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Audiocodes 310 HD Admin Guide

The document provides an administrator's manual for the AudioCodes 310HD IP phone. It contains information on configuring and managing the phone through its LCD interface and web-based interface. The document includes sections on network settings, SIP accounts, call progress tones, speed dials, firmware updates, and other phone parameters and features. Configuration options are explained for areas like signaling protocols, codecs, media streaming, dialing, services, and regional settings. The manual also describes creating configuration files and directories to customize the phone's operation.

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0% found this document useful (0 votes)
118 views

Audiocodes 310 HD Admin Guide

The document provides an administrator's manual for the AudioCodes 310HD IP phone. It contains information on configuring and managing the phone through its LCD interface and web-based interface. The document includes sections on network settings, SIP accounts, call progress tones, speed dials, firmware updates, and other phone parameters and features. Configuration options are explained for areas like signaling protocols, codecs, media streaming, dialing, services, and regional settings. The manual also describes creating configuration files and directories to customize the phone's operation.

Uploaded by

Alpeshlad
Copyright
© © All Rights Reserved
We take content rights seriously. If you suspect this is your content, claim it here.
Available Formats
Download as PDF, TXT or read online on Scribd
You are on page 1/ 90

AudioCodes 300HD Series of High Definition IP Phones

HD VoIP 310HD IP Phone

Administrator’s Manual
310HD IP Phone
Version 1.4.0
Administrator's Manual Contents

Table of Contents
1 Introduction ................................................................................................................. 9
2 LCD-Based Management .......................................................................................... 11
2.1 Accessing the Administration Submenus ........................................................................11
2.2 Configuring Network Settings ..........................................................................................12
2.3 Configuring SIP Accounts ................................................................................................13
2.4 Restoring Defaults ...........................................................................................................14
2.5 Restarting Phone .............................................................................................................15
2.6 Viewing Network Status ...................................................................................................15
2.7 Viewing Phone Model and Firmware Version ..................................................................16
3 Web-based Management .......................................................................................... 17
3.1 Accessing the Web Interface ...........................................................................................17
3.2 Getting Started with the Web ...........................................................................................18
3.3 Quick Setup .....................................................................................................................19
3.4 Changing LCD Display Language ...................................................................................20
3.5 Defining Speed Dials .......................................................................................................20
3.6 Regional Call Progress Tones and Ring Tones ...............................................................22
3.6.1 Selecting the Region for Call Progress Tones ...................................................................22
3.6.2 Uploading Ring Tones .......................................................................................................23
3.7 Configuring LAN Interface ...............................................................................................24
3.8 VoIP Settings ...................................................................................................................25
3.8.1 Configuring Signaling Protocol ..........................................................................................25
3.8.2 Configuring Dialing.............................................................................................................28
3.8.3 Configuring Media Streaming ............................................................................................30
3.8.4 Configuring Voice...............................................................................................................31
3.8.5 Configuring Line ................................................................................................................. 32
3.8.6 Configuring Services ..........................................................................................................33
3.8.7 Configuring Volume ...........................................................................................................35
3.9 Configuring Date and Time ..............................................................................................37
3.10 Firmware and Configuration Management ......................................................................38
3.10.1 Configuring Automatic Update of Firmware and Configuration File ..................................38
3.10.2 Updating (Manually) Configuration File .............................................................................40
3.10.3 Recovering Firmware .........................................................................................................41
3.11 Changing Login Username and Password ......................................................................42
3.12 Restoring Phone Defaults ................................................................................................43
3.13 Restarting Phone .............................................................................................................43
3.14 Configuring TR-069 Remote Management ......................................................................44
3.15 Viewing Status Information ..............................................................................................45
3.15.1 Viewing LAN Information ...................................................................................................45
3.15.2 Viewing Call History ........................................................................................................... 46
3.15.3 Viewing Phone’s Version Number .....................................................................................46
3.16 Diagnostics ......................................................................................................................47
3.16.1 Configuring System Logging..............................................................................................47
3.16.2 Configuring Packet Recording ...........................................................................................48
4 Configuration File Parameters ................................................................................. 49
4.1 Overview ..........................................................................................................................49
4.1.1 File Syntax ......................................................................................................................... 49
4.1.2 Linking Additional Files using “Include” .............................................................................49

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310HD IP Phone

4.2 Configuration File Parameters .........................................................................................51


4.2.1 System Parameters ...........................................................................................................51
4.2.2 Network Parameters ..........................................................................................................54
4.2.3 Provisioning Parameters ....................................................................................................56
4.2.4 TR-069 Management Parameters .....................................................................................59
4.2.5 VoIP Parameters................................................................................................................60
4.2.5.1 Line Parameters .................................................................................................60
4.2.5.2 Gain Parameters ................................................................................................61
4.2.5.3 Codec Parameters..............................................................................................64
4.2.5.4 Media Streaming Parameters.............................................................................65
4.2.5.5 Dialing Parameters .............................................................................................66
4.2.5.6 SIP Signaling Parameters ..................................................................................67
4.2.5.7 Voice ...................................................................................................................74
4.2.5.8 Supplementary Services Parameters .................................................................74
4.2.6 Regional Settings Parameters ...........................................................................................76
4.2.7 Packet Recording (Debugging) Parameters ......................................................................78
4.2.8 LCD Display Parameters ...................................................................................................78
4.3 Creating a Corporate Directory File .................................................................................79
4.4 Creating a Speed Dial File ...............................................................................................80
5 Built-in Test Plan ....................................................................................................... 81
5.1 Test Preparation ..............................................................................................................81
5.2 Keypad Test.....................................................................................................................82
5.3 LED Test ..........................................................................................................................82
5.4 Handset Test ...................................................................................................................83
5.5 Headset Test ...................................................................................................................84
5.6 Speaker Test ...................................................................................................................85
5.7 MAC Address and Serial Number Verification Test .........................................................86
6 Specifications ............................................................................................................ 87

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Administrator's Manual Contents

List of Figures
Figure 3-1: Web Login Window ........................................................................................................................ 17
Figure 3-2: Web Interface Areas ...................................................................................................................... 18
Figure 3-3: Quick Setup Page .......................................................................................................................... 19
Figure 3-4: Language Page .............................................................................................................................. 20
Figure 3-5: Speed Dial Page ............................................................................................................................ 20
Figure 3-6: Tones Page .................................................................................................................................... 22
Figure 3-7: Tones Page .................................................................................................................................... 23
Figure 3-8: LAN Settings Page ......................................................................................................................... 24
Figure 3-9: Signaling Protocol Page ................................................................................................................. 25
Figure 3-10: Dialing Page ................................................................................................................................. 28
Figure 3-11: Media Streaming Page.................................................................................................................30
Figure 3-12: Voice Page ................................................................................................................................... 31
Figure 3-13: Voice Page ................................................................................................................................... 32
Figure 3-14: Services Page .............................................................................................................................. 33
Figure 3-15: Volume Settings Page..................................................................................................................35
Figure 3-16: Date and Time Page .................................................................................................................... 37
Figure 3-17: Automatic Update Page ............................................................................................................... 39
Figure 3-18: Configuration File Page................................................................................................................40
Figure 3-19: Load new Configuration File ........................................................................................................ 40
Figure 3-20: System Authorization Page .......................................................................................................... 42
Figure 3-21: Restore Defaults Page ................................................................................................................. 43
Figure 3-22: Submit Confirmation Box ............................................................................................................. 43
Figure 3-23: Restart System Page ................................................................................................................... 43
Figure 3-24: Confirmation Box .......................................................................................................................... 43
Figure 3-25: TR-069 Page ................................................................................................................................ 44
Figure 3-26: Network Status Page ................................................................................................................... 45
Figure 3-27: Call History Page ......................................................................................................................... 46
Figure 3-28: System Information Page ............................................................................................................. 46
Figure 3-29: System Logging Page .................................................................................................................. 47
Figure 3-30: Recording Page ........................................................................................................................... 48
Figure 5-1: Testing Keypad .............................................................................................................................. 82
Figure 5-2: Testing Keypad – Characters Disappear ....................................................................................... 82
Figure 5-3: Successful Completion of Keypad Test ......................................................................................... 82
Figure 5-4: Off hook Handset Test ................................................................................................................... 83
Figure 5-5: Recording Test ............................................................................................................................... 83
Figure 5-6: Playing Test.................................................................................................................................... 83
Figure 5-7: Headset Test .................................................................................................................................. 84
Figure 5-8: Headset Recording Test ................................................................................................................ 84
Figure 5-9: Headset Playing Test ..................................................................................................................... 84
Figure 5-10: Speaker Test ................................................................................................................................ 85
Figure 5-11: Speaker Recording Test .............................................................................................................. 85
Figure 5-12: Speaker Playing Test ................................................................................................................... 85
Figure 5-13: MAC Address and Serial Number ................................................................................................ 86

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310HD IP Phone

List of Tables
Table 3-1: Language Parameter Description ................................................................................................... 20
Table 3-2: Region CPT Parameter Description ................................................................................................ 22
Table 3-3: Ring Tone Parameter Description ................................................................................................... 23
Table 3-4: LAN Parameters Description ........................................................................................................... 24
Table 3-5: SIP Parameters Description ............................................................................................................ 26
Table 3-6: Dialing Parameters Description ....................................................................................................... 28
Table 3-7: Media Streaming Parameters Description ...................................................................................... 30
Table 3-8: Voice Parameters Description ......................................................................................................... 31
Table 3-9: Line Parameters Description ........................................................................................................... 32
Table 3-10: Supplementary Services Parameters Description ........................................................................34
Table 3-11: Volume Parameters Description ................................................................................................... 35
Table 3-12: Date and Time Parameters Description ........................................................................................ 37
Table 3-13: Automatic Provisioning Parameters Description ........................................................................... 39
Table 3-14: Login Username and Password Parameters Description ............................................................. 42
Table 3-15: TR-069 Parameters Description .................................................................................................... 44
Table 3-16: System Logging Parameters Description ...................................................................................... 47
Table 3-17: Packet Recording Parameters Description ................................................................................... 48
Table 4-1: System Configuration File Parameters ........................................................................................... 51
Table 4-2: Network Configuration File Parameters .......................................................................................... 54
Table 4-3: Provisioning Configuration File Parameters .................................................................................... 56
Table 4-4: TR-069 Management Configuration File Parameters .....................................................................59
Table 4-5: Line Configuration File Parameters ................................................................................................. 60
Table 4-6: Gain Configuration File Parameters ................................................................................................ 61
Table 4-7: Codec Configuration File Parameters ............................................................................................. 64
Table 4-8: Media Streaming Configuration File Parameters ............................................................................65
Table 4-9: Dialing Configuration File Parameters ............................................................................................ 66
Table 4-10: SIP Configuration File Parameters ................................................................................................ 67
Table 4-11: Voice Configuration File Parameters ............................................................................................ 74
Table 4-12: Supplementary Services Configuration File Parameters ..............................................................74
Table 4-13: Regional Configuration File Parameters ....................................................................................... 76
Table 4-14: Packet Recording (Debugging) Configuration File Parameters ....................................................78
Table 4-15: LCD Display Configuration File Parameters ................................................................................. 78
Table 6-1: 310HD IP Phone Specifications ...................................................................................................... 87

Administrator's Manual 6 Document #: LTRT-09905


Administrator's Manual Notices

Notice
This manual provides the system administrator a description for setting up and configuring the 310HD
IP Phone.

Information contained in this document is believed to be accurate and reliable at the time of printing.
However, due to ongoing product improvements and revisions, AudioCodes cannot guarantee
accuracy of printed material after the Date Published nor can it accept responsibility for errors or
omissions. Updates to this document and other documents can be viewed by registered customers at
http://www.audiocodes.com/downloads.

© Copyright 2010 AudioCodes Ltd. All rights reserved.


This document is subject to change without notice.

Date Published: July-27-2010

Trademarks
AudioCodes, AC, AudioCoded, Ardito, CTI2, CTI², CTI Squared, HD VoIP, HD VoIP Sounds
Better, InTouch, IPmedia, Mediant, MediaPack, NetCoder, Netrake, Nuera, Open Solutions
Network, OSN, Stretto, TrunkPack, VMAS, VoicePacketizer, VoIPerfect, VoIPerfectHD, What’s
Inside Matters, Your Gateway To VoIP and 3GX are trademarks or registered trademarks of
AudioCodes Limited. All other products or trademarks are property of their respective owners.

WEEE EU Directive
Pursuant to the WEEE EU Directive, electronic and electrical waste must not be disposed of with
unsorted waste. Please contact your local recycling authority for disposal of this product.

Customer Support
Customer technical support and service are provided by AudioCodes’ Distributors, Partners, and
Resellers from whom the product was purchased. For Customer support for products purchased
directly from AudioCodes, contact [email protected].

Abbreviations and Terminology


Each abbreviation, unless widely used, is spelled out in full when first used, and only Industry
standard terms are used throughout this manual. The symbol 0x indicates hexadecimal notation.

Version 1.4 7 July 2010


310HD IP Phone

Related Documentation

Document Name

300HD IP Phone Release Notes


310HD IP Phone Quick Guide
310HD IP Phone User’s Manual

Regulatory Information
The Compliancy and Regulatory Information can be viewed at
http://www.audiocodes.com/library.

Administrator's Manual 8 Document #: LTRT-09905


Administrator's Manual 1. Introduction

1 Introduction
This manual is intended for the system administrator who is responsible for setting up and
configuring the 310HD IP phone.
AudioCodes 310HD IP Phone is based on AudioCodes proprietary High Definition (HD) voice
technology, providing clarity and a rich audio experience in Voice-over-IP (VoIP) calls. The
310HD IP Phone is a fully-featured telephone that provides 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 310HD IP Phone offers a wide variety of management and configuration tools:
„ Phone’s LCD display user interface - easy-to-use, menu-driven display screen, providing
basic phone configuration and status capabilities
„ Embedded Web server - provides a user-friendly Web interface that runs on a Web browser
(Microsoft® Internet Explorer is the recommended browser).
„ Configuration file - text-based file (created using any plain text editor such as Microsoft’s
Notepad) containing configuration parameters and which is loaded to the phone using the
Web interface or a TFTP, FTP, HTTP or HTTPS server.
„ TR-069 for remote configuration and management
„ CLI over Telnet
For a detailed description on hardware installation and for operating the phone’s call features,
refer to the User’s Manual.

Version 1.4 9 July 2010


310HD IP Phone

Reader’s Notes

Administrator's Manual 10 Document #: LTRT-09905


Administrator's Manual 2. LCD-Based Management

2 LCD-Based Management
The IP phone provides a liquid crystal display (LCD) based screen, offering an intuitive, menu-
driven interface for configuring the phone.
This chapter provides step-by-step procedures for configuring the following administration
settings using the phone’s LCD interface:
„ Network settings (refer to Section 2.2 on page 12)
„ SIP accounts (refer to Section 2.3 on page 13)
„ Restoring factory defaults (refer to Section 2.4 on page 14)
„ Restarting the phone (refer to Section 2.5 on page 15)
These administration tasks are performed in the phone’s Administration menu (refer to Section
2.1).
This chapter also describes how to view the phone’s current network settings (refer to Section 2.6
on page 15) and firmware version (refer to Section 2.7 on page 16).

2.1 Accessing the Administration Submenus


The submenus of the Administration menu are password protected. When attempting to access
a submenu, the phone prompts you to enter the access password.

¾ To access the Administration menu:


1. In idle display, press the MENU key; the menu list is displayed:

2. Select the required Administration submenu (using the and Navigation keys), and
then press the ENTER key; the Administration submenus are displayed.
3. Select a submenu; you are prompted for a password:

4. Enter the password, and then press the ENTER key.

Notes:

• The default password is “1234”. To change the login password, use the
phone’s Web interface (refer to Section 3.11 on page 42).
• Once you have entered the password, the access session is applied to all
the submenus.

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310HD IP Phone

2.2 Configuring Network Settings


The phone’s LAN connection interface can be manually defined (static IP address) or
automatically configured using a DHCP server from where the LAN IP address is obtained.

¾ To configure the phone’s LAN connection type:


1. Access the Network Settings screen (MENU key > Administration menu > Network
Settings).

The following screen appears:

2. Choose one of the following IP addressing schemes (using the and Navigation keys):
• Static IP
• Automatic IP
3. Press the ENTER key. If you selected Static IP, then continue with Step 4; otherwise, skip to
Step 5.
4. If you selected the Static IP option, the following screen appears:

a. Select the required network parameter (IP Address, Netmask, Gateway, Primary
DNS, and/or Secondary DNS).
b. Press the ENTER key; the screen prompts you to enter the IP address of the selected
network parameter (e.g., static IP address below):

c. Press the ENTER key; the IP address is applied


d. Repeat steps 4.a through 4.c for each required network parameter.

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Administrator's Manual 2. LCD-Based Management

5. Press the left Navigation key until you are prompted to save settings, as shown below:

6. Select the Yes option (using the and Navigation keys), and then press the ENTER
key; the network settings are saved.

2.3 Configuring SIP Accounts


The SIP Accounts submenu allows you to configure parameters related to the phone’s SIP
account.

¾ To configure the phone’s SIP account:


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

The following screen appears:

2. Define the SIP proxy IP address:


a. Select the SIP Proxy Addr. option (using the and Navigation keys), and then
press the ENTER key: the following screen appears:

b. Select one of the following options (using the and Navigation keys):
♦ Disable: disables SIP proxy
♦ Enable: enables SIP proxy
An asterisk appears alongside the selected option.

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310HD IP Phone

c. Press the ENTER key; you are prompted to enter the IP address:

d. Enter the IP address, and then press the ENTER key; you are returned to the SIP
Accounts options.
3. Define the remaining SIP proxy parameters, by selecting (using the and Navigation
keys) the required parameter:
• User ID
• Display Name: caller ID
• Auth Name: authentication name
• Auth Password: authentication password
For each parameter, once you have entered the required value, press the ENTER key.
4. Select the Save option (using the and Navigation keys), and then press the ENTER
key; your Sip account settings are saved.

2.4 Restoring Defaults


The procedure below describes how to restore the phone to factory defaults.

¾ To restore the phone to default settings:


1. Select the Restore Defaults submenu (MENU key > Administration menu > Restore
Defaults):

A confirmation message appears, prompting you to confirm factory reset:

2. Select the Yes option (using the and Navigation keys), and then press the ENTER
key; the phone is restored to factory defaults.

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Administrator's Manual 2. LCD-Based Management

2.5 Restarting Phone


The procedure below describes how to restart the phone.

¾ To restart the phone:


1. Select the Restart submenu (MENU key > Administration menu > Restart):

A confirmation message appears, prompting you to confirm phone reset:

2. Select the Yes option (using the and Navigation keys), and then press the ENTER
key; the phone resets.

2.6 Viewing Network Status


You can view the phone’s network parameters. These include the following:
„ IP address
„ Subnet mask
„ Default gateway
„ Primary and secondary DNS addresses
„ MAC address

¾ To view the phone’s network status:


1. From the idle screen, press the MENU key, select the Status menu, and then the Network
Status submenu.

The following screen appears:

2. Scroll through the list (using the and Navigation keys) to view the required network
parameters.

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310HD IP Phone

2.7 Viewing Phone Model and Firmware Version


You can view the phone’s model number and software (firmware) version.

¾ To view the phone’s model and firmware version:


1. From the idle screen, press the MENU key, select the Status menu, and then the Network
Versions submenu.

The following screen appears:

2. Press the ENTER key; the phone’s model and firmware version is displayed.

Administrator's Manual 16 Document #: LTRT-09905


Administrator's Manual 3. Web-based Management

3 Web-based Management
This chapter describes the phone’s embedded Web server interface for configuring the phone.

Note: Where Web parameters can also be configured using the Configuration file, a
table is provided listing the corresponding Configuration file parameter. In such
cases, the description of the parameter (Web and Configuration file parameters)
appears only in the Chapter on Configuration file parameters (see Chapter 4).

3.1 Accessing the 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).

¾ To access the phone’s Web interface:


1. Connect the LAN port of your phone to the IP network (using the Cable or ADSL modem
from your Internet Service Provider).
2. Determine the phone’s IP address obtained from the DHCP server, using the phone’s LCD
screen as described in Section 2.1 on page 11 (in the ‘IP Address’ field).
3. Open a Web browser, and then in the URL address field, enter the phone’s IP address (for
example, http://192.168.1.2:8000 or https://192.168.1.2), as displayed below:

The Web login window appears:

Figure 3-1: Web Login Window

4. Alternatively, if your DHCP and DNS servers are synchronized you can access the phone
Web browser by using the following method:
http://310hd-<MAC Address>.<Domain Name>
(e.g. http://310hd-001122334455.corp.YourCompany.com)

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310HD IP Phone

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

Note: The administrator’s default login user name and password are “admin” and
“1234” respectively. To change the login credentials, refer to Section 3.11 on
page 42.

3.2 Getting Started with the Web


The areas of the Web interface are shown below:

Figure 3-2: Web Interface Areas

The Web interface is composed of the following main areas:


„ Toolbar: displays AudioCodes logo and provides the following buttons:

• Home: opens the Home page

• Log off: closes the Web interface


„ Navigation bar: provides tabs for accessing the configuration menus:
• Configuration: provides menus for configuring the phone.
• Management: provides menus for various management tasks such as firmware
upgrade and changing the login username and password.
• Status & Diagnostics: provides menus for displaying information on the status of the
phone, such as call history.
„ Navigation tree: tree-like, hierarchical structure of menus pertaining to the selected tab on
the Navigation bar.
„ Configuration pane: displays the configuration parameters pertaining to a selected menu in
the Navigation tree.

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Administrator's Manual 3. Web-based Management

3.3 Quick Setup


The Web interface allows you to quickly configure the main parameters required for basic phone
functioning. This is provided by the ‘Quick Setup’ page, as described below.

¾ To quickly setup your phone:


1. Access the ‘Quick Setup’ page (Configuration tab > Quick Setup menu > Quick Setup).

Figure 3-3: Quick Setup Page

2. For a description of the parameters on this page, refer to the following:


• Parameters under the LAN Setup group, refer to Section 3.7 on page 24.
• Parameters under the SIP Proxy and Registrar group, refer to Section 3.8.1 on page
25
• Parameters under the Line Settings group, refer to Section 3.8.5 on page 32

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310HD IP Phone

3.4 Changing LCD Display Language


You can choose your preferred language for the LCD display.

¾ To choose a language:
1. Access the ‘Language’ page (Configuration tab > Personal Settings menu > Language).

Figure 3-4: Language Page

2. Select the language according to the parameter in the table below, and then click Submit;
the phone reboots and changes the LCD display language accordingly.

Table 3-1: Language Parameter Description

Web Parameter Configuration File Parameter Description

Phone Display Language personal_settings/language Refer to Section 4.2.8

3.5 Defining Speed Dials


You can add up to 10 speed dials to your phone.

Note: The phone’s speed dials can also be defined in a simple text-based editor,
placed on a server (e.g., HTTP or FTP/TFTP), and then uploaded to the phone
using a configuration file. For a detailed description, refer to Section 4.3.

¾ To add a speed dial:


1. Access the ‘Speed Dial’ page (Configuration tab > Personal Settings menu > Speed Dial).

Figure 3-5: Speed Dial Page

2. In the ‘Number’ field corresponding to the phone’s Speed Dial key (in the ‘Button’ column),
enter the speed dial number to which you want to assign the Speed Dial key.
3. Click Submit.

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Administrator's Manual 3. Web-based Management

¾ To delete speed dials:


• Deleting selected speed dial entries: select the ‘Delete’ check boxes corresponding to
the speed dials that you want to delete, and then click Submit.
• All speed dials: Click Delete All, and then at the prompt, click OK.
• To clear (unselect) all your selected ‘Delete’ check boxes, and then click Reset.

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310HD IP Phone

3.6 Regional Call Progress Tones and Ring Tones


You can suite your phone’s call progress tones to the location in which the phone is located. In
addition, instead of using the default ring tone, you can upload a different ring tone file and use
that ring tone for indicating incoming calls. This ring tone can only be selected using the phones
LCD interface.

3.6.1 Selecting the Region for Call Progress Tones


Follow the procedure below for selecting the region in which your phone is located. This is
important for suiting your phone for the call progress tones of the country in which the phone is
located.

¾ To select the geographical location of your phone:


1. Access the ‘Tones’ page (Configuration tab > Personal Settings menu > Tones).

Figure 3-6: Tones Page

2. From the ‘Current Location’ drop-down list, select the country in which your phone is located.
3. Click Submit.

Table 3-2: Region CPT Parameter Description

Web Parameter Configuration File Parameter Description

Current Location voip/regional_settings/selected_country Refer to Section 4.2.6


(includes additional
configuration file
parameters for CPT)

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Administrator's Manual 3. Web-based Management

3.6.2 Uploading Ring Tones


Follow the procedure below for uploading new ring tones.

Notes:

• The ring tone file must be in WAV file format (A/Mu-Law, 8-kHz audio
sample rate and 8-bit audio sample size or PMC 16-kHz audio sample rate
and 16-bit audio sample size, Intel PCM encoding).
• If you want the phone to use an uploaded ring tone, you need to use the
phone’s LCD screen to select it (refer to the phone’s User’s Manual).

¾ To upload a ring tone:


1. Access the ‘Tones’ page (Configuration tab > Personal Settings menu > Tones).

Figure 3-7: Tones Page

2. For uploading a ring tone, perform the following:


a. In the ‘Ringing Tone Name’ field, enter the name of the ring tone file that you want to
upload.
• Click the Browse button, navigate to the folder in which the ring tone file is located,
select the file, and then click Open; the file name and path is displayed in the ‘File
Location’ field.
• Click Submit; the file is loaded to the phone and displayed in the Ring Tone list.

¾ To delete Ring Tones:


• Deleting selected ring tones: select the ‘Delete’ check boxes corresponding to the ring
tone that you want to delete, and then click Submit.

Table 3-3: Ring Tone Parameter Description

Web Parameter Configuration File Parameter Description

Ringing Tone Name provisioning/ring_tone_uri Refer to Section 4.2.3


File Location

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310HD IP Phone

3.7 Configuring LAN Interface


The phone’s LAN configuration includes defining the method for obtaining an IP address as well
as VLAN settings. 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.

¾ To define the phone’s LAN settings:

1. Access the ‘LAN Settings’ page (Configuration tab > Network Connections menu > LAN
Settings).

Figure 3-8: LAN Settings Page

2. Configure the LAN settings according to the parameters in the table below, and then click
Submit.

Table 3-4: LAN Parameters Description

Web Parameter Configuration File Parameter Description

IP Type network/lan_type Refer to Section 4.2.2


IP Address network/lan/fixed_ip/ip_address
Subnet Mask network/lan/fixed_ip/netmask
Default Gateway Address network/lan/fixed_ip/gateway
Primary DNS network/lan/fixed_ip/primary_dns
Secondary DNS network/lan/fixed_ip/secondary_dns
VLAN Settings
VLAN Discovery Mode network/lan/vlan/mode Refer to Section 4.2.2
VLAN ID network/lan/vlan/id
VLAN Priority network/lan/vlan/priority

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3.8 VoIP Settings


The Voice Over IP menu allows you to configure the following VoIP settings:
„ Signaling protocol (refer to Section 3.8.1 on page 25)
„ Dialing (refer to Section 3.8.2 on page 28)
„ Media streaming (refer to Section 3.8.3 on page 30)
„ Voice (refer to Section 3.8.4 on page 30)
„ Line (refer to Section 3.8.5 on page 32)
„ Services (refer to Section 3.8.6 on page 33)
„ Volume (refer to Section 3.8.7 on page 35)

3.8.1 Configuring Signaling Protocol


The ‘Signaling Protocol’ page allows you to define various SIP signaling parameters.

¾ To define the phone’s SIP settings:

1. Access the ‘Signaling Protocol’ page (Configuration tab > Voice Over IP menu > Signaling
Protocols).

Figure 3-9: Signaling Protocol Page

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310HD IP Phone

2. Configure SIP according to the parameters in the table below, and then click Submit.

Table 3-5: SIP Parameters Description

Web Parameter Configuration File Parameter Description

SIP General Parameters


SIP Transport Protocol voip/signalling/sip/transport_protocol Refer to Section 4.2.5.6
SIP Local Port voip/signalling/sip/port
Gateway Name voip/signalling/sip/proxy_gateway
PRACK Mode voip/signalling/sip/prack/enabled
Enable RPORT voip/signalling/sip/rport/enabled
Include PTIME in SDP voip/signalling/sip/sdp_include_ptime
Enable Keep Alive using voip/signalling/sip/keepalive_options/en
OPTIONS abled
Keep Alive Period voip/signalling/sip/keepalive_options/ti
meout
Connect Media on 180 voip/signalling/sip/connect_media_on_1
Response 80
Block Caller ID on Outgoing voip/signalling/sip/block_callerid_on_o
Calls utgoing_calls
Incoming Anonymous Call voip/signalling/sip/anonymous_calls_bl
Blocking ocking
SIP Proxy and Registrar Parameters
Use SIP Proxy voip/signalling/sip/use_proxy Refer to Section 4.2.5.6
Proxy IP Address or Host voip/signalling/sip/proxy_address
Name
Proxy Port voip/signalling/sip/proxy_port
Maximum Number of voip/signalling/sip/proxy_timeout
Authentication Retries
Use SIP Proxy IP and Port for voip/signalling/sip/use_proxy_ip_port_f
Registration or_registrar
Use SIP Registrar voip/signalling/sip/sip_registrar/enabled
Registrar IP Address or Host voip/signalling/sip/sip_registrar/addr
Name
Registrar Port voip/signalling/sip/sip_registrar/port
Registration Expires voip/signalling/sip/proxy_timeout
Use SIP Outbound Proxy voip/signalling/sip/sip_outbound_proxy/
enabled
Outbound Proxy IP Address voip/signalling/sip/sip_outbound_proxy/
or Host Name addr
Outbound Proxy Port voip/signalling/sip/sip_outbound_proxy/
port
Use Redundant Proxy voip/signalling/sip/redundant_proxy/ena
bled
Redundant Proxy Address voip/signalling/sip/redundant_proxy/ad
dress

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Web Parameter Configuration File Parameter Description

Redundant Proxy Port voip/signalling/sip/redundant_proxy/por


t
Redundant Proxy Keep Alive voip/signalling/sip/redundant_proxy/kee
Period palive_period
Switch back to Primary SIP voip/signalling/sip/redundant_proxy/sy
proxy when available mmetric_mode
SIP Timers
Retransmission Timer T1 voip/signalling/sip/sip_t1 Refer to Section 4.2.5.6
Retransmission Timer T2 voip/signalling/sip/sip_t2
Retransmission Timer T4 voip/signalling/sip/sip_t4
INVITE Timer voip/signalling/sip/sip_invite_timer
Session-Expires voip/signalling/sip/session_timer
Min-SE voip/signalling/sip/min_session_interval
SIP QoS
Type of Service (ToS) voip/signalling/sip/tos Refer to Section 4.2.5.6

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3.8.2 Configuring Dialing


The dialing parameters can be configured in the ‘Dialing’ page, as described below.

¾ To define the dialing parameters:


1. Access the ‘Dialing’ page (Configuration tab > Voice Over IP menu > Dialing).

Figure 3-10: Dialing Page

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

Table 3-6: Dialing Parameters Description

Web Parameter Configuration File Parameter Description

Dialing Timeout voip/dialing/timeout Refer to Section 4.2.5.5


Phone Number Length voip/dialing/phone_number_max_size
Enable Dialing Complete Key voip/dialing/dial_complete_key/enabled
Dialing Complete Key voip/dialing/dial_complete_key/key
Dial Tone Timeout voip/dialing/dialtone_timeout
Reorder Tone Timeout voip/dialing/warning_tone_timeout
No Answer Call Timeout voip/dialing/unanswered_call_timeout
Howler Tone Timeout voip/dialing/offhook_tone_timeout
Secondary Dial Tone voip/dialing/secondary_dial_tone/enabl
ed
Secondary Dial Tone Key voip/dialing/secondary_dial_tone/key_s
equence

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Web Parameter Configuration File Parameter Description

DTMF Transport Mode voip/media/out_of_band_dtmf


Digit Map voip/signalling/sip/digit_map
Dial Plan voip/signalling/sip
/number_rules
Automatic Dialing
Activate voip/dialing/auto_dialing/enabled Refer to Section 4.2.5.5
Timeout voip/dialing/auto_dialing/timeout
Destination Phone Number voip/dialing/auto_dialing/destination

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3.8.3 Configuring Media Streaming


The media streaming parameters are configured in the ‘Media Streaming’ page, as described
below.

¾ To define the media streaming parameters:


1. Access the ‘Media Streaming’ page (Configuration tab > Voice Over IP menu > Media
Streaming).

Figure 3-11: Media Streaming Page

2. Configure media streaming according to the parameters in the table below, and then click
Submit.

Table 3-7: Media Streaming Parameters Description

Web Parameter Configuration File Parameter Description

RTP Port Range voip/media/media_port Refer to Section 4.2.5.4


DTMF Relay RFC 2833 voip/media/dtmf_payload
Payload Type
Quality of Service (QoS)
Type of Service (ToS) voip/media/media_tos Refer to Section 4.2.5.4
Codecs
Codec Type voip/codec/codec_info /%d/name Refer to Section 4.2.5.3
Packetization Time voip/codec/codec_info /%d/ptime
G.723 Bitrate
G.723 Bitrate voip/codec/g723_bitrate Refer to Section 4.2.5.3

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3.8.4 Configuring Voice


The voice parameters are configured in the ‘Voice’ page, as described below.

¾ To define the voice parameters:


1. Access the ‘Voice’ page (Configuration tab > Voice Over IP menu > Voice).

Figure 3-12: Voice Page

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

Table 3-8: Voice Parameters Description

Web Parameter Configuration File Parameter Description

Gain Control
Enable Automatic Gain Control voip/audio/gain/automatic_gain_contro Refer to Section 4.2.5.2
l/enabled
Automatic Gain Control voip/audio/gain/automatic_gain_contro
Direction l/direction
Target Energy voip/audio/gain/automatic_gain_contro
l/target_energy
Jitter Buffer
Minimum Delay voip/audio/jitter_buffer/min_delay Refer to Section 4.2.5.7
Optimization Factor voip/audio/jitter_buffer/optimization_fa
ctor
Silence Compression
Enable Silence Compression voip/audio/silence_compression/enabl Refer to Section 4.2.5.7
ed

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3.8.5 Configuring Line


Before you can make a call on your phone, you must configure a phone line.

¾ To define the lines:


1. Access the ‘Line Settings’ page (Configuration tab > Voice Over IP menu > Line
Settings).

Figure 3-13: Voice Page

2. Configure the line according to the parameters in the table below, and then click Submit.

Table 3-9: Line Parameters Description

Web Parameter Configuration File Parameter Description

Line Activate voip/line/%d/enabled Refer to Section 4.2.5.1


User ID voip/line/%d/id
Display Name voip/line/%d/description
Authentication User Name voip/line/%d/auth_name
Authentication Password voip/line/%d/auth_password

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3.8.6 Configuring 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).

Note: The Busy Lamp Field (BLF) feature will be supported in the next applicable
release.

¾ To define services:
1. Access the ‘Services’ page (Configuration tab > Voice Over IP menu > Services).

Figure 3-14: Services Page

2. Configure the services according to the parameters in the table below, and then click
Submit.

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Table 3-10: Supplementary Services Parameters Description

Web Parameter Configuration File Parameter Description

Application Server Type voip/services/application_server_type Refer to Section 4.2.5.8


Call Waiting
Activate voip/services/call_waiting/enabled Refer to Section 4.2.5.8
Call Waiting SIP Reply voip/services/call_waiting/sip_reply
Call Forward
Activate voip/services/call_forward/line/0/enabl Refer to Section 4.2.5.8
ed
Call Forward Type voip/services/call_forward/line/0/type
Forward on No Reply voip/services/call_forward/line/0/timeo
Timeout ut
DND (Do Not Disturb)
Activate voip/services/do_not_disturb/enabled Refer to Section 4.2.5.8
Message Waiting Indication
Voice Mail Number voip/services/msg_waiting_ind/voice_ Refer to Section 4.2.5.8
mail_number
Activate voip/services/msg_waiting_ind/enable
d
Subscribe To MWI voip/services/msg_waiting_ind/subscr
ibe
MWI Server IP Address or voip/services/msg_waiting_ind/subscr
Host Name ibe_address
MWI Serve Port voip/services/msg_waiting_ind/subscr
ibe_port
MWI Subscribe Expiry Time voip/services/msg_waiting_ind/expirai
tion_timeout
General Parameters
Stutter Tone Duration voip/services/msg_waiting/stutter_ton Refer to Section 4.2.5.8
e_duration
Out of Service Behavior voip/services/out_of_service_bahavio
r
Automatic Disconnect voip/dialing/automatic_disconnect

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3.8.7 Configuring Volume


You can configure various volume parameters such as speaker, tones, ringer, analog/digital input
and output gain.

¾ To define volume settings:


1. Access the ‘Services’ page (Configuration tab > Voice Over IP menu > Volume Settings).

Figure 3-15: Volume Settings Page

2. Configure the volume settings according to the parameters in the table below, and then click
Submit.

Table 3-11: Volume Parameters Description

Web Parameter Configuration File Parameter Description

Additional Speaker Gain voip/audio/gain/additional_speaker_g Refer to Section 4.2.5.2


ain
Tone Volume voip/audio/gain/tone_signal_level
Ringer Volume voip/audio/gain/ringer_signal_level
Hands-free Gain Parameters
Hands Free Digital Output voip/audio/gain/handsfree_digital_out Refer to Section 4.2.5.2
put_gain
Hands Free Digital Input voip/audio/gain/handsfree_digital_inp
ut_gain
Hands Free Analog Output voip/audio/gain/handsfree_analog_out
put_gain

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Web Parameter Configuration File Parameter Description

Hands Free Analog Input voip/audio/gain/handsfree_analog_inp


ut_gain
Handset Gain Parameters
Handset Digital Output voip/audio/gain/handset_digital_outpu Refer to Section 4.2.5.2
t_gain
Handset Digital Input voip/audio/gain/handset_digital_input
_gain
Handset Analog Output voip/audio/gain/handset_analog_outp
ut_gain
Handset Analog Input voip/audio/gain/handset_analog_input
_gain
Handset Analog Sidetone voip/audio/gain/handset_analog_sidet
Gain one_gain
Headset Gain Parameters
Headset Digital Output voip/audio/gain/headset_digital_outpu Refer to Section 4.2.5.2
t_gain
Headset Digital Input voip/audio/gain/headset_digital_input
_gain
Headset Analog Output voip/audio/gain/headset_analog_outp
ut_gain
Headset Analog Input voip/audio/gain/headset_analog_input
_gain
Headset Analog Sidetone voip/audio/gain/headset_analog_sidet
Gain one_gain

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3.9 Configuring Date and Time


Generally, a phone retrieves the date and time from a Network Time Protocol (NTP) server when
it connects to the Internet. Alternatively, the date and time can be configured manually. NTP is a
protocol for distributing the Coordinated Universal Time (UTC) by means of synchronizing the
clocks of computer systems over packet-switched, variable-latency data networks.

¾ To define the date and time:


1. Access the ‘Date and Time’ page (Configuration tab > Advanced Applications menu >
Date and Time).

Figure 3-16: Date and Time Page

2. Configure the date and time according to the parameters in the table below, and then click
Submit.

Table 3-12: Date and Time Parameters Description

Web Parameter Configuration File Parameter Description

Localization
Time Zone system/ntp/gmt_offset Refer to Section 4.2.1
Daylight Saving Time
Active system/ntp/daylight_saving/activate Refer to Section 4.2.1
Start Time system/ntp/daylight_saving/start_date
End Time system/ntp/daylight_saving/end_date
Offset system/ntp/daylight_saving/offset
NTP & Time Settings
Active system/ntp/enabled Refer to Section 4.2.1
Primary Server system/ntp/primary_server_address
Secondary Server system/ntp/secondary_server_addres
s
Update Interval system/ntp/sync_time

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3.10 Firmware and Configuration Management


The Web interface allows you to perform the following:
„ Automatic update of firmware and configuration files
„ Manual update of firmware and configuration files

3.10.1 Configuring Automatic Update of Firmware and Configuration File


The IP phone offers a built-in mechanism for automatically upgrading its software image and
updating its configuration. This method is used to upgrade the phone firmware and update its
configuration, by remotely downloading an updated software image and configuration file.
The automatic update mechanism helps you keep your software image and configuration up-to-
date, by performing routine checks for newer software versions and configuration files, as well as
allowing you to perform manual checks.
The automatic update mechanism is as follows:
1. Before connecting the phone, verify that the provisioning server is running and that the
firmware and configuration files are located in the correct location.
2. Connect your phone to the IP network, and then connect the phone to the power outlet.
3. 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.
4. 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.
5. 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 and the version is later than the current one. When configuration update is
needed, the phone reboots.

Notes:
• In the DHCP Discover message, the phone publishes its model name in
Option fields 60 and 77 (e.g. 310HD). If the administrator wants to provide
different provisioning information to different phone models, the
administrator can set up a policy in the DHCP server according to the phone
model name.
• If the phone for some reason is powered off during the firmware upgrade
process, the phone is unusable and the recovery process must be
performed (refer to Section 3.10.3 on page 41).
• 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.
• An additional auto-provisioning mechanism is supported if the provisioning
environment does not provide all the required information (e.g. DHCP
options). For more information, refer to the document 3xxHD Advanced
Support for Auto-provisioning Application Note.

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¾ To define automatic update:


1. Access the ‘Automatic Update page (Management tab > Automatic Update menu >
Automatic Update).

Figure 3-17: Automatic Update Page

2. Configure automatic update of firmware and configuration files according to the parameters
in the table below, and then click Submit.

Table 3-13: Automatic Provisioning Parameters Description

Web Parameter Configuration File Parameter Description

Provisioning Method provisioning/method Refer to Section 4.2.3


Firmware URL provisioning/firmware/url
Configuration URL provisioning/configuration/url
DHCP Option Value provisioning/url_option_value
Check Period provisioning/period/type
Every provisioning/period/hourly/hours_inte
(Check Period = Hourly) rval
Every day at provisioning/period/daily/time
Every provisioning/period/weekly/day
(Check Period = Weekly)
provisioning/period/weekly/time
Random Provisioning Time provisioning/random_provisioning_ti
me

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3.10.2 Updating (Manually) Configuration File


The phone enables you to view, save, and load its configuration file to backup and restore the
current configuration.

¾ To manually update the configuration file:


1. Access the ‘Configuration File’ page (Management tab > Manual Update menu >
Configuration File). The current configuration file settings are displayed in the text pane.

Figure 3-18: Configuration File Page

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

Figure 3-19: 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.

Note: The configuration file name must have the extension *.cfg.

¾ 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.

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3.10.3 Recovering Firmware


If the phone is powered off for some reason during the firmware upgrade process, the phone
becomes unusable.

¾ To recover the phone firmware:


1. Ensure that your DHCP server supports Options 66 (TFTP server address) and 67 (firmware
file), and that these are configurable.
2. Before connecting the phone, verify that the TFTP server is running and the firmware file for
recovery is located in the correct location.
3. Connect your phone to the IP network, and then connect the phone to the power outlet;
b. 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.
c. The phone, in the DHCP Discover message sends its model name in the DHCP Option
77 field. The DHCP server, according to the phone model, sets the appropriate firmware
file name in the DHCP Option 67 field sent to the phone
(e.g. 310HD_1.2.2_build_5.img).
d. The phone then upgrades to the recovery firmware.
e. After the firmware upgrade process completes, the phone boots up successfully.

Note: In Recovery Mode, the phone is unable to communicate with a TFTP server
located on a Linux machine or any other operating system (OS) that is not
Microsoft Window XP.

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3.11 Changing Login Username and Password


You can change the phone’s login user name and password. This is the login required for
accessing the Web interface and the Administration menu of the phone’s LCD screen. The
default user name and password is “admin” and “1234” respectively.

¾ To change the login username and password:


1. Access the ‘System Authorization’ page (Management tab > Administration menu >
Users).

Figure 3-20: System Authorization Page

2. In the ‘Username’ field, enter a user name.


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 box appears.
5. Click OK.

Table 3-14: Login Username and Password Parameters Description

Web Parameter Configuration File Parameter Description

Username system/user_name Refer to Section 4.2.1


Password system/password

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3.12 Restoring Phone Defaults


You can restore all your phone’s settings to factory default settings, as described in the procedure
below.

¾ To restore the phone to factory defaults:


1. Access the ‘Restore Defaults’ page (Management tab > Administration menu > Restore
Defaults).

Figure 3-21: Restore Defaults Page

2. Click the Submit button; a confirmation box appears prompting you to confirm.

Figure 3-22: Submit Confirmation Box

3. Click OK.

3.13 Restarting Phone


You can use the Web interface to restart your phone.

¾ To restart the phone:


1. Access the ‘Restart System’ page (Management tab > Administration menu > Restart
System).

Figure 3-23: Restart System Page

2. Click the Restart button; a confirmation box appears prompting you to confirm.

Figure 3-24: Confirmation Box

3. Click OK.

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3.14 Configuring TR-069 Remote Management


To connect to the remote management server the TR-069 client on the phone should be
configured as following:

¾ To configure the TR-69 parameters:


1. Access the ‘TR-069’ page (Management tab > Remote Management menu > TR-069).

Figure 3-25: TR-069 Page

2. Configure TR-069 according to the parameters in the table below.


3. Click Submit, and then OK at the confirmation box prompt.

Table 3-15: TR-069 Parameters Description

Web Parameter Configuration File Parameter Description

Activate management/tr069/enabled Refer to Section 4.2.4


TR-069 Key management/tr069/feature_key
TR-069 ACS URL management/tr069/acs_url
TR-069 User management/tr069/user_name
TR-069 Password management/tr069/password
TR-069 Request Connection management/tr069/connection_reques
User t/user_name
TR-069 Request Connection management/tr069/connection_reques
Password t/password
Informed management/tr069/inform/enabled
Informed Interval management/tr069/inform/interval

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3.15 Viewing Status Information


The Web interface allows you to view a variety of status information about your phone.

Note: Currently, the ‘VoIP Status’ page (Status & Diagnostics tab > System Status
menu > VoIP Status) does not exist.

3.15.1 Viewing LAN Information


You can view various LAN details such as IP address and default gateway.

¾ To view LAN status information:


„ Access the ‘Network Status’ page (Status & Diagnostics tab > System Status menu >
Network Status).

Figure 3-26: Network Status Page

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3.15.2 Viewing Call History


You can view a list of received calls, missed calls, and dialed numbers.

¾ To view call history log:


1. Access the ‘Call History’ page (Status & Diagnostics tab > History menu > Call History).

Figure 3-27: Call History Page

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.
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.

3.15.3 Viewing Phone’s Version Number


You can view the phone’s model name as well as the firmware version currently running on the
phone.

¾ To view the phone’s model and version number:


„ Access the ‘System Information’ page (Status & Diagnostics tab > System Information
menu > Versions).

Figure 3-28: System Information Page

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3.16 Diagnostics

3.16.1 Configuring System Logging


System logging allows you to debug the different software components of the phone.

¾ To define the logging parameters:


1. Access the ‘System logging’ page (Status & Diagnostics tab > Diagnostics menu >
Logging).

Figure 3-29: System Logging Page

2. Configure system logging according to the parameters in the table below, and then click
Submit.

Table 3-16: System Logging Parameters Description

Web Parameter Configuration File Parameter Description

Activate system/watchdog/enabled Refer to Section 4.2.1


Server IP Address or Host system/syslog/server_address
Name
Server Port system/syslog/server_port
Category system/syslog/sip_log_filter

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3.16.2 Configuring Packet Recording


Recording parameters allows you to debug the voice activity of the phone.

¾ To define the recording parameters:


1. Access the ‘Recording’ page (Status & Diagnostics tab > Diagnostics menu >
Recording).

Figure 3-30: Recording Page

2. Configure packet recording according to the parameters in the table below, and then click
Submit.

Table 3-17: Packet Recording Parameters Description

Web Parameter Configuration File Parameter Description

Remote IP Address or Host voip/packet_recording/remote_ip Refer to Section 4.2.7


Name
Remote Port voip/packet_recording/remote_port
Enable DSP Recording /voip/packet_recording/enabled
Enable RTP Recording voip/packet_recording/rtp_recording/e
nabled
Enable EC Debug Recording voip/packet_recording/ec_debug_reco
rding/enabled
Enable Network Recording voip/packet_recording/network_recor
ding/enabled
Enable TDM Recording voip/packet_recording/tdm_recording/
enabled

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4 Configuration File Parameters


This section describes the Configuration file and the parameters that you can set in the
Configuration file.

4.1 Overview
The Configuration file can be loaded to the phone using the automatic provisioning mechanism, or
manually from your local PC using the Web interface (refer to Section 3.10.2). The subsections
below describe the Configuration file syntax and linking additional Configuration files to a
Configuration file.

4.1.1 File Syntax


The Configuration file can be created using a standard ASCII, text-based program such as
Notepad. The Configuration file is a *.cfg file with the file name being the phone’s MAC address:
<phone’s MAC address>.cfg.
The syntax of the Configuration file is as follows:

<parameter name>=<value>

Ensure that the Configuration file adheres to the following guidelines:


„ No spaces on either side of the equals (=) sign.
„ Each parameter must be on a new line.
Below is an example of part of a Configuration file:

system/type=310HD
voip/line/0/enabled=1
voip/line/0/id=1234
voip/line/0/description=320HD
voip/line/0/auth_name=1234
voip/line/0/auth_password=4321

4.1.2 Linking Additional Files using “Include”


The Configuration file allows you to include links (URL and/or file name) to other Configuration files
that provide additional parameter settings. This is especially useful in deployments with multiple
phones, where the phones share common configuration but where each phone has some unique
settings. In such a scenario, a phone’s Configuration file can include unique parameter settings as
well as links to additional Configuration files with settings common to all phones.
Linking additional files is achieved by using the include function in the phone’s Configuration file.
For example, the below Configuration file provides links to additional Configuration files (shown in
bolded font):

system/type=310HD
include 310HD_<MAC>_voip.cfg
include vlan_conf.cfg
include network_conf.cfg
include provisioning_conf.cfg

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In addition, the Configuration file can provide URL paths (FTP, TFTP, HTTP, or HTTPS) to where
the additional files are located, as shown in the example below (shown in bolded font):

system/type=310HD
include http://10.10.10.10/310HD_<MAC>_voip.cfg
include https://remote-pc/vlan_conf.cfg
include tftp://10.10.10.10/310HD_<MAC>_network.cfg
include ftp://remote-pc/provisining_conf.cfg

Note: If no URL is provided in the Configuration file, the files are retrieved according to
the provisioning information (e.g. DHCP Options 66/67 or 160).

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Administrator's Manual 4. Configuration File Parameters

4.2 Configuration File Parameters


The configuration file parameters are described in this subsection.

Note: The optional values of the Configuration file parameters are enclosed in square
brackets while its corresponding Web values arewritten outside the square
brackets, for example, [1] Enable.

4.2.1 System Parameters


Table 4-1: System Configuration File Parameters

Parameter Description

Note: To add a value to these parameters, enter system/ followed by the parameter name, equal sign and
then the value (e.g. system/type=310HD).
system/type The phone model.
The default value 310HD.
system/user_name The phone user name.
The default value is admin.
Note: This parameter is applicable only to the Web and Telnet interfaces.
system/password The encrypted phone password.
The default value is 1234.
To generate an encrypted password for the system/password parameter, run
the password generator (passwd_gen) with the required password, as shown
below. This application is available on Windows and Linux operating systems.
The resultant string is the encrypted password for system/password.

Note: This parameter is applicable only to the Web and Telnet interfaces, and
LCD display.
system/watchdog/enable Enables the system watch dog.
d ƒ [0] Disable
ƒ [1] Enable (default)
Note: It is recommended to leave this parameter at its default value.
system/syslog/log_watch Enables the watchdog logs to be sent to the address specified in
dog_events system/syslog/server_address.
ƒ [0] (default) - Disable
ƒ [1] - Enable

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Parameter Description

system/syslog/sip_log_fi Filters the type of the VoIP application logging.


lter ƒ [0] (default) = Logging is disabled
ƒ [1] - SIP Call Control
ƒ [2] - SIP Stack
ƒ [3] - SIP Call Control and SIP Stack
ƒ [4] - User Application
ƒ [5] - SIP Call Control and User Application
ƒ [6] - SIP Stack and User Application
ƒ [7] - SIP Call Control, SIP Stack and User Application
system/syslog/server_ad The IP address (in dotted-decimal notation) of the computer you are using to
dress run the Syslog server (e.g. WireShark). The Syslog server is an application
designed to collect the logs and error messages generated by the phone. The
default IP address is 0.0.0.0.
system/syslog/server_po Defines the UDP port of the Syslog server.
rt
The valid range is 0 to 65,535. The default port is 514.
system/syslog/mode Defines the output direction of the Syslog information.
ƒ [NONE] (default) - Disable
ƒ [UDP] - Toward the network
Network Time Protocol (NTP) Server
system/ntp/enabled Enables the NTP server from which the phone retrieves the date and time.
ƒ [0] Disable (default)
ƒ [1] Enable – obtains the time information from a configured NTP server
system/ntp/primary_serv Defines the address of the main NTP server (this can be a domain name, for
er_address example, tick.nap.com.ar).
system/ntp/secondary_s Defines the address of the secondary NTP server.
erver_address
system/ntp/gmt_offset The GMT offset.
The format of this value is + or - xx:yy (e.g., :+02:00). The default offset is
00:00.
system/ntp/sync_time This sub-section defines how often the phone must perform an update with the
NTP server.
ƒ [days] - defines the number of days
ƒ [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
system/ntp/sync_time/da The number of days.
ys
The valid range is 0 to 7. The default of days is 0.
system/ntp/sync_time/ho The number of hours.
urs
The valid range is 0 to 24. The default is 12.

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Parameter Description

Daylight Saving
system/ntp/daylight_savi Determines whether the phone automatically detects the Daylight Saving
ng/activate setting for selected time zones.
ƒ [Disable] Disable (default)
ƒ [Enable] Enable
system/ntp/daylight_savi This sub-section defines the starting day for the daylight saving offset.
ng/start_date ƒ [month] - defines the specific month in a year
ƒ [day] - defines the specific day in a month
ƒ [hour] - defines the specific hour in a day
ƒ [minute] - defines the specific minute in an hour
For example: To configure the phone to start daylight savings with a specific
offset on February 22nd at 14:30, set the following:
system/ntp/daylight_saving/start_date/month=2
system/ntp/daylight_saving/start_date/day=22
system/ntp/daylight_saving/start_date/hour=14
system/ntp/daylight_saving/start_date/minute=30
system/ntp/daylight_savi The month in a year.
ng/start_date/month
The valid range is 1 to 12.
system/ntp/daylight_savi The day in a month.
ng/start_date/day
The valid range is 1 to 31.
system/ntp/daylight_savi The hour in the day.
ng/start_date/hour
The valid range is 0 to 23.
system/ntp/daylight_savi The minute in an hour.
ng/start_date/minute
The valid range is 0 to 59.
system/ntp/daylight_savi This sub-section defines the ending day for the daylight saving offset.
ng/end_date ƒ [month] - defines the specific month in a year
ƒ [day] - defines the specific day in a month
ƒ [hour] - defines the specific hour in a day
ƒ [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
system/ntp/daylight_savi The month in a year.
ng/end_date/month
The valid range is 1 to 12.
system/ntp/daylight_savi The day in a month.
ng/end_date/day
The valid range is 1 to 31.
system/ntp/daylight_savi The hour in the day
ng/end_date/hour
The valid range is 0 to 23.

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Parameter Description

system/ntp/daylight_savi The minute in an hour.


ng/end_date/minute
The valid range is 0 to 59.
system/ntp/daylight_savi The offset value for the daylight saving.
ng/offset
The valid range is 0 to 180. The default offset is 60.

4.2.2 Network Parameters


Table 4-2: Network Configuration File Parameters

Parameter Description

Note: To add a value to these parameters, enter network/ followed by the parameter name, equal sign and
then the value (e.g. network/lan_type=DHCP).
network/lan_type Defines the IP addressing method:
ƒ [DHCP] Automatic IP DHCP (default) - Phone’s IP address is acquired
automatically from a DHCP server
ƒ [STATIC] Static IP - Phone’s IP address is defined manually
network/lan/fixed_ip This sub-section defines the parameters which are relevant in case lan_type is
configured to “STATIC”.
network/lan/fixed_ip/ip_a The LAN IP address.
ddress
Note: This parameter is applicable only when the phone is assigned a static IP
address.
network/lan/fixed_ip/net The subnet mask address.
mask
Note: This parameter is applicable only when the phone is assigned a static IP
address.
network/lan/fixed_ip/gate The IP address of the default gateway.
way
Note: This parameter is applicable only when the phone is assigned a static IP
address.
Domain Name Server (DNS)
network/lan/fixed_ip/prim The primary DNS server address.
ary_dns
Note: This parameter is applicable only when the phone is assigned a static IP
address.
network/lan/fixed_ip/sec The secondary DNS server address. The phone connects to this server if the
ondary_dns primary DNS server is unavailable.
Note: This parameter is applicable only when the phone is assigned a static IP
address.

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Parameter Description

VLAN
network/lan/vlan/mode Determines the VLAN mode of operation.
ƒ [Disable] Disable
ƒ [Manual] Manual Configuration of LAN - Static configuration of VLAN ID
and priority
ƒ [Automatic] Automatic Configuration of VLAN/CDP (default) - VLAN
discovery mechanism based on Cisco Discovery Protocol (CDP)
network/lan/vlan/id The VLAN ID.
The valid range is 0 to 4096. The default VLAN ID is 0.
network/lan/vlan/priority The priority of traffic pertaining to this VLAN.
The valid range is 0 to 7 (where 7 is the highest priority). The default VLAN
priority is 0.

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4.2.3 Provisioning Parameters


Table 4-3: Provisioning Configuration File Parameters

Parameter Description
Note: To add a value to these parameters, enter provisioning/ followed by the parameter name equals the
value (e.g. provisioning/method=dynamic).
provisioning/method Defines the provisioning method:
ƒ [Disable] Disable - Automatic update is disabled. The phone attempts to
upgrade its firmware and configuration
ƒ [Dynamic] DHCP Options (Dynamic URL) (default) - Using DHCP options
160 or 66/67 for provisioning
ƒ [Static] Static URL - Using Static URL for provisioning
provisioning/firmware/url The static URL for checking the firmware file. The URL must be entered using
one of the following syntax options:
ƒ <protocol>://<server IP address or host name>
ƒ <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:
ƒ tftp://192.168.2.1 – retrieved firmware file is 310HD.img
ƒ 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”.
provisioning/configuratio The static URL for checking the configuration file. The URL must be entered
n/url using one of the following syntax options:
ƒ <protocol>://<server IP address or host name>
ƒ <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:
ƒ http://192.168.2.1 - configuration file name is <MAC Address>.cfg, for
example, 001122334455.cfg
ƒ https://192.168.2.1/310HD_<MAC>_conf.cfg - retrieved configuration file
name is 310HD_<MAC Address>_conf.cfg, for example,
310HD_001122334455_conf.cfg
Note: This parameter is applicable only when method is configured to “Static”.
provisioning/period/type Defines the period type for automatic provisioning:
ƒ [hourly] Hourly - Sets an interval in hours.
ƒ [daily] Daily (default) - Sets an hour in the day.
ƒ [weekly] Weekly - Sets a day in the week and an hour in the day.
ƒ [powerup] On Power-up Only - The phone tries to upgrade only after
power-up.
provisioning/period/hour The interval in hours for automatically checking for new firmware and
ly/hours_interval 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”.

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Parameter Description

provisioning/period/daily The hour in the day for automatically checking for new firmware and
/time 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”.
provisioning/period/week The day in the week for automatically checking for new firmware and
ly/day configuration files.
ƒ [Sunday] Sunday (default)
ƒ [Monday] Monday
ƒ [Tuesday] Tuesday
ƒ [Wednesday] Wednesday
ƒ [Thursday] Thursday
ƒ [Friday] Friday
ƒ [Saturday] Saturday
Note: This parameter is applicable only when type is configured to “weekly”.
provisioning/period/week The hour in the day for automatically checking for new firmware and
ly/time configuration files.
The format of this value is: hh:mm, where hh is hour and mm is minutes. For
example: 00:30
The default time is 00:00.
Note: This parameter is applicable only when type is configured to “weekly”.
provisioning/url_option_ Determines the DHCP option number to be used for receiving the URL for
value provisioning.
The default value is 160.
The phone supports DHCP Option 160 for complete URL and Options 66/67
for TFTP usage. Option 160 has the highest priority and if absent, Options
66/67 are used.
The following syntax is available for DHCP option 160:
ƒ <protocol>://<server IP address or host name>
ƒ <protocol>://<server IP address or host name>/<firmware file name>
ƒ <protocol>://<server IP address or host name>/<firmware file
name>;<configuration file name>
ƒ <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:
ƒ ftp://192.168.2.1 – retrieved firmware file is 310HD.img and the
configuration file name is <MAC address>.cfg. For example,
001122334455.cfg
ƒ 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
ƒ http://192.168.2.1/different_firmware_name.img;
310HD_<MAC>_conf.cfg - retrieved firmware file is
different_firmware_name.img and the configuration file name is
310HD_<MAC address>_conf.cfg. For example,
310HD_001122334455_conf.cfg

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Parameter Description

ƒ https://192.168.2.1/310HD_<MAC>_conf.cfg - In this case the retrieved


firmware file is 310HD.img and the configuration file name is 310HD_<MAC
Address>_conf.cfg. For example, 310HD_001122334455_conf.cfg
The following syntax is available for DHCP Options 66/67:
ƒ Option 66 must be a valid IP address or host name of a TFTP server only.
ƒ Option 67 must be the firmware name.
If Option 67 is absent, the phone requests for the 310HD.img image file. For
example:
ƒ Option 66: 192.168.2.1 or myTFTPServer
ƒ Option 67: 310HD_1.2.2_build_5.img
Notes:
ƒ This parameter is applicable only when method is configured to “Dynamic”.
ƒ It is recommended to leave the parameter at its default value to avoid
conflict with other DHCP options settings.
provisioning/random_pr Defines the maximum random number to start the provisioning process.
ovisioning_time
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.
provisioning/XXX_uri The URI that allows the administrator to retrieve relevant information from
separate files. The URI’s can include specific protocol and path of the
provisioning server or alternatively, the files can be retrieved from the server
whose information was provided during provisioning (e.g. DHCP options 66/67
or 160).
For example, in the below setting, YYY.ext is retrieved from the server whose
information was provided during provisioning (e.g. DHCP options 66/67 or 160:
provisioning/XXX_uri=YYY.ext
Alternatively, the full URL and protocol can be set as follows:
provisioning/XXX_uri=<protocol>://<full path>/YYY.ext
For example:
ƒ provisioning/XXX_uri=tftp://10.10.10.10/YYY.ext
ƒ provisioning/XXX_uri=ftp:// 10.10.10.10/YYY.ext
ƒ provisioning/XXX_uri=http:// 10.10.10.10/YYY.ext
ƒ provisioning/XXX_uri=https:// 10.10.10.10/YYY.ext
provisioning/ring_tone_u The URI for retrieving the ring tones file. The ring tones can be compressed to
ri *.zip or *.tgz files and provided to the phone during provisioning.
For example: provisioning/ring_tone_uri=tones.tgz
Notes:
ƒ The ringtone file is downloaded only after boot up, and not periodically.
ƒ If the tones file is new, the phone updates the information, but does not
reboot.

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Parameter Description

provisioning/corporate_d The URI for retrieving the corporate directory. The corporate directory must be
irectory_uri included in a separate file that can be downloaded to the phone during
provisioning.
For example: provisioning/corporate_directory_uri=corporate_dir.txt
Notes:
ƒ The corporate directory file is downloaded after boot up and periodically.
ƒ If the corporate directory file is new, the phone updates the information,
does not reboot.
ƒ For creating a Corporate Directory file, refer to Section 4.3.
provisioning/speed_dial_ The URI for retrieving the speed dial list. The speed dial list must be included
uri in a separate file that can be downloaded to the phone during provisioning.
For example: provisioning/speed_dial_uri=speed_dial_list.txt
Notes:
ƒ The speed dial file is downloaded after boot up and periodically.
ƒ If the speed dial file is new, the phone reboots.
ƒ For creating a Speed Dial file, refer to Section 4.4.

4.2.4 TR-069 Management Parameters


Table 4-4: TR-069 Management Configuration File Parameters

Parameter Description

Note: To add a value to these parameters, enter management/ followed by the parameter name, equal sign
and then the value (e.g. management/tr069/enabled=0).
management/tr069/enabl Enables TR-069.
ed ƒ [0] Disable (default)
ƒ [1] Enable
Note: This feature can be enabled only if the valid key is provided in the
feature_key field.
management/tr069/featur Feature key to enable the TR-069.
e_key

management/tr069/acs_u URL for connecting the device to the ACS.


rl
Note: This parameter must be in the form of a valid HTTP or HTTPS URL.
management/tr069/user_ Username used to authenticate the device when making a connection to the
name ACS.
management/tr069/pass Password used to authenticate the device when making a connection to the
word ACS.
management/tr069/conne Username to authenticate an ACS making a Connection Request to the
ction_request/user_nam device.
e
management/tr069/conne Password to authenticate an ACS making a Connection Request to the device.
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Parameter Description

management/tr069/infor Determines whether or not the device must periodically send device
m/enabled information to the ACS using the Inform method call.
ƒ [0] Disable
ƒ [1] Enable (default)
management/tr069/infor The duration (in seconds) of the interval for which the device must attempt to
m/interval connect with the ACS and call the Inform method if TR-069 is enabled
(management/tr069/inform/enabled).
The valid range is 0 to 65535. The default is 3600.

4.2.5 VoIP Parameters

4.2.5.1 Line Parameters

Table 4-5: Line Configuration File Parameters

Parameter Description

voip/line/%d/enabled Enables the line.


ƒ [0] Disable
ƒ [1] Enable
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.
voip/line/%d/description Arbitrary name to intuitively identify the line and that is displayed to remote
parties as your caller ID.
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.
voip/line/%d/auth_passwo Password provided to you from the VoIP Service Provider. This is used when
rd sending a response to Unauthorized or Proxy Authentication Requested
(401/407).
The authentication password can be up to 35 characters.
voip/line/%d/do_not_distur Activates the Do Not Disturb (DnD) feature, if enabled, using the parameter
b/activated voip/services/do_not_disturb/enabled.
ƒ [0] (default) - Activate
ƒ [1] - Deactivate
Notes: DnD can also be activated using the LCD screen interface (more
common).

Note: The variable %d refers to the line number, however, only line 0 is relevant since the
phone supports only one extension.

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4.2.5.2 Gain Parameters

Table 4-6: Gain Configuration File Parameters

Parameter Description

voip/audio/gain/additional_ Additional parameter for speaker gain configuration.


speaker_gain ƒ [0] 0dB
ƒ [1] 1dB
ƒ [2] 2dB
ƒ [3] 3dB (default)
voip/audio/gain/tone_signa Call progress tone volume. This volume can be modified on-the-fly by
l_level pressing the phone’s VOLUME key in certain scenarios.
The valid range is 1 to 31 (-dB). The default value is 10 (-10dB).
voip/audio/gain/ringer_sign Ringing tone volume. This volume can be modified on-the-fly by pressing the
al_level phone’s VOLUME key when the phone is in idle state.
The valid range is 0 to 63:
ƒ [0] - Mute
ƒ [1] - -31 dB
ƒ [32] - 0 dB
ƒ [63] - 31 dB
Hands-free Gain Parameters
Note: Values are in decibels (dB) and represented as follows:
ƒ Negative values: use the word “minus” (e.g. =minus9db).
ƒ Positive values: use the word “plus” (e.g. =plus9db).
Decimal places: use underscore instead of period (e.g. plus19_5db).
voip/audio/gain/handsfree_ Digital output gain (in dB).
digital_output_gain
The valid range is (-32) to 31 (dB), where -32 is mute. The default value is 7
(7dB).
voip/audio/gain/handsfree_ Digital input gain (in dB).
digital_input_gain
The valid range is (-32) to 31 (dB), where -32 is mute. The default value is 0
(0dB).
voip/audio/gain/handsfree_ Analog output gain (in dB).
analog_output_gain
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]
The default value is 0db.

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Parameter Description

voip/audio/gain/handsfree_ Analog input gain (in dB).


analog_input_gain
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]
The default value is plus33db.
Handset Gain Parameters
Note: Values are in decibels (dB) and represented as follows:
ƒ Negative values: use the word “minus” (e.g. =minus9db).
ƒ Positive values: use the word “plus” (e.g. =plus9db).
Decimal places: use underscore instead of period (e.g. plus19_5db).
voip/audio/gain/handset_di Digital output gain (in dB).
gital_output_gain
The valid range is (-32) to 31 (dB), where -32 is mute. The default value is 0
(0dB).
voip/audio/gain/handset_di Digital input gain (in dB).
gital_input_gain
The valid range is (-32) to 31 (dB), where -32 is mute. The default value is 0
(0dB).
voip/audio/gain/handset_a Analog output gain (in dB).
nalog_output_gain
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]
The default value is minus9db.
voip/audio/gain/handset_a Analog input gain (in dB).
nalog_input_gain
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]
The default value is plus19_5db.
voip/audio/gain/handset_a Analog side tone gain (in db).
nalog_sidetone_gain
Valid values: [minus9db], [minus12db], [minus15db], [minus18db],
[minus21db], [minus24db], [minus27db], [MUTE]
The default value is minus12db.

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Parameter Description

Headset Gain Parameters


Note: Values are in decibels (dB) and represented as follows:
ƒ Negative values: use the word “minus” (e.g. =minus9db).
ƒ Positive values: use the word “plus” (e.g. =plus9db).
Decimal places: use underscore instead of period (e.g. plus19_5db).
voip/audio/gain/headset_di Digital output gain (in dB).
gital_output_gain
The valid range is (-32) to 31 (dB), where -32 is mute. The default value is 0
(0dB).
voip/audio/gain/headset_di Digital input gain (in dB).
gital_input_gain
The valid range is (-32) to 31 (dB), where -32 is mute. The default value is 0
(0dB).
voip/audio/gain/headset_an Analog output gain (in dB).
alog_output_gain
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]
The default value is minus12db.
voip/audio/gain/headset_an Analog input gain (in dB).
alog_input_gain
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]
The default value is plus33db.
voip/audio/gain/headset_an Analog side tone gain (in db).
alog_sidetone_gain
Valid values: [minus9db], [minus12db], [minus15db], [minus18db],
[minus21db], [minus24db], [minus27db], [MUTE]
The default value is minus12db.
Automatic Gain Control (AGC)
voip/audio/gain/automatic_ Enables the AGC. AGC automatically adjusts the phone’s voice volume to
gain_control/enabled compensate for weak or loud signals.
ƒ [0] Disable (default)
ƒ [1] Enable
voip/audio/gain/automatic_ Determines whether AGC is located before the encoder input (CTL_LOCAL)
gain_control/direction or after the decoder output (CTL_REMOTE).
ƒ [CTL_REMOTE] For Remote User (default) - AGC is located after the
Decoder output
ƒ [CTL_LOCAL] For Local User - AGC is located before the Encoder input
voip/audio/gain/automatic_ The required output energy (in dBm) of the AGC.
gain_control/target_energy
The valid range is -31 to 0 (dB). The default value is -19.

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4.2.5.3 Codec Parameters

Table 4-7: Codec Configuration File Parameters

Parameter Description

voip/codec/codec_info/%d/ Determines the codecs that you want to implement and their priority. Up to
enabled five codecs can be configured, where the first codec (i.e., voip/codec/0/…)
has the highest priority. To make a call, at least one codec must be
configured. In addition, for best performance it is recommended to select as
many codecs as possible.
When you start a call to a remote party, your available codecs are compared
with the remote party's to determine the codec to use. If there is no codec
that both parties have made available, the call attempt fails. Note that if more
than one codec is common to both parties, you cannot force which of the
common codecs are used by the remote party's client. If you do wish to force
the use of a specific codec, configure the list with only that specific codec.
The %d variable stands for the priority:
ƒ [0] - Disabled
ƒ [1] (default) - Enabled
voip/codec/codec_info Name of the codec. The variable %d depicts the index number of the codec
/%d/name entry and its priority, where the first codec (i.e.
voip/codec/codec_info/0/name=…) has the highest priority. The default
order of codec priority from highest to lowest is G.722, G.711 Mu-Law,
G.711 A-Law, G.729, and then G.723.
ƒ [G722] G.722
ƒ [PCMA] G.711 A-Law
ƒ [PCMU] G.711 Mu-Law
ƒ [G729] G.729
ƒ [G723] G.723
For example, voip/codec/codec_info/0/name=G722.
voip/codec/codec_info Packetization time - length of the digital voice segment that each packet
/%d/ptime holds.
The default is 20 millisecond packets, excluding G.723 which is 30
millisecond packets.
voip/codec/g723_bitrate Low or high bit rate for G.723.
ƒ [LOW] Low
ƒ [HIGH] High (default)
voip/codec/g722_bitrate G.722 bit rate.
ƒ [G722_64K] (default)
ƒ [G722_56K]
ƒ [G722_48K]
Note: Currently, only 64bps is supported.

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4.2.5.4 Media Streaming Parameters

Table 4-8: Media Streaming Configuration File Parameters

Parameter Description

voip/media/dtmf_payload Defines the RTP payload type used for RFC 2833 DTMF relay packets.
The valid range is 96 to 127. The default value is 101.
voip/media/media_port Defines the starting port range for Real Time Protocol (RTP) voice transport.
The valid range is 1024 to 65535. The default value is 4000.
voip/media/rtp_mute_on Mute sending RTP packets to remote in HOLD state.
_hold ƒ [0] - Disabled. RTP packets are sent to remote end when in HOLD state.
ƒ [1] (default) - Enabled. RTP packets are not sent to remote end when in
HOLD state.
voip/media/out_of_band_ DTMF transport mode.
dtmf ƒ [INBAND] Inband
ƒ [RFC2833] RFC 2833 (default)
ƒ [VIA_SIP] Via SIP
Quality of Service (QoS)
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).

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4.2.5.5 Dialing Parameters

Table 4-9: Dialing Configuration File Parameters

Parameter Description

voip/dialing/timeout The duration (in seconds) of allowed inactivity between dialed digits. When you
work with a proxy, the number you have dialed before the dialing process has
timed out is sent to the proxy as the user ID to be called. This is useful for
calling a remote party without creating a speed dial entry (assuming the remote
party is registered with the proxy).
The valid range is 0 to 10. The default value is 5.
voip/dialing/phone_num The maximum length of shortcut numbers that you can enter and the maximum
ber_max_size number of digits that you can dial
The valid range is 0 to 32. The default value is 19.
voip/dialing/dial_complet Enables the feature for defining a key to indicate that dialing has completed.
e_key/enabled 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.
ƒ [0] Disable
ƒ [1] Enable (default)
Note: This parameter is available only if the parameter
voip/dialing/dial_complete_key/enabled is set to 1.
voip/dialing/dial_complet Defines the Dialing Complete key.
e_key/key
The valid value is a single character. The default value is the pound (#) key.
voip/dialing/dialtone_tim The maximum duration of the dial tone (in seconds) after which the dial tone
eout stops and a reorder tone is played.
The valid range is 0 to 300. The default value is 30.
voip/dialing/warning_ton The maximum duration of the reorder tone (in seconds) after which the reorder
e_timeout tone stops and a howler tone is played.
The valid range is 0 to 300. The default value is 40.
voip/dialing/unanswered Timeout before the phone automatically sends a Cancel message. When the
_call_timeout phone makes a call and the other side doesn’t answer, the phone sends a
Cancel after this timeout
The valid range is 0 to 300. The default value is 60.
voip/dialing/offhook_ton The duration (in seconds) of the howler tone. If the limit is exceeded, the
e_timeout howler tone stops. The howler tone indicates that the phone has been left in an
off-hook state.
The valid range is 0 to 300. The default value is 120.
voip/dialing/secondary_d Enables the secondary dial tone.
ial_tone/enabled ƒ [0] Disable - Phone doesn't use secondary dial tone.
ƒ [1] Enable (default) - Phone plays secondary dial tone if the secondary dial
tone key is pressed (first digit).
voip/dialing/secondary_d Secondary dial tone is played if this is the first key pressed.
ial_tone/key_sequence The valid range is 0 to 9. The default value is 9.
Note: This parameter is available only if the parameter
voip/dialing/secondary_dial_tone/enabled is set to 1.

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Parameter Description

voip/dialing/automatic_di Determines whether the phone automatically goes idle (i.e. on-hook) when the
sconnect last remaining call is disconnected. This is only relevant when the speaker or
headset is used.
ƒ [0] Disable
ƒ [1] Enable(default)
Automatic Dialling
voip/dialing/auto_dialing/ Determines whether automatic dialing is enabled (i.e., phone number is
enabled automatically dialed when you off-hook the phone).
ƒ [0] Disable (default)
ƒ [1] Enable
voip/dialing/auto_dialing/ Timeout (in seconds) before automatic dialing occurs after the phone is off-
timeout hooked. When set to 0, automatic dialing is performed immediately.
The valid range is 0 to 120. The default value is 15.
voip/dialing/auto_dialing/ The number that is automatically dialed when the phone is off-hooked.
destination
The valid value can be up to 32 characters.

4.2.5.6 SIP Signaling Parameters

Table 4-10: SIP Configuration File Parameters

Parameter Description

voip/signalling/sip/transp Determines the transport layer for outgoing SIP calls initiated by the phone.
ort_protocol ƒ [UDP] UDP (default)
ƒ [TCP] TCP
ƒ [TLS] TLS
voip/signalling/sip/port Defines the local SIP port (UDP, TCP or TLS) for SIP messages.
The valid range is 1024 to 65535. The default value is 5060.
voip/signalling/sip/proxy Assigns a name to the phone. The name is used as the host part of the SIP
_gateway URI in the From header.
Notes:
ƒ Ensure that the name you choose is the one with which the Proxy is
configured to identify the phone.
ƒ If not specified, the phone's IP address is used (default).
voip/signalling/sip/prack/ Determines whether the phone sends PRACK (Provisional Acknowledgment)
enabled messages upon receipt of 1xx SIP reliable responses.
ƒ [0] Disable
ƒ [1] Enable (default)
voip/signalling/sip/rport/ Determines whether the phone adds the ‘rport’ parameter to the relevant SIP
enabled message (in the SIP Via header).
ƒ [0] Disable (default)
ƒ [1] Enable

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Parameter Description

voip/signalling/sip/sdp_i Determines whether the phone adds the PTIME parameter to the SDP
nclude_ptime message body.
ƒ [0] Disable (default)
ƒ [1] Enable
voip/signalling/sip/keepa Determines whether keep-alive is performed using SIP OPTIONS messages
live_options/enabled sent to the Proxy.
ƒ [0] Disable (default)
ƒ [1] Enable
voip/signalling/sip/keepa Defines the Proxy keep-alive time interval (in seconds) between Keep-Alive
live_options/timeout messages.
The valid range is 0 to 86400. The default value is 300.
voip/signalling/sip/conne Determines whether the media is connected upon receipt of SIP 180, 183, or
ct_media_on_180 200 messages. When the parameter is disabled, media is connected upon
receipt of 183 and 200 messages only.
ƒ [0] Disable (default)
ƒ [1] Enable
voip/signalling/sip/block When enabled, the outgoing INVITE message is sent with an anonymous From
_callerid_on_outgoing_c header and P-Asserted-Identity header.
alls ƒ [0] Disable (default)
ƒ [1] Enable
For example:
ƒ FROM header contains anonymous URI:
From: "Anonymous" sip:[email protected]
ƒ P-Asserted-Identity header:
P-Asserted-Identity: "1001" [email protected]
voip/signalling/sip/anony When enabled, incoming INVITE messages with anonymous From header are
mous_calls_blocking rejected.
ƒ [0] Disable (default)
ƒ [1] Enable
For example: From:"Anonymous"<sip:[email protected]>
The phone responds with a SIP 403 “Forbidden” response.
voip/signalling/sip/auth_r Defines the number of times authenticated register messages are re-sent if
etries 401 or 407 SIP responses with a different “nonce” are received.
The valid range is 0 to 100. The default value is 4.
voip/signalling/sip/displa Sets the Display Name in the “To” and “From” fields of the SIP REGISTER
y_name_in_registration_ message.
msg/enabled ƒ [0] Disable (default)
ƒ [1] Enable

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Parameter Description

voip/signalling/sip/digit_ Enables the administrator to predefine possible formats (or patterns) for the
map 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 ’|’.
ƒ Using ‘|’ separator: The following constructs can be used in each
numbering scheme:
9 Digit: A digit from "0" to "9".
9 DTMF: A digit, or one of the symbols "A", "B", "C", "D", "#", or "*".
Extensions may be defined.
9 Wildcard: The symbol "x" which matches any digit ("0" to "9").
9 * Range: One or more DTMF symbols enclosed between square
brackets ("[" and "]").
9 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 "]".
9 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[2-9]xxxxxxxxx|91[2-
9]xxxxxxxxx|9[2-9]xxxxxx|*xx|[8]xxxx|[2-7]xxx
This example includes the following rules:
9 [2-9]11: 911 rule: 211, 311, 411, 511, 611, 711, 811, 911 are dialed
immediately
9 0: Local operator rule: After dialing "0" the phone waits T seconds and
then completes the call automatically
9 100: Auto-attendant default extension
9 101: Voicemail default extension
9 011xxx.: International rule without prefix
9 9011xxx.: International rule with prefix
9 1[2-9]xxxxxxxxx: LD rule without prefix
9 91[2-9]xxxxxxxxx: LD rule with prefix
9 9[2-9]xxxxxx: Local call with prefix
9 *xx: 2-digit star codes
9 [1-7]xx: A regular 3 digit extension that does not start with 9 or 8 is
dialed immediately
9 [2-7]xx: A regular 3 digit extension that does not start with 9 or 8 or 1 is
dialed immediately
9 [2-7]xxx: A regular 4 digit extension that does not start with 9 or 8 or 1
is dialed immediately
9 [8]xxx: A 3 digit extension prefixed with an 8 (routes calls directly to
voicemail of extension xxx)
9 [8]xxxx: A 4 digit extension prefixed with an 8 (routes calls directly to
voicemail of extension xxxx)
9 T: Refers to the Dialing Timeout.
ƒ 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.

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Parameter Description

voip/signalling/sip This parameter works in conjunction with the parameter


/number_rules 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.
SIP Proxy and Registrar
voip/signalling/sip/use_p Determines whether to use a SIP Proxy server.
roxy ƒ [0] Disable (default)
ƒ [1] Enable
voip/signalling/sip/proxy The IP address or host name of the SIP proxy server.
_address
The default value is 0.0.0.0.
voip/signalling/sip/proxy The UDP or TCP port of the SIP proxy server.
_port The valid range is 1024 to 65535. The default value is 5060.
voip/signalling/sip/proxy The SIP proxy server registration timeout (in seconds).
_timeout The valid range is 0 to 86400. The default value is 3600.
voip/signalling/sip/sip_re Determines whether the phone registers to a separate SIP Registrar server.
gistrar/enabled ƒ [0] Disable (default)
ƒ [1] Enable
voip/signalling/sip/use_p Determines whether to use the SIP proxy’s IP address and port for registration.
roxy_ip_port_for_registr When enabled, there is no need to configure the address of the registrar
ar separately.
ƒ [0] Disable (default)
ƒ [1] Enable
voip/signalling/sip/sip_re The IP address or host name of the Registrar server.
gistrar/addr
The default value is 0.0.0.0.
voip/signalling/sip/sip_re The UDP or TCP port of the Registrar server.
gistrar/port The valid range is 1024 to 65535. The default value is 5060.
voip/signalling/sip/sip_o Determines whether an outbound SIP proxy server is used (all SIP messages
utbound_proxy/enabled are sent to this server as the first hop).
ƒ [0] Disable (default)
ƒ [1] Enable
voip/signalling/sip/sip_o The IP address of the outbound proxy. If this parameter is set, all outgoing
utbound_proxy/addr messages (including Registration messages) are sent to this Proxy according
to the Stack behavior.
The default value is 0.0.0.0.
voip/signalling/sip/sip_o The port on which the outbound proxy listens.
utbound_proxy/port The valid range is 1024 to 65535. The default value is 5060.

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Parameter Description

voip/signalling/sip/redun Enables the redundant proxy mechanism.


dant_proxy/enabled ƒ [0] Disable (default) - Phone doesn't use redundant proxy (default).
ƒ [1] Enable - Phone registered to redundant proxy if the primary proxy does
not respond to SIP signaling messages.
voip/signalling/sip/redun The IP address or host name of the redundant proxy.
dant_proxy/address
The default value is 0.0.0.0.
Note: This parameter is applicable only if the parameter
use_redundant_proxy is set to 1.
voip/signalling/sip/redun The UDP or TCP port of the redundant proxy server.
dant_proxy/port
The valid range is 1024 to 65535. The default value is 5060.
Note: This parameter is applicable only if the parameter
use_redundant_proxy is set to 1.
voip/signalling/sip/redun Defines the interval in seconds for sending keep-alive messages to the proxy.
dant_proxy/keepalive_pe
The valid range is 0 to 300. The default value is 60.
riod
Note: This parameter is applicable only if the parameter
use_redundant_proxy is set to 1.

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Parameter Description

voip/signalling/sip/redun Determines the proxy redundancy mode.


dant_proxy/symmetric_ ƒ [0] (default) - Asymmetric mode.
mode
ƒ [1] - Symmetric mode.
The Redundant Proxy feature allows the configuration of a backup SIP proxy
server to increase QoS stability. Once this feature is enabled, the phone
identifies cases where the primary proxy does not respond to SIP signaling
messages. In these scenarios, the phone registers to the redundant proxy and
the phone seamlessly continues normal functionality, without the user noticing
any connectivity failure or malfunction with the primary proxy.
The Redundant Proxy feature includes two operational modes:
ƒ Asymmetric mode: the primary proxy is assigned a higher priority for
registration than the redundant proxy. In the Asymmetric mode, once the
phone is registered to the primary proxy, it sends keep-alive messages
(using SIP OPTIONS messages) to the primary proxy. If the primary proxy
does not respond, the phone registers to the redundant proxy, but
continues sending keep-alive messages to the primary proxy. If the primary
proxy responds to these keep-alive messages, the phone re-registers to the
primary proxy. Therefore, the phone assigns the primary proxy a higher
priority for registration.
ƒ Symmetric mode: both proxies are assigned the same priority for
registration. In the Symmetric mode, once the phone is registered to a
proxy, it sends keep-alive messages to this proxy. The phone switches
proxies only once the proxy to which it has registered does not respond.
Therefore, the phone assigns both proxies the same priority for registration
In both modes, the following applies:
ƒ If the phone is not registered (i.e., if the proxy server¬–redundant or
primary–to which the phone currently tries to register does not respond), the
phone attempts to register to an alternative proxy. These attempts continue
until the phone successfully registers.
ƒ If this feature is enabled and the user reboots the phone, the phone
registers to the last proxy to which it was trying to register, and not
necessarily to the primary proxy.
Note: This parameter is applicable only if the parameter
use_redundant_proxy is set to 1.

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Parameter Description

SIP Timers
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):
ƒ The first retransmission is sent after 500 msec.
ƒ The second retransmission is sent after 1000 (2*500) msec.
ƒ The third retransmission is sent after 2000 (2*1000) msec.
ƒ The fourth retransmission and subsequent retransmissions until SIPMaxRtx
are sent after 4000 (2*2000) msec..
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 200.
Note: The time interval between subsequent retransmissions of the same SIP
message starts with SipT1Rtx and is multiplied by two until SipT2Rtx.
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 1000.
voip/signalling/sip/sip_in The SIP INVITE timer according to RFC 3261.
vite_timer The valid range is 0 to 65535. The default value is 32000.
voip/signalling/sip/sessi The time (in seconds) at which an element considers the call timed out if no
on_timer 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.
voip/signalling/sip/min_s The minimum value for the session interval that the application is willing to
ession_interval accept.
The valid range is 0 to 65535. The default value is 900.
SIP QoS
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).

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4.2.5.7 Voice

Table 4-11: Voice Configuration File Parameters

Parameter Description

voip/audio/jitter_buffer/m The initial and minimal delay of the adaptive jitter buffer mechanism, which
in_delay compensates for network problems. The value should be set according to the
expected average jitter in the network (in milliseconds).
The valid range is 0 to 300. The default value is 35.
voip/audio/jitter_buffer/o The adaptation rate of the jitter buffer mechanism. Higher values cause the
ptimization_factor jitter buffer to respond faster to increased network jitter.
The valid range is 0 to 13. The default value is 7.
voip/audio/echo_cancell Enables echo cancellation.
ation/enabled ƒ [0] - Disable
ƒ [1] (default) – Enable
Note: Disabling echo cancellation should be done for testing purposes only.
voip/audio/silence_comp Enables silence compression for reducing network bandwidth consumption.
ression/enabled ƒ [0] Disable (default)
ƒ [1] Enable

4.2.5.8 Supplementary Services Parameters

Table 4-12: Supplementary Services Configuration File Parameters

Parameter Description

voip/services/application Defines the type of the Application server to which the device is registered.
_server_type ƒ [Generic] Generic (default)
ƒ [Asterisk] Asterisk
ƒ [BSFT] Broadsoft
Note: Currently, this parameter is not applicable (only Generic is supported).
voip/services/out_of_ser Determines whether a reorder tone is played instead of a dial tone if you
vice_bahavior configured a Registrar IP address and the registration failed.
ƒ [NONE] No Tone
ƒ [REORDER_TONE] Reorder Tone (default)
Call Waiting
voip/services/call_waitin Enables the Call Waiting feature.
g/enabled ƒ [0] Disable
ƒ [1] Enable (default)
voip/services/call_waitin Determines the SIP response that is sent when another call arrives while a call
g/sip_reply is in progress:
ƒ [RINGING] Ringing - 180 Ringing
ƒ [QUEUED] Queued (default) - 182 Queued
Call Forward
voip/services/call_forwar Enables the Call Forward feature.
d/line/0/enabled ƒ [0] Disable
ƒ [1] Enable (default)

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Parameter Description

voip/services/call_forwar Determines the condition upon which incoming calls are forwarded to another
d/line/0/type destination:
ƒ [Unconditional] Unconditional - incoming calls are forwarded
independently of the status of the line.
ƒ [Busy] Busy - incoming calls are forwarded only if the phone is busy.
ƒ [No_Reply] No Reply (default) - incoming calls are forwarded only if the
phone does not answer before a user-defined timeout.
voip/services/call_forwar If calls are forwarded when the condition is No-Reply, then this parameter
d/line/0/timeout defines the time (in seconds) after which incoming calls are forwarded when
this is no reply.
The valid range is 0 to 7200. The default value is 6.
voip/services/call_forwar The destination to which the call is directed when call forward is activated.
d/line/0/destination
voip/services/call_forwar Activates call forwarding, if it has been enabled (using the parameter
d/line/0/active voip/services/call_forward/line/0/enabled).
ƒ [0] (default) - Disable
ƒ [1] - Enable
Note: Call forwarding can also be activated using the LCD screen (common).
Message Waiting Indication (MWI)
voip/services/msg_waitin Enables the MWI feature.
g_ind/enabled ƒ [0] Disable
ƒ [1] Enable (default)
voip/services/msg_waitin Determines whether the phone registers to an MWI server.
g_ind/subscribe ƒ [0] Disable (default)
ƒ [1] Enable
voip/services/msg_waitin The port number of the MWI server.
g_ind/subscribe_port The valid range is 1024 to 65535. The default value is 5060.
voip/services/msg_waitin The IP address or host name of the MWI server.
g_ind/subscribe_address
The default value is 0.0.0.0.
voip/services/msg_waitin The interval between the MWI Subscribe messages.
g_ind/expiraition_timeou The valid range is 0 to 86400. The default value is 3600.
t
voip/services/msg_waitin Defines the duration for which a stutter tone is played when you have unheard
g/stutter_tone_duration messages.
The valid range is 0 to 7200. The default value is 2500.
voip/services/msg_waitin Defines the extension number for accessing your voice mail messages.
g_ind/voice_mail_numbe
The valid value is up to 64 characters.
r
Do Not Disturb (DND)
voip/services/do_not_dis Enables the Do not Disturb feature.
turb/enabled ƒ [0] Disable
ƒ [1] Enable (default)

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4.2.6 Regional Settings Parameters


Table 4-13: Regional Configuration File Parameters

Parameter Description

voip/regional_settings/se Defines the country in which your phone is located. The behavior and
lected_country parameters of analog telephones lines vary between countries. The set of Call
Progress Tones are all location-specific. The phone automatically selects the
correct regional settings according to this setting.
Supported countries:
ƒ [Israel] Israel
ƒ [China] China
ƒ [France] France
ƒ [Germany] Germany
ƒ [Netherlands] Netherlands
ƒ [UK] UK
ƒ [Brazil] Brazil
ƒ [Italy] Italy
ƒ [Argentina] Argentina
ƒ [Portugal] Portugal
ƒ [USA] USA
voip/regional_settings/us Enables the user-defined CPT. When this parameter is enabled, the
e_config_file_values selected_country parameter is not relevant and the below Call Progress
Tones values can be determined by the user.
ƒ [0] (default) - Disable
ƒ [1] - Enable
Call Progress Tones (CPT)
Note: Up to 10 CPT’s can be configured (voip/regional_settings/call_progress_tones/0…9).
voip/regional_settings/ca Enables the specific CPT.
ll_progress_tones/%d/en ƒ [0] - Disable
abled
ƒ [1] - Enable
voip/regional_settings/ca Defines the name of the CPT.
ll_progress_tones/%d/na
me
voip/regional_settings/ca Defines the cadence type of the tone.
ll_progress_tones/%d/ca ƒ [0] - Continuous signal
dence
ƒ [1] - Cadence signal
ƒ [2] - Burst signal
voip/regional_settings/ca Defines the low frequency (in Hz) of the tone.
ll_progress_tones/%d/fre
The valid value range is 300 to 1980 Hz, in steps of 1 Hz. Unused frequencies
quency_a
must be set to zero.
voip/regional_settings/ca Defines the high frequency (in Hz) of the tone.
ll_progress_tones/%d/fre
The valid value range is 300 to 3000 Hz, in steps of 1 Hz. Unused frequencies
quency_b
must be set to zero.
voip/regional_settings/ca Output level of the low frequency tone (in -dBm) in Call Progress generation.
ll_progress_tones/%d/fre
The valid range is 0 to 63, where 63 is mute.
quency_a_level

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Administrator's Manual 4. Configuration File Parameters

Parameter Description

voip/regional_settings/ca Output level of the low frequency tone (in -dBm) in Call Progress generation.
ll_progress_tones/%d/fre
The valid range is 0 to 63, where 63 is mute.
quency_b_level
voip/regional_settings/ca tone_on_0 to tone_on_3.
ll_progress_tones/%d/to
If the signal is Cadence or Burst, then this value represents the on duration. If
ne_on_0
a Continuous tone, then this value represents the minimum detection time. The
units are in 10 msec.
The valid range is 0 to 10000.
voip/regional_settings/ca tone_off_0 to tone_on_3.
ll_progress_tones/%d/to
If the signal is Cadence, then this value represents the off duration. The units
ne_off_0
are in 10 msec. If it is not used, then set it to zero. If the signal is Burst, only
tone_off 0 is relevant. It represents the off time that is required from the end of
the signal to the detection time.
The valid range is 0 to 10000.

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4.2.7 Packet Recording (Debugging) Parameters


Table 4-14: Packet Recording (Debugging) Configuration File Parameters

Parameter Description

voip/packet_recording/e Activates the packet recording mechanism.


nabled ƒ [0] Disable (default)
ƒ [1] Enable
voip/packet_recording/re The IP address (in dotted-decimal notation) of the remote computer to which
mote_ip 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.
voip/packet_recording/re Defines the UDP port of the remote computer to which the recorded packets
mote_port are sent.
The valid range is 1024 to 65535. The default value is 50000.
voip/packet_recording/rt Activates the DSP RTP recording.
p_recording/enabled ƒ [0] Disable (default)
ƒ [1] Enable
voip/packet_recording/ec Activates the Echo Canceller Debug recording.
_debug_recording/enabl ƒ [0] Disable (default)
ed
ƒ [1] Enable
voip/packet_recording/n Activates the DSP network (TDM Out) recording.
etwork_recording/enable ƒ [0] Disable (default)
d
ƒ [1] Enable
voip/packet_recording/td Activates the DSP TDM (TDM In) recording.
m_recording/enabled ƒ [0] Disable (default)
ƒ [1] Enable

4.2.8 LCD Display Parameters


Table 4-15: LCD Display Configuration File Parameters

Parameter Description

personal_settings/langua Determines the LCD display user interface language.


ge ƒ [English] English (default)
ƒ [Spanish] Spanish

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Administrator's Manual 4. Configuration File Parameters

4.3 Creating a Corporate Directory File


The Configuration file can include a link to a user-defined Corporate Directory file, using the
provisioning/corporate_directory_uri parameter. This allows you to upload a corporate directory
to the phone.
The Corporate Directory file includes a list of contacts and their phone numbers. The file must be a
simple text file that can be created using an Excel document and saved as a CSV file.
The syntax of the corporate directory file must be as follows:

<full name>,<office>,<home>,<mobile>,<user defined 1>


,<user defined 2>

For example:

John Smith,1234,98765432,574685746,8888,9999

If not all phone numbers are required, the relevant field must be left empty. For example, in the
below directory entry, the home and user-defined numbers are absent:
John Weiss,1234,,574685746

John Smith,1234,,574685746

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4.4 Creating a Speed Dial 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 dials file must be as follows:

<memory key serial number>,<memory key phone number>,<type>

The type variable represents speed dial and must be set to “0”.
For example:

1,4418,0
2,4403,0
3,039764432,0
4,4391,0
.
.
12,1234,0

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Administrator's Manual 5. Built-in Test Plan

5 Built-in Test Plan


The 310HD IP Phone provides a built-in test for verifying its operational integrity. These tests can
be performed before deployment and in cases of troubleshooting. These tests are all performed
in one continuous sequence, where each test is followed by another consecutively, in the order
listed below:

„ Keypad and off-hook/on-hook test


„ LED test
„ Handset microphone (recording/speaking) and handset receiver (playing/listening) test
„ Headset microphone (recording/speaking) and headset headphone/receiver
(playing/listening) test
„ Speaker microphone (recording/speaking) and speaker receiver (playing/listening) test
„ MAC address and serial number verification

5.1 Test Preparation


Before you can start the built-in test, you need to cable the phone as follows:
1. Connect the phone’s LAN port to a switch, using a LAN cable.
2. Ensure that the DHCP server is functioning.
3. Connect a headset to the phone.
4. Power on the phone and wait until initialization is complete.
5. From this stage onwards, follow the procedures described in the subsequent sections and in
consecutive order. In other words, continue with the procedure described in Section 5.2 on
page 82.

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310HD IP Phone

5.2 Keypad Test


The Keypad test checks the responsiveness and correct functioning of the keys on the phone’s
keypad.

¾ To test the keypad:


1. Using the keypad, dial the number “0123456789” and then press the star (“*”) key; the
phone’s LCD screen displays the following:

Figure 5-1: Testing Keypad

2. Press all keys except the keys 0-9, *, and #; for every key pressed, its corresponding
character displayed on the LCD disappears.

Figure 5-2: Testing Keypad – Characters Disappear

3. Continue with the LED test in Section 5.3 on page 82.

5.3 LED Test


Upon the successful completion of the Keypad test (in the previous section), the LCD screen
displays the message “All LEDs are on. Press #” and the phone’s single LED lights up red, as
shown in the figure below:

Figure 5-3: Successful Completion of Keypad Test

„ Continue with Handset test in Section 5.4 on page 83.

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Administrator's Manual 5. Built-in Test Plan

5.4 Handset Test


Upon successful completion of the LED test (in the previous section), perform the Handset test.
This test verifies the correct functioning of the handset, which includes the following:
„ Handset’s microphone (transmitter) for speaking
„ Handset’s receiver (speaker) for listening

¾ To test the handset:


1. Press the pound (#) key; the LCD screen displays the message ”Handset test. Off hook to
start”.

Figure 5-4: Off hook Handset Test

2. Off hook the phone; the LCD screen displays the message “Recording”.

Figure 5-5: Recording Test

3. Speak into the handset microphone; after about five seconds, the LCD screen displays the
message ”Playing”.

Figure 5-6: Playing Test

The voice that was recorded when you spoke into the handset microphone is now played
from the handset receiver.

4. Continue with the Headset test in Section 5.5 on page 84.

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310HD IP Phone

5.5 Headset Test


Upon successful completion of the Handset test (in the previous section), perform the Headset
test. This test verifies the correct functioning of the headset (for hands-free operation), which
includes the following:
„ Headset’s microphone (transmitter) for speaking
„ Headset’s headphone (receiver) for listening

¾ To test the headset:


1. After about five seconds after your recorded voice is played in the Handset test (in the
previous section), the LCD screen displays the message ”Headset test. Any key to start”.

Figure 5-7: Headset Test

2. Press any key; the LCD screen displays the message ”Recording”.

Figure 5-8: Headset Recording Test

3. Speak into the headset’s microphone; after about five seconds, the LCD screen displays
”Playing” and the voice that was recorded when you spoke into the headset’s microphone is
now played from the headset’s receiver.

Figure 5-9: Headset Playing Test

4. Continue with the Speaker test in Section 5.6 on page 85.

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Administrator's Manual 5. Built-in Test Plan

5.6 Speaker Test


Upon successful completion of the Headset test (in the previous section), perform the Speaker
test. This test verifies the correct functioning of the speakers, which includes the following:
„ Speaker microphone (transmitter) for speaking (recording)
„ Speaker receiver for listening (playing)

¾ To test the speaker:


1. After about five seconds after your recorded voice is played in the Headset test (in the
previous section), the LCD screen displays the message ”Speaker/Mic test. Any key to start”.

Figure 5-10: Speaker Test

2. Press any key; the LCD screen displays the message ”Recording”.

Figure 5-11: Speaker Recording Test

3. Speak into the phone’s speaker microphone; after about five seconds, the LCD screen
displays the message ”Playing”, and the voice that was recorded when you spoke into the
speaker’s microphone is now played from the speaker’s receiver.

Figure 5-12: Speaker Playing Test

4. Continue with the MAC Address and Serial Number Verification test in Section 5.7 on page
86.

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310HD IP Phone

5.7 MAC Address and Serial Number Verification Test


The MAC Address and Serial Number Verification test allows you to verify that the phone’s MAC
address and serial number are correct.

¾ To verify the MAC address and serial number:


1. After about five seconds after your recorded voice is played in the Speaker test (in the
previous section), the LCD screen displays a MAC address and serial number:

Figure 5-13: MAC Address and Serial Number

2. Check that the LAN MAC address and serial number are correct.
3. Press any key to exit the phone’s built-in test.

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Administrator's Manual 6. Specifications

6 Specifications
Table 6-1: 310HD IP Phone Specifications

Feature Details

VoIP Signaling Protocols ƒ SIP: RFC 3261, RFC 2327 (SDP)


Data Protocols ƒ IPv4, TCP, UDP, ICMP, ARP, DNS and DNS SRV for SIP Signaling
ƒ SIP over TLS (SIPS)
ƒ 802.1p/Q for Traffic Priority and QoS
ƒ ToS (Type of Service) field, indicating desired QoS DHCP Client
ƒ DHCP Client
ƒ NTP Client
Media Processing ƒ Voice Coders: G.711, G.723.1, G.729A/B, G.722
ƒ Acoustic Echo Cancelation: G.168-2004 compliant, 64-msec tail length
ƒ Adaptive Jitter Buffer 300 msec
ƒ Voice Activity Detection
ƒ Comfort Noise Generation
ƒ Packet Lost Concealment
ƒ RTP/RTCP Packetization (RFC 3550, RFC 3551)
ƒ DTMF Relay (RFC 2833)
Telephony Features ƒ Call Hold / Un Hold
ƒ Call Transfer
ƒ 3-way Conference (with local mixing)
ƒ Redial
ƒ Caller ID Notification
ƒ Call Waiting Indication
ƒ Message Waiting Indication (including MWI LED)
ƒ Local and Corporate Directories
ƒ Automatic On-hook Dialing
ƒ CWRR (Call Waiting Reminder Ring)
ƒ Secondary Dial Tone
ƒ Call Logs: Missed/Received Calls and Dialed Numbers
ƒ Speed Dial
ƒ Dial Plan
ƒ Call Forward (Unconditional / Busy / No answer)
ƒ Redundant SIP Proxy Mechanism
Configuration/ ƒ TR-069 for remote configuration and management
Management ƒ LCD Display User Interface Language Support (Various Languages)
ƒ Web-based Management (HTTPS)
ƒ Auto-Provisioning (via TFTP, FTP, HTTP, and HTTPS) for firmware and
configuration file upgrade
ƒ DHCP options (66, 67, and 160) for Auto-provisioning
ƒ DHCP options (12, 60, and 77) for device information

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Feature Details

Hardware ƒ LCD screen: 2 * 16 characters


ƒ Connectors interfaces:
9 2 x RJ-45 ports (10/100BaseT Ethernet) for WAN and LAN
9 RJ-9 port (jack) for Headset
9 RJ-9 port (jack) for Handset
ƒ Mounting:
9 Wall mounting
9 Adjustable angle tilt for desktop mounting
ƒ Power:
9 DC jack adapter 12V
9 Power supply AC 100 ~ 240V
9 PoE: IEEE802.3af (optional)
ƒ Keys:
9 10 x speed dial keys
9 VOICE MAIL message hotkey
9 4-way navigation keys with ENTER Key
9 MENU
9 DIRECTORY
9 REDIAL
9 HOLD
9 MUTE
9 TRANSFER
9 CONFERENCE
9 FORWARD
9 DND
9 VOLUME control key
9 HEADSET
9 SPEAKER
Headset Compatibility ƒ GN.com 2100 – both single and dual
ƒ Jabra BIZ 2400 – both single and dual
ƒ GN.com 9350
ƒ Plantronics H261N
ƒ Plantronics H251N

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Administrator's Manual 6. Specifications

Reader’s Notes

Version 1.4 89 July 2010


AudioCodes 300HD Series of High Definition IP Phones

HD VoIP 310HD IP Phone

Administrator’s Manual
310HD IP Phone
Version 1.4.0

www.audiocodes.com

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