTP / FTP / HTTP / HTTPS in DHCP Options 66/67

This feature lets network administrators enable phones to be automatically provisioned when the

phones are plugged in, using TFTP / FTP / HTTP / HTTPS in DHCP Options 66/67.

A.3.8 Enabling Agents to Sign in with Phone Numbers

This feature lets the call center administrator power up all phones without setting a valid SIP account.

When an agent then wants to use their phone, they register to the network with their phone number.

To enable the feature using configuration file:

◼ Use the table below as reference.

Table 25: Enabling Agents to Sign in with Phone Numbers

Parameter

Description

[system/login_sk_before_signed_in]

Determines whether or not to enable agents sign

in with phone numbers.

◼ [0] Disable (default)

◼ [1] Enable

A. Configuring Phones in Server-Specific Deployments

400HD Series IP Phones

- 239 -

A.3.9 Locking Agents' Phones' Alphabetical Keys

Only applies to the 420HD and 405/405HD models.

This feature lets call center network administrators lock agents' phones' alphabetical (non-

numerical) keys so that only numerical keys are available to them. This feature provides call centers

the option to limit agents to work-specific tasks. The feature reduces private activity on the part of

agents. Agents cannot, for example, add contacts to a personal directory.

When this feature is enabled, agents can only use numbers. Only two menus are available in agents'

phone screens:

◼ Status

◼ Administration

To lock alphabetical keys using configuration file:

◼ Use the table below as reference.

Table 26: Locking Agents Phones Alphabetical Keys

Parameter

Description

[voip/block_non_numeric_key]

Determines whether or not to lock Agents'

phones' alphabetical keys and only allow Agents

to use numerical keys.

◼ [0] Disable (default)

◼ [1] Enable

A.3.10 Playing a Beep on an Incoming Call

This feature lets call center network administrators configure a beep to be played when a call comes

in if auto-answer is configured. The beep is played on both speaker and headset. Agents will know

from the beep that they have an incoming call in which to attend.

To configure the auto-answer feature, see Section 22.7.

To configure playing a beep using configuration file:

◼ Use the table below as reference.

Table 27: Playing a Beep on an Incoming Call

Parameter

Description

[voip/auto_answer/headset_beep/enabled]

Determines whether or not to play a beep on an

incoming call, on the headset.

◼ [0] Disable (default)

◼ [1] Enable

[voip/auto_answer/speakerphone_beep/enabled]

Determines whether or not to play a beep on an

incoming call, on the speaker.

◼ [0] Disable (default)

◼ [1] Enable

A. Configuring Phones in Server-Specific Deployments

400HD Series IP Phones

- 240 -

A.3.11 Enabling Proactive Mute

This feature lets call center network administrators enable a proactive mute when calls come in so

that when they come in, callers cannot hear the agents until the agents unmute by pressing the Mute

button. The feature can protect call centers from agent conduct that might be offensive to callers.

Agents may for example pass an offensive remark to one another about a caller whose call is coming

in, without realizing the caller can hear.

To enable proactive mute using configuration file:

◼ Use the table below as reference.

Table 28: Enabling Proactive Mute

Parameter

Description

[voip/proactive_mute/enabled]

Determines whether or not to enable proactive

mute.

◼ [0] Disable (default)

◼ [1] Enable

A.3.12 Configuring Automatic Answer

You can configure a supplementary service on the phones called Automatic Answer. The feature is

configured using the configuration file. Use the table below as a configuration reference.

Table 29: Automatic Answer

Parameter

Description

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.

A. Configuring Phones in Server-Specific Deployments

400HD Series IP Phones

- 241 -

A.3.13 Regulating the 'Logged out' Message

This feature lets call center network administrators enable/limit the length of time the 'Logged out'

message is displayed in the phone's idle screen after agents log out.

When agents log out, the 'Logged out' message will only be displayed in the phone's idle screen for

the length of time, in seconds, configured by the call center network administrator. After the

configured time lapses, the message disappears from the screen.

Administrators can also disable the feature.

To regulate the feature using the configuration file:

1. Open the Configuration File page in the Web interface (Management tab > Manual Update >

Configuration File).

2. Configure the 'logged_out_message_timer' parameter using the table below as reference.

Table 30: Regulating the 'Logged out' Message

Parameter

Description

[voip/services/ACD/logged_out_message_timer]

◼ [-1] Disabled (default)

◼ [0] No 'Logged out' message is displayed.

◼ [>1] The time, in seconds, that lapses before the

'Logged out' message, displayed in the phone's

idle screen after an agent logs out, disappears.

This value also enables the feature.

A.3.14 3PCC (Third Party Call Control)

The 3PCC feature lets an agent control their phone remotely from a computer application.

3PCC always supports the following functions:

◼ MakeCall (call initiation/setup)

◼ Release

◼ Hold

◼ Retrieve

◼ Transfer

◼ Consult

◼ Conference

◼ DTMF

To configure 3PCC using configuration file:

◼ Use the table below as reference.

Table 31: 3PCC Parameters

Parameter

Description

[voip/talk_event/enabled]

◼ [0] Disable (default)

◼ [1] Enable

voip/talk_event must be enabled for 3PCC to function.

A. Configuring Phones in Server-Specific Deployments

400HD Series IP Phones

- 242 -

A.3.14.1 Enabling 3PCC Calls

This feature complies with the RFC 3725 standard for 3PCC in SIP for 'Black Holed' and 'Non SDP'.

See the RFC for detailed information.

To enable 3PCC calls using the configuration file:

1. Open the Configuration File page in the Web interface (Management tab > Manual Update >

Configuration File).

2. Configure the capability using the table below as reference.

Table 32: Enabling 3PCC Calls

Parameter

Description

[3PCC/make_call/enabled]

◼ [0] = Disable (default)

◼ [1] = Enable

A. Configuring Phones in Server-Specific Deployments

400HD Series IP Phones

- 243 -

A.3.15 Disabling Handset Mode

Administrators can disable handset mode using the configuration file parameter

'voip/handset_mode/enabled' whose default is enabled.

Some call centers don't want agents to work with any device other than headsets. In this case, their

administrators can change the parameter default to disabled.

Some call centers don't even connect the handsets to the phones. In this case, even though

the handsets are not physically connected to the phones, administrators should disable the

new parameter.

To disable handset mode:

◼ Use the table below as reference.

Table 33: BroadSoft Server - Shared Call Appearance Add

Parameter

Description

[voip/handset_mode/enabled]

When disabled [0], the handset becomes

unavailable. Configure either:

◼ [1] = Enabled (default)

◼ [0] = Disabled

A.3.16 Changing Phone Screen Backlight Timeout

Call center administrators can use a configuration file parameter to change phone screens’ backlight

timeout. A phone screen's backlight timeout can also be changed on the phone by the user, unless

the call center administrator has disabled the possibility with the configuration file parameter

personal_settings/menu/backlight_timeout/enabled (see Section 26 for more information about

disabling phone hard keys and softkeys).

To change the backlight timeout:

◼ Use the table below as reference.

Table 34: Backlight Timeout

Parameter

Description

[system/lcd/backlight/timeout]

Range: 0-6

0 = Always On

1 = 10 seconds (default)

2 = 20 seconds

3 = 30 seconds

4 = 40 seconds

5 = 50 seconds

6 = 60 seconds

A. Configuring Phones in Server-Specific Deployments

400HD Series IP Phones

- 244 -

A.3.17 Displaying a Message on Agents' Phones

Call center administrators can use a configuration file parameter to define a message that will be

displayed on agents' phones, for example: 'Reminder: Your calls might be recorded'. Agents will then

see this message, together with the date (in month/day format), displayed in their screens when

their phones are in idle state.

To display a message on Agents' phones:

◼ Use the table below as reference.

Table 35: Displaying a Message on Agents' Phones

Parameter

Description

[system/display/message_on_screen]

Defines a message that will be displayed in agents'

phone screens together with the date (in month/day

format)

A.3.18 Configuring a Redundant (Backup) Genesys Server

A phone can be registered on two Genesys servers simultaneously, to provide immediate backup.

The feature enables quick transition to the redundant backup server; redundant proxy usage is

available all the time. The phone is registered on the redundant server in the same way as it is

registered on the primary server.

To register a phone on the redundant server:

1. Open the Signaling Protocol page (Configuration tab > Voice Over IP menu > Signaling

Protocols).

Figure 142: Registering a Phone on the Redundant Genesys Server

2. Scroll down to the ‘Redundant Proxy Mode’ and 'Secondary Proxy Address' parameters.

Configure them using the table below as reference, and then click Submit.

Table 36: Redundant Genesys Server - Parameters

Parameter

Description

Redundant Proxy Mode

From the dropdown, select Simultaneously; you're now in Dual Registration

mode; when in this mode, the value of the parameter 'Retransmission Timer

T1' is taken from the configuration file parameter

'voip/signalling/sip/redundant_proxy/dual_reg/t1' rather than from the

'Retransmission Timer T1' parameter (see Table 25 for more information about

this parameter).

Secondary Proxy Address

Provide the IP address of the redundant proxy; the phone then registers on

both servers from the outset, instead of transitioning from one to another.

A. Configuring Phones in Server-Specific Deployments

400HD Series IP Phones

- 245 -

A.3.18.1 Configuring Retransmission Timer T1 Using the Configuration File

Configuration of the T1 retransmission timer is only relevant when in dual registration mode, i.e.,

after configuring a redundant Genesys server, as shown in the previous section.

Configuring Retransmission Timer T1 using the configuration file:

3. Open the Configuration File page in the Web interface (Management tab > Manual Update >

Configuration File).

4. Configure the capability using the table below as reference.

Table 37: Retransmission Timer T1 - Parameter

Parameter

Description

[voip/signalling/sip/redundant_proxy/dual_reg/t1]

Only relevant if dual registration / redundancy server

is configured. Allows quicker retransmission of SIP

messages than the Web interface parameter

'Transmission Timer T1' and overrides it if configured

(see Table 25 for more information).

Default: 20 milliseconds. Range: 20-200.

A.4 Genband: KBS Softswitch Solution

Network administrators can configure the following Kandy Business Solutions (KBS) softswitch

solution features in the KBS Portal:

◼ Shared Line Appearance (SLA)

◼ Call pickup

◼ Busy Lamp Fields (BLFs)

◼ Remote conference

A. Configuring Phones in Server-Specific Deployments

400HD Series IP Phones

- 246 -

A.4.1 Configuring Shared Line Appearance

When a call comes in on a shared line, all phones ring in the SLA group. When answered by someone

in the group, all other users in the group can see there’s an active call on the line. When there’s an

active call on the line, no other phone can initiate a call on the line. When a call is put on hold, the

caller hears music; other users in the group can see the call is on hold (color indication or flashing).

When a call is on hold, the same phone or another phone can retrieve the call.

To configure an SLA group:

1. Open the Genband portal in your web browser and from the ‘Provision’ menu, select Call

Answer Groups.

Figure 143: Call Answer Groups

2. In the Call Answer Groups screen that opens, click Add Group.

Figure 144: Call Answer Groups - Add Group

3. From the ‘Type’ dropdown, select the option Shared Line Appearance.

A. Configuring Phones in Server-Specific Deployments

400HD Series IP Phones

- 247 -

Figure 145: Call Answer Group - Type

4. In the ‘Group Name’ field, enter the name of a group, for example, SLA_Group_1, as shown in

the figure below.

5. Under the Group Members section of the screen, select the members of the group. You

cannot select members highlighted yellow; they’ve already been assigned to groups. Select

from members that aren’t highlighted yellow. In the example shown in the figure below,

Regular_11, Regular_11 and Regular_11 are selected.

6. Click in the field ‘Primary Account’ and in the Primary Account section of the screen that

opens, select the primary account. In the example shown in the figure below, AUDC (first

name) Regular_6 (last name) is selected; the ‘Primary Account’ field is immediately populated

with the selection.

Figure 146: Call Answer Group – Group Name

7. Click Save Group; the SLA group is created.

A. Configuring Phones in Server-Specific Deployments

400HD Series IP Phones

- 248 -

A.4.2 Configuring Call Pickup

When configuring a call pickup group, basic configuration options determine:

◼ the numbers that route into a call pickup group

◼ whether or not vertical service code (VSC) dialing can be used by group members

◼ group members

Advanced configuration options allow you to specify:

◼ the maximum number of group members

◼ the maximum number of call queues

◼ whether or not SIP dialog event package subscriptions are enabled

To configure a Call Pickup group:

1. Open the Genband portal in your web browser and from the ‘Provision’ menu, select Call

Answer Groups.

Figure 147: Call Answer Groups

A. Configuring Phones in Server-Specific Deployments

400HD Series IP Phones

- 249 -

2. Click Add Group.

Figure 148: Call Answer Group

3. From the ‘Type’ dropdown, select Call Pickup.

4. Enter a Group Name, for example, Call_Pickup_GRP, as shown in the preceding figure.

5. Click the ‘Number’ field; all available numbers are displayed under ‘Numbers’, as shown in the

preceding figure.

6. Select the phone number to route into the group, and then click Accept Number; the number

is displayed in the ‘Number’ field, as shown in the figure below.

Figure 149: Number

7. From the ‘Alias Status’ dropdown, select Enabled to enable VSC dialing, as shown in the

preceding figure.

A. Configuring Phones in Server-Specific Deployments

400HD Series IP Phones

- 250 -

8. Click Manage Advanced to configure advanced options.

Figure 150: Advanced

9. Enter a number in the ‘Max Group Size’ field to specify the maximum number of members

allowed in the group.

10. Enter a number in the ‘Max Queue Size’ field to specify the maximum number of calls allowed

in the queue for the group.

11. From the ‘Active Subscriptions’ dropdown, select Enabled to enable SIP dialog event package

subscriptions.

12. To add members to the group, select the checkbox beside the name of each person to be

included in the GROUP MEMBERS list.

13. Click Save Group.

A. Configuring Phones in Server-Specific Deployments

400HD Series IP Phones

- 251 -

A.4.3 Setting up a Remote Conference

More than three participants can be added to a remote conference call. By contrast, a 'local'

conference only supports a maximum of three. The feature must be enabled on Genband’s server.

For detailed information about the feature’s capabilities, see RFC 4579, Session Initiation Protocol

(SIP) - Call Control - Conferencing for User Agents. IGS-1882

To set up the remote conference feature using the Web interface:

1. In the Web interface, open the Services page (Configuration > Voice Over IP > Services).

Figure 151: Services - Conference

2. From the 'Type' dropdown under 'Application Server', select GENBAND, as shown in the

preceding figure. The feature must be enabled on Genband’s server for it to be supported.

1. From the 'Mode' dropdown under 'Conference', select Remote, as shown in the preceding

figure.

2. In the 'Remote Conference Media Server' field, enter the address of the server hosting the

remote conference, for example, conference@SIP proxy address of Genband’s server, as

shown in the preceding figure.

3. Click Submit; the user can now establish a remote conference from the phone (see the

phone's User's Manual for detailed information).

To set up the remote conference feature using the configuration file:

◼ Use the table below as reference.

Table 38: Remote Conference Parameters

Parameter

Description

[voip/services/application_server_type]

Set to GENBAND.

[voip/services/conference/conf_ms_addr]

Set the address of the server hosting the remote

conference: conference@SIP proxy address of

Genband’s server

[voip/services/conference/mode]

Set the mode to REMOTE.

A. Configuring Phones in Server-Specific Deployments

400HD Series IP Phones

- 252 -

A.4.4 Configuring Parameters Mandatory for Phones in a Genband

Environment

Network administrators must configure the parameters shown in this section which are mandatory

for phones deployed in a Genband environment.

To configure parameters mandatory for phones in a Genband environment, using the

configuration file:

◼ Use the table below as reference.

Table 39: Parameters Mandatory for Phones in a Genband Environment

Parameter

Description

[voip/services/application_server_type]

Configure GENBAND

[voip/services/SLA/type]

Configure GENBAND_SCA

[voip/services/busy_lamp_field/enabled]

Configure 1 (enabled)

[voip/signalling/sip/transport_protocol]

Configure TCP

[voip/signalling/sip/registration_failed_timeout]

Configure 30

[voip/signalling/sip/sip_outbound_proxy/enabled]

Configure 1 (enabled)

[voip/signalling/sip/use_proxy]

Configure 1 (enabled)

[voip/signalling/sip/proxy_address]

Configure %SIP_Proxy_Address%

[voip/signalling/sip/sip_outbound_proxy/addr]

Configure %Outbound_Proxy_Address%

[voip/cwrr_enabled]

Configure 0 (disabled)

[voip/lhcwrr_enabled]

Configure 0 (disabled)

[voip/media/allow_multiple_rtp]

Configure 1 (enabled)

[voip/distribute_multiple_line_registrations/enable]

Configure 1 (enabled)

voip/services/sla/barging/enable

Configure 1 (default=0)

voip/sla/group/0/description

Configure Group1

voip/sla/group/0/enabled

Configure 1 (default=0)

voip/sla/group/1/description

Configure Group2

voip/sla/group/1/enabled

Configure 1 (default=0)

BLF service must be activated for normal operation of BLF, SLA, Call Pick Up and Call Park

services on phones deployed in a Genband environment, as shown below.

Figure 152: Services - BLF Support

B. Configuring Automatic Call Distribution (ACD)

400HD Series IP Phones

- 253 -

B Configuring Automatic Call Distribution (ACD)

• AudioCodes' IP Phones seamlessly interwork with Genesys SIP Server to support ACD

functionality. The phones support two different ACD methods: a legacy method

referred to as 'Genesys' in the Web interface, and an enhanced method based on

BroadSoft’s ACD capabilities, referred to as 'BROADSOFT' in the Web interface.

• For optimal ACD functionality with Genesys SIP Server, the BroadSoft-based ACD

method must be chosen.

This appendix shows how to enable the ACD (Automatic Call Distribution) feature on the phone using

either the Web interface or the configuration file. The feature automatically distributes incoming

calls to agents' phones on the basis of agent availability and unavailability.

In contact centers, ACD is a key feature of CTI (Computer Telephony Integration). The feature

automatically distributes incoming calls to a specific group of terminals that contact center agents

use. Most ACD functionality is the SIP server's responsibility; however, users must inform the Call

Center SIP server on the following events:

◼ Whenever the call center representative logs in or out on the phone. This information is

included in a SIP SUBSCRIBE message.

◼ Whenever the call center representative indicates whether they are ready or not to take a

call. When the BroadSoft server is configured, the user can also specify the reason for their

unavailability e.g., Lunch break. All this information is included in a SIP NOTIFY message.

◼ Whenever the user is busy with After Call Work (ACW) (only relevant when a BroadSoft SIP

server is configured). This information is included in a SIP NOTIFY message.

All the above actions can be performed on the phone (see the phone's User's Manual). The Call

Center SIP server then uses the above presence information to automatically distribute calls

between agents based on their availability.

ACD systems allow companies that handle a large number of incoming phone calls to direct the

callers to a company employee who is able to talk at the earliest opportunity.

The feature is typically implemented in contact centers encountering large numbers of incoming

customer calls that must be distributed to available agents to provide immediate support to callers.

The feature automatically directs incoming calls to agents working in the contact center whose

presence status is 'Ready' rather than not ready. The feature's main benefit is to reduce the time

customers are kept waiting and thereby improve service.

AudioCodes' IP phones seamlessly interwork with Genesys' SIP server to support the ACD feature.

Once an agent signs in on their phone to ACD, their status is set to 'Ready' and synchronized with

Genesys' Server. Incoming calls are directed to an agent whenever their status becomes 'Ready'.

• This feature applies to the 405/405HD, 420HD, 430HD and 440HD phone models.

• The Web interface screenshots shown in this section are of the 420HD model. They're

identical to those of the 405/405HD, 430HD and 440HD phone models.

B. Configuring Automatic Call Distribution (ACD)

400HD Series IP Phones

- 254 -

To configure the ACD server using the Web interface:

1. In the Web interface, access the ACD page (Configuration tab > Advanced Applications > ACD):

Figure 153: ACD

For optimal ACD functionality with Genesys, select the BROADSOFT option from the 'Server

Type' dropdown.

Figure 154: ACD – Unavailable Reason Code

2. Configure the parameters using the table below as reference.

B. Configuring Automatic Call Distribution (ACD)

400HD Series IP Phones

- 255 -

To configure the ACD server using the configuration file:

◼ Define a path and configure the parameters using the table below as reference.

Table 40: ACD Parameters

Parameter

Description

Active

[voip/services/ACD/enabled]

Enable/Disable the status of the ACD feature for

combined ACD/Hoteling

◼ [0] Disable (default)

◼ [1] Enable

Server Type

[voip/services/ACD/server_type]

From the 'Server Type' drop-down list, select GENESYS

or BROADSOFT.

Select the BROADSOFT option for optimal ACD

functionality with Genesys.

When you select BROADSOFT, after logging into the ACD

server, the ACW (After Call Work) softkey will be

displayed on the phone and the Missed softkey

displayed in the command menu along with the

FORWARD and DND options.

Note that administrators can optionally hide the ACW

softkey (see the next parameter).

[voip/services/ACD/show_acw_softkey/enabled]

Allows administrators to hide the ACW softkey. After

logging into the call center's Automatic Call Distributor

(ACD) server, the ACW softkey is by default displayed on

the phone [1]. Administrators can change this default

and hide the softkey [0].

◼ [0] Disable

◼ [1] Enable (default)

The default softkey layout is ACW, READY/Not Ready

(when the user is logged on), Login/Logout and

Command menu items including Missed and DND.

When the ACW softkey is disabled, the display stays the

same but without the ACW softkey.

[system/dnd/show_softkey]

Removes the DND item from the Call Menu where it's

displayed by default when ACD is enabled.

◼ [0] Removes DND from the Call Menu

◼ [1] Displays DND in the Call Menu (default)

[system/forward/show_softkey]

Removes the Forward item from the Call Menu where

it's displayed by default when ACD is enabled.

◼ [0] Removes Forward from the Call Menu

◼ [1] Displays Forward in the Call Menu (default)

Use Sip Server As ACD Server

[voip/services/ACD/server_use_sip_server]

From the drop-down, choose Enable.

◼ [0] Disable

◼ [1] Enable

Expire Time

[voip/services/ACD/expire_time]

The server registration timeout, in seconds.

Range: 0 to 86400. Default: 3600.

Server Address

[voip/services/ACD/server_address]

Displayed only when 'Use SIP Server As ACD Server' is

set to Disable (see previous). Defines the IP address of

the ACD server. Default: 0.0.0.0

B. Configuring Automatic Call Distribution (ACD)

400HD Series IP Phones

- 256 -

Parameter

Description

Server Port

[voip/services/ACD/server_port]

Displayed only when 'Use SIP Server As ACD Server' is

set to Disable (see previous). Defines the port of the

ACD server. Default: 80

User Name

[system/user_name]

Enter the agent's User Name. The agent will use this

name when logging in to ACD to define or change

availability status.

Password

[system/password]

Enter a password if necessary.

State After Login1

[voip/services/ACD/state_after_login]

The call center's network administrator can select:

◼ Ready

◼ Not Ready (default)

◼ Not Set

If set to Ready, each phone in the call center will

automatically be set to a state of readiness to take

incoming calls immediately after the call center's agents

If set to Not Ready, agents can log in and then manually

configure their readiness status on the phone's screen,

giving them time to perform personal tasks before

beginning work.

If set to Not Set, the status of the phone after login will

be controlled by the server. For example, if the server is

set to be in 'Ready' status following login, the phone will

be in 'Ready' status when the user logs in.

First Notify Close Enabled

[voip/services/ACD/first_notify_close/enabled]

◼ [0] When an agent logs in, the ACD server is notified

that the agent is Ready (available) to take calls.

◼ [1] (Default) When an agent logs in, the ACD server

is notified that the agent is Not Ready (unavailable)

to take calls. This gives agents time to get organized.

[voip/services/ACD/logged_out_message_timer]

For detailed information, see Appendix A.3.13.

Unavailable Reason Code

[voip/services/ACD/unavailable_reason/0-

9/code]

Up to 10 reasons can be defined (0-9).

Specifies the code that is sent in the SIP NOTIFY

message to the Call Center SIP server to indicate the

specific reason for the Call Center representative’s

unavailability.

This parameter is relevant when the 'Server Type'

parameter (see above) is BroadSoft.

Reason Name

[voip/services/ACD/unavailable_reason/0-

9/name]

Describes the unavailability reason code (configured

above). For example, 'Lunch'.

This parameter is relevant when the 'Server Type'

parameter (see above) is BroadSoft.

[voip/services/enhanced_ACD/enabled]

ACD feature status (ACD only).

◼ [0] Disable

◼ [1] Enable

This parameter is only relevant to Genesys Call Centers.

B. Configuring Automatic Call Distribution (ACD)

400HD Series IP Phones

- 257 -

B.1 Softkey Display and Command Menu Options

The following tables show the different softkey display states and command menu options that are

available according to the user's login state and the configured SIP server.

Table 41: BroadSoft-Softkey Display States and Command Menu Options

State

Softkey #0

Softkey #1

Softkey #2

Softkey #3

Idle

ACD Disabled