Yealink | SIP-T4X | Specifications | Yealink SIP-T4X Specifications

v
Copyright © 2014 YEALINK NETWORK TECHNOLOGY CO., LTD.
Copyright © 2014 Yealink Network Technology CO., LTD. All rights reserved. No parts of this
publication may be reproduced or transmitted in any form or by any means, electronic or
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damages arising from use of an illegally modified or altered publication.
THE SPECIFICATIONS AND INFORMATION REGARDING THE PRODUCTS IN THIS GUIDE ARE
SUBJECT TO CHANGE WITHOUT NOTICE. ALL STATEMENTS, INFORMATION, AND
RECOMMENDATIONS IN THIS GUIDE ARE BELIEVED TO BE ACCURATE AND PRESENTED
WITHOUT WARRANTY OF ANY KIND, EXPRESS OR IMPLIED. USERS MUST TAKE FULL
RESPONSIBILITY FOR THEIR APPLICATION OF PRODUCTS.
YEALINK NETWORK TECHNOLOGY CO., LTD. MAKES NO WARRANTY OF ANY KIND WITH
REGARD TO THIS GUIDE, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE. Yealink Network Technology
CO., LTD. shall not be liable for errors contained herein nor for incidental or consequential
damages in connection with the furnishing, performance, or use of this guide.
Hereby, Yealink Network Technology CO., LTD. declares that this phone is in conformity
with the essential requirements and other relevant provisions of the CE, FCC.
This device is marked with the CE mark in compliance with EC Directives 2006/95/EC and 2004/108/EC.
This device is compliant with Part 15 of the FCC Rules. Operation is subject to the following two conditions:
1. This device may not cause harmful interference.
2. This device must accept any interference received, including interference that may cause undesired
operation.
Note: This equipment has been tested and found to comply with the limits for a Class B digital device,
pursuant to part 15 of the FCC Rules. These limits are designed to provide reasonable protection against
harmful interference in a residential installation. This equipment generates, uses and can radiate radio
frequency energy and, if not installed and used in accordance with the instructions, may cause harmful
interference to radio communications. However, there is no guarantee that interference will not occur in a
particular installation. If this equipment does cause harmful interference to radio or television reception,
which can be determined by turning the equipment off and on, the user is encouraged to try to correct the
interference by one or more of the following measures:
1. Reorient or relocate the receiving antenna.
2. Increase the separation between the equipment and receiver.
3. Connect the equipment into an outlet on a circuit different from that to which the receiver is connected.
4. Consult the dealer or an experienced radio/TV technician for help.
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electrical and electronic equipment should understand the meaning of the crossed-out
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collect such WEEE separately.
We are striving to improve our documentation quality and we appreciate your feedback. Email
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vii
Yealink SIP-T42G IP phone firmware contains third-party software under the GNU General Public License
(GPL). Yealink uses software under the specific terms of the GPL. Please refer to the GPL for the exact terms
and conditions of the license.
The original GPL license, source code of components licensed under GPL and used in Yealink products can
be downloaded online:
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About This Guide
Thank you for choosing the SIP-T42G IP Phone, an ultra-elegant gigabit IP Phone which is
exquisitely designed to provide business telephony features, such as Call Hold, Call
Transfer, Busy Lamp Field, Multicast Paging and Conference over an IP network.
This guide provides everything you need to quickly use your new phone. First, verify
with your system administrator that the IP network is ready for phone configuration. Also
be sure to read the Packaging Contents and Regulatory Notices sections in this guide
before you set up and use the SIP-T42G IP phone.
Shared Line, Network Directory and Network Call Log features are hidden for IP phones
in the neutral firmware, which are designed for the BroadWorks environment. Please
contact your system administrator for more information.
Note
Topics provided in this guide include:

Chapter 1 Overview

Chapter 2 Getting Started

Chapter 3 Customizing Your Phone

Chapter 4 Basic Call Features

Chapter 5 Advanced Phone Features
This section describes the changes to this guide for each release and guide version.
The following section is new:

Optional Accessories on page 10
Major updates have occurred to the following sections:

DSS Keys on page 46

Appendix A - Time Zones on page 133
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User Guide for the SIP-T42G IP Phone
The following sections are new:

BLF List on page 101

Call Completion on page 74
Major updates have occurred to the following sections:

LED Instructions on page 4

Ring Tones on page 26

Anonymous Call on page 96

Anonymous Call Rejection on page 97

Multicast Paging on page 108

Troubleshooting on page 121

Appendix A - Time Zones on page 133
Major updates have occurred to the following sections:

Documentations on page 8

Packaging Contents on page 9
The following section is new:

Call Completion on page 74
Major updates have occurred to the following sections:

LED Instructions on page 4

Backlight on page 19

Anonymous Call on page 96
Major updates have occurred to the following sections:
vi

Icon Instructions on page 3

DSS Keys on page 46

Dial Plan on page 58
About This Guide

Anonymous Call on page 96

Troubleshooting on page 121
Major updates have occurred to the following sections:

Phone Status on page 14

Directory on page 28

Remote Phone Book on page 39
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User Guide for the SIP-T42G IP Phone
viii
Table of Contents
About This Guide ...................................................................... v
In This Guide .................................................................................................................... v
Summary of Changes ...................................................................................................... v
Changes for Release 73, Guide Version 73.40 ............................................................ v
Changes for Release 73, Guide Version 73.16 ........................................................... vi
Changes for Release 72, Guide Version 72.2 ............................................................. vi
Changes for Release 72, Guide Version 72.1 ............................................................. vi
Changes for Release 71, Guide Version 71.120 ......................................................... vi
Changes for Release 71, Guide Version 71.90 .......................................................... vii
Table of Contents ..................................................................... ix
Overview .................................................................................. 1
Hardware Component Instructions ................................................................................. 1
Icon Instructions ............................................................................................................... 3
LED Instructions ................................................................................................................ 4
User Interfaces ................................................................................................................. 5
Phone User Interface .................................................................................................... 6
Web User Interface ....................................................................................................... 6
Documentations ............................................................................................................... 8
Getting Started ......................................................................... 9
Packaging Contents ......................................................................................................... 9
Optional Accessories.................................................................................................. 10
Phone Installation........................................................................................................... 11
Phone Initialization ......................................................................................................... 14
Phone Status................................................................................................................... 14
Basic Network Settings .................................................................................................. 16
Registration .................................................................................................................... 17
Idle Screen ..................................................................................................................... 17
Customizing Your Phone ......................................................... 19
General Settings ............................................................................................................ 19
Backlight...................................................................................................................... 19
Language .................................................................................................................... 20
Time & Date................................................................................................................. 20
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User Guide for the SIP-T42G IP Phone
Administrator Password ............................................................................................. 22
Key as Send ................................................................................................................ 23
Phone Lock .................................................................................................................. 23
Audio Settings ................................................................................................................ 26
Volume......................................................................................................................... 26
Ring Tones ................................................................................................................... 26
Contact Management ................................................................................................... 28
Directory ...................................................................................................................... 28
Local Directory ............................................................................................................ 30
Blacklist........................................................................................................................ 38
Remote Phone Book .................................................................................................... 39
Call History Management ............................................................................................. 42
System Customizations .................................................................................................. 43
Headset Use ............................................................................................................... 43
DSS Keys ..................................................................................................................... 46
Account Management ............................................................................................... 56
Dial Plan ...................................................................................................................... 58
Emergency Number.................................................................................................... 63
Live Dialpad ................................................................................................................ 64
Hotline ......................................................................................................................... 65
Basic Call Features ................................................................. 67
Placing Calls ................................................................................................................... 68
Answering Calls ............................................................................................................. 70
Ending Calls ................................................................................................................... 71
Redialing Numbers ........................................................................................................ 71
Recent Call In Dialing .................................................................................................... 72
Auto Answer ................................................................................................................... 73
Auto Redial ..................................................................................................................... 73
Call Completion ............................................................................................................. 74
Recall .............................................................................................................................. 75
Call Mute ........................................................................................................................ 76
Call Hold/Resume .......................................................................................................... 76
Do Not Disturb (DND) .................................................................................................... 77
Call Forward................................................................................................................... 81
Call Transfer ................................................................................................................... 86
Call Waiting .................................................................................................................... 87
Conference..................................................................................................................... 88
Local Conference........................................................................................................ 88
Network Conference .................................................................................................. 89
Call Park ......................................................................................................................... 91
Call Pickup...................................................................................................................... 92
Anonymous Call ............................................................................................................. 96
Anonymous Call Rejection ............................................................................................. 97
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Table of Contents
Advanced Phone Features ..................................................... 99
Busy Lamp Field (BLF).................................................................................................... 99
BLF List .......................................................................................................................... 101
Call Recording.............................................................................................................. 103
Hot Desking .................................................................................................................. 105
Intercom ....................................................................................................................... 106
Outgoing Intercom Calls .......................................................................................... 106
Incoming Intercom Calls ........................................................................................... 107
Multicast Paging .......................................................................................................... 108
Sending RTP Stream ................................................................................................. 109
Receiving RTP Stream ............................................................................................... 112
Music on Hold .............................................................................................................. 113
Automatic Call Distribution (ACD) .............................................................................. 114
Messages ..................................................................................................................... 116
Voice Mail.................................................................................................................. 116
Message Waiting Indicator (MWI) .......................................................................... 117
Troubleshooting .....................................................................121
Regulatory Notices ................................................................129
Service Agreements .................................................................................................... 129
Limitations of Liability .................................................................................................. 129
Safety Instructions ........................................................................................................ 129
Appendix A - Time Zones ......................................................133
Index ......................................................................................137
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User Guide for the SIP-T42G IP Phone
xii
Overview
This chapter provides the overview of the SIP-T42G IP phone. Topics include:

Hardware Component Instructions

Icon Instructions

LED Instructions

User Interfaces

Documentations
If you require additional information or assistance with your new phone, contact your
system administrator.
The main hardware components of the SIP-T42G IP phone are the LCD screen and the
keypad.
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User Guide for the SIP-T42G IP Phone
Hardware component instructions of the SIP-T42G IP phone are:
Item
Description
Shows information about calls, messages, soft keys,
time, date and other relevant data:
• Call information—caller ID, call duration
①
LCD Screen
• Icons (for example,
)
• Missed call text or second incoming caller information
• Prompt text (for example, "Saving config file!")
• Time and date
②
Power Indicator LED
③
Line Keys
Indicates phone power and some feature statuses.
Use these keys to activate up to twelve accounts and
assign various features.
Mute Key
Toggles and indicates mute feature.
Headset Key
Toggles and indicates the headset mode.
Message Key
Accesses voice mails.
Redial Key
Redials a previously dialed number.
⑤
Speakerphone Key
Toggles the hands-free speakerphone mode.
⑥
Volume Key
⑦
Keypad
④
Adjusts the volume of the handset, headset, speaker,
and ringer.
Provides the digits, letters, and special characters in
context-sensitive applications.
Scroll through the displayed information.
⑧
Confirms actions or answers incoming calls.
Cancels actions or rejects incoming calls.
⑨
Soft Keys
⑩
Speaker
Label automatically to identity their context-sensitive
features.
Provides ringer and hands-free (speakerphone) audio
output.
Secures the handset in the handset cradle when the IP
⑪
Hookswitch Tab
phone is mounted vertically. For more information on
how to adjust the hookswitch tab, refer to Yealink Wall
Mount Quick Installation Guide for T4X IP Phones .
Picking up the handset from the handset cradle, the
⑫
Hookswitch
hookswitch bounces and the phone connects to the line,
laying the handset down on the handset cradle, the
phone disconnects from the line.
2
Overview
Icons appearing on the LCD screen are described in the following table:
Icon
Description
Network is unavailable
Registered successfully
Register failed
Registering
Hands-free speakerphone mode
Handset mode
Headset mode
Voice Mail
Auto Answer
Do Not Disturb
Call Hold
Call Mute
Ringer volume is 0
Phone Lock
Alphanumeric input mode
Numeric input mode
Multi-lingual lowercase letters input mode
Multi-lingual uppercase letters input mode
Multi-lingual uppercase and lowercase letters
input mode
Call Forward/Forwarded Calls
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User Guide for the SIP-T42G IP Phone
Icon
Description
Missed Calls
Received Calls
Placed Calls
Recording box is full
A call cannot be recorded
Recording starts successfully
Recording cannot be started
Recording cannot be stopped
Power Indicator LED
LED Status
Description
Solid red
The phone is initializing.
Fast flashing red (300ms)
The phone is ringing.
Slow flashing red (1s)
The phone receives a voice mail.
The phone is powered off.
The phone is idle.
Off
The phone is busy.
The call is placed on hold or is held.
The call is muted.
Line key LED
LED Status
Solid green
4
Description
The line is seized.
The line is in conversation.
Fast flashing green
The line receives an incoming call.
Slow flashing green
The call is placed on hold.
Off
The line is inactive.
Overview
Line key LED (configured as a BLF key or BLF List key)
LED Status
Description
Solid green
The monitored user is idle.
Fast flashing red (200ms)
The monitored user receives an incoming call.
Solid red
The monitored user is busy.
The call is parked against the monitored user’s
Slow flashing red (1s)
phone number.
The monitored user’s conversation is placed on
hold (This status requires server support).
Off
The monitored user does not exist.
Line key LED (used as a page switch key)
LED Status
Solid red
Solid green
Fast flashing green
Fast flashing red
Off
Note
Description
There is a parked call on the line key of the
non-current page.
There is a call on the line key of the non-current
page.
The line receives an incoming call on the
non-current page.
The monitored user receives an incoming call on
the non-current page.
Line keys on every page are idle.
The above introduces the default LED status. The statuses of the power indicator LED and
BLF key are configurable via web user interface. For more information, refer to
Yealink_SIP-T4X_IP_Phones_Administrator_Guide .
Two ways to customize configurations of your SIP-T42G IP phone:

The user interface on the IP phone.

The user interface in a web browser on your PC.
The hardware components keypad and LCD screen constitute the phone user interface,
which allows the user to execute all call operation tasks and basic configuration
changes directly on the phone. In addition, you can use the web user interface to
access all configuration settings. In many cases, it is possible to use either the phone
user interface and/or the web user interface interchangeably. However, in some cases,
it is only possible to use one or the other interface to operate the phone and change
settings.
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User Guide for the SIP-T42G IP Phone
You can customize your phone by pressing the Menu soft key to access the phone user
interface. The Advanced option is only accessible to the administrator, and the default
administrator password is “admin” (case-sensitive). For more information on
customizing your phone with the available options from the phone user interface, refer
to Customizing Your Phone on page 19.
In addition to the phone user interface, you can also customize your phone via web user
interface. In order to access the web user interface, you need to know the IP address of
your new phone. To obtain the IP address, press the OK key on the phone when the
phone is idle. Enter the IP address (e.g., http://192.168.0.10 or 192.168.0.10) in the
address bar of a web browser on your PC. The default administrator user name and
password are both “admin” (case-sensitive).
The options you can use to customize the IP phone via phone user interface and/or via
web user interface are listed in the following table:
Options
Phone User Interface
Web User Interface
√
√
Status
--IP Address
--MAC
--Firmware
--Network
--Phone
--Accounts
Basic Phone Settings
6
--Backlight
√
--Language
√
--Time & Date
√
--Administrator Password
√
--Key as Send
√
-- Phone Lock
√
--Ring Tones
√
--Contact Management
×
--Directory
×
--Local Directory
√
--Blacklist
√
--Remote Phone Book
×
--Call History Management
√
--DSS Keys
√
√
Overview
Options
Phone User Interface
--Account Registration
√
--Dial Plan
×
--Emergency Number
×
--Live Dialpad
×
--Hotline
√
Web User Interface
Basic Call Features
--Recent Call In Dialing
×
--Auto Answer
√
--Auto Redial
√
--Call Completion
√
--Recall
√
-- Do Not Disturb (DND)
√
--Call Forward
√
--Call Transfer
√
--Call Waiting
√
--Conference
×
--Call Park
√
--Call Pickup
√
--Anonymous Call
√
--Anonymous Call Rejection
√
√
Advanced Phone Features
--Busy Lamp Field (BLF)
√
--BLF List
×
--Call Recording
√
--Hot Desking
√
--Intercom
√
--Multicast Paging
×
--Music on Hold
×
--ACD
√
--Messages
√
√
SIP Account
--User Options
×
--Activation
√
--Label
√
--Display Name
√
--Register Name
√
--User Name
√
--Password
√
--SIP Server1/2
√
--Server Option
--Registrar Port
√
×
×
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User Guide for the SIP-T42G IP Phone
Options
Note
Phone User Interface
--Outbound Status
√
--Outbound Proxy
√
--NAT Traversal
×
--STUN Status
√
--STUN Server
√
Web User Interface
The table above lists most of the feature options. Please refer to the relevant sections for
more information.
The following table shows documentations available for the SIP-T42G IP phone.
Name
Contents
Where found
Language
In the package
English
On the website
English
Basic call features
Quick Start Guide
and phone
customizations
Phone/Web user
interface settings
User Guide
Basic call features
and advanced
phone features
Note
8
You can also download the latest documentations online:
http://www.yealink.com/SupportDownloadfiles_detail.aspx?CateId=312&flag=142 .
Getting Started
This chapter provides the following basic installation instructions and information for
obtaining the best performance with the SIP-T42G IP phone. Topics include:

Packaging Contents

Phone Installation

Phone Initialization

Phone Status

Basic Network Settings

Registration

Idle Screen
If you require additional information or assistance with your new phone, contact your
system administrator.
The following components are included in your SIP-T42G IP phone package:

SIP-T42G IP Phone
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User Guide for the SIP-T42G IP Phone

Phone Stand

Handset & Handset Cord

Ethernet Cable

Quick Start Guide
Check the list before installation. If you find anything missing, contact your system
administrator.
The following items are optional accessories for your SIP-T42G IP phone. You need to
purchase them separately if required.
10
Getting Started

Wall Mount Bracket
Power Adapter


Headset

Wireless Headset Adapter EHS36
If your phone is already installed, proceed to Phone Initialization on page 14.
This section introduces how to install the phone:
1)
Attach the stand or the optional wall mount bracket
2)
Connect the handset and optional headset
3)
Connect the network and power
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User Guide for the SIP-T42G IP Phone
1)
Attach the stand or the optional wall mount bracket
Desk Mount Method
Wall Mount Method (Optional)
Note
The wall mount bracket should be purchased separately. For more information on how to
attach the wall mount bracket, refer to Yealink Wall Mount Quick Installation Guide for
SIP-T4X IP Phones.
2)
Note
Connect the handset and optional headset
The headset and wireless headset adapter EHS36 should be purchased separately.
EHS36 is used for connecting the phone and a compatible wireless headset. For more
information on how to use the EHS36, refer to Yealink EHS36 User Guide.
12
Getting Started
3)
Connect the network and power
You have two options for power and network connections. Your system administrator will
advise you which one to use.

AC power (Optional)

Power over Ethernet (PoE)
AC Power (Optional)
To connect the AC power:
1.
Connect the DC plug on the power adapter to the DC5V port on the phone and
connect the other end of the power adapter into an electrical power outlet.
2.
Connect the included or a standard Ethernet cable between the Internet port on the
phone and the one on the wall or switch/hub device port.
Power over Ethernet
With the included or a regular Ethernet cable, the SIP-T42G IP phone can be powered
from a PoE-compliant switch or hub.
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User Guide for the SIP-T42G IP Phone
To connect the PoE:
1.
Connect the Ethernet cable between the Internet port on the phone and an
available port on the in-line power switch/hub.
Note
If in-line power is provided, you don’t need to connect the phone to the power adapter.
Make sure the switch/hub is PoE-compliant.
The phone can also share the network with another network device such as a PC
(personal computer). This is an optional connection.
Important! Do not remove power to the phone while it is updating firmware and
configurations.
After your phone is powered on, the system boots up and performs the following steps:
Automatic Phone Initialization
The phone finishes the initialization by loading the saved configuration. The LCD screen
displays "Initializing…please wait" during this process.
DHCP (Dynamic Host Configuration Protocol)
The phone attempts to contact a DHCP server in your network to obtain valid IPv4
network settings (e.g., IP address, subnet mask, default gateway address and DNS
address) by default.
Note
If your network does not use DHCP, proceed to Basic Network Settings on page 16.
You can view phone status via phone user interface or web user interface.
Available information of phone status includes:
14
Getting Started

Network status (e.g., IPv4 status, IP address mode and MAC address).

Phone status (e.g., device model, hardware version, firmware version, product ID
and device certificate status).

Note
Account status (e.g., register status of SIP accounts).
You can view device certificate status via phone user interface only.
To view the phone status via phone user interface:
1.
Press
, or press Menu->Status.
2.
Press
or
to scroll through the list and view the specific information.
To view the phone status via web user interface:
1.
Open a web browser on your computer.
2.
Enter the IP address in the browser’s address bar, and then press the Enter key.
3.
Enter the user name (admin) and password (admin) in the login page.
4.
Click Confirm to login.
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User Guide for the SIP-T42G IP Phone
The phone status is displayed on the first page of the web user interface.
If your phone cannot contact a DHCP server for any reason, you need to configure
network settings manually. The IP phone can support either or both IPv4 and IPv6
addresses.
To configure the IP address mode via phone user interface:
1.
Press Menu->Settings->Advanced Settings (default password: admin)
->Network->WAN Port.
2.
Press
or
, or the Switch soft key to select IPv4, IPv6 or IPv4 & IPv6 from the IP
Mode field.
3.
Press the Save soft key to accept the change or the Back soft key to cancel.
To configure a static IPv4 address via phone user interface:
1.
Press Menu->Settings->Advanced Settings (default password: admin)
->Network->WAN Port.
16
2.
Press
to select IPv4 and then press the Enter soft key.
3.
Press
to select Static IP Client and then press the Enter soft key.
4.
Enter the desired values in the IPv4, Subnet Mask, Default Gateway, IPv4 Pri.DNS
Getting Started
and IPv4 Sec.DNS fields respectively.
5.
Press the Save soft key to accept the change or the Back soft key to cancel.
To configure a static IPv6 address via phone user interface:
1.
Press Menu->Settings->Advanced Settings (default password: admin)
->Network->WAN Port.
2.
Press
to select IPv6 and then press the Enter soft key.
3.
Press
to select Static IP and then press the Enter soft key.
4.
Enter the desired values in the IPv6 IP, Prefix, Default Gateway, IPv6 Pri.DNS and IPv6
Sec.DNS fields respectively.
5.
Note
Press the Save soft key to accept the change or the Back soft key to cancel.
Wrong network settings may result in inaccessibility of your phone and may also have an
impact on your network performance. For more information on these parameters,
contact your system administrator.
Generally, your phone will be deployed with multiple other phones. In this case, your
system administrator will configure the phone parameters beforehand, so that after you
start up your phone, the phone will be registered and ready for use. The SIP-T42G IP
phone supports up to 12 accounts. If your phone is not registered, you may have to
register it. For more information on how to register your phone, refer to Account
Management on page 56.
If the phone has successfully started up, the idle LCD screen will be displayed as below.
17
User Guide for the SIP-T42G IP Phone
The idle screen displays the label of the current account, time and date, and four soft
keys.
18
Customizing Your Phone
You can customize your SIP-T42G IP phone by personally configuring certain settings, for
example, time & date and ring tones. You can add contacts to the phone’s local
directory manually or from call history. You can also personalize different ring tones for
different callers.
This chapter provides basic operating instructions for customizing your phone. Topics
include:

General Settings

Audio Settings

Contact Management

Call History Management

System Customizations
If you require additional information or assistance with your new phone, contact your
system administrator.
You can configure the backlight to adjust the brightness of the LCD screen. Backlight
status on the LCD screen can be configured from the following options:

Always On: Backlight is on permanently.

15s, 30s, 60s, 120s, 300s, 600s or 1800s: Backlight is turned off when the phone is
inactive after the designated time (in seconds).
To configure the backlight via phone user interface:
1.
Press Menu->Settings->Basic Settings->Display->Backlight.
2.
Press
or
, or the Switch soft key to select the desired value from the
Backlight Time field.
3.
Press the Save soft key to accept the change or the Back soft key to cancel.
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User Guide for the SIP-T42G IP Phone
Backlight is configurable via web user interface at the path Settings->Preference.
The default language of the phone user interface is English. If the language of your web
browser is not supported by the phone, the web user interface will use English by
default. You can change the language for the phone user interface and the web user
interface respectively.
To change the language for the phone user interface:
1.
Press Menu->Settings->Basic Settings->Language.
2.
Press
3.
Press the Save soft key to accept the change.
or
to select the desired language.
Text displayed on the phone user interface will change to the selected language.
To change the language for the web user interface:
1.
Click on Settings->Preference.
2.
Select the desired language from the pull-down list of Language.
3.
Click Confirm to accept the change.
Text displayed on the web user interface will change to the selected language.
The time and date are displayed on the LCD screen when the phone is idle. You can
configure the phone to obtain the time and date from the Simple Network Time Protocol
(SNTP) server automatically, or configure the date and time manually. If the phone
cannot obtain the time and date from the SNTP server, contact your system
20
Customizing Your Phone
administrator for more information.
To configure the SNTP settings via phone user interface:
1.
Press Menu->Settings->Basic Settings->Time & Date->SNTP Settings.
2.
Press
or
, or the Switch soft key to select the time zone that applies to your
area from the Time Zone field.
The default time zone is "+8 China(Beijing)".
3.
Enter the domain names or IP addresses in the NTP Server1 and NTP Server2 fields
respectively.
4.
Press
or
, or the Switch soft key to select the desired value from the
Daylight Saving field.
5.
Note
Press the Save soft key to accept the change or the Back soft key to cancel.
Please refer to Appendix A - Time Zones for the list of available time zones on the IP
phone.
To configure the time and date manually via phone user interface:
1.
Press Menu->Settings->Basic Settings->Time & Date->Manual Settings.
2.
Enter the specific time and date.
3.
Press the Save soft key to accept the change.
The date and time displayed on the LCD screen will change accordingly.
To configure the time and date format via phone user interface:
1.
Press Menu Settings->Basic Settings->Time & Date->Time & Date Format.
2.
Press
or
, or the Switch soft key to select the desired time format (12 Hour
or 24 Hour) from the Time Format field.
3.
Press
or
, or the Switch soft key to select the desired date format from the
21
User Guide for the SIP-T42G IP Phone
Date Format field.
4.
Press the Save soft key to accept the change or the Back soft key to cancel.
There are 7 available date formats. For example, for the date format “WWW DD MMM”,
“WWW” represents the abbreviation of the weekday, “DD” represents the two-digit day,
and “MMM” represents the first three letters of the month.
The date formats available:
Date Format
Example (2014-7-26)
WWW MMM DD
Sat Jul 26
DD-MMM-YY
26-Jul-14
YYYY-MM-DD
2014-07-26
DD/MM/YYYY
26/07/2014
MM/DD/YY
07/26/14
DD MMM YYYY
26 Jul 2014
WWW DD MMM
Sat 26 Jul
Time and date are configurable via web user interface at the path Settings->Time &
Date.
The Advanced Settings option is only accessible to the administrator. The default
administrator password is “admin”. For security reasons, you should change the default
administrator password as soon as possible.
To change the administrator password via phone user interface:
1.
Press Menu->Settings->Advanced Settings (default password: admin) ->Set
Password.
2.
Enter the current password in the Current PWD field.
3.
Enter the new password in the New PWD field.
4.
Re-enter the new password in the Confirm PWD field.
5.
Press the Save soft key to accept the change or the Back soft key to cancel.
Administrator password is configurable via web user interface at the path
Security->Password.
22
Customizing Your Phone
You can set the "#" or "*" to perform as a send key while dialing.
To configure key as send via phone user interface:
1.
Press Menu->Features->Key as Send.
2.
Press
or
, or the Switch soft key to select # or * from the Key as Send field,
or select Disabled to disable this feature.
3.
Press the Save soft key to accept the change or the Back soft key to cancel.
Key as send is configurable via web user interface at the path Features->General
Information.
You can lock your phone temporarily when you are not using it. This feature helps to
protect your phone from unauthorized use.
Phone lock consists of the following:
Menu Key:
The Menu soft key is locked. You cannot access the menu of the
phone until unlocked.
Function Keys:
The function keys are locked. You cannot use the MESSAGE,
REDIAL, MUTE, OK, X, navigation keys, soft keys and line keys
until unlocked.
All Keys:
All keys are locked, except the Volume, Headset,
Speakerphone and digit keys. You are only allowed to dial
emergency numbers, reject or answer incoming calls, and
cancel a call.
Note
The emergency number setting, if desired, must be made before lock activation. For
more information, refer to Emergency Number on page 63.
23
User Guide for the SIP-T42G IP Phone
To activate the phone lock via phone user interface:
1.
Press Menu->Settings->Advanced Settings (default password: admin) -> Phone
Lock.
2.
Press
or
, or the Switch soft key to select Enabled from the Phone Lock
Enable field.
3.
Press
or
, or the Switch soft key to select the desired type from the Lock
Type field.
4.
Press the Save soft key to accept the change.
Long press
to lock the phone immediately when the phone is idle.
The LCD screen prompts “Phone locked.” and displays the icon
.
You can configure the interval (in seconds) to automatically lock the phone instead of
long pressing
.
To configure the interval of automatic phone lock via web user interface:
Note
1.
Click on Features->Phone Lock.
2.
Enter the desired time in the Phone Lock Time Out (0~3600s) field.
3.
Click Confirm to accept the change.
The default timeout is 0 second, that is, you can long press
to lock the phone only.
Interval of automatic phone lock is configurable via web user interface only.
24
Customizing Your Phone
To unlock the phone, you must know the phone unlock PIN of the phone. The default
phone unlock PIN is “123”.
To change the phone unlock PIN via phone user interface:
1.
Press Menu->Settings->Basic Settings->Change PIN.
2.
Enter the desired values in the Current PIN, New PIN and Confirm PIN fields
respectively.
3.
Note
Press the Save soft key to accept the setting or the Back soft key to cancel.
The unlock PIN length must be within 15 digits.
To unlock the phone via phone user interface:
1.
Press any locked key, the LCD screen prompts “Please Enter PIN”.
2.
Enter the PIN in the Phone Unlock PIN field.
3.
Press the OK soft key to unlock the phone.
The
icon disappears from the LCD screen.
You can long press
or wait for a period of time (if configured) to lock the
phone again.
Note
You can also unlock the phone by entering administrator password. When you enter the
administrator password to unlock the phone, the phone will turn to the Reset Phone PIN
interface.
To deactivate the phone lock via phone user interface:
1.
Press Menu->Settings->Advanced Settings (default password: admin) ->Phone
Lock.
25
User Guide for the SIP-T42G IP Phone
2.
Press
or
, or the Switch soft key to select Disabled from the Phone Lock
Enable field.
3.
Press the Save soft key to accept the change.
Phone lock is configurable via web user interface at the path Features->Phone Lock.
You can press the volume key to adjust the ringer volume when the phone is idle. You
can also press the volume key to adjust the receiver volume of currently engaged audio
devices (handset, speakerphone or headset) when the phone is in use.
To adjust the volume when the phone is idle:
1.
Note
Press
to adjust the ringer volume.
If the ringer volume is adjusted to minimum, the
icon will appear on the LCD screen.
To adjust the volume when the phone is during a call:
1.
Press
to adjust the volume of currently engaged audio device (handset,
speakerphone or headset).
Ring tones are used to indicate incoming calls. You can select different ring tones to
distinguish different accounts registered on your phone, or to distinguish your phone
from your neighbor’s.
26
Customizing Your Phone
To select a ring tone for the phone via phone user interface:
1.
Press Menu->Settings->Basic Settings->Sound->Ring Tones->Common.
2.
Press
3.
Press the Save soft key to accept the change or the Back soft key to cancel.
or
to select the desired ring tone.
A ring tone for the phone is configurable via web user interface at the path
Settings->Preference->Ring Type.
To select a ring tone for the account via phone user interface:
1.
Press Menu->Settings->Basic Settings ->Sound->Ring Tones.
2.
Press
or
to select the desired account and then press the Enter soft key.
3.
Press
or
to select the desired ring tone.
If Common is selected, this account will use the ring tone selected for the phone.
4.
Press the Save soft key to accept the change or the Back soft key to cancel.
A ring tone for the account is configurable via web user interface at the path
Account->Basic->Ring Type.
To upload a custom ring tone for your phone via web user interface:
1.
Click on Settings->Preference.
27
User Guide for the SIP-T42G IP Phone
2.
Click Browse to locate a ring tone file (the file format must be *.wav) from your local
system.
3.
Note
Click Upload to upload the file.
The ring tone for an incoming call on the phone may be different. For example, when the
phone receives an incoming call from a contact stored in the local directory, it will play
the ring tone assigned to the contact in the local directory (refer to Adding Contacts). If
no ring tone is assigned to the contact, the phone will play the ring tone assigned to the
associated group (refer to Adding Groups). Otherwise, the phone will play the ring tone
assigned to the account. If no ring tone is assigned to the contact, the associated group
or account, the phone will play the ring tone assigned to the phone.
All custom ring tone files must be no greater than 100KB. Uploading custom ring tones for
your phone is configurable via web user interface only.
This section provides the operating instructions for managing contacts. Topics include:

Directory

Local Directory

Blacklist

Remote Phone Book
Directory provides easy access to frequently used lists. The lists may contain Local
Directory, History, Remote Phone Book and LDAP.
To configure the directory via web user interface:
1.
Click on Directory->Setting.
2.
In the Directory block, select the desired list from the Disabled column and then
click
.
The selected list appears in the Enabled column.
28
Customizing Your Phone
3.
Repeat step 2 to add more lists to the Enabled column.
4.
To remove a list from the Enabled column, select the desired list and click
5.
To adjust the display order of enabled lists, select the desired list and click
or
.
.
The LCD screen will display the list(s) in the adjusted order.
6.
Note
Click Confirm to accept the change.
Directory is configurable via web user interface only.
To view the directory via phone user interface:
1.
Press the Directory soft key when the phone is idle.
The LCD screen displays the enabled list(s) in the directory.
If there is only one list in the directory, press the Directory soft key to enter this list
directly.
Note
If remote phone book and LDAP are not configured in advance, you cannot see remote
phone book and LDAP lists on the phone user interface. For more information on remote
phone book, refer to Remote Phone Book on page 39. For more information on the LDAP,
refer to Yealink_SIP-T4X_IP_Phones_Administrator_Guide.
29
User Guide for the SIP-T42G IP Phone
The built-in phone directory can store the names and phone numbers of your contacts.
You can store up to 1000 contacts and 50 groups in your phone's local directory. There
are four default groups in the local directory: All Contacts, Company, Family and Friend.
You can add new groups and contacts, edit, delete or search for a contact, or simply
dial a contact number from the local directory.
To add a group to the local directory:
1.
Press the Directory soft key.
The IP phone enters the local directory directly as there is only Local Directory
enabled in the directory by default.
If the Local Directory is removed from the directory, press Menu->Directory->Local
Directory to enter the local directory.
2.
Press the Add soft key.
3.
Enter the desired group name in the Name field.
4.
Press
or
, or the Switch soft key to select the desired ring tone from the Ring
field.
If Auto is selected, this group will use the ring tone specified for the contact. If no
ring tone is specified for the contact, it will then play the ring tone specified for the
account. For more information on the ring tone for the account, refer to Ring Tones
on page 26.
5.
Press the Add soft key to accept the change or the Back soft key to cancel.
You can also edit or delete the Company, Family, Friend and any newly added contact
groups.
30
Customizing Your Phone
To edit a group in the local directory:
1.
Press the Directory soft key.
The IP phone enters the local directory directly as there is only Local Directory
enabled in the directory by default.
If Local Directory is removed from the directory, press Menu->Directory->Local
Directory to enter the local directory.
2.
Select the desired contact group.
3.
Press the Option soft key and then select Detail from the prompt list.
4.
Press
5.
Press the Save soft key to accept the change or the Back soft key to cancel.
or
to highlight the group information and then edit.
To delete a group from the local directory:
1.
Press the Directory soft key.
The IP phone enters the local directory directly as there is only Local Directory
enabled in the directory by default.
If Local Directory is removed from the directory, press Menu->Directory->Local
Directory to enter the local directory.
2.
Select the desired contact group.
3.
Press the Option soft key and then select Delete from the prompt list.
31
User Guide for the SIP-T42G IP Phone
The LCD screen prompts the following warning:
4.
Press the OK soft key to confirm the deletion or the Cancel soft key to cancel.
You can also delete all groups by pressing the Option soft key, and then select Delete
All.
You can add contacts to the local directory in one of the following ways:

Manually

From call history

From a remote phone book
Adding Contacts Manually
To add a contact to the local directory manually:
1.
Press the Directory soft key.
The IP phone enters the local directory directly as there is only Local Directory
enabled in the directory by default.
If Local Directory is removed from the directory, press Menu->Directory->Local
Directory to enter the local directory.
2.
Select the desired contact group and press the Enter soft key.
3.
Press the Add soft key.
4.
Enter the name and the office, mobile or other numbers in the corresponding fields.
5.
Press
or
, or the Switch soft key to select the desired account from the
Account field.
If Auto is selected, the phone will use the first available account when placing calls
to the contact from the local directory.
6.
Press
or
, or the Switch soft key to select the desired ring tone from the Ring
field.
If Auto is selected, the phone will play the ring tone assigned to the group when
receiving incoming calls from this contact.
32
Customizing Your Phone
7.
Note
Press the Add soft key to accept the change or the Back soft key to cancel.
If the contact already exists in the directory, the LCD screen will prompt “Contact name
existed!”.
Adding Contacts from Call History
To add a contact to the local directory from call history:
1.
Press the History soft key.
2.
Press
3.
Press the Option soft key, and then select Add to Contacts from the prompt list.
4.
Enter the contact name.
5.
Press the Save soft key to accept the change.
or
to highlight the desired entry.
The entry is successfully saved to the local directory.
Adding Contacts from Remote Phone Book
To add a contact to the local directory from a remote phone book:
1.
Press Directory->Remote Phone Book.
If Remote Phone Book is removed from the directory, press
Menu->Directory->Remote Phone Book to enter the remote phone book.
2.
Select the desired remote group and press the Enter soft key.
3.
Press
4.
Press the Option soft key, and then select Add to Contacts from the prompt list.
5.
Press the Save soft key to save the contact to the local directory.
or
to highlight the desired entry.
If the contact already exists in the local directory, the LCD screen will prompt
"Overwrite the original contact?". Press the OK soft key to overwrite the original
contact in the local directory or the Cancel soft key to cancel.
For more information on remote phone book operating, refer to Remote Phone Book on
page 39.
To edit a contact in the local directory:
1.
Press the Directory soft key.
The IP phone enters the local directory directly as there is only Local Directory
33
User Guide for the SIP-T42G IP Phone
enabled in the directory by default.
If Local Directory is removed from the directory, press Menu->Directory->Local
Directory to enter the local directory.
2.
Select the desired contact group and press the Enter soft key.
3.
Press
4.
Press the Option soft key, and then select Detail from the prompt list.
5.
Press
6.
Press the Save soft key to accept the change or the Back soft key to cancel.
or
or
to highlight the desired contact.
to highlight the contact information and then edit.
To delete a contact from the local directory:
1.
Press the Directory soft key.
The IP phone enters the local directory directly as there is only Local Directory
enabled in the directory by default.
If Local Directory is removed from the directory, press Menu->Directory->Local
Directory to enter the local directory.
2.
Select the desired contact group and press the Enter soft key.
3.
Press
4.
Press the Option soft key, and then select Delete from the prompt list.
or
to highlight the desired contact.
The LCD screen prompts the following warning:
5.
Press the OK soft key to confirm the deletion or the Cancel soft key to cancel.
You can also delete all contacts by pressing the Option soft key, and then select Delete
All.
To place a call to a contact from the local directory:
1.
Press the Directory soft key.
The IP phone enters the local directory directly as there is only Local Directory
enabled in the directory by default.
34
Customizing Your Phone
If Local Directory is removed from the directory, press Menu->Directory->Local
Directory to enter the local directory.
2.
Select the desired contact group and press the Enter soft key.
3.
Press
4.
Do one of the following:
-
or
to highlight the desired contact.
If only one number for the contact is stored in the local directory, press the Send
soft key to dial out the number.
-
If multiple numbers for the contact are stored in the local directory, press the
Send soft key to display a list of numbers.
Press
or
to highlight the desired number.
Press the Send soft key to dial out the number.
To search for a contact in the local directory:
1.
Press the Directory soft key.
The IP phone enters the local directory directly as there is only Local Directory
enabled in the directory by default.
If Local Directory is removed from the directory, press Menu->Directory->Local
Directory to enter the local directory.
2.
Press the Search soft key.
3.
Enter a few continuous characters of the contact name or continuous numbers of the
contact number (office, mobile or other number) using the keypad.
The contacts whose name or phone number matches the characters entered will
appear on the LCD screen. You can dial from the result list.
You can search for a contact from the desired lists when the phone is in the dialing
screen. The lists can be Local Directory, History, Remote Phone Book and LDAP.
To configure search source list in dialing via web user interface:
1.
Click on Directory->Setting.
2.
In the Search Source List In Dialing block, select the desired list from the Disabled
column and click
.
The selected list appears in the Enabled column.
35
User Guide for the SIP-T42G IP Phone
3.
Repeat step 2 to add more lists to the Enabled column.
4.
To remove a list from the Enabled column, select the desired list and click
5.
To adjust the display order of the enabled lists, select the desired list and click
or
.
.
The LCD screen will display search results in the adjusted order.
6.
Note
Click Confirm to accept the change.
Search source list in dialing is configurable via web user interface only.
To search for an entry in the enabled search source lists:
1.
Pick up the handset, press the speakerphone or press the line key.
2.
Enter a few continuous characters of the entry’s name or continuous numbers of the
entry’s phone number (office, mobile or other number) using the keypad.
The entries in the enabled search source lists whose name or phone number
matches the characters entered will appear on the LCD screen. You can press
or
to scroll to the desired entry, and then place a call to the entry.
You can manage your phone’s local directory via phone user interface or web user
36
Customizing Your Phone
interface. But you can only import or export the contact list via web user interface.
To import an XML contact list file via web user interface:
1.
Click on Directory->Local Directory.
2.
Click Browse to locate a contact list file (the file format must be *.xml) from your
local system.
3.
Click Import XML to import the contact list.
The web user interface prompts "The original contact will be covered, Continue?".
4.
Click OK to complete importing the contact list.
To import a CSV contact list file via web user interface:
1.
Click on Directory->Local Directory.
2.
Click Browse to locate a contact list file (the file format must be *.csv) from your
local system.
3.
(Optional.) Check the Show Title checkbox.
It will prevent importing the title of the contact information which is located in the
first line of the CSV file.
4.
Click Import CSV to import the contact list.
5.
(Optional.) Mark the On radio box in the Delete Old Contacts field.
It will delete all existing contacts while importing the contact list.
6.
(Optional.) Select the contact information you want to import into the local directory
from the pull down list of Index.
37
User Guide for the SIP-T42G IP Phone
At least one item should be selected to be imported into the local directory.
7.
Click Import to complete importing the contact list.
To export a contact list via web user interface:
1.
Click on Directory->Local Directory.
2.
Click Export XML (or Export CSV).
3.
Click Save to save the contact list to your local system.
Note
Importing/exporting contact lists is available via web user interface only.
The built-in phone directory can store names and phone numbers for a blacklist. You
can store up to 30 contacts, add, edit, delete or search for a contact in the blacklist
directory, and even call a contact from the blacklist directory. Incoming calls from
blacklist directory contacts will be rejected automatically.
To add a contact to the blacklist directory manually:
1.
Press Menu->Directory->Blacklist.
2.
Press the Add soft key.
3.
Enter the name and the office, mobile or other numbers in the corresponding fields.
4.
Press
or
Account field.
38
, or the Switch soft key to select the desired account from the
Customizing Your Phone
If Auto is selected, the phone will use the first available account when placing calls
to the contact from the blacklist directory.
5.
Press the Add soft key to accept the change or the Back soft key to cancel.
To add a contact to the blacklist directory from the local directory:
1.
Press Directory soft key.
The IP phone enters the local directory directly as there is only Local Directory
enabled in the directory by default.
If Local Directory is removed from the directory, press Menu->Directory->Local
Directory to enter the local directory.
2.
Select the desired contact group and press the Enter soft key.
3.
Press
4.
Press the Option soft key and then select Add to Blacklist from the prompt list.
or
to highlight the desired contact.
The LCD screen prompts "Move selected to blacklist?".
5.
Press the OK soft key to confirm the setting.
For operating instructions on editing, deleting, placing calls to and/or searching for
contacts in the blacklist, refer to the operating instructions of Local Directory on page 29.
You can add new contacts to the local directory, search for a contact, or simply dial a
contact number from the remote phone book.
You can configure your new phone to access up to 5 remote phone books. The phone
supports up to 5000 remote phone book entries. For the access URL of the remote phone
book, contact your system administrator.
To configure an access URL for a remote phone book via web user interface:
1.
Click on Directory->Remote Phone Book.
2.
Enter the access URL in the Remote URL field.
39
User Guide for the SIP-T42G IP Phone
3.
Enter the name in the Display Name field.
4.
Click Confirm to accept the change.
Note
An access URL for a remote phone book is configurable via web user interface only.
To access your remote phone book via phone user interface:
1.
Press Directory->Remote Phone Book.
If Remote Phone Book is removed from the directory, press
Menu->Directory->Remote Phone Book to enter the remote phone book.
2.
Select the desired remote group, and then press the Enter soft key.
The phone then connects to the remote phone book and proceeds to load it. The
contacts in the remote phone book are displayed on the LCD screen.
3.
Press the Back soft key to back to the previous interface.
To search for a contact in the remote phone book:
1.
Press Menu->Directory->Remote Phone Book.
If Remote Phone Book is added to the directory, press Directory->Remote Phone
Book to enter the remote phone book.
2.
Select the desired remote group, and then press the Enter soft key to load the
remote phone book.
3.
40
Press the Search soft key.
Customizing Your Phone
4.
Enter a few continuous characters of the contact name or continuous numbers of the
contact number using the keypad.
The contacts whose name or phone number matches the characters entered will
appear on the LCD screen. You can place a call from the result list.
To place a call from the remote phone book:
1.
Press Menu->Directory->Remote Phone Book.
If Remote Phone Book is added to the directory, press Directory->Remote Phone
Book to enter the remote phone book.
2.
Select the desired remote group, and then press the Enter soft key to load the
remote phone book.
3.
Select the desired contact in the remote phone book.
4.
Press the Send soft key.
In addition, you can enable the phone to present the caller identity stored in the remote
phone book when receiving a call.
To enable the presentation of a caller identity stored in remote phone book via web user
interface:
1.
Click on Directory->Remote Phone Book.
2.
Select Enabled from the pull-down list of Incoming/Outgoing Call lookup.
3.
Enter the desired refresh period in the Update Time Interval (Seconds) field.
The default value is 21600 seconds.
4.
Click Confirm to accept the change.
41
User Guide for the SIP-T42G IP Phone
The SIP-T42G IP phone maintains call history lists of Placed Calls, Received Calls, Missed
Calls and Forwarded Calls. Call history lists support up to 400 entries. You can view call
history, place a call, add a contact or delete an entry from the call history list.
History record feature is enabled by default. If you don’t want to save the call history,
you can disable the feature
To disable history record via phone user interface:
1.
Press Menu->Features->History Setting.
2.
Press
or
, or the Switch soft key to select Disabled from the History Record
field.
3.
Press the Save soft key to accept the change or the Back soft key to cancel.
To view call history:
1.
Press the History soft key.
The LCD screen displays all call records.
2.
Press
or
to switch between all, placed, received, missed and forwarded
or
to select the desired entry.
call lists.
3.
Press
4.
Press the Option soft key, and then select Detail from the prompt list.
The detailed information of the entry appears on the LCD screen.
To place a call from the call history list:
1.
Press the History soft key.
2.
Press
or
to switch between all, placed, received, missed and forwarded
or
to select the desired entry.
call lists.
3.
Press
4.
Press the Send soft key.
To add a contact to the local directory (or blacklist directory) from the call history list:
1.
Press the History soft key.
2.
Press
or
to switch between all, placed, received, missed and forwarded
or
to select the desired entry.
call lists.
3.
42
Press
Customizing Your Phone
4.
Press the Option soft key, and then select Add to Contacts (or Add to Blacklist) from
the prompt list.
5.
Enter the desired values in the corresponding fields.
6.
Press the Save soft key to accept the change.
For more information, refer to Contact Management on page 28.
To delete an entry from the call history list:
1.
Press the History soft key.
2.
Press
or
to switch between all, placed, received, missed and forwarded
or
to select the desired entry.
call lists.
3.
Press
4.
Press the Delete soft key.
To delete all entries from the call history list:
1.
Press the History soft key.
2.
Press
or
to switch between all, placed, received, missed and forwarded
call lists.
3.
Press the Option soft key, and then select Delete All from the prompt list.
4.
Press the OK soft key.
The LCD screen prompts "Delete all the call records?".
5.
Press the OK soft key to confirm the deletion or the Cancel soft key to cancel.
If you want to use a headset, physically connect your headset and activate the headset
mode for use. For more information on physically connecting a headset, refer to Phone
Installation on page 11.
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User Guide for the SIP-T42G IP Phone
To activate the headset mode:
1.
Press
on the phone.
The headset key LED illuminates solid green when the headset mode is activated.
Press the line key or the Answer soft key to answer an incoming call. The call will be
connected to your headset automatically.
Enter the desired number and press the Send soft key, the phone will then place a
call using the headset automatically. For more information on using the headset to
place a call, refer to Placing Calls on page 68.
To deactivate the headset mode:
1.
Press
again on the phone.
The headset key LED goes out when the headset mode is deactivated.
You can use headset in priority when headset prior feature is enabled. This feature is
especially useful for permanent or full-time headset users.
To enable headset prior via web user interface:
44
1.
Click on Features->General Information.
2.
Select Enabled from the pull-down list of Headset Prior.
3.
Click Confirm to accept the change.
Customizing Your Phone
To use headset prior, you should activate the headset mode in advance:
1.
Physically connect the headset.
2.
Press
Note
to activate the headset mode.
If headset prior is enabled, the headset mode will not be deactivated until you press the
HEADSET key again.
If headset prior is disabled, the headset mode can be deactivated by pressing the
speakerphone key or the HEADSET key.
Headset prior is configurable via web user interface only.
You can use two headsets when dual headset feature is enabled. To use this feature,
you must physically connect headsets to the headset jack and handset jack respectively.
Once the phone connects to a call, the headset connected to the headset jack will have
full-duplex capabilities, while the one connected to the handset jack will only be able to
listen.
To enable dual headset via web user interface:
Note
1.
Click on Features->General Information.
2.
Select Enabled from the pull-down list of Dual-Headset.
3.
Click Confirm to accept the change.
Dual headset is configurable via web user interface only.
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User Guide for the SIP-T42G IP Phone
There are two types of DSS keys: Line Keys and Programmable Keys. Details will be
introduced in the following. The SIP-T42G IP phone supports 15 line keys.
You can assign predefined functionalities to line keys. You can also define a label for a
line key feature which will appear on the LCD screen. Line keys allow you to quickly
access features such as recall and voice mail. The line key LEDs will indicate the
monitored status when line keys are assigned with particular features, such as BLF. The
default key type of line key 1-12 is Line. The default key type of line key 13-15 is N/A,
which indicates that this line key provides no functionality until configuration.
To assign functionality to a line key via phone user interface:
1.
Press Menu->Features->DSS Keys.
2.
Select the desired line key, and then press the Enter soft key.
3.
Select the desired key type from the Type field.
4.
(Optional.) Select the desired key event type from the Key Type field.
5.
(Optional.) Select the desired line from the Account ID field.
6.
(Optional.) Enter a string that will appear on the LCD screen in the Label field.
7.
(Optional.) Enter the corresponding value in the Value field.
8.
(Optional.) Enter the corresponding value in the Extension field.
9.
Press the Save soft key to accept the change or the Back soft key to cancel.
Line key is configurable via web user interface at the path DSSKey->Line Key.
Note
When the phone is idle, you can also long press the line key to configure it directly on the
phone.
You can assign functionalities to 15 line keys. These 15 line keys are indicated on three
pages. When you assign functionality to line key 1-6, labels of line keys are indicated on
the first page.
The following figure shows an example:
When you assign functionalities to line key 7-15, labels of the line key 6-10 are indicated
on the second page, and labels of the line key 11-15 are indicated on the third page.
The right bottom line key is used to switch pages.
46
Customizing Your Phone
The following figure shows an example:
You can configure the page icon and page switch key LED to indicate different statuses.
This feature is disabled by default.
The following table lists the page icon and status:
Icons/Status
Indicates the current page of line keys.
1
2
Description
3
Indicates the non-current page of line keys.
The BLF monitored user receives an incoming call on the
Fast flashing
non-current page.
The line on the non-current page receives an incoming call.
There is a hold call on the line on the non-current page.
For more information on the page switch key LED status, refer to LED Instructions on
page 4.
To configure the page icon to indicate status via web user interface:
Note
1.
Click on DSSKey->Line Key.
2.
Select Enabled from the pull-down list of Enable Page Tips.
3.
Click Confirm to accept the change.
The page icon status is configurable via web user interface only.
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User Guide for the SIP-T42G IP Phone
Line key features are explained in the following subchapters in detail:

Line

Speed Dial

Voice Mail

Directed Pickup

Group Pickup

DTMF

Prefix

Local Group

XML Group

XML Browser

LDAP

Conference

Forward

Transfer

Hold

DND

Group Listening

Zero Touch

URL

Phone Lock

Directory
For the features not listed above, refer to Basic Call Features on page 67 and Advanced
Phone Features on page 99. For more information, contact your system administrator.
Line
You can use this key feature to accept incoming calls, place active calls on hold or
resume a held call. It performs in the same way as a hard line key.
Dependencies: Type (Line)
Account ID (the account this feature will be applied to)
Label (key label displayed on the LCD screen)
Value (Default/Lock/Float)
Usage: When receiving an incoming call, the DSS key LED flashes green:
48
1.
Press the DSS key to accept the incoming call.
2.
Press the DSS key to place a new call and the active call is placed on hold.
3.
Press the DSS key again to resume the held call.
Customizing Your Phone
If you select Lock from the Value field, the line keys located in the same position of the
other page will be automatically assigned the line key feature with Lock filled in the
Value field. Then you can use this key on every page. This feature is only applicable to
line key 1-3.
If you select Float from the Value field, the phone will automatically switch to the page
the line key located when the line key receives an incoming call.
If you select Default from the Value field, you can only use this key on the current page.
Speed Dial
You can use this key feature to speed up dialing the numbers frequently used or hard to
remember.
Dependencies: Type (Speed Dial)
Account ID (the account this feature will be applied to)
Label (key label displayed on the LCD screen)
Value (the number you want to dial out)
Usage: Press the DSS key to dial out the number specified in the Value field, using the
account selected from the Account ID field.
Voice Mail
You can use this key feature to quickly connect a voice mail. For more information, refer
to Voice Mail on page 116.
Dependencies: Type (Key Event)
Key Type (Voice Mail)
Account ID (the account this feature will be applied to)
Label (key label displayed on the LCD screen)
Value (the voice mail access code)
Usage: Press the DSS key to dial out the voice mail access code. Then follow the voice
prompt to listen to voice mails.
Directed Pickup
You can use this key feature to answer someone else’s incoming call on the phone.
Dependencies: Type (Key Event)
Key Type (DPickup)
Account ID (the account this feature will be applied to)
Label (key label displayed on the LCD screen)
Value (the directed pickup code followed by the target phone number)
Usage: Press the DSS key on your phone when the target phone number receives an
incoming call. The call is then answered on your phone.
Group Pickup
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User Guide for the SIP-T42G IP Phone
You can use this key feature to answer incoming calls in a group that is associated with
their own group.
Dependencies: Type (Key Event)
Key Type (GPickup)
Account ID (the account this feature will be applied to)
Label (key label displayed on the LCD screen)
Value (the group pickup feature code)
Usage: Press the DSS key on your phone when a phone number in the group receives
an incoming call. The call is answered on your phone.
DTMF
You can use this key feature to send the specification of arbitrary key sequences via
DTMF.
Dependencies: Type (Key Event)
Key Type (DTMF)
Label (key label displayed on the LCD screen)
Value (DTMF sequence)
Note
DTMF sequence can only contain "0-9", "*", "#" and "A-D".
Usage: Press the DSS key during an active call to send the key sequence specified in the
Value field.
Prefix
You can use this key feature to add a specified prefix number before the dialed number.
Dependencies: Type (Key Event)
Key Type (Prefix)
Label (key label displayed on the LCD screen)
Value (the prefix number)
Usage: Press the DSS key when the phone is idle, the phone will then enter the
pre-dialing screen and display the prefix number that you specified in the Value field.
You can enter the remaining digits and then dial out.
Local Group
You can use this key feature to quickly access a contact group in the local directory. For
more information, refer to Local Directory on page 29.
Dependencies: Type (Key Event)
Key Type (Local Group)
Local Group (the contact group name you want to access)
50
Customizing Your Phone
Label (key label displayed on the LCD screen)
Usage: Press the DSS key to access the contact group specified in the Local Group field.
XML Group
You can use this key feature to quickly access a remote group in your remote phone
book. You should configure remote phone book in advance. For more information, refer
to Remote Phone Book on page 39.
Dependencies: Type (Key Event)
Key Type (XML Group)
PhoneBook Name (the remote group name you want to access if the
remote phone book is configured)
Label (key label displayed on the LCD screen)
Usage: Press the DSS key to access the remote group specified in the XML Group field.
XML Browser
You can use this key feature to quickly access a XML browser. The XML browser allows
you to create custom services which meet your functional requirements on the server.
You can customize practical applications, such as weather report, stock information,
Google search, etc.
Dependencies: Type (Key Event)
Key Type (XML Browser)
Label (key label displayed on the LCD screen)
Value (the access URL for XML browser)
Usage: Press the DSS key to access the XML browser specified in the Value field.
LDAP
You can use this key feature to quickly access a LDAP search screen.
Dependencies: Type (Key Event)
Key Type (LDAP)
Label (key label displayed on the LCD screen)
Usage:
1.
Press the DSS key to access the LDAP search screen.
2.
Enter a few continuous characters of the contact name or continuous numbers of the
contact number using the keypad.
The contacts whose name or phone number matches the characters entered will
appear on the LCD screen.
Conference
You can use this key feature to set up a conference call. For more information, refer to
Conference on page 88.
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User Guide for the SIP-T42G IP Phone
Dependencies: Type (Key Event)
Key Type (Conf)
Label (key label displayed on the LCD screen)
Value (the number you want to add to the conference)
Usage: Press the DSS key during an active call to set up a conference with the number
specified in the Value field.
Note
If the Value field is left blank, the DSS key performs the same as the Conf soft key during
a call.
Forward
You can use this key feature to forward an incoming call to someone else. For more
information, refer to Call Forward on page 81.
Dependencies: Type (Key Event)
Key Type (Forward)
Label (key label displayed on the LCD screen)
Value (the number you want to forward to)
Usage: Press the DSS key to forward an incoming call to the number specified in the
Value field.
Note
If the Value field is left blank, the DSS key performs the same as the Forward soft key
when receiving an incoming call.
When call forward is activated, the forward key LED will illuminate solid green.
Transfer
When there is an active call on the phone, you can use this key feature to handle the call
differently depending on the transfer mode assigned to the DSS key.
Dependencies: Type (Key Event)
Key Type (Transfer)
Label (key label displayed on the LCD screen)
Value (the number you want to transfer to)
Usage:

When the transfer mode via DSS key is Blind Transfer, press the DSS key to
complete the blind transfer to the number specified in the Value field.

When the transfer mode via DSS key is Attended Transfer, press the DSS key to dial
out the number specified in the Value field, and then perform the attended or
semi-attended transfer.

52
When the transfer mode via DSS key is New Call, press the DSS key to place a new
Customizing Your Phone
call to the number specified in the Value field.
Note
Transfer mode via DSS key is configurable via web user interface at the path
Features->Transfer->Transfer Mode Via Dsskey.
If the Value field is left blank, the DSS key performs the same as the Transfer soft key
during a call.
Hold
You can use this key feature to place an active call on hold or retrieve a held call.
Dependencies: Type (Key Event)
Key Type (Hold)
Label (key label displayed on the LCD screen)
Usage:
1.
Press the DSS key during an active call to place the call on hold.
2.
Press the DSS key again to retrieve the held call.
DND
You can use this key feature to activate or deactivate the DND mode. You can also use
this key feature to access the custom DND screen. For more information, refer to Do Not
Disturb (DND) on page 77.
Dependencies: Type (Key Event)
Key Type (DND)
Label (key label displayed on the LCD screen)
Usage:
When DND is in phone mode:
1.
Press the DSS key to activate DND.
2.
Press the DSS key again to deactivate DND.
When DND is in custom mode:
1.
Press the DSS key to access the custom DND screen. You can activate or deactivate
DND for one or all accounts.
Note
When DND is activated, the DND key LED will illuminate solid green, and the incoming
calls will be rejected automatically.
Group Listening
You can use this key feature to activate the Speakerphone and Handset/Headset mode
at the same time. It is suitable for the group conversations which have more than one
person present at one end. You are able to speak and listen through the
handset/headset, while the others nearby can only listen through the speaker.
Dependencies: Type (Key Event)
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User Guide for the SIP-T42G IP Phone
Key Type (Group Listening)
Label (key label displayed on the LCD screen)
Usage:
1.
During a call, press the DSS key to activate the group listening mode.
You can then speak and listen through the handset/headset, while other people at
your side can only listen through the speaker at the same time.
2.
Press the DSS key again to deactivate the group listening mode.
Zero Touch
You can use this key feature to quickly configure auto provision and network
parameters.
Dependencies: Type (Key Event)
Key Type (Zero Touch)
Label (key label displayed on the LCD screen)
Usage:
1.
Press the DSS key to access the zero touch screen.
2.
Press the OK soft key within a few seconds.
3.
Configure the network parameters in the corresponding fields.
4.
Press the Next soft key.
5.
Configure the auto provision parameters in the corresponding fields.
6.
Press the OK soft key.
The phone will reboot to update configurations.
URL
You can use this key feature to trigger the phone to send an HTTP GET request
containing a specific URL.
Dependencies: Type (URL)
Label (key label displayed on the LCD screen)
Value (the URL contained in the HTTP GET request)
Usage: Press the DSS key to trigger the phone to send an HTTP GET request containing
the URL specified in the Value field.
Phone Lock
You can use this key feature to immediately lock your phone instead of long
pressing
. For more information, refer to Phone Lock on page 23.
Dependencies: Type (Key Event)
Key Type (Phone Lock)
Label (key label displayed on the LCD screen)
Usage: Press the DSS key to immediately lock your phone instead of long
pressing
54
.
Customizing Your Phone
Directory
You can use this key feature to easily access frequently used lists. For more information,
refer to Directory on page 28.
Dependencies: Type (Key Event)
Key Type (Directory)
Label (key label displayed on the LCD screen)
Usage: Press the DSS key to immediately access frequently used lists.
Note
The DSS key performs the same function as the Directory soft key when the phone is idle.
You can customize the soft keys, navigation keys and function keys.
To customize programmable keys via web user interface:
1.
Click on DSSKey->Programmable Key.
2.
Customize specific features for these keys.
3.
Click Confirm to accept the change.
You can click Reset to default to reset custom settings to defaults.
Then you can press the keys on the phone to perform the features you configured.
For example:
Switch Account Up
You can press this key to switch default account.
Dependencies: Type (Switch Account Up)
Usage: Press the programmable key to scroll up the account list to select the desired
default account.
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User Guide for the SIP-T42G IP Phone
Switch Account Down
You can press this key to switch default account.
Dependencies: Type (Switch Account Down)
Usage: Press the programmable key to scroll down the account list to select the desired
default account.
Note
Programmable keys are configurable via web user interface only.
You can register one or multiple accounts on the SIP-T42G IP phone. You can also
configure each line key to associate with an account or configure multiple line keys to
associate with an account.
To register an account via phone user interface:
1.
Press Menu->Settings->Advanced Settings (default password: admin) ->Accounts.
2.
Select the desired account and then press the Enter soft key.
3.
Select Enabled from the Activation field.
4.
Enter the desired values in Label, Display Name, Register Name, User Name,
Password and SIP Server1/2 fields respectively. Contact your system administrator
for more information.
5.
Press the Save soft key to accept the change or the Back soft key to cancel.
You can repeat steps 2 to 5 to register more accounts.
The following figures demonstrate single or multiple accounts registered on the phone:
Single account:
56
Customizing Your Phone
Multiple accounts:
To disable an account via phone user interface:
1.
Press Menu->Settings->Advanced Settings (default password: admin) ->Accounts.
2.
Select the desired account and then press the Enter soft key.
3.
Select Disabled from the Activation field.
4.
Press the Save soft key to accept the change or the Back soft key to cancel.
Registering an account is configurable via web user interface at the path
Account->Register.
You can configure multiple line keys to associate with an account. This enhances call
visualization and simplifies call handling.
If this is the case, the LCD screen will resemble the following figure:
Incoming calls to this account will be distributed evenly among the available line keys.
Outgoing calls will be distributed similarly.
Your phone can be configured to have a combination of accounts with a single line key
and accounts with multiple line keys.
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User Guide for the SIP-T42G IP Phone
Dial plan is a string of characters that governs the way your SIP-T42G IP phone
processes the inputs received from your phone keypad. The SIP-T42G IP phone supports
the following dial plan features:

Replace Rule

Dial-now

Area Code

Block Out
Basic expression syntaxes you need to know:
Syntax
Description
The dot "." can be used as a placeholder or multiple placeholders for
.
any character. Example:
"12." would match "123", "1234", "12345", "12abc", etc.
x
An "x" can be used as a placeholder for any character. Example:
"12x" would match "121", "122", "123", "12a", etc.
The square brackets "[]" can be used as a placeholder for a single
[]
character which matches any of a set of characters. Example:
"91[5-7]1234" would match "9151234", "9161234", "9171234".
The parentheses "( )" can be used to group together patterns, for
()
instance, to logically combine two or more patterns. Example:
"([1-9])([2-7])3" would match "923", "153", "773",etc.
The “$” should be followed by the sequence number of a parenthesis.
The “$” plus the sequence number means the whole character or
characters placed in the parenthesis. The number directs to the right
parenthesis when there are more than one. Example:
$
A replace rule configuration, Prefix: "001(xxx)45(xx)", Replace:
"9001$145$2". When you dial out "0012354599" on your phone, the IP
phone will replace the number with "90012354599". “$1” means 3 digits
in the first parenthesis, that is, “235”. “$2” means 2 digits in the second
parenthesis, that is, “99”.
You can configure one or more replace rules (up to 100) to remove the specified string
and replace it with another string. You can configure a pattern with wildcards (refer to
the expression syntax in the table above), so that any string that matches the pattern
will be replaced. This feature is convenient for you to dial out a long number. For
example, a replace rule is configured as "Prefix: 1" and "Replace: 1234", when you try to
dial out the number “1234”, you just need to enter “1” on the phone and then press the
Send soft key.
58
Customizing Your Phone
To add a replace rule via web user interface:
1.
Click on Settings->Dial Plan->Replace Rule.
2.
Enter the string (e.g., 1) in the Prefix field.
3.
Enter the string (e.g., 1234) in the Replace field.
4.
Enter the desired line ID in the Account field or leave it blank.
5.
Click Add to add the replace rule.
When you enter the number “1” using the keypad and then press the Send soft key,
the phone will dial out “1234” instead.
Note
The valid values for the Account field can be one or more digits among 1-12. Every two
digits must be separated by a comma. For example, when you enter the value “1, 2” in
the Account field, this replace rule will apply to account 1 and account 2.
If you leave the Account field blank or enter 0, the replace rule will apply to all accounts.
To edit a replace rule via web user interface:
1.
Click on Settings->Dial Plan->Replace Rule.
2.
Select the desired replace rule by checking the checkbox.
3.
Edit values in the Prefix and Replace fields.
4.
Enter the desired line ID in the Account field or leave it blank.
5.
Click Edit to accept the change.
To delete one or more replace rules via web user interface:
1.
Click on Settings->Dial Plan->Replace Rule.
2.
Select the one or more replace rules by checking the checkbox(es).
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User Guide for the SIP-T42G IP Phone
3.
Note
Click Del to delete the replace rule(s).
Replace rule is configurable via web user interface only.
You can configure one or more dial-now rules (up 100) on your phone. When the dialed
number matches the dial-now string, the number will be dialed out automatically. For
example, a dial-now rule is configured as "2xx", any entered three-digit string beginning
with 2 will then be dialed out automatically on the phone.
To add a dial-now rule via web user interface:
1.
Click on Settings->Dial Plan->Dial-now.
2.
Enter the desired value (e.g., 2xx) in the Rule field.
3.
Enter the desired line ID in the Account field or leave it blank.
For more information on the valid values for the Account field, refer to Replace Rule
on page 58.
4.
Click Add to add the dial-now rule.
When you enter the number “234” using the keypad, the phone will dial out “234”
automatically without the pressing of any key.
Note
You can also edit or delete the dial-now rule, refer to Replace Rule on page 58 for more
information.
Dial-now rule is configurable via web user interface only.
60
Customizing Your Phone
Delay Time for Dial-Now Rule
You can configure the interval for dial-now rules. That is, you can configure your phone
to automatically dial out the phone number which matches a dial-now rule, after the
designated period of time.
To configure the delay time for dial-now rule via web user interface:
1.
Click on Features->General Information.
2.
Enter the time between 1 and 14 (seconds) in the Time-Out for Dial-Now Rule field.
3.
Click Confirm to accept the change.
Note
Delay time for dial-now rule is configurable via web user interface only.
Area codes are also known as Numbering Plan Areas (NPAs). They usually indicate
geographical areas in a country. This feature is necessary when dialing a phone
number outside the code area. For example, an area code is configured as "Code: 0592,
Min Length: 1, Max Length: 15". When you dial out the number "56789" (the length of the
number is between 1 and 15), the phone will add the area code and dial out the
number "059256789". You can only configure one area code rule on your phone.
To configure the area code and lengths via web user interface:
1.
Click on Settings->Dial Plan->Area Code.
2.
Enter the desired values in the Code, Min Length (1-15) and Max Length (1-15)
fields.
3.
Enter the desired line ID in the Account field or leave it blank.
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User Guide for the SIP-T42G IP Phone
For more information on the valid values for the Account field, refer to Replace Rule
on page 58.
4.
Note
Click Confirm to accept the change.
The default values of minimum and maximum length are 1 and 15 respectively.
Area code is configurable via web user interface only.
You can block specific numbers (up to 10) from being dialed on your phone. When you
dial a block out number on your phone, the dialing will fail and the LCD screen will
prompt "Forbidden Number".
To add a block out number via web user interface:
62
1.
Click on Settings->Dial Plan->Block Out.
2.
Enter the desired value in the BlockOut Number field.
3.
Enter the desired line ID in the Account field or leave it blank.
Customizing Your Phone
For more information on the valid value of the Account field, refer to Replace Rule
on page 58.
4.
Note
Click Confirm to add the block out number.
Block out number is configurable via web user interface only.
Public telephone networks in countries around the world have a single emergency
telephone number (emergency services number), that allows a caller to contact local
emergency services for assistance when necessary. The emergency telephone number
may differ from country to country. It is typically a three-digit number so that it can be
easily remembered and dialed quickly. Some countries have a different emergency
number for each of the different emergency services.
You can specify the emergency telephone numbers on the IP phone for contacting the
emergency services in an emergency situation.
Note
Contact your local phone service provider for available emergency numbers in your
area.
To specify emergency numbers via web user interface:
1.
Click on Features->Phone Lock.
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User Guide for the SIP-T42G IP Phone
2.
Enter the emergency number in the Emergency field.
For multiple emergency numbers, enter a comma between every two numbers. The
default emergency numbers are 112, 911 and 110.
3.
Note
Click Confirm to accept the change.
Emergency number is configurable via web user interface only.
You can enable live dialpad on the SIP-T42G IP phone, which enables the IP phone to
automatically dial out a phone number without the pressing of the send key. You can
also configure a delay, where the phone will dial out the phone number automatically
after the designated period of time.
To enable live dialpad via web user interface:
64
1.
Click on Settings->Preference.
2.
Select Enabled from the pull-down list of Live Dialpad.
Customizing Your Phone
3.
Enter the desired delay time in the Inter Digit Time (1~14s) field.
The default delay time is 4s.
4.
Note
Click Confirm to accept the change.
Live dialpad is configurable via web user interface only.
You can dial a hotline number immediately upon lifting the handset, pressing the
speakerphone key or the line key. You can also configure a delay, where the phone will
dial out the hotline number automatically after the designated period of time.
To configure the hotline number via phone user interface:
1.
Press Menu->Features->Hot Line.
2.
Enter the desired number in the Number field.
3.
Enter the delay time (in seconds) in the Hotline Delay field.
The valid values range from 0 to10.
4.
Press the Save soft key to accept the change or the Back soft key to cancel.
Hotline is configurable via web user interface at the path Features->General
Information.
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User Guide for the SIP-T42G IP Phone
66
Basic Call Features
The SIP-T42G IP phone is designed to be easily used like a regular phone on a public
switched telephone network (PSTN). You can place calls, answer calls, transfer a call to
someone else, or conduct a conference call.
This chapter provides basic operating instructions for the SIP-T42G IP phone. Topics
include:

Placing Calls

Answering Calls

Ending Calls

Redialing Numbers

Recent Call In Dialing

Auto Answer

Auto Redial

Call Completion

Recall

Call Mute

Call Hold/Resume

Do Not Disturb (DND)

Call Forward

Call Transfer

Call Waiting

Conference

Call Park

Call Pickup

Anonymous Call

Anonymous Call Rejection
If you require additional information or assistance with your new phone, contact your
system administrator.
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User Guide for the SIP-T42G IP Phone
You can place a call in one of three ways using your SIP-T42G IP phone:

Using the handset

Using the speakerphone

Using the headset
You can also dial the number first, and then choose the way you want to speak to the
other party.
You can also search and dial a contact from call history, local directory or remote phone
book. For more information, refer to Contact Management on page 28 and Call History
Management on page 42.
During a call, you can alternate between Speakerphone, Headset, or Handset mode by
pressing the Speakerphone key, the Headset key, or by picking up the handset.
The call duration of the active call is visible on the LCD screen. In the figure below, the
call to the number”1006” has lasted 12 seconds.
To place a call using the handset:
1.
Pick up the handset.
2.
Enter the desired number using the keypad.
3.
Press
,
, or the Send soft key.
The # key is configured as a send key by default. You can also set the * key as the send
key, or set neither. For more information, refer to the Key as Send on page 23.
Note
You can also dial using the SIP URI or IP address. To obtain the IP address of a phone,
press the OK key when the phone is idle. The maximum length of SIP URI or IP address
length is 32 characters. For example, SIP URI: 3606@sip.com, IP: 192.168.1.15.
Your phone may not support direct IP dialing. Contact your system administrator for more
information.
To place a call using the hands-free speakerphone mode:
Do one of the following:
-
With the handset on-hook, press
or the line key to obtain a dial tone.
Enter the desired number using the keypad.
Press
68
,
or the Send soft key.
Basic Call Features
-
With the handset on-hook, enter the desired number using the keypad.
Press
,
,
or the Send soft key.
To place a call using the headset:
Do one of the following:
-
With the optional headset connected, press
to activate the headset mode.
Press the line key to obtain a dial tone.
Enter the desired number using the keypad.
Press
-
,
, or the Send soft key.
With the optional headset connected, press
to activate the headset mode.
Enter the desired number using the keypad.
Press
Note
,
, or the Send soft key.
To permanently use the headset mode, refer to Headset Prior on page 44.
The SIP-T42G IP phone can handle multiple calls at a time. However, only one active call
(the call that has audio associated with it) can be in progress at any given time. The
SIP-T42G IP phone can handle a maximum of 6 calls at one time.
To place multiple calls:
You can have more than one call on your SIP-T42G IP phone. To place a new call during
an active call, do one of the following:
-
Press the line key. The active call is placed on hold.
Enter the desired number using the keypad.
Press
-
,
, or the Send soft key.
Press the Hold soft key to place the original call on hold.
Press the New Call soft key.
Enter the desired number using the keypad.
Press
You can press
,
, or the Send soft key.
or
to switch between calls, and then press the Resume soft key
to retrieve the desired call.
Note
If multiple accounts are registered on the phone, you can first press the desired line key
in the idle screen or press the Line soft key in the dialing screen, and then you can use
the selected account to place a call.
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User Guide for the SIP-T42G IP Phone
When you are not in another call, you can answer a call in one of three ways:

Using the handset

Using the speakerphone

Using the headset
Note
You can reject incoming calls by pressing the X key, or the Reject soft key. You can also
activate Do Not Disturb mode to reject the incoming calls without ringing on your phone.
For more information, refer to Do Not Disturb (DND) on page 77.
You can forward incoming calls to someone else by pressing the Forward soft key. For
more information, refer to Call Forward on page 81.
Answering When Not in Another Call
Call duration and destination will always appear on the LCD screen for the active call.
To answer a call using the handset:
1.
Pick up the handset.
To answer a call using the hands-free speakerphone mode:
Do one of the following:
-
Press
.
-
With the handset on-hook and the headset mode deactivated, press the Answer
soft key.
-
With the handset on-hook and the headset mode deactivated, press the line key
(the line key LED flashes green).
To answer a call using the headset:
Do one of the following:
-
Press
-
With the headset mode activated, press the Answer soft key.
-
With the headset mode activated, press the line key (the line key LED flashes
green).
70
.
Basic Call Features
Answering When in Another Call
If you have an active call, and an incoming call arrives on the phone, do one of the
following:
-
Press the Answer soft key.
The incoming call is answered and the original call is placed on hold.
-
Press
to access the new call.
Press
or the Answer soft key.
The incoming call is answered and the original call is placed on hold.
To end a call:
Do one of the following:
-
If you are using the handset, press the End Call soft key or hang up the handset.
-
If you are using the headset, press the End Call soft key.
-
If you are using the speakerphone, press
Note
or the End Call soft key.
When a call is placed on hold, you cannot press
to end it. You should press the
Resume soft key to resume the call, and then press
to end it.
To redial the last dialed number from your phone:
1.
Press
twice.
A call to your last dialed number is attempted.
To redial a previously dialed number from your phone:
1.
Press
2.
Press
when the phone is idle.
or
to select the desired entry from the placed calls list, and then
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User Guide for the SIP-T42G IP Phone
press
or the Send soft key.
To view the placed calls list when the phone is in the pre-dialing interface, you should
enable recent call in dialing in advance.
To enable recent call in dialing via web user interface:
1.
Click on Directory->Setting.
2.
Select Enabled from the pull-down list of Recent Call In Dialing.
3.
Click Confirm to accept the change.
Note
Recent call in dialing is configurable via web user interface only.
To view placed calls list when the phone is in the pre-dialing interface:
1.
Pick up the handset, press the speakerphone or press the line key.
The LCD screen displays the placed calls list.
72
Basic Call Features
You can use auto answer to automatically answer an incoming call on a line. Auto
answer is configurable on a per-line basis.
To configure auto answer via phone user interface:
1.
Press Menu->Features->Auto Answer.
2.
Press
or
, or the Switch soft key to select the desired line from the Account
or
, or the Switch soft key to select Enabled from the Status field.
field.
3.
Press
4.
Press the Save soft key to accept the change or the Back soft key to cancel.
The
icon appears on the LCD screen.
Auto answer is configurable via web user interface at the path Account->Basic.
Note
Auto answer is only applicable when there is no other call in progress on the phone.
You can enable auto redial to automatically redial a phone number when the called
party is busy. You can also configure the number of auto redial attempts and the time to
wait between redial attempts.
To configure auto redial via phone user interface:
1.
Press Menu->Features->Auto Redial.
2.
Press
or
, or the Switch soft key to select Enabled from the Auto Redial
field.
3.
Enter the desired time (in seconds) in the Redial Interval field.
The default time interval is 10 seconds.
4.
Enter the desired number of redial attempts in the Redial Times field.
The default value is 10.
5.
Press the Save soft key to accept the change or the Back soft key to cancel.
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User Guide for the SIP-T42G IP Phone
Auto redial is configurable via web user interface at the path Features->General
Information.
To use auto redial:
When the called party is busy, the LCD screen prompts the following:
1.
Press the OK soft key to activate auto redial.
The LCD screen prompts the following:
2.
Wait for the designated period of time or press the OK soft key to redial the phone
number.
The phone will retry as many times as configured until the called party is idle.
You can use call completion to notify the caller who failed to reach a desired party when
the party becomes available to receive a call.
To configure call completion via phone user interface:
1.
Press Menu->Features->Call Completion.
2.
Press
or
, or the Switch soft key to select Enabled from the Call
Completion field.
3.
Press the Save soft key to accept the change or the Back soft key to cancel.
Call completion is configurable via web user interface at the path Features->General
Information.
74
Basic Call Features
To use call completion:
When the called party is busy, the following prompt will appear on the LCD screen of the
phone:
1.
Press the OK soft key, the phone returns to the idle screen and call completion is
activated.
When the called party becomes idle, the following prompt appears on the LCD
screen of the phone:
2.
Note
Press the OK soft key to redial the number.
Call completion is not available on all servers. For more information, contact your system
administrator.
You can press a recall key to place a call back to the last incoming call.
To configure a recall key via phone user interface:
1.
Press Menu->Features->DSS Keys.
2.
Select a desired DSS key.
3.
Press
or
, or the Switch soft key to select Key Event from the Type field.
4.
Press
or
, or the Switch soft key to select ReCall from the Key Type field.
5.
(Optional.) Enter a string that will appear on the LCD screen in the Label field.
6.
Press the Save soft key to accept the change or the Back soft key to cancel.
Recall key is configurable via web user interface at the path DSSKey->Line Key.
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User Guide for the SIP-T42G IP Phone
You can mute the microphone of the active audio device during an active call so that
the other party cannot hear you. Call mute applies to all modes (Handset, Headset and
Speakerphone).
To mute a call:
1.
Press
during an active call.
The LCD screen indicates that the call is now muted. The mute key LED illuminates
solid red when the call is muted.
To un-mute a call:
1.
Press
again to un-mute the call.
The mute key LED goes out.
You can place an active call on hold. Only one active call can be in progress at any time.
Other calls can be made and received while placing the original call on hold. When you
place a call on hold, your IP PBX may play music to the other party while waiting.
To place a call on hold:
1.
Press the Hold soft key during a call.
The line key LED flashes green. The LCD screen indicates that the call is on hold.
Note
The phone will beep softly every 30 seconds to remind you that you still have a call on
hold.
To resume a held call:
1.
76
Press the Resume soft key.
Basic Call Features
Multiple Calls on Hold:
If multiple calls are placed on hold, do one of the following:
Press
-
or
to switch between the calls, and then press the Resume soft key
to retrieve the desired call.
Press the corresponding line key to retrieve the call.
-
If more than one call is placed on hold, a numbered prompt appears on the LCD screen,
for example "2/3", indicating that this is the second call out of three calls.
You can use DND to reject incoming calls automatically on the phone. The prompt
message "n New Missed Call(s)" ("n" indicates the number of missed calls) will appear
on the LCD screen, and callers will receive a busy message. All calls you receive while
DND is enabled are logged to your missed calls list.
Note
The prompt message will display only if Missed Call Log for the line is enabled. Missed
call log is configurable via web user interface at the path Account->Basic.
Do not disturb is local to the phone, and may be overridden by the server settings. For
more information, contact your system administrator.
You can enable/disable DND for the phone system, or you can customize DND for each
or all accounts. Two DND modes:

Phone (default): DND is effective for the phone system.

Custom: DND can be configured for each or all accounts
You can receive incoming calls from authorized numbers when DND is enabled.
To configure the DND mode via web user interface:
1.
Click on Features->Forward & DND.
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User Guide for the SIP-T42G IP Phone
Note
2.
In the DND block, mark the desired radio box in the Mode field.
3.
Click Confirm to accept the change.
DND mode is configurable via web user interface only.
To activate DND in phone mode:
1.
Press the DND soft key when the phone is idle.
The
icon on the idle screen indicates that DND is enabled.
Incoming calls will be rejected automatically and "n New Missed Call(s)" ("n"
indicates the number of missed calls) will prompt on the LCD screen.
Note
78
When DND and busy forward are enabled in phone mode, calls will be sent to the
configured destination number. For more information on busy forward, refer to Call
Forward on page 80.
Basic Call Features
To activate DND in custom mode for a specific account:
1.
Press the DND soft key when the phone is idle.
The LCD screen displays a list of accounts registered on the phone.
2.
Press
or
to select the desired account and then press the Enter soft key.
3.
Press
or
, or the Switch soft key to select Enabled from the DND Enables
field.
4.
Press the Save soft key to accept the change.
If you activate DND for the default account, the associated line icon will change
to
, and the
icon will appear on the status bar.
If you activate DND for the non-default account, only the associated line icon will
change to
.
Incoming calls on the specific line will be rejected automatically, and the LCD
screen prompts "n New Missed Call(s)" ("n" indicates the number of missed calls).
Note
When DND and busy forward are enabled for a specific account, calls from the specific
account will be sent to the configured destination number. For more information on busy
forward, refer to Call Forward on page 80.
To activate DND in custom mode for all accounts:
1.
Press the DND soft key when the phone is idle.
The LCD screen displays a list of accounts registered on the phone.
2.
Press the All On soft key to activate DND for all accounts.
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User Guide for the SIP-T42G IP Phone
3.
Press the Save soft key to accept the change.
The
icon appears on the idle screen, and all line icons change to
.
Incoming calls will be rejected automatically and "n New Missed Call(s)" ("n"
indicates the number of missed calls) will be prompted on the LCD screen.
To configure the DND authorized numbers via web user interface:
1.
Click on Features->Forward & DND.
2.
Select Enabled from the pull-down list of DND Emergency.
3.
Enter the numbers in the DND Authorized Numbers field.
For multiple numbers, enter a comma between every two numbers.
4.
Click Confirm to accept the change.
When DND is enabled on the phone, the phone can still receive incoming calls from
the numbers specified in the DND Authorized Numbers field.
Note
DND authorized number is configurable via web user interface only.
When the phone receives a voice mail, the voice mail prompt window will pop up by
default, if you want to disable the feature, contact your system administrator for more
information.
80
Basic Call Features
You can configure your phone to forward incoming calls to another party through static
forwarding. You can also forward incoming calls while your phone is ringing; refer to the
dynamic forwarding.
Static Forwarding
Three types of static forwarding:

Always Forward: Incoming calls are immediately forwarded.

Busy Forward: Incoming calls are immediately forwarded if the phone is busy.

No Answer Forward: Incoming calls are forwarded if not answered after a period
of time.
You can enable/disable call forward for the phone, or you can customize call forward for
each or all accounts. Two call forward modes:

Phone (default): Call forward is effective for the phone system.

Custom: Call forward can be configured for each or all accounts.
To configure the call forward mode via web user interface:
Note
1.
Click on Features->Forward & DND.
2.
In the Forward block, mark the desired radio box in the Mode field.
3.
Click Confirm to accept the change.
Call forward mode is configurable via web user interface only.
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User Guide for the SIP-T42G IP Phone
To enable call forward in phone mode:
1.
Press Menu->Features->Call Forward.
2.
Press
or
to select the desired forwarding type, and then press the Enter
soft key.
3.
Depending on your selection:
a.) If you select Always Forward:
1) Press
or
, or the Switch soft key to select Enabled from the Always
Forward field.
2) Enter the destination number you want to forward all incoming calls to in
the Forward to field.
3) (Optional.) Enter the always forward on code or off code respectively in the
On Code or Off Code field.
b.) If you select Busy Forward:
1) Press
or
, or the Switch soft key to select Enabled from the Busy
Forward field.
2) Enter the destination number you want to forward all incoming calls to
when the phone is busy in the Forward to field.
3) (Optional.) Enter the busy forward on code or off code respectively in the
On Code or Off Code field.
c.) If you select No Answer Forward:
1) Press
or
, or the Switch soft key to select Enabled from the No
Answer Forward field.
2) Enter the destination number you want to forward all unanswered incoming
calls to in the Forward to field.
82
Basic Call Features
3) Press
or
, or the Switch soft key to select the ring time to wait
before forwarding from the After Ring Time field.
The default ring time is 12 seconds.
4) (Optional.) Enter the no answer forward on code or off code respectively in
the On Code or Off Code field.
4.
Press the Save soft key to accept the change or the Back soft key to cancel.
The
icon on the idle screen indicates that the call forward is enabled.
To enable call forward in custom mode:
1.
Press Menu->Features->Call Forward.
2.
Press
or
to select the desired account, and then press the Enter soft key.
3.
Press
or
to select the desired forwarding type, and then press the Enter
soft key.
4.
Depending on your selection:
a.) If you select Always Forward, you can enable it for a specific account.
1) Press
or
, or the Switch soft key to select Enabled from the Always
Forward field.
2) Enter the destination number you want to forward all incoming calls to in
the Forward to field.
3) (Optional.) Enter the always forward on code or off code respectively in the
On Code or Off Code field.
You can also enable always forward for all accounts. After always forward was
enabled for a specific account, do the following:
1) Press
or
to highlight the Always Forward field.
2) Press the All Lines soft key.
The LCD screen prompts “Copy to all lines?”.
3) Press the OK soft key to accept the change or the Cancel soft key to cancel.
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User Guide for the SIP-T42G IP Phone
b.) If you select Busy Forward, you can enable it for a specific account.
1) Press
or
, or the Switch soft key to select Enabled from the Busy
Forward field.
2) Enter the destination number you want to forward incoming calls to when
the phone is busy in the Forward to field.
3) (Optional.) Enter the busy forward on code or off code respectively in the
On Code or Off Code field.
You can also enable busy forward for all accounts. After busy forward was
enabled for a specific account, do the following:
1) Press
or
to highlight the Busy Forward field.
2) Press the All Lines soft key.
The LCD screen prompts “Copy to all lines?”.
3) Press the OK soft key to accept the change or the Cancel soft key to cancel.
c.) If you select No Answer Forward, you can enable it for a specific account.
1) Press
or
, or the Switch soft key to select Enabled from the No
Answer Forward field.
2) Enter the destination number you want to forward all unanswered incoming
calls to in the Forward to field.
3) Press
or
, or the Switch soft key to select the ring time to wait
before forwarding from the After Ring Time field.
The default ring time is 12 seconds.
4) (Optional.) Enter the no answer forward on code or off code respectively in
the On Code or Off Code field.
You can also enable no answer forward for all accounts. After no answer
forward was enabled for a specific account, do the following:
1) Press
84
or
to highlight the No Answer Forward field.
Basic Call Features
2) Press the All Lines soft key.
The LCD screen prompts “Copy to all lines?”.
3) Press the OK soft key to accept the change or the Cancel soft key to cancel.
5.
Press the Save soft key to accept the change or the Back soft key to cancel.
If you activate call forward for the default account, the associated line icon will change
to
, and the
icon will appear on the status bar.
If you activate call forward for the non-default account, only the associated line icon will
change to
.
Call forward is configurable via web user interface at the path Features->Forward &
DND.
Note
You can also enter the SIP URI or IP address in the Forward to field. For more information
on using the SIP URI or IP address, refer to Placing Calls on page 68.
Call forward is local to the phone, and may be overridden by the server settings. Call
forward on code or off code may be different between servers. For more information,
contact your system administrator.
To disable call forward in phone mode:
1.
Press Menu->Features->Call Forward.
2.
Press
or
to select the desired forwarding type, and then press the Enter
or
, or the Switch soft key to select Disabled to disable the call
soft key.
3.
Press
forward.
4.
Press the Save soft key to accept the change.
To disable call forward in custom mode for a specific account:
1.
Press Menu->Features->Call Forward.
2.
Press
or
to select the desired account, and then press the Enter soft key.
3.
Press
or
to select the desired forwarding type, and then press the Enter
or
, or the Switch soft key to select Disabled to disable the call
soft key.
4.
Press
forward.
5.
Press the Save soft key to accept the change.
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User Guide for the SIP-T42G IP Phone
Dynamic Forwarding
To forward an incoming call to another party:
1.
When the phone is ringing, press the Forward soft key.
2.
Enter the number you want to forward the incoming call to.
3.
Press
,
, or the Send soft key.
The LCD screen prompts a call forward message.
Note
When the phone forwards a call, the forward call prompt window will pop up by default,
if you want to disable the feature, contact your system administrator for more
information.
You can transfer a call to another party in one of three ways:

Blind Transfer: Transfer a call directly to another party without consulting.

Semi-Attended Transfer: Transfer a call when the target phone is ringing.

Attended Transfer: Transfer a call with prior consulting.
To perform a blind transfer:
1.
Press the Transfer soft key during a call.
2.
Enter the number you want to transfer the call to.
3.
Press the Transfer soft key to complete the transfer.
Then the call is connected to the number to which you are transferring.
To perform a semi-attended transfer:
86
1.
Press the Transfer soft key during a call.
2.
Do one of the following:
-
Enter the number you want to transfer the call to.
-
Press the Directory soft key, and then select Local Directory. Select the desired
Basic Call Features
group and search for the contact (Directory should be configured in advance.
Refer to Directory on page 28 for more information).
-
Press the Directory soft key, and then select History. Select the desired list and
press
or
to select the entry (Directory should be configured in
advance. Refer to Directory on page 28 for more information).
-
Press the Directory soft key, and then select Remote Phone Book. Select the
desired group and search for the contact (Directory and Remote Phone Book
should be configured in advance. Refer to Directory on page 28 and Remote
Phone Book on page 39 for more information).
3.
Press
or
to dial out.
4.
Press the Transfer soft key to complete the transfer when receiving ringback.
To perform an attended transfer:
1.
Press the Transfer soft key during a call.
2.
Do one of the following:
-
Enter the number you want to transfer the call to.
-
Press the Directory soft key, and then select Local Directory. Select the desired
group and search for the contact (Directory and Remote Phone Book should be
configured in advance. Refer to Directory on page 28 for more information).
-
Press the Directory soft key, and then select History. Select the desired list and
press
or
to select the entry (Directory should be configured in
advance. Refer to Directory on page 28 for more information).
-
Press the Directory soft key, and then select Remote Phone Book. Select the
desired group and search for the contact (Directory and Remote Phone Book
should be configured in advance. Refer to Directory on page 28 and Remote
Phone Book on page 39 for more information).
3.
Enter the number you want to transfer the call to.
4.
Press
5.
After the party answers the call, press the Transfer soft key to complete the transfer.
or
to dial out.
If you are using a handset, the transfer can be completed by hanging up the handset.
You can cancel the transfer before the call is connected by pressing the Cancel soft key.
You can enable or disable call waiting on the phone. If call waiting is enabled, you can
receive another call while there is already an active call on the phone. Otherwise,
another incoming call is automatically rejected by the phone with a busy message
when there is an active call on the phone. You can also enable or disable the phone to
play a warning tone when receiving another call.
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User Guide for the SIP-T42G IP Phone
To configure call waiting via phone user interface:
1.
Press Menu->Features->Call Waiting.
2.
Press
or
, or the Switch soft key to select Enabled from the Call Waiting
or
, or the Switch soft key to select Enabled from the Play Tone field.
field.
3.
Press
4.
(Optional.) Enter the call waiting on code or off code respectively in the On Code or
Off Code field.
5.
Press the Save soft key to accept the change or the Back soft key to cancel.
Call waiting is configurable via web user interface at the path Features->General
Information.
You can create a conference with other two parties using the phone’s local conference.
You can create a conference between an active call and a call on hold by pressing the
Conf soft key. The SIP-T42G IP phone also supports network conference.
Note
Network conference is not available on all servers. For more information, contact your
system administrator.
The SIP-T42G IP phone supports up to 3 parties (including yourself) in a conference call.
This is the default method of conference called Local Conference.
To set up a local conference call:
1.
Place a call to the first party.
2.
When the first party answers the call, press the Conf soft key to place a new call.
The active call is placed on hold.
3.
88
Enter the number of the second party and press
,
, or the Send soft key.
Basic Call Features
4.
When the second party answers the call, press the Conf soft key again to join all
parties in the conference.
To join two calls in a conference:
1.
Place two calls using two different accounts on the phone (for example, place the
first call using account 1, and then place the second call using account 2).
2.
Press
or
to select the call for conference and ensure that the call is active
(for example, select the call on account 1).
3.
Press the Conf soft key to join the two calls in the conference on account 1.
During the conference call, you can do the following actions:
-
Press the Hold soft key to place the conference on hold.
-
Press the Split soft key to split the conference call into two individual calls on hold.
-
Press the Manage soft key, and then press

or
to select the desired party:
Press the Far Mute soft key to forbid the party from speaking. The muted party
can hear everyone, but no one can hear the muted party.

Press the Remove soft key to remove the party from the conference call.

Press the New Call soft key to place a new call.

Press the Back soft key to return to the previous interface.
-
Press
to mute the conference call.
-
Press the End Call soft key to drop the conference call.
You can use network conference on the SIP-T42G IP phone to conduct a conference with
multiple participants.
This feature allows you to perform the following:

Join two calls together into a conference call.

Invite another party into an active conference call.
To use this feature, contact your system administrator for the network conference URI in
advance.
To configure network conference via web user interface:
1.
Click on Account->Advanced.
2.
Select Network Conference from the pull-down list of Conference Type.
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User Guide for the SIP-T42G IP Phone
3.
Enter the conference URI (e.g., conference@example.com) in the Conference URI
field.
4.
Note
Click Confirm to accept the change.
Network conference is configurable via web user interface only.
To set up a network conference call:
1.
Place a call to the first party.
2.
Press the Conf soft key to place a new call.
The active call is placed on hold.
3.
Enter the number of the second party and press
,
, or the Send soft key.
4.
When the second party answers the call, press the Conf soft key to add the second
party to the conference.
5.
Press the Conf soft key to place a new call.
The conference is placed on hold.
6.
Enter the number of the new party and then press
,
, or the Send soft key.
7.
When the new party answers the call, press the Conf soft key to add the new party
to the conference.
8.
Repeat steps 5 to 7 until you have added all intended parties.
The procedures to set up a network conference call on specific servers may be different
from that introduced above. Contact your system administrator for more information.
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Basic Call Features
You can use call park to place a call on hold, and then retrieve the call from another
phone in the system (for example, a phone in another office or conference room). You
can park an active call by pressing the call park key on the phone. If the call is parked
successfully, the response is either a voice prompt confirming that the call was parked,
or a visible prompt on the LCD screen.
Note
Call park is not available on all servers. Contact your system administrator for more
information.
To configure a call park key via phone user interface:
1.
Press Menu->Features->DSS Keys.
2.
Select the desired DSS key.
3.
Press
or
, or the Switch soft key to select Key Event from the Type field.
4.
Press
or
, or the Switch soft key to select Call Park from the Key Type field.
5.
Press
or
, or the Switch soft key to select the desired line from the Account
ID field.
6.
(Optional.) Enter the string that will appear on the LCD screen in the Label field.
7.
Enter the call park code in the Value field.
8.
Press the Save soft key to accept the change or the Back soft key to cancel.
Call park key is configurable via web user interface at the path DSSKey->Line Key.
To use call park:
1.
User on phone A places a call to phone B.
2.
User on phone A wants to take the call in a conference room for privacy, and so
presses the call park key on phone A.
3.
User on phone A walks to an available conference room where the phone is
designated as phone C. The user dials the call park retrieve code to retrieve the
parked call.
The system establishes call between phone C and B.
Note
The call park code and call park retrieve code are predefined on the system server.
Contact your system administrator for more information.
If the parked call is not retrieved within a period of time assigned by the system, the
phone performing call park will receive a call back.
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User Guide for the SIP-T42G IP Phone
You can use call pickup to answer someone else’s incoming call on your phone. The
SIP-T42G IP phone supports directed call pickup and group call pickup. Directed call
pickup is used for picking up a call that is ringing at a target phone number. Group call
pickup is used for picking up a call that is ringing at any phone number in a certain
group. The pickup group should be predefined, contact your system administrator for
more information.
You can pick up an incoming call by using the DPickup/GPickup soft key. To use call
pickup, you need to configure the call pickup code beforehand on a global or per-line
basis via web user interface.
Note
If there are many incoming calls at the same time, pressing the GPickup soft key on the
phone will pick up the call that rings first.
Directed Call Pickup
To enable directed call pickup and configure the directed call pickup code on a global
basis via web user interface:
1.
Click on Features->Call Pickup.
2.
Select Enabled from the pull-down list of Directed Call Pickup.
3.
Enter the directed call pickup code in the Directed Call Pickup Code field.
4.
Click Confirm to accept the change.
To configure the directed call pickup code on a per-line basis via web user interface:
92
1.
Click on Account.
2.
Select the desired account from the pull-down list of Account.
Basic Call Features
3.
Click on Advanced.
4.
Enter the directed call pickup code in the Directed Call Pickup Code field.
5.
Click Confirm to accept the change.
To pick up a call directly:
1.
Pick up the handset and press the More soft key.
The DPickup soft key appears on the LCD screen.
1.
Press the DPickup soft key on your phone when the target phone receives an
incoming call.
2.
Enter the phone number which is receiving an incoming call.
3.
Press the DPickup soft key again.
The call is answered on your phone.
You can also configure a DSS key as directed pickup key via phone user interface or
web user interface. Once configured, you can pick up a call by pressing the directed
pickup key directly.
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User Guide for the SIP-T42G IP Phone
Group Call Pickup
To enable group call pickup and configure the group call pickup code on a global basis
via web user interface:
1.
Click on Features->Call Pickup.
2.
Select Enabled from the pull-down list of Group Call Pickup.
3.
Enter the group call pickup code in the Group Call Pickup Code field.
4.
Click Confirm to accept the change.
To configure the group call pickup code on a per-line basis via web user interface:
94
1.
Click on Account.
2.
Select the desired account from the pull-down list of Account.
3.
Click on Advanced.
Basic Call Features
4.
Enter the group call pickup code in the Group Call Pickup Code field.
5.
Click Confirm to accept the change.
To pick up a call in the group:
1.
Pick up the handset.
The GPickup soft key appears on the LCD screen.
2.
Press the GPickup soft key on your phone when a phone in the group receives an
incoming call.
The call is answered on your phone.
You can also configure a DSS key as group pickup key via phone user interface or web
user interface. Once configured, you can pick up a call by pressing the group pickup
key directly.
Note
The directed call pickup code and group call pickup code are predefined on the system
server. Contact your system administrator for more information.
The call pickup code configured on a per-line basis takes precedence over that
configured on a global basis.
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User Guide for the SIP-T42G IP Phone
You can use anonymous call to block your identify and phone number from appearing
to the called party when you call someone. For example, you want to call to consult
some services, but don’t want to be harassed. Anonymous call is configurable on a
per-line basis. You can also configure the phone to send anonymous call on/off code to
the server to activate/deactivate anonymous call on the server side.
Note
Anonymous call is not available on all servers. Contact your system administrator for the
anonymous call on code and off code.
To configure anonymous call via phone user interface:
1.
Press Menu->Features->Anonymous Call.
2.
Press
or
, or the Switch soft key to select the desired line from the Account
or
, or the Switch soft key to select Enabled from the Local
ID field.
3.
Press
Anonymous field.
4.
(Optional.) Press
or
, or the Switch soft key to select the desired value from
the Send Anonymous Code field.
5.
(Optional.) Enter the anonymous call on code and off code respectively in the On
Code and Off Code field.
The phone will send the configured on code or off code depending on your
selection when you enable or disable anonymous call feature on the phone.
6.
Press the Save soft key to accept the change or the Back soft key to cancel.
Anonymous call is configurable via web user interface at the path Account->Basic.
To place an anonymous call:
1.
Using the specific line on the phone to place a call to phone B.
The LCD screen of phone B prompts an incoming call from anonymity.
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Basic Call Features
You can use anonymous call rejection to reject incoming calls from anonymous callers.
Anonymous call rejection automatically rejects incoming calls from callers who
deliberately block their identities and numbers from being displayed. Anonymous call
rejection is configurable on a per-line basis. You can also configure the phone to send
anonymous call rejection on/off code to the server to activate/deactivate anonymous
call rejection on the server side.
To configure anonymous call rejection via phone user interface:
1.
Press Menu->Features->Anonymous Call.
2.
Press
or
, or the Switch soft key to select the desired line from the Account
ID field.
3.
Press
or
to scroll to the Local Anonymous Rejection field.
4.
Press
or
, or the Switch soft key to select Enabled from the Local
Anonymous Rejection field.
5.
(Optional.) Press
or
, or the Switch soft key to select the desired value from
the Send Anonymous Rejection Code field.
6.
(Optional.) Enter the anonymous call rejection on code and off code respectively in
the On Code and Off Code field.
The phone will send the configured on code or off code depending on your
selection when you enable or disable local anonymous rejection feature on the
phone.
7.
Press the Save soft key to accept the change or the Back soft key to cancel.
Anonymous call rejection is configurable via web user interface at the path
Account->Basic.
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User Guide for the SIP-T42G IP Phone
98
Advanced Phone Features
This chapter provides operating instructions for the advanced features of the SIP-T42G IP
phone. Topics include:

Busy Lamp Field (BLF)

BLF List

Call Recording

Hot Desking

Intercom

Multicast Paging

Music on Hold

Automatic Call Distribution (ACD)

Messages
If you require additional information or assistance with your new phone, contact your
system administrator.
You can use BLF to monitor a specific line for status changes on the phone. For example,
you can configure a BLF key on the phone to monitor the status of a friend’s line (busy or
idle). The BLF key LED illuminates solid red when the friend’s line is in use. For more
information on BLF key LED indications, refer to LED Instructions on page 4.
You can press a BLF key to dial out the monitored phone number when the monitored
line is idle. You can receive a visual or/and an audio alert (if enabled), and also pick up
calls that are received on the monitored line. For more information, contact your system
administrator.
To configure a BLF key via phone user interface:
1.
Press Menu->Features->DSS Keys.
2.
Select the desired DSS key.
3.
Press
or
, or the Switch soft key to select BLF from the Type field.
4.
Press
or
, or the Switch soft key to select the desired line from the Account
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User Guide for the SIP-T42G IP Phone
ID field.
5.
(Optional.) Enter a string that will appear on the LCD screen in the Label field.
6.
Enter the phone number or extension you want to monitor in the Value field.
7.
(Optional.) Enter the pickup code in the Extension field.
8.
Press the Save soft key to accept the change or the Back soft key to cancel.
BLF key is configurable via web user interface at the path DSSKey->Line Key.
You can enable audio alert feature for BLF pickup on the phone. This allows the
monitoring phone to play a warning tone when the monitored line receives an incoming
call. You can also enable visual alert feature for BLF pickup on the phone. This allows the
LCD screen of the monitoring phone to display the caller ID when the monitored line
receives an incoming call.
To enable visual and audio alert features via web user interface:
1.
Click on Features->Call Pickup.
2.
Select Enabled from the pull-down list of Visual Alert for BLF Pickup.
3.
Select Enabled from the pull-down list of Audio Alert for BLF Pickup.
4.
Click Confirm to accept the change.
Note
Visual and audio alert features are configurable via web user interface only.
When the monitored line receives an incoming call, the following occurs on your phone:
100

The phone plays a warning tone (if enabled).

The BLF key LED flashes red.

The caller ID appears on the LCD screen (if enabled).
Advanced Phone Features
In the following figure, the LCD screen shows an incoming call from 1001 on the
monitored line.
Note
If your phone is locked and the type of the phone lock is configured as Function Keys or
All keys, you cannot use the Pickup, Send, New Call and Cancel soft keys until unlocked.
When there is an active call on the IP phone, you can transfer the active call to the
monitored phone number directly by pressing the BLF key. The phone handles the active
call differently depending on the transfer mode on DSS key. For more information on the
transfer mode on DSS key, refer to transfer in DSS Keys section on page 46.
You can use the BLF List feature to monitor a list of users defined by your system
administrator. For example, your system administrator enables BLF List, and creates a
BLF List URI (e.g., BLFList@example.com) including a list of user1, user2 on the server.
The BLF List keys on the IP phone can present the status of user1 and user2. The key LEDs
illuminate either flashing or solid depending on the status of those users. For more BLF
List key LED indications, refer to LED Instructions on page 4.
You can use the BLF List keys in the following ways:

When the monitored user is idle, press the BLF list key to dial out the phone number.

When there is already an active call on the IP phone, you can transfer the active
call to the monitored user by pressing the BLF List key. The phone handles the
active call differently depending on the transfer mode on DSS key. For more
information on the transfer mode on DSS key, refer to Transfer on page 52.

When the monitored user receives an incoming call, press the BLF list key to pick up
the call directly. Before picking up an incoming call, ensure that the directed
pickup code has been configured in advance. If the directed pickup code is not
configured, the phone will place a call to the monitored user instead of picking up
the incoming call of the monitored user when you press the BLF List key.

When there is a conversation on the monitored user, press the BLF list key to barge
in and set up a conference call. Before barging in an active call, ensure that the
barge-in code has been configured in advance. If the BLF list barge in code is not
configured, the phone will place a call to the monitored user instead of barging in
an active call of the monitored user when you press the BLF List key.

When a call is being parked against the monitored phone, press the BLF List key to
retrieve the parked call from the monitored user. Before retrieving the parked call,
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User Guide for the SIP-T42G IP Phone
ensure that the BLF List Retrieve call parked Code has been configured in advance.
If the code is not configured, the phone will place a call to the monitored user
instead of retrieving the parked call when you press the BLF List key.
To configure BLF List settings via web user interface:
1.
Click on Account->Advanced.
2.
Select the desired account from the pull-down list of Account.
3.
Enter the BLF List URI in the BLF List URI field.
4.
(Optional.) Enter the directed pickup code in the BLF List code field.
5.
(Optional.) Enter the barge-in code in the BLF List Barge In Code field.
6.
(Optional.) Enter the call park retrieve code in the BLF List Retrieve call parked
Code field.
7.
Note
Click Confirm to accept the change.
For more information on BLF List URI/BLF List Code/BLF List Barge In Code/BLF List Retrieve
Call Parked Code, contact your system administrator.
According to the response message from the server, the IP phone will automatically
configure the BLF List keys beginning from the first unused line key. Once any linekey is
seized, the IP phone will skip to configure the next line key.
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Advanced Phone Features
You can receive a visual or/and an audio alert (if enabled) on your phone when the
monitored user receives an incoming call. For more information, refer to Busy Lamp Field
(BLF) on page 99.
Note
The pickup code is used in the following order of preference: BLF List Code
(Account->Advanced)>Directed Call Pickup Code (Account->Advanced)>Directed Call
Pickup Code (Features->Call Pickup). If all of them are not configured, pressing the BLF
List key will directly call the monitored user when he/she receives an incoming call. For
more information about pickup Code, refer to Call Pickup on page 92.
You can record calls by pressing a record key on the phone. The SIP-T42G IP phone
supports record and URL record.
Two ways of call recording:
Record: The phone sends a SIP INFO message containing a specific header “Record:

on/off” to trigger a recording.
URL Record: The phone sends an HTTP URL request to trigger a recording. Contact

your system administrator for the predefined URL.
Note
Call record is not available on all servers. Contact your system administrator for more
information.
To configure a record key via phone user interface:
1.
Press Menu->Features->DSS Keys.
2.
Select the desired DSS key.
3.
Press
or
, or the Switch soft key to select Key Event from the Type field.
4.
Press
or
, or the Switch soft key to select Record from the Key Type field.
5.
(Optional.) Enter a string that will appear on the LCD screen in the Label field.
6.
Press the Save soft key to accept the change or the Back soft key to cancel.
To configure a URL record key via phone user interface:
1.
Press Menu->Features->DSS Keys.
2.
Select the desired DSS key.
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User Guide for the SIP-T42G IP Phone
3.
Press
or
, or the Switch soft key to select URL Record from the Type field.
4.
(Optional.) Enter a string that will appear on the LCD screen in the Label field.
5.
Enter the URL (e.g., http://10.1.2.224/phonerecording.cgi) in the Value field.
6.
Press the Save soft key to accept the change or the Back soft key to cancel.
Record and URL Record keys are configurable via web user interface at the path
DSSKey->Line Key.
The Record and URL Record keys control the recording function, and are available:

During an active call

When calls are on hold or muted

During a blind or attended transfer

During a conference call

When the phone prompts you to answer an incoming call
The Record and URL Record keys are not available when:

There are no connected calls on your phone

You place a new call
To record a call:
1.
Press the Record or URL Record key during a call.
If the recording starts successfully, the recording icon will appear on the LCD screen
and the Record or URL Record key LED will flash green.
2.
Press the Record or URL Record key again to stop recording.
The recording icon disappears from the LCD screen and the Record or URL Record
key LED goes out.
Recording status indications you need to know:
Circumstance
Icons on the LCD screen
A recording is started
appears on the LCD screen
A recording cannot be started
appears for 1 second
A recording cannot be stopped
appears for 1 second, then goes back
The recording box is full
appears for 1 second
The call cannot be recorded
appears for 1 second
You can listen to the recordings stored on your server system. For example, you can dial
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Advanced Phone Features
an access code to listen to the recordings.
Note
The way in which you listen to the recordings may be different depending on the server.
Contact your system administrator for more information.
Hot desking originates from the definition of being the temporary physical occupant of
a work station or surface by a particular employee. A primary motivation for hot
desking is cost reduction. This feature is regularly used in places where not all the
employees are in the office at the same time, or not in the office for very long, which
means that actual personal offices would be often vacant, consuming valuable space
and resources.
You can use hot desking on the SIP-T42G IP phone to log out of existing accounts and
then log into a new account. As a result, many users can share the phone resource at
different times. To use this feature, you need to configure a hot desking key in advance.
Note
Hot desking is not available on all servers. Contact your system administrator for more
information.
To configure a hot desking key via phone user interface:
1.
Press Menu->Features->DSS Keys.
2.
Select the desired DSS key.
3.
Press
or
, or the Switch soft key to select Key Event from the Type field.
4.
Press
or
, or the Switch soft key to select Hot Desking from the Key Type
field.
5.
(Optional.) Enter a string that will appear on the LCD screen in the Label field.
6.
Press the Save soft key to accept the change or the Back soft key to cancel.
Hot desking key is configurable via web user interface at the path DSSKey->Line Key.
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User Guide for the SIP-T42G IP Phone
To use hot desking:
1.
Press the hot desking key when the phone is idle.
The LCD screen prompts the following warning:
2.
Press the OK soft key, registration configurations of all accounts on the phone will be
cleared immediately.
The login wizard will be displayed as below:
3.
Enter the login information in each field.
4.
Press the Save soft key to login or the Back soft key to cancel.
Intercom is a useful feature in an office environment to quickly connect with the operator
or the secretary. You can press the intercom key to automatically connect with a remote
extension for outgoing intercom calls, and the remote extension will automatically
answer incoming intercom calls.
Note
Intercom is not available on all servers. Contact your system administrator for more
information.
To configure an intercom key via phone user interface:
106
1.
Press Menu->Features->DSS Keys.
2.
Select the desired DSS key.
3.
Press
4.
Select the desired line from the Account ID field.
or
, or the Switch soft key to select Intercom from the Type field.
Advanced Phone Features
5.
(Optional.) Enter the string that will appear on the LCD screen in the Label field.
6.
Enter the remote extension number in the Value field.
7.
Press the Save soft key to accept the change or the Back soft key to cancel.
Intercom key is configurable via web user interface at the path DSSKey->Line Key.
To place an intercom call:
1.
Press the intercom key when the phone is idle.
The phone is automatically connected to the extension specified in the Value field.
2.
Press the intercom key again or the Cancel soft key to end the intercom call.
The SIP-T42G IP phone supports automatically to answer an incoming intercom call by
default. The phone automatically plays a warning tone when it receives an incoming
intercom call. In addition, you can enable the phone to mute the microphone when it
automatically answers an incoming intercom call. You can also enable the phone to
automatically answer an incoming intercom call while there is already an active call on
the phone. The active call is then placed on hold.
Intercom features you need to know:
Intercom Feature
Accept Intercom
Intercom Mute
Intercom Tone
Description
Enable or disable the IP phone to automatically
answer an incoming intercom call.
Enable or disable the IP phone’s microphone for
intercom calls.
Enable or disable the IP phone to play a warning
tone when it receives an incoming intercom call.
Enable or disable the IP phone to automatically
Intercom Barge
answer an incoming intercom call while there is
already an active call on the phone.
To configure intercom features via phone user interface:
1.
Press Menu->Features->Intercom.
2.
Make the desired changes.
3.
Press the Save soft key to accept the change or the Back soft key to cancel.
These specific parameters are configurable via web user interface at the path
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User Guide for the SIP-T42G IP Phone
Features->Intercom.
Accept Intercom
You can enable or disable the phone to automatically answer an incoming intercom call.
If Accept Intercom is enabled, the phone will automatically answer an incoming
intercom call. If Accept Intercom is disabled, the phone will reject incoming intercom
calls and send a busy message to the caller. Accept Intercom is enabled by default.
Note
Your administrator can set a period of delay time before the phone automatically
answers intercom calls. Contact your system administrator for more information.
Intercom Mute
You can mute or un-mute the phone’s microphone for intercom calls automatically. If
Intercom Mute is enabled, the microphone will be muted for intercom calls. If Intercom
Mute is disabled, the microphone will work for intercom calls. Intercom Mute is disabled
by default.
Intercom Tone
You can enable or disable the phone to play a warning tone when receiving an
intercom call. If Intercom Tone is enabled, the phone will play a warning tone before
answering the intercom call. If Intercom Tone is disabled, the phone will automatically
answer the intercom call without warning. Intercom Tone is enabled by default.
Intercom Barge
You can enable or disable the phone to automatically answer an incoming intercom call
while there is already an active call on the phone. If Intercom Barge is enabled, the
phone will automatically answer the intercom call and place the active call on hold. If
Intercom Barge is disabled, the phone will handle an incoming intercom call like a
waiting call. Intercom Barge is disabled by default.
You can use multicast paging to quickly and easily forward out time sensitive
announcements to people within the multicast group. You can configure a multicast
paging key or a paging list key on the phone, which allows you to send a Real Time
Transport Protocol (RTP) stream to the pre-configured multicast address(es) without
involving SIP signaling. You can configure the phone to receive an RTP stream from
pre-configured multicast listening address(es) without involving SIP signaling. You can
specify up to 10 multicast listening addresses.
108
Advanced Phone Features
To configure a multicast paging key via phone user interface:
1.
Press Menu->Features->DSS Keys.
2.
Select the desired DSS key.
3.
Press
or
, or the Switch soft key to select Key Event from the Type field.
4.
Press
or
, or the Switch soft key to select Paging from the Key Type field.
5.
(Optional.) Enter the string that will appear on the LCD screen in the Label field.
6.
Enter the multicast IP address and port number (e.g., 224.5.6.20:10008) in the Value
field.
The valid multicast IP addresses range from 224.0.0.0 to 239.255.255.255.
7.
Press the Save soft key to accept the change or the Back soft key to cancel.
Multicast paging key is configurable via web user interface at the path DSSKey->Line
Key.
To configure a paging list key via phone user interface:
1.
Press Menu->Features->DSS Keys.
2.
Select the desired DSS key.
3.
Press
or
, or the Switch soft key to select Key Event from the Type field.
4.
Press
or
, or the Switch soft key to select Paging List from the Key Type
field.
5.
(Optional.) Enter the string that will appear on the LCD screen in the Label field.
6.
Press the Save soft key to accept the change or the Back soft key to cancel.
Paging list key is configurable via web user interface at the path DSSKey->Line Key.
To configure paging list via phone user interface:
1.
Press the paging list key when the phone is idle.
2.
Press
or
to select a desired paging group.
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User Guide for the SIP-T42G IP Phone
The default tag is Empty if it is not configured before.
3.
Press the Option soft key.
4.
Press the Edit soft key.
5.
Enter the multicast IP address and port number (e.g., 224.5.6.20:10008) in the
Address field.
The valid multicast IP addresses range from 224.0.0.0 to 239.255.255.255.
6.
Enter the group name in the Label field.
7.
Press the Save soft key to accept the change.
8.
Repeat the step 2-7, you can add more paging groups.
Paging list is configurable via web user interface at the path Directory->Multicast IP.
To delete paging group via phone user interface:
1.
Press the Paging list key when the phone is idle.
2.
Press
3.
Press the Option soft key.
4.
Press the Delete soft key.
5.
Press the OK soft key to accept the change or the Cancel soft key to cancel.
or
to select a desired group.
If you want to delete all paging group, you can press the Delete All soft key.
You can also configure the phone to use a default codec for sending multicast RTP
stream via web user interface.
To configure a default codec for multicast paging via web user interface:
110
1.
Click on Features->General Information.
2.
Select the desired codec from the pull-down list of Multicast Codec.
Advanced Phone Features
The default codec is G722.
3.
Note
Click Confirm to accept the change.
If G722 codec is used for multicast paging, the LCD screen will display the
indicate that it is providing high definition voice.
icon to
Default codec for multicast paging is configurable via web user interface only.
To send RTP stream via a multicast paging key:
1.
Press the multicast paging key when the phone is idle.
The phone sends RTP to a preconfigured multicast address (IP: Port). Any phone in
the local network then listens to the RTP on the preconfigured multicast address (IP:
Port). For both sending and receiving of the multicast RTP, there is no SIP signaling
involved.
The multicast paging key LED illuminates solid green.
The following figure shows a multicast RTP session on the phone:
Note
2.
Press the Hold soft key to place the current multicast RTP session on hold.
3.
Press the End Call soft key to cancel the multicast RTP session.
Multicast RTP is one way only from the sender to the multicast address(es) (receiver). For
outgoing RTP multicasts, all other existing calls on the phone will be placed on hold.
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User Guide for the SIP-T42G IP Phone
To send RTP stream via a paging key list:
1.
Press the paging list key when the phone is idle.
2.
Press
or
3.
Press
or the Paging soft key to send RTP.
4.
Press the Hold soft key to place the current multicast RTP session on hold.
5.
Press the End Call soft key to cancel the multicast RTP session.
to select the desired group.
You can configure the phone to receive a Real Time Transport Protocol (RTP) stream from
the pre-configured multicast address(es) without involving SIP signaling. You can
specify up to 10 multicast addresses that the phone listens to on the network.
How the phone handles incoming multicast paging calls depends on Paging Barge and
Paging Priority Active parameters configured via web user interface.
Paging Barge
The paging barge parameter defines the priority of the voice call in progress. If the
priority of an incoming multicast paging call is lower than that of the active call, it will be
ignored automatically. If Disabled is selected from the pull-down list of Paging Barge,
the voice call in progress will take precedence over all incoming multicast paging calls.
Valid values in the Paging Barge field:

1 to 10: Define the priority of the active call, 1 with the highest priority, 10 with the
lowest.

Disabled: The voice call in progress will take precedence over all incoming paging
calls.
Paging Priority Active
The paging priority active parameter decides how the phone handles incoming
multicast paging calls when there is already a multicast paging call on the phone. If
enabled, the phone will ignore incoming multicast paging calls with lower priorities,
otherwise, the phone will answer incoming multicast paging calls automatically and
place the previous multicast paging call on hold. If disabled, the phone will
automatically ignore all incoming multicast paging calls.
To configure multicast listening addresses via web user interface:
112
1.
Click on Directory->Multicast IP.
2.
Select the desired value from the pull-down list of Paging Barge.
Advanced Phone Features
3.
Select the desired value from the pull-down list of Paging Priority Active.
4.
Enter the multicast IP address(es) and port number (e.g., 224.5.6.20:10008) which
the phone listens to for incoming RTP multicast in the Listening Address field.
5.
Enter the label in the Label field.
Label will appear on the LCD screen when receiving the multicast RTP stream.
6.
Note
Click Confirm to accept the change.
The priorities of listening addresses are predefined: 1 with the highest priority, 10 with the
lowest.
Both the multicast paging sender and receiver’s phones play a warning tone when
establishing a multicast paging call.
Multicast listening addresses are configurable via web user interface only.
Music on hold (MoH) is the business practice of playing recorded music to fill the silence
that would be heard by the party placed on hold. To use this feature, you should specify
a SIP URI pointing to a Music on Hold Server account. When a call is placed on hold, the
phone will send a SIP INVITE message to the Music on Hold Server account. The Music
on Hold Server account automatically answers the SIP INVITE messages and
immediately plays audio from some source located anywhere (LAN, Internet) to the
held party. Contact your system administrator for the SIP URI.
To configure music on hold server via web user interface:
1.
Click on Account.
2.
Select the desired account from the pull-down list of Account.
3.
Click on Advanced.
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User Guide for the SIP-T42G IP Phone
4.
Enter the SIP URI (e.g., sip:moh@sip.com) in the Music Server URI field.
5.
Click Confirm to accept the change.
When you have placed a call on hold, the held party can hear the music.
Note
For this feature to function, all involved parties cannot use encrypted RTP (SRTP).
Music on hold server is configurable via web user interface only.
ACD is often used in offices for customer service, such as call center. The ACD system
handles large volumes of incoming calls from callers who have no need to talk to a
specific person but who require assistance from any of multiple persons at the earliest
opportunity. ACD on the SIP-T42G IP phone allows the ACD system to distribute large
volumes of incoming calls to the registered ACD users. To use this feature, you should
configure an ACD key in advance.
Note
Make sure ACD is enabled on your IP phone. For more information on enabling ACD,
contact your system administrator.
To configure an ACD key via phone user interface:
114
1.
Press Menu->Features->DSS Keys.
2.
Select the desired DSS key.
Advanced Phone Features
3.
Press
or
, or the Switch soft key to select ACD from the Type field.
4.
(Optional.) Enter the string that will appear on the LCD screen in the Label field.
5.
Press the Save soft key to accept the change or the Back soft key to cancel.
ACD key is configurable via web user interface at the path DSSKey->Line Key.
To log into the ACD system:
1.
Press the ACD key when the phone is idle.
The LCD screen prompts you the following information:
User ID: the identity used to log into the queue.
Password: the password used to log into the queue.
2.
Note
Press the Login soft key to log in.
Contact your system administrator for the User ID and Password to access the ACD
system.
After configuring an ACD key, you can press the ACD key to log into the ACD system.
After logging in, you are ready to receive calls from the ACD system. You can press the
ACD key to show your current ACD user status. You can press the Available/Unavailable
soft key to change your ACD user status. ACD user status synchronizes on both IP phone
and ACD system. When you set the ACD user status to be available, the ACD key LED
illuminates solid green, and then the server begins distributing calls to your phone.
When you set the ACD user status to be unavailable, the ACD key LED flashes green,
and then the server temporarily stops distributing calls to your phone. To log out of the
ACD system, press the Logout soft key.
Note
It is recommended you configure no more than one ACD key per phone. At any time, at
most one ACD key can be in progress on your phone.
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User Guide for the SIP-T42G IP Phone
You can leave voice mails for someone else using the SIP-T42G IP phone. You can also
listen to voice mails that are stored in a voice mailbox. When receiving a new voice mail,
the phone will play a warning tone, and the power indicator LED will slow flash red. The
LCD screen will prompt that the phone receives a new message and display an icon.
If the voice mail pop-up message box disappears, it won't pop up again unless the user
receives a new voice mail or the user re-registers the account that has unread voice
mail(s).
Note
Voice mail is not available on all servers.
You can configure the phone not to display the pop-up prompt, contact your system
administrator for more information.
To leave a voice mail:
You can leave a voice mail for someone else when he/she is busy or inconvenient to
answer the call. Follow the voice prompt from the system server to leave a voice mail,
and then hang up.
To configure voice mail access codes via phone user interface:
1.
Press Menu->Message->Voice Mail->Set Voice Mail.
2.
Press the navigation keys to highlight the account which you want to set.
3.
Press the 123 soft key to select the proper input mode and then enter the voice mail
access code (e.g., *88).
4.
Note
116
Press the Save soft key to accept the change or the Back soft key to cancel.
Voice mail access codes must be predefined on the system server. Contact your system
administrator for the more information.
Advanced Phone Features
To listen to voice mails:
1.
When the LCD screen prompts that the phone receives a new voice mail and the
power indicator LED slow flashes red, press
or the Connect soft key to dial out
the voice mail access code.
2.
Note
Follow the voice prompt to listen to your voice mails.
Before listening to voice mails, make sure the voice mail access code has been
configured.
When all new voice mails are retrieved, the power indicator LED will go out.
To view the voice mail via phone user interface:
1.
Press Menu->Message->Voice Mail->View Voice Mail.
The LCD screen displays the amount of new and old voice mails.
2.
Select an account and then press the Connect soft key to listen to voice mails.
The SIP-T42G IP phone supports MWI feature when receiving a new voice message. If
someone leaves you a voice mail, you will receive a message waiting indicator. MWI
will be indicated in three ways: a warning tone, an indicator message (including a
voice mail icon) on the LCD screen, and the power indicator LED slow flashes red. This
will be cleared when you retrieve all voice mails or delete them.
The MWI service is unsolicited for some servers, so the SIP-T42G IP phone only handles
the MWI messages sent from the server. But for other servers, the MWI service is
solicited, so the SIP-T42G IP phone must enable subscription for MWI.
Note
MWI service is not available on all servers. Contact your system administrator for more
information.
The MWI subscription parameters you need to know:
Options
Subscribe for MWI
Description
Enable or disable a subscription for MWI service.
Period of MWI subscription. The IP phone sends a
MWI Subscription Period
refresh SUBSCRIBE request before initial SUBSCRIBE
expiration.
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User Guide for the SIP-T42G IP Phone
Options
Description
Enable or disable a subscription to the voice mail
Subscribe MWI To Voice
number for MWI service.
Mail
To use this feature, you should also configure the
voice mail number.
Note
The phone will send SUBSCRIBE messages for the MWI service to the account or the
voice number MWI service depending on the server. Contact your system administrator
for more information.
To configure subscribe for MWI via web user interface:
1.
Click on Account.
2.
Select the desired account from the pull-down list of Account.
3.
Click on Advanced.
4.
Select Enabled from the pull-down list of Subscribe for MWI.
5.
Enter the period time in the MWI Subscription Period (Seconds) field.
6.
Click Confirm to accept the change.
The IP phone will subscribe to the account number for MWI service by default.
118
Advanced Phone Features
To enable Subscribe MWI to Voice Mail via web user interface:
1.
Click on Account.
2.
Select the desired account from the pull-down list of Account.
3.
Click on Advanced.
4.
Select Enabled from the pull-down list of Subscribe MWI To Voice Mail.
5.
Enter the desired voice mail number in the Voice Mail field.
6.
Click Confirm to accept the change.
The IP phone will subscribe to the voice mail number for MWI service using
Subscribe MWI to Voice Mail.
Note
MWI subscription is configurable via web user interface only.
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User Guide for the SIP-T42G IP Phone
120
Troubleshooting
This chapter provides general troubleshooting information to help you solve the
problems you might encounter when using your SIP-T42G IP phone.
If you require additional information or assistance with your new phone, contact your
system administrator.
Why is the LCD screen blank?

Ensure that the phone is properly plugged into a functional AC outlet.

Ensure that the phone is plugged into a socket controlled by a switch that is on.

If the phone is plugged into a power strip, try to plug it directly into a wall outlet
instead.

If the phone is powered from PoE, ensure that you use a PoE-compliant switch or
hub.
Why does the phone display "Network Unavailable"?

Ensure that the Ethernet cable is plugged into the Internet port on the phone and
the Ethernet cable is not loose.

Ensure that the switch or hub in your network is operational.
Why doesn’t the phone display time and date correctly?
Check if you have configured the phone to obtain the time and date from the SNTP
server automatically. If the phone fails to connect to the SNTP server, you need to
configure the time and date manually.
How do I find the basic information of the IP phone?
Press the OK key when the IP phone is idle to check the basic information of the IP phone,
such as IP address and firmware version. For more basic information, refer to Phone
Status on page 14.
How to obtain the MAC address of a phone when the phone is not powered on?
Three ways to obtain the MAC address of a phone:

You can ask your supplier for the shipping information sheet which includes MAC
addresses according to the corresponding PO (Purchase Order).

You can find the MAC address on the label of the carton box.

You can also find the MAC address from the phone’s bar code on the back of the
phone.
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User Guide for the SIP-T42G IP Phone
Why can’t I get a dial tone?

Check for any loose connections and that the phone has been installed properly.
For the installation instructions, refer to Phone Installation on page 11.

Switch between the Handset, Headset (if present) or Hands-Free Speakerphone to
check whether the dial tone is present for one of the audio modes.
If the dial tone exists on another audio mode, connect a different handset or
headset to isolate the problem.
Why doesn’t the phone ring?
Check the ringer volume on the phone. To adjust the ringer volume setting, press the
Volume key when the phone is on-hook and idle. For more information, refer to Volume
on page 26.
Why can’t I receive calls?

Check the SIP registration with your system administrator.

Check that DND (Do Not Disturb) mode is deactivated on your phone. Refer to Do
Not Disturb (DND) on page 77.

Check that call forward is disabled on the phone. Refer to Call Forward on page81.

Check whether the caller number is stored in the blacklist directory. Refer to Blacklist
on page 38.
Why is my handset not working?
Check that the handset cord is fully connected to both the handset jack on the phone
and handset. Refer to Phone Installation on page 11.
Why is my headset not working?

Check that the headset cord is fully connected to the headset jack on the phone.
Refer to Phone Installation on page 11.

Check that the headset mode is activated. Refer to Headset Use on page 43.

Check that the headset volume is adjusted to an appropriate level. Refer to Volume
on page 26.
What is the difference between user name, register name and display name?
Both user name and register name are defined by the server. A user name is used to
identify the account, while a register name matched with a password is used for
authentication if the server requires. Display name is the caller ID that will be displayed
on the callee’s LCD screen. Server configuration may override the local configuration.
122
Troubleshooting
Why does the phone play a tone when hold? How to disable it?
When there is a call is on hold, the phone will play a hold tone every 30 seconds. Play
hold tone is enabled by default. Play hold tone and the interval of playing a hold tone
are configurable via web user interface only.
To configure the call hold tone and call hold tone delay via web user interface:
1.
Click on Features->General Information.
2.
Select Enabled or Disabled from the pull-down list of Play Hold Tone.
3.
Enter the desired time in the Play Hold Tone Delay field.
4.
Click Confirm to accept the change.
How to change the user password?
To change the user password via web user interface:
1.
Click on Security->Password.
2.
Select User from the pull-down list of User Type.
3.
Enter the new user password in the New Password field and Confirm Password field.
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User Guide for the SIP-T42G IP Phone
4.
Click Confirm to accept the change.
You can also contact your system administrator for help.
Note
If logging into the web user interface of the phone with user credentials, you need to
enter the current user password in the Old Password field.
User password is configurable via web user interface only.
How to make a call using SRTP?
You can enable SRTP to encrypt the audio stream(s) of phone calls. The parties
participating in the call should enable SRTP. You can configure SRTP on per-line basis.
To enable SRTP via web user interface:
Note
1.
Click on Account.
2.
Select the desired account from the pull-down list of Account.
3.
Click on Advanced.
4.
Select the desired type from the pull-down list of RTP Encryption (SRTP).
5.
Click Confirm to accept the change.
SRTP is not available on all servers. Contact your system administrator for more
information.
SRTP is configurable via web user interface only.
124
Troubleshooting
How to reboot the phone?
To reboot the phone via web user interface:
1.
Click on Settings->Upgrade.
2.
Click Reboot to reboot the IP phone.
Note
Any reboot of the phone may take a few minutes.
How to export PCAP trace?
We may need you to provide a PCAP trace to help analyze your problem.
To export a PCAP trace via web user interface:
1.
Click on Settings->Configuration.
2.
Click Start to begin capturing signal traffic.
3.
Recreate the error to be documented in the trace.
4.
Click Stop to stop the capture.
5.
Click Export to open file download window, and then save the file to your local
system.
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User Guide for the SIP-T42G IP Phone
How to export system log?
We may need you to provide the phone configurations to help analyze your problem. In
some instance, you may need to import configurations to your phone.
To export the system log to a local PC via web user interface:
1.
Click on Settings->Configuration.
2.
Select 6 from the pull-down list of System Log Level.
The default system log level is 3.
3.
Click Confirm to accept the change.
4.
Mark the Local radio box in the Export System Log field.
5.
Click Export to open file download window, and then save the system log file to
your local system.
You can also export the system log to a syslog server. Contact your system administrator
for more information.
Note
It is recommended to reset the syslog level to 3 after exporting the system syslog.
How to export/import the phone configurations?
We may need you to provide the phone configurations to help analyze your problem. In
some instance, you may need to import configurations to your phone.
To export the phone configurations via web user interface:
1.
126
Click on Settings->Configuration.
Troubleshooting
2.
In the Export or Import Configuration block, click Export to open file download
window, and then save the file to your local system.
To import the phone configurations via web user interface:
1.
Click on Settings->Configuration.
2.
In the Export or Import Configuration block, click Browse to locate a configuration
file from your local system.
3.
Note
Click Import to import the configuration file.
The file format of configuration file must be *.bin.
How to upgrade firmware?
To upgrade firmware via web user interface:
1.
Click on Settings->Upgrade.
2.
Click Browse to locate the required firmware from your local system.
3.
Click Upgrade to upgrade the firmware.
The web user interface prompts “Firmware of the SIP Phone will be updated. It will
take 5 minutes to complete. Please don't power off!”.
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User Guide for the SIP-T42G IP Phone
4.
Click OK to confirm upgrading.
How to reset the phone?
Reset the phone to factory configurations after you have tried all troubleshooting
suggestions but do not solve the problem. You need to note that all customized settings
will be overwritten after reset.
To reset the phone via phone user interface:
1.
Press Menu->Settings->Advanced Settings (default password: admin) ->Reset to
Factory.
2.
Press the Enter soft key.
The LCD screen prompts the following warning:
3.
Press the OK soft key.
The LCD screen prompts the following:
Resetting…
4.
Please Wait!
The LCD screen prompts the following:
Welcome
5.
Initializing...please wait
The phone will be reset to factory settings sucessfully after startup.
Note
128
Reset of your phone may take a few minutes. Do not power off until the phone has started
up successfully.
Regulatory Notices
Contact your Yealink Authorized Reseller for information on service agreements
applicable to your product.
TO THE FULL EXTENT ALLOWED BY LAW, YEALINK EXCLUDES FOR ITSELF AND ITS
SUPPLIERS ANY LIABILITY, WHETHER BASED IN CONTRACT OR TORT (INCLUDING
NEGLIGENCE), FOR INCIDENTAL, CONSEQUENTIAL, INDIRECT, SPECIAL, OR PUNITIVE
DAMAGES OF ANY KIND, OR FOR LOSS OF REVENUE OR PROFITS, LOSS OF BUSINESS,
LOSS OF INFORMATION OR DATA, OR OTHER FINANCIAL LOSS ARISING OUT OF OR IN
CONNECTION WITH THE SALE, INSTALLATION, MAINTENANCE, USE, PERFORMANCE,
FAILURE, OR INTERRUPTION OF ITS PRODUCTS, EVEN IF YEALINK OR ITS AUTHORIZED
RESELLER HAS BEEN ADVISED OF THE POSSIBILITY OF SUCH DAMAGES, AND LIMITS ITS
LIABILITY TO REPAIR, REPLACEMENT, OR REFUND OF THE PURCHASE RICE PAID, AT
YEALINK’S OPTION. THIS DISCLAIMER OF LIABILITY FOR DAMAGES WILL NOT BE
AFFECTED IF ANY REMEDY PROVIDED HEREIN SHALL FAIL OF ITS ESSENTIAL PURPOSE.
Save these instructions. Read these safety instructions before use!
The following basic safety precautions should always be followed to reduce the risk of
fire, electrical shock, and other personal injury.
General Requirements

Before you install and use the device, read the safety instructions carefully
and observe the situation during operation.

During the process of storage, transportation, and operation, please always
keep the device dry and clean.

During the process of storage, transportation, and operation, please avoid
collision and crash of the device.

Please do not attempt to dismantle the device by yourself. In case of any
discrepancy, please contact the appointed maintenance center for repair.

Without prior written consent, no organization or individual is permitted to
make any change to the structure or the safety design of the device. Yealink
is under no circumstances liable to consequences or legal issues caused by
such changes.

Please refer to the relevant laws and statutes while using the device. Legal
rights of others should also be respected as well.
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User Guide for the SIP-T42G IP Phone
Environmental Requirements

Place the device at a well-ventilated place. Do not expose the device under
direct sunlight.

Keep the device dry and free of dust.

Place the device on a stable and level platform.

Please do not place heavy objects on the device in case of damageand
deformation caused by the heavy load.

Keep at least 10 cm between the device and the closest object for heat
dissipation.

Do not place the device on or near any inflammable or fire-vulnerable object,
such as rubber-made materials.

Keep the device away from any heat source or bare fire, such as a candle or
an electric heater.

Keep the device away from any household appliance with a strong
magnetic field or electromagnetic field, such as a microwave oven or a
refrigerator.
Operating Requirements

Do not let a child operate the device without guidance.

Do not let a child play with the device or any accessory in case of accidental
swallowing.

Please only use the accessories provided or authorized by the manufacturer.

The power supply of the device must meet the requirements of the input
voltage of the device. Please use the provided surge protection power
socket only.

Before plugging or unplugging any cable, make sure that your hands are
completely dry.

Do not spill liquid of any kind on the product or use the equipment near
water, for example, near a bathtub, washbowl, kitchen sink, wet basement
or near a swimming pool.

Do not tread on, pull, or over-bend any cable in case of malfunction of the
device.

During a thunderstorm, stop using the device and disconnect it from the
power supply. Unplug the power plug and the Asymmetric Digital Subscriber
Line (ADSL) twisted pair (the radio frequency cable) to avoid lightning strike.

If the device is left unused for a rather long time, disconnect it from the
power supply and unplug the power plug.

When there is smoke emitted from the device, or some abnormal noise or
smell, disconnect the device from the power supply, and unplug the power
plug immediately. Contact the specified maintenance center for repair.

Do not insert any object into equipment slots that is not part of the product or
auxiliary product.

Before connecting a cable, connect the grounding cable of the device first.
Do not disconnect the grounding cable until you have disconnected all other
cables.
Cleaning Requirements
130
Regulatory Notices

Before cleaning the device, stop using it and disconnect it from the power
supply.

Use a piece of soft, dry and anti-static cloth to clean the device.

Keep the power plug clean and dry. Using a dirty or wet power plug may
lead to electric shock or other perils.
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User Guide for the SIP-T42G IP Phone
132
Appendix A-Time Zones
Time Zone
−11:00
−10:00
−09:30
−09:00
−08:00
−08:00
−08:00
−07:00
−07:00
−07:00
−07:00
−06:00
−06:00
−06:00
−06:00
−05:00
−05:00
−05:00
−05:00
−04:30
−04:00
−04:00
−04:00
−04:00
−04:00
−04:00
−03:30
−03:00
−03:00
−03:00
−03:00
−02:30
−02:00
−01:00
0
0
0
0
0
0
0
0
+01:00
+01:00
+01:00
+01:00
+01:00
+01:00
+01:00
+01:00
+01:00
Time Zone Name
Samoa
United States-Hawaii-Aleutian
French Polynesia
United States-Alaska Time
Canada(Vancouver, Whitehorse)
Mexico(Tijuana, Mexicali)
United States-Pacific Time
Canada(Edmonton, Calgary)
Mexico(Mazatlan, Chihuahua)
United States-Mountain Time
United States-MST no DST
Canada-Manitoba(Winnipeg)
Chile(Easter Islands)
Mexico(Mexico City, Acapulco)
United States-Central Time
Bahamas(Nassau)
Canada(Montreal, Ottawa, Quebec)
Cuba(Havana)
United States-Eastern Time
Venezuela(Caracas)
Canada(Halifax, Saint John)
Chile(Santiago)
Paraguay(Asuncion)
United Kingdom-Bermuda(Bermuda)
United Kingdom(Falkland Islands)
Trinidad&Tobago
Canada-New Foundland(St.Johns)
Denmark-Greenland(Nuuk)
Argentina(Buenos Aires)
Brazil(no DST)
Brazil(DST)
Newfoundland and Labrador
Brazil(no DST)
Portugal(Azores)
GMT
Greenland
Denmark-Faroe Islands(Torshavn)
Ireland(Dublin)
Portugal(Lisboa, Porto, Funchal)
Spain-Canary Islands(Las Palmas)
United Kingdom(London)
Morocco
Albania(Tirane)
Austria(Vienna)
Belgium(Brussels)
Caicos
Chad
Spain(Madrid)
Croatia(Zagreb)
Czech Republic(Prague)
Denmark(Kopenhagen)
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User Guide for the SIP-T42G IP Phone
Time Zone
+01:00
+01:00
+01:00
+01:00
+01:00
+01:00
+01:00
+01:00
+02:00
+02:00
+02:00
+02:00
+02:00
+02:00
+02:00
+02:00
+02:00
+02:00
+02:00
+02:00
+02:00
+02:00
+03:00
+03:00
+03:00
+03:30
+04:00
+04:00
+04:00
+04:00
+04:00
+04:30
+05:00
+05:00
+05:00
+05:00
+05:30
+05:45
+06:00
+06:00
+06:30
+07:00
+07:00
+08:00
+08:00
+08:00
+08:00
+08:45
+09:00
+09:00
+09:00
+09:30
+09:30
+10:00
+10:00
+10:00
134
Time Zone Name
France(Paris)
Germany(Berlin)
Hungary(Budapest)
Italy(Rome)
Luxembourg(Luxembourg)
Macedonia(Skopje)
Netherlands(Amsterdam)
Namibia(Windhoek)
Estonia(Tallinn)
Finland(Helsinki)
Gaza Strip(Gaza)
Greece(Athens)
Israel(Tel Aviv)
Jordan(Amman)
Latvia(Riga)
Lebanon(Beirut)
Moldova(Kishinev)
Russia(Kaliningrad)
Romania(Bucharest)
Syria(Damascus)
Turkey(Ankara)
Ukraine(Kyiv, Odessa)
East Africa Time
Iraq(Baghdad)
Russia(Moscow)
Iran(Teheran)
Armenia(Yerevan)
Azerbaijan(Baku)
Georgia(Tbilisi)
Kazakhstan(Aktau)
Russia(Samara)
Afghanistan(Kabul)
Kazakhstan(Aqtobe)
Kyrgyzstan(Bishkek)
Pakistan(Islamabad)
Russia(Chelyabinsk)
India(Calcutta)
Nepal(Katmandu)
Kazakhstan(Astana, Almaty)
Russia(Novosibirsk, Omsk)
Myanmar(Naypyitaw)
Russia(Krasnoyarsk)
Thailand(Bangkok)
China(Beijing)
Singapore(Singapore)
Australia(Perth)
Russia(Irkutsk, Ulan-Ude)
Eucla
Korea(Seoul)
Japan(Tokyo)
Russia(Yakutsk,Chita)
Australia(Adelaide)
Australia(Darwin)
Australia(Sydney, Melbourne, Canberra)
Australia(Brisbane)
Australia(Hobart)
Appendix A-Time Zones
Time Zone
+10:00
+10:30
+11:00
+11:00
+11:30
+12:00
+12:00
+12:45
+13:00
+13:30
+14:00
Time Zone Name
Russia(Vladivostok)
Australia(Lord Howe Islands)
New Caledonia(Noumea)
Russia(Srednekolymsk Time)
Norfolk Island
New Zealand(Wellington, Auckland)
Russia(Kamchatka Time)
New Zealand(Chatham Islands)
Tonga(Nukualofa)
Chatham Islands
Kiribati
135
User Guide for the SIP-T42G IP Phone
136
Index
E
A
About This Guide
v
Account Management 56
Account Registration 56
Adding Contacts 32
Administrator password 22
Anonymous Call 96
Anonymous Call Rejection 97
Answering Calls 70
Area Code 61
Attaching Stand 11
Audio Settings 26
Auto Answer 73
Auto Redial 73
Automatic Call Distribution 114
B
Backlight 19
Basic Network Settings
Blacklist 38
Block Out 62
Busy Lamp Field (BLF)
BLF List 101
16
99
C
Call completion 74
Call Forward
80
Call Mute 76
Call History Management 42
Call Hold/Resume 76
Call Park 91
Call Pickup 92
Call Recording 101
Call Transfer 86
Call Waiting 87
CE Mark Warning vi
Conference 88
Contact Management 28
Copyright v
Customizing Your Phone 19
D
Declaration of Conformity vi
Deleting Contacts 34
Dial-now 60
Dial Plan 58
Do Not Disturb (DND) 77
Documentation 8
DSS Keys 46
Directory
28
Editing Contacts 33
Emergency Number
63
Ending Calls 71
G
General Settings 19
Getting Started
9
H
Hardware Component Instructions 1
Headset Use 43
Headset Mode Activation/Deactivation
44
Hotline
65
Hot Desking 105
I
Icon Instructions 3
Idle screen 17
Importing/Exporting Contact lists
In This Guide v
Incoming Intercom Calls 107
Intercom 106
36
K
Key as Send
23
L
Language 20
LED Instructions 4
Limitation of Liability 129
Line Keys 46
Live Dialpad 64
Local Conference 88
Local Directory 29
M
Message Waiting Indicator
117
Messages 116
Multicast Paging 108
Multiple Line Keys per Account 57
Music on Hold 113
N
Network Conference
Network Connection
89
11
P
Packaging Contents 9
Phone Initialization 14
137
User Guide for the SIP-T42G IP Phone
Phone Installation 11
Phone Status 14
Phone Lock 23
Phone User Interface 6
Placing Calls 68
Placing Calls to Contacts 34
Programmable Keys 55
O
Outgoing Intercom Calls 106
Overview 1
Optional Accessories
10
R
Recall 74
Recent Call In Dialing 72
Redialing Numbers 71
Registration 17
Regulatory Notices
129
Remote Phone Book 39
Replace Rule 58
Ring Tones 26
S
Safety Instructions 129
Search Source List in Dialing
Searching for Contacts 35
Service Agreements 129
Summary of Changes v
System Customizations 43
T
Time & Date 20
Time Zones 133
Troubleshooting 121
V
Voice Mail 116
Volume 26
W
Warranty vi
Web User Interface
WEEE Warning vii
138
6
35
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