Talkaphone VOIP-600H User Manual

VOIP-600 Series Phone

Configuration / Operation Manual

VOIP-600 Series Phone

Configuration and Operation Manual

v1.0.1

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Configuration / Operation Manual

Contents

1. Introduction ....................................................................................................................................................6

1.1. Overview of the Manual .............................................................................................................................6

1.2. Intended Audience .....................................................................................................................................6

1.3. Objective .....................................................................................................................................................6

1.4. Typographic Conventions ...........................................................................................................................6

1.5. Related Documents ....................................................................................................................................7

1.6. Technical Support / Help Desk ...................................................................................................................7

2. Overview .........................................................................................................................................................8

2.1. VOIP-600 Series Phone Overview ...............................................................................................................8

3. Getting Started ...............................................................................................................................................9

3.1. Pre-requisites..............................................................................................................................................9

3.2. Configuration Using the Web GUI ..............................................................................................................9

3.3. Configuration Using the Serial Console ................................................................................................... 10

4. Using the Web GUI ...................................................................................................................................... 12

4.1. Web GUI Layout ....................................................................................................................................... 12

4.2. Web Application Menu ............................................................................................................................ 14

4.3. Home ....................................................................................................................................................... 15

4.4. Network ................................................................................................................................................... 16

4.5. VoIP .......................................................................................................................................................... 19

4.6. Devices ..................................................................................................................................................... 27

4.7. Digital Outputs ......................................................................................................................................... 36

4.8. Voice Messages ....................................................................................................................................... 41

4.9. Self Diagnostics and Reporting ................................................................................................................ 45

4.10. Authentication ......................................................................................................................................... 47

4.11. Maintaining the Phone ............................................................................................................................ 49

5. Operations ................................................................................................................................................... 61

5.1. Calling Emergency/Information Numbers ............................................................................................... 61

5.2. Answering the Emergency/Information Call ........................................................................................... 61

5.3. Terminating Calls ..................................................................................................................................... 61

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5.4. Activating/Deactivating Auxiliary Outputs .............................................................................................. 62

5.5. Activating/Deactivating the “Help on the Way” LED............................................................................... 62

5.6. Activating/Deactivating Speaker and Microphone ................................................................................. 62

5.7. Adjusting Volume .................................................................................................................................... 62

5.8. Playing Voice Messages ........................................................................................................................... 62

5.9. Recording Voice Messages ...................................................................................................................... 63

5.10. Previewing Voice Messages ..................................................................................................................... 63

5.11. Feedback Tones ....................................................................................................................................... 64

5.12. Paging Operation ..................................................................................................................................... 64

5.13. Line Level Recording ................................................................................................................................ 64

5.14. Silent Monitoring ..................................................................................................................................... 64

5.15. Rebooting the Phone ............................................................................................................................... 65

5.16. Factory Default Settings .......................................................................................................................... 65

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Acronyms

Definitions

ADA

Americans with Disabilities Act

AEC

Acoustic Echo Cancellation

AGC

Automatic Gain Control

ANSI

The American National Standards Institute a private non-profit organization that oversees the development of voluntary consensus standards for products, services, processes, systems, and personnel in the United States.

AUX

Auxiliary Input/Output. An Auxiliary Input accepts a contact closure from an external device, such as a Vehicle Detector, Door Switch, Scream Alert™, and card swipe. An auxiliary output produces a contact closure to an external device, such as a strobe light and motorized garage gate.

BABT

British Approvals Board of Telecommunications

CNG/VAD

Comfort Noise Generator/Voice Activity Detector. It is used to reduce the transmission rate during inactive speech periods while maintaining an acceptable level of output quality.

CSA

Canadian Standards Association

The CE marking certifies that a product has met EU consumer safety, health or environmental requirements.

DHCP

Dynamic Host Configuration Protocol — protocol for assigning dynamic IP addresses to devices on a network.

DNS

Domain Name Server

DTMF

Dual Tone Multi Frequency signaling is used for telecommunication signaling over telephone lines.

FCC

Federal Communications Commission

FTP

File Transfer Protocol

GUI

Graphical User Interface

G.711

G.711 is codec also known as Pulse Code Modulation (PCM). It is the ITU-T international standard for encoding telephone audio on a 64 kbps channel.

G.723

G.723 is an ITU-T standard speech codec.

G.729

G.729 is an audio data compression algorithm. It is the ITU-T international standard for encoding telephone audio on 8 kbps channel.

IVR

Interactive Voice Response

IP-PBX

It is an IP based switch for call handling through public and private exchanges.

Internet Explorer

IETF

The Internet Engineering Task Force (IETF) develops and promotes Internet standards.

PCBA

Printed Circuit Board Assembly

PoE

Power over Ethernet, IEEE 802.3af standard.

Acronyms and Abbreviations

The following acronyms and abbreviations are commonly used throughout the document:

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Acronyms

Definitions

QoS

Quality of Service is of particular concern for the continuous transmission of high-bandwidth video and multimedia information.

SIP

Session Initiation Protocol is a signaling protocol, widely used for setting up and tearing down multimedia communication sessions over network.

TIA

Telecommunications Industry Association

Underwriters Laboratories

VoIP

Voice over Internet Protocol

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Item

Convention

Sample

Acronyms

All uppercase

SIP

Chapter titles

Title caps

See Chapter 3 Getting Started

Command-line commands and options (switches)

All lowercase, bold

ifconfig command /a option

Device names

All uppercase

VOIP-600

Directories

All lowercase

/flash

Error message names

Initial caps

Update failed

File names

Title caps (internal caps in short file names are acceptable for readability)

MainLogFile.txt BackupLogFile.txt

Menu names

Bold; title caps

Insert menu

Programs and applications

Usually title caps

HyperTerminal

Toolbar button names

Usually title caps (follow the interface); bold

Apply Reset

URLs

All lowercase; break long URLs before a forward slash, if necessary to break; do not hyphenate.

http://www.talkaphone.com/

User input

Usually title caps; bold

Enter Password

1. Introduction

1.1. Overview of the Manual

This manual provides detailed instructions for the configuration and operation of VOIP-600 Series Phones. It is recommended to read this instructional manual completely before performing any configuration.

1.2. Intended Audience

This manual is targeted towards systems administrators, or any person who would configure and maintain VOIP-600 Series Phones. Fundamental knowledge in computer networking and Voice over Internet Protocol (VoIP) technologies is recommended for understanding this manual.

1.3. Objective

This manual provides a detailed examination of the features included in VOIP-600 Series Phones. It guides an administrator through the configuration and optimization of phone features. While configuration of the VOIP-600 Series Phones is covered in detail, configuration of other peripheral VoIP network elements is beyond the scope of this document.

1.4. Typographic Conventions

The following guidelines are used as typographic conventions in this user manual:

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1.5. Related Documents

VOIP-600 Quick Installation Guide, v1.0 VOIP-600 Avaya IP Office Integration Guide, v1.0 VOIP-600 Cisco UCM Integration Guide, v1.0 VOIP-600 Alcatel-Lucent OmniPCX Integration Guide, v1.0

1.6. Technical Support / Help Desk

For technical assistance beyond the scope of this document, contact your distributor or Talkaphone Technical Support for further information.

Talk-A-Phone Co. 7530 North Natchez Avenue Niles, Illinois 60714

Phone: 773.539.1100 Fax: 773.539.1241

E-mail: support@talkaphone.com Web: www.talkaphone.com

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2. Overview

2.1. VOIP-600 Series Phone Overview

The VOIP-600 series phones are outdoor-rated, ADA-compliant hands-free Voice over IP (VoIP) Emergency/Information Phones for use in locations such as: parking facilities, college campuses, medical centers, and industrial parks. VOIP-600 series phones are SIP1 compliant.

The VOIP-600 series of phones are categorized into various device modes based on the number of buttons,

keypad, handset and speaker/microphone present on the phone’s faceplate.

These device modes are also given different marketing names. A model with “Keypad” is denoted with a letter “K” in its model number. A model with two buttons is denoted with a letter “D” in its model number. For

example, VOIP-600 corresponds to “Single button Emergency VoIP phone”. VOIP-600D is “Dual button Emergency VoIP phone”. VOIP-600K is “Two-button Emergency VoIP Phone with Keypad”.

Some of the functionality described below may not be available depending on the phone device mode. The rest of the document may use the term “phone” to refer to a “VOIP-600 series” phone.

NOTE: Not all features of the VOIP-600 series phones are available when used in conjunction with analog telephony adapter devices or with the Public Switched Telephone Network (PSTN).

RFC 3261 and RFC 2833 for DTMF delay

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3. Getting Started

3.1. Pre-requisites

Prior to configuring a VOIP-600 Series Phone, ensure the unit is powered on and connected to the network. The VOIP-600 can be configured from a computer with either a TCP/IP network connection or a DB-9 Serial

port. The VOIP-600 supports straight-through serial connections for basic programming. For access to the full configurable feature set, a modern web browser will be required.

The VOIP-600 currently supports access from:

 Internet Explorer 8 or greater  Firefox 3.5 or greater

3.2. Configuration Using the Web GUI

1. Ensure both the VOIP-600 and your PC are connected to the Local Area Network. A direct

connection to the VOIP-600 will require the use of a crossover network cable.

VOIP-600 Series Phones are pre-configured with the following default settings:

IP Address: 192.168.1.10 Username: admin Password: admin@123

2. Configure the IP address of your PC to be on the same subnet as the VOIP-600. For example, 192.168.1.3

3. Open a supported web browser and direct it to the IP address of the VOIP-600. For example, enter the

following URL: http://192.168.1.10. The browser prompts for authentication:

Figure 1 Authentication Required

4. Enter the default Username and Password. After authentication is successful, you are redirected to

the Home page.

5. Further configuration of VOIP-600 settings are examined in Section 4: Using the Web GUI.

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3.3. Configuration Using the Serial Console

Basic settings can be configured on VOIP-600 Series Phones through a serial console connection. Knowledge of Linux shell commands is recommended for serial console configuration.

Most commonly, the serial console connection can be used to set the IP address, as described below.

1. Connect a straight-through serial cable (DB9 Male to DB9 Female) from the serial port on the phone to an

available COM port on the PC, noting which port was utilized.

2. Open your preferred serial console application (e.g., HyperTerminal or TeraTerm) on the PC and

specify the following settings:

Communications Port: <Determined in Step 1> Baud rate (bits per second): 115200 Data bits: 8 Parity: None Stop bits: 1 Flow control: None

3. Once connected, you will be presented with the following prompt:

root:/>

4. The /flash directory contains persistent files. It is highly recommended to switch to the /flash directory

by entering the following command, and pressing Enter.

Note the space between ‘cd’ and ‘/flash’:

root:/> cd /flash

5. Create a file where configuration parameters can be entered. For example, we create the config.txt

file by entering the following command, and pressing Enter.

Note the space between ‘vi’ and ‘config.txt’:

root:flash/> vi config.txt

6. While in the vi application, press i to force vi into “Insert Mode.”

7. The body of the config.txt file can now be created.

To set a new static IP address enter the following data: NetworkMode = 3 IPAddress = <IP address, e.g. 192.168.1.10> Netmask = <netmask, e.g. 255.255.255.0> Gateway = <default gateway, e.g. 192.168.1.1>

8. Once the configuration options are entered, save the changes and exit vi by pressing Escape, then

entering :wq

To exit without saving changes press Escape, then enter :q!

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9. With the configuration file created, the final step is to apply these settings to the VOIP-600. Enter the

following command, and press Enter.

root:flash/> configApp admin admin@123 config.txt

10. A reboot is required after serial console configuration. Enter the following command, and press Enter.

root:flash/> reboot

11. All configurable options are available in Appendix E: Sample Serial Configuration File.

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4. Using the Web GUI

This section describes various configurable options available from the Web Interface of the phone. It is the primary interface for configuring and monitoring the phone. You must assign an IP address (Refer to 4.4.1 IP Settings) before configuring the features of the phone.

4.1. Web GUI Layout

This section describes the layout of the Web GUI.

Figure 2 Using Web GUI

The Web GUI layout is divided into four sections:

1. Header: The Header section displays Talkaphone Logo and product title “VoIP Phone”. The right side of

the header has the Refresh, Help, and Logout buttons. The Refresh button is used to reload the content section. The Help button provides help content from the Talkaphone website. The Logout button logs out the user. The Apply button will not be grayed out if the content pane (3) contains configurable parameters. The Apply button is used to save configuration changes to phone.

2. Menu: The menu page provides links to different menus and submenus available on the phone.

3. Content: The top part of the content section is used for Page Title (3a) and Status (3b) for any

operation. Bottom part of content pane (3c) is used for displaying configurable fields when a menu or a submenu is selected. All the fields (text boxes/dropdown) display allowed field ranges.

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4. Footer: The Footer section displays the Talkaphone copyright statement.

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To update the phone configuration, the user must press the Apply button. After clicking Apply the status will

be displayed in the “Status” section of the Content section (3b).

Note: When a configuration change is in the process of being applied, an message appears. An message appears for the successful update.

An message appears if the input is incorrect. Incorrect inputs are marked with red.

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4.2. Web Application Menu

This section provides a preview of the phone’s Web Menu hierarchy/organization.

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Figure 3 Application Menu

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Field Name

Description

Device Mode

This indicates the type of device. For example, 2BHSK indicates a device having two buttons, a handset, a keypad, and a speaker. Refer to Appendix D: VOIP-600 Device Modes for details on other device modes.

Firmware Version

Indicates the version of the phone’s firmware

Bootloader version

Indicates the version of the bootloader that is used during startup (booting) of the phone. The bootloader version is independent of the firmware version.

MAC Address

Indicates the number assigned as the phone’s unique MAC address.

4.3. Home

The Home page displays phone information and contact information for Talkaphone.

Figure 4 Home

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4.4. Network

4.4.1. IP Settings

The IP Settings page allows you to configure the IP settings and the hostname of the phone.

Figure 5 IP Settings

To set IP settings:

1. Select DHCP – Automatic Configuration OR Static IP - Manual Configuration.

2. If DHCP is selected for configuration of the IP address, then the VOIP-600 should be rebooted after

clicking ‘Apply’.

3. Enter network details IP Address, Subnet Mask, Default Gateway of your network for Static IP -

Manual Configuration.

4. Enter DNS Server to resolve domain names.

5. Enter Hostname. Use unique identifier for each device on the network.

6. Click Apply to save settings.

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4.4.2. SIP Settings

This page is used to configure the phone number, SIP registrars and outbound proxy details.

Figure 6 SIP Settings

To configure SIP settings:

1. Enter the Phone Number (number used to call from other SIP endpoints).

2. Enter the Fully Qualified Domain Name or the IP Address of the SIP Gateway/Proxy. Secondary

and Tertiary FQDN or IP Addresses can be assigned for failover. There is a 30 character limit for this field. If Secondary and Tertiary Registrars are not used, then voip500.local is the default placeholder.

3. Check or Uncheck Register to enable or disable the SIP registration.

4. Enter SIP registrar details: (Secondary and Tertiary details can be assigned for failover)

 Username: This is an optional field however it is required when Authentication is enabled on the

SIP registrar. Valid entries include alphanumeric and special characters (‘*’, ’.’, ’-‘, No spaces allowed)

 Password: This is an optional field however it is required when Authentication is enabled on the

SIP registrar. Valid entries include alphanumeric and special characters (‘*’, ’.’, ’-‘, No spaces allowed)

 IP Address and Port: Configure the IP Address and Port of the SIP registrar on which you want

to register the phone. Default port is 5060. Valid port range is 1024-49151.

 Re-registration Time: Configure the registration refresh time. On expiration of this time interval,

the phone automatically sends a new register request to the registrar. Default re-registration time

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is 3600 seconds. Valid range is 10-14400 seconds.

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5. Enter Outbound Proxy details: If a separate outbound proxy is not being used please enter the SIP

registrar information again. (Secondary and Tertiary details can be assigned for failover)

 Username: This is an optional field however it is required when Authentication is enabled on the

SIP registrar. Valid entries include alphanumeric and special characters (‘*’, ’.’, ’-‘, No spaces allowed)

 Password: This is an optional field however it is required when Authentication is enabled on the

SIP registrar. Valid entries include alphanumeric and special characters (‘*’, ’.’, ’-‘, No spaces allowed)

 IP Address and Port: Configure the IP Address and Port of the SIP registrar on which you want

to register the phone. Default port is 5060. Valid port range is 1024-49151.

 Re-registration Time: Configure the registration refresh time. On expiration of this time interval,

the phone automatically sends a new register request to the registrar. Default re-registration time is 3600 seconds. Valid range is 10-14400 seconds.

6. Click Apply. The following type of registration status appears based on your selection:

 Registering as <Phone Number> @ <Domain Name> - Registration is in progress with SIP registrar.  Registered as <Phone Number> @ <Domain Name> - Registration is completed and phone is able

to receive incoming calls on the registered phone number.

 Unregistered as <Phone Number> @ <Domain Name> - Phone is not registered for incoming calls.

Outbound calls can still be established via outbound proxy.

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4.5. VoIP

4.5.1. Number Lists

A phone can have seven lists of phone numbers. Each list can contain up to six numbers for round robin dialing. If the first number dialed is busy or does not answer after a specified number of rings, the phone dials the next number in the phone number list. The phone continues trying the numbers in the list in order until it reaches the last number programmed in the list. If the last phone number is also busy or does not answer after the specified number of rings, the phone should go back to dialing the first number, then the second number, etc. until the call is completed, or until the time limit defined by Call Conversation Timer expires.

The numbers from these lists are dialed on call initiation when the associated buttons are pressed or when the associated Auxiliary Inputs are activated (Refer to 4.6.5 Auxiliary Inputs).

Figure 7 Number Lists

To set numbers in the number lists:

1. Use the drop-down menu to select the number list to be configured (List1 is the default list assigned to

Button 1.)

2. Provide up to six numbers. Numbers can be in the format of plain numbers/aliases or as number@ip

address. Valid range is up to 30 characters.

3. Click Apply to save settings.

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4.5.2. Phone Settings

The Phone Settings page allows you to configure the VoIP protocol for communication, auto answer mode of operation, prioritization between phone and paging operation, and caller ID settings.

Figure 8 Phone Settings

To configure the phone settings:

1. Select Enable if you want the VOIP-600 to answer any incoming calls automatically (only applicable to

VOIP-600 phones with a built-in hands-free speaker). When this option is set to Disable, incoming calls are answered by either picking up the handset or by

pressing any button configured as a hook switch. If the VOIP-600 does not have a handset option and cannot be configured with a hook switch button, a manual answer method will not be available.

2. Select and assign the priority between Phone Mode and Paging Mode. When the “phone mode” has

priority over the “paging mode,” the incoming paging request is processed when the phone is in idle

state. When the phone mode has a higher priority, all paging operations stop and phone operations are processed. When the paging mode has a higher priority, all phone operations stop when the paging mode has a higher priority and the paging request is processed.

3. Select Enable or Disable to transmit a Caller ID over the network or to peers. This field allows

alphanumeric characters only. Some phone systems require Caller ID assignment from within the phone system management.

4. Click Apply to save settings.

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4.5.3. Audio Settings

The Audio Settings page allows you to configure settings for preferred VoIP codec, enable/disable audio processing modules, outbound DTMF duration, line level output gain, and the microphone and speaker parameters.

Figure 9 Audio Settings

To configure audio settings:

1. Select G.711 PCM a-Law @ 64kbps, G.711 PCM u-Law @ 64kbps, G.729A or G.723.1a for the

preferred VoIP codec. The selected codec is used for outgoing call negotiation. (Default is G.711 PCM uLaw @ 64kbps.)

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2. Check or uncheck VAD/CNG, AEC, or AGC to enable/disable the audio processing modules.

 VAD/CNG (Voice Activity Detection / Comfort Noise Generation) is enabled by default. VAD/CNG is

used to reduce background noise and also produces as low but audible comfort noise.

 AEC (Acoustic Echo Cancellation) is enabled by default. AEC is used to remove the echo from voice

communications in a speakerphone environment.

 AGC (Automatic Gain Control) is disabled by default. AGC can be used where audio input levels

vary widely in an effort to enhance/reduce the audio input to the phone.

3. Select the Outbound DTMF2 duration of 51 msec, 60 msec, 102 msec, or Custom to be used for

sending DTMF packets during a call. Select Disable if Outbound DTMF is not required. This

configuration option is only available for a device mode with a “K” designator (e.g., 1BKH, 4BHSK).

4. Click the drop-down menu to select Line Gain for the Line Level Output parameter. Each level

increases/decreases the line gain in steps of approximately 3dB.

5. Enable or disable the Speaker and/or Microphone. Configuration changes made to the Speaker

parameters will affect the built-in hands-free speaker and the optional handset speaker. Configuration changes made to the Microphone parameters will affect the built-in hands-free microphone and the optional handset microphone.

6. The Speaker Gain and the Microphone Gain can be individually adjusted. Each level

increases/decreases gain in steps of approximately 3dB.

7. During an active call, the Speaker/Microphone gains can be adjusted during an active call using

DTMF Operation codes (Refer to Appendix A: Remote Operation Key Codes – Quick Reference: Speaker Enable/Disable and Microphone Enable/Disable). Gain adjustments made via DTMF only takes effect for the instance of the phone call where the DTMF Operation Code was applied.

8. Click Apply to save settings.

Out-of-band RTP

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4.5.4. Call Parameters

The Call Parameters page is used to configure the mandatory and the optional call related parameters such as call progress tones, answer/disconnect key, welcome tone, and timers.

Figure 10 Call Parameters

To set the call parameters:

1. Select Enable/Disable to allow/mute call progress tones to be heard by the user until the call is answered

by the remote side.

2. Click the drop-down menu to set a key to answer a call and/or disconnect a call from the remote side.

You can assign a key within the range of 0-9, *, #, or Disable.

3. Select Enable or Disable to enable or disable the “Welcome Tone”. When enabled, a tone is played

to the remote side when a call is placed to the VOIP-600 phone that auto-answers the call. The “Welcome Tone” assists with distinguishing ETP-400 Series Phones from the VOIP-600 Series Phones.

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4. Configure required timers:

 Provisional Timer is the time interval for which the phone waits for the call initiation

acknowledgement from the remote side. This timer is started when the call is initiated. This timer is useful in situations where the remote phone is unreachable for unknown reasons. The default value is set to five seconds. This timer is used only during round robin dialing.

 Ringer Timer is the number of rings (one ring = six seconds) the phone waits before dialing the next

number. The default value is five rings. This timer is used only during round robin dialing.

 Hang-up Timer is the time interval for which the phone waits before initiating a new call after

hanging up the current call. The default value is 0.5 seconds.

 Local Inter-digit Timer is the time interval between two consecutive dialed digits. The inter-digit

timer is started when the first digit is entered and is restarted after each new digit is entered. When the user dials a number, a call gets initiated after the expiry of this timer. This timer also applies during local key code translations for performing the operations. The default value is five seconds.

 Remote Inter-digit Timer is the time interval for which the phone is expecting a digit to be received

from remote side during key press to answer and for DTMF Operation Codes3 that require remote key codes for activation/deactivation. The default value is five seconds.

5. Configure optional timers:

 Call Conversation Timer is the time interval after which the call is terminated automatically. The

default value is 12 minutes. If the call conversation time is disabled the conversation lasts for an indefinite duration.

 Ringback or Busy Timer is the time interval for which the phone waits on the call queue before

dialing the next number from the number list. The default value is 15 seconds. The phone waits for indefinite time if this timer is disabled. This timer is used only during round robin dialing.

 Hang-up on Silence Timer is the time interval after which a call is terminated automatically if the

voice or audio channel is idle. The default value is 30 seconds.

6. Click Apply to save settings.

For example, the remote key code sequence 123 is used for activating Aux output #1.During a call, the remote side presses 12, now the phone is expecting key 3 to be received before the expiry of this timer, if key 3 is pressed before expiry of this timer, a single beep is heard at the remote side followed by Aux Output #1 Activation at the local end. If not, the phone considers this as an unsuccessful operation and will play two beeps to the remote side.

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4.5.5. Paging Settings

The VOIP-600 Series Phone can be used as a paging device by a WEBS Contact® Server. The page can be output through either the built-in hands-free speaker or the line level output.

Figure 11 Paging Settings

To configure paging settings:

1. Select Speaker or Line Level Output to send the incoming audio page to the built-in hands-free speaker

or the line level output. The speaker option is only available for device modes with an “S” designator (e.g.,

1BS, 4BHSK).

2. Click the drop-down menu to adjust the Output Gain for all incoming pages before sending to the

selected audio paging output. Each level adjustment corresponds to approximately 3dB.

3. Click ‘Yes’ to Automatically activate paging for inbound phone calls to the VOIP-600.

4. Enter the Keycodes to activate and deactivate paging during an inbound phone call to the VOIP-

600.

5. The phone displays Unregistered or Registered @ <WEBS_Contact_Server_IP_Address> for

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Registration Status on the WEBS Contact® Server.

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6. If it deregistration is required from the WEBS Contact® Server in error situations where the WEBS

Contact® Server is unable to deregister the device, select Deregister check box to clear the registration status.

7. Click Apply to save settings.

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2(i)

2(ii)

2(iii)

4.6. Devices

4.6.1. Buttons

The Buttons configuration page allows for programming of the buttons depending on the device mode.

Figure 12 Buttons

To configure buttons:

1. Select the button mode for each button you want to configure. The mode for Button #1 is not configurable

and is always set to Autodial. Buttons #2, #3, and #4 can be configured to Autodial or Hook-switch. Optionally, Buttons #3 and #4 can

be configured as Volume Up/Volume Down/Play Voice Message. The number of buttons available for configuration will be based on the Device Mode (e.g., Device Mode 2BS will have Button #1 and #2 available for configuration).

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Configuration / Operation Manual

2. When a Button is configured for Autodial mode, configure the following:

i. Use the drop-down menu to assign the Number List. Phone Numbers from the Number List are

dialed in round robin fashion when the corresponding button is pressed.

ii. Select the Call Priority. This parameter assigns a priority to the call that is generated when the

corresponding button is pressed. A call with a “Call Priority” of 1 terminates all the existing calls

having the call priority in the range of 2–10.

iii. Select the Network Priority4 to assign a network priority to the placed call. The value is utilized by

the network to prioritize voice traffic. The default value is 46.

3. Click Apply to save settings.

Note Button #1 has the highest call priority and this cannot be changed. Pressing Button #1 cancels all

operations and places a call.

Network priority is the same as Differentiated Services Code Point (DSCP). Do not change this unless absolutely required as it might degrade the voice quality of the phone.

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Configuration / Operation Manual

2 4 3

4.6.2. Hook Switch

The Hook switch configuration page configures the numbers to be dialed when the handset goes off-hook. This configuration page is only available for the 0BH device mode.

Figure 13 Hookswitch

To configure the Hook switch:

1. Click the drop-down menu to assign a Number List. Phone Numbers from the Number List are dialed in

round robin fashion when the handset goes off-hook.

2. Select the Call Priority. This parameter assigns a priority to the call that is generated when the

handset goes off-hook. A call with a “Call Priority” of 1 terminates all the existing calls having the call priority in the range of 2–10.

3. Select the Network Priority5 to assign a network priority to the placed call. The value is utilized by

the network to prioritize the voice traffic. The default value is 46.

4. Click Apply to save settings.

Network priority is the same as Differentiated Services Code Point (DSCP). Do not change this unless absolutely required as it might degrade the voice quality of the phone.

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Talkaphone VOIP-600H User Manual

Specifications and Main Features

Frequently Asked Questions

User Manual