Shared Line, Busy Lamp Field List, Network Directory and Network Call Log features are
hidden for IP phones in the neutral firmware, which are designed for BroadWorks
environment. Please contact your system administrator for more information.
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.
If this is your first time using the SIP-T42G IP phone, we recommend that you first refer to
Appendix A - Time Zones .................................................. 129
Index ................................................................................. 131
ix
User Guide for the SIP-T42G IP Phone
x
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
Item
Description
①
LCD Screen
Shows information about calls, messages, soft keys,
time, date and other relevant data:
• Call information—caller ID, call duration
• 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
Indicates phone power and some feature statuses.
③
Line Keys
Use these keys to activate up to three accounts and
assign various features.
④
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
Adjusts the volume of the handset, headset, speaker,
and ringer.
⑦
Keypad
Provides the digits, letters, and special characters in
context-sensitive applications.
⑧
Scroll through the displayed information.
Confirms actions or answers incoming calls.
Cancels actions or rejects incoming calls.
⑨
Soft Keys
Label automatically to identity their context-sensitive
features.
Hardware component instructions of the SIP-T42G IP phone are:
2
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
Keypad 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
Overview
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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
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.
Off
The phone is powered off.
The phone is idle.
The phone is busy.
The call is placed on hold or is held.
The call is mute.
LED Status
Description
Solid green
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.
Power Indicator LED
Line key LED
4
Line key LED (configured as a BLF key)
LED Status
Description
Solid green
The monitored user is idle.
Fast flashing green (200ms)
The monitored user receives an incoming call.
Slow flashing green (500ms)
The monitored user is dialing.
The monitored user is talking.
The monitored user’s conversation is placed on
hold.
Slow flashing green (1s)
The call is parked against the monitored user ’s
phone number.
Off
The monitored user does not exist.
LED Status
Description
Solid red
There is a parked call on the line key of the
non-current page.
Solid green
There is a call on the line key of the non-current
Fast flashing green
The line receives an incoming call on the
non-current page.
Fast flashing red
The monitored user receives an incoming call on
the non-current page.
Off
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
2. Press to select IPv4 and press the Enter soft key.
3. Press to select Static IPv4 Client and press the Enter soft key.
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User Guide for the SIP-T42G IP Phone
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.
4. Enter the desired values in the IPv4 Address, Subnet Mask, Default Gateway,
Primary DNS and Secondary 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:
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
Not all of the supported languages are available for selection. The available languages
depend on the language packs currently loaded to the IP phone. Please contact your
system administrator for more information on the available languages of your new
phone.
Backlight is configurable via web user interface at the path Settings->Preference.
The default language of the phone user interface is English. The phone will detect and
use the same language as which of your web browser for the web user interface. 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.
Note
To change the language for the phone user interface:
2. Enter the desired values in the Current PIN, New PIN and Confirm PIN fields
respectively.
3. Press the Save soft key to accept the setting or the Back soft key to cancel.
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User Guide for the SIP-T42G IP Phone
You can also unlock the keypad by entering administrator password. When you enter the
administrator password to unlock the keypad, the phone will turn to the Reset Phone PIN
interface.
To unlock the keypad via phone user interface:
1. Press any locked key, the LCD screen prompts “Please Enter PIN”.
2. Enter the PIN in the PIN field.
3. Press the Confirm soft key to unlock the keypad.
The icon disappears from the LCD screen.
You can long press or wait for a period of time (if configured) to lock the
keypad again.
Note
To deactivate the keypad lock via phone user interface:
3. Press the Save soft key to accept the change or the Back soft key to cancel.
To select a ring tone for the account via web user interface:
1. Click on Account.
2. Select the desired account from the pull-down list of Account.
3. Click on Basic.
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User Guide for the SIP-T42G IP Phone
A ring tone for the account is configurable via web user interface only.
4. Select the desired ring tone from the pull-down list of Ring Type.
If Common is selected, this account will use the ring tone selected for the phone.
Refer to the above instruction.
Note
5. Click Confirm to accept the change.
To upload a custom ring tone for your phone via web user interface:
1. Click on Settings->Preference.
2. Click Browse to locate a ring tone file (the file format must be *.wav) from your local
system.
28
3. 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 and account, the phone
will play the ring tone assigned to the phone.
All custom ring tone files must be within 100KB. Uploading custom ring tones for your
phone is configurable via web user interface only.
Note
This section provides the operating instructions for managing contacts. Topics include:
Directory
Local Directory
Customizing Your Phone
Blacklist
Remote PhoneBook
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.
3. Repeat the 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 .
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User Guide for the SIP-T42G IP Phone
Directory is configurable via web user interface only.
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 PhoneBook on page 40. For more information on the LDAP,
refer to
2. Select the desired account and then press the Enter soft key.
3. Select Disable from the Account Status 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 resembles 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.
58
Customizing Your Phone
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", etc.
()
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: "9([5-7])(.)", Replace: "5$2". When
you enter "96123" to dial out on your phone, the number will be replaced
as "5123" and then dialed out. “$2” means the characters in the second
parenthesis, that is, “123”.
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:
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
(expression syntax refer to 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: 1234567", when you try
to dial out the number “1234567”, you just need to enter “1” on the phone and then
press the Send soft key.
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User Guide for the SIP-T42G IP Phone
Valid values of Account field can be one or more digits among 1, 2, and 3. 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 add a replace rule via web user interface:
1. Click on Settings->Dial Plan->Replace Rule.
2. Enter the string (e.g., 2xx) 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.
Note
5. Click Add to add the replace rule.
When you enter the number “234” using the keypad and then press the Send soft
key, the phone will dial out “1234” instead.
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).
60
Note
You can also edit or delete the dial-now rule, refer to Replace Rule on page 59 for more
information.
Dial-now rule is configurable via web user interface only.
Replace rule is configurable via web user interface only.
Customizing Your Phone
3. Click Del to delete the replace rule(s).
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", then any entered three-digit string
beginning with 2 will 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., 1001) in the Rule field.
3. Enter the desired line ID in the Account field or leave it blank.
For more information on the valid value of the Account field, refer to Replace Rule
on page 59.
4. Click Add to add the dial-now rule.
When you enter the number “1001” using the keypad, the phone will dial out “1001”
automatically without pressing any key.
Note
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User Guide for the SIP-T42G IP Phone
Delay time for dial-now rule is configurable via web user interface only.
Delay Time for Dial-Now Rule
You can configure the interval for dial-now rules. That is, you can configure your phone
to dial out the phone number automatically, which matches a dial-now rule, after the
specified 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 within 1-14 (in seconds) in the Time-Out for Dial-Now Rule field.
Note
3. Click Confirm to accept the change.
Area codes are also known as Numbering Plan Areas (NPAs). They usually indicate
geographical areas in a country. This feature is necessary only when dialing the number
outside the code area. For example, area code is configured as "Code: 011, Min Length:
4, Max Length: 11". When you dial out the number "56789" (the length of the number is
between 4 and 11), the phone will add the area code and dial out the number
"01156789".
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.
62
For more information on the valid value of the Account field, refer to Replace Rule
The default values of minimum and maximum length are 1 and 15 respectively.
Area code is configurable via web user interface only.
on page 59.
4. Click Confirm to accept the change.
Note
Customizing Your Phone
You can block specific numbers (up 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:
1. Click on Settings->Dial Plan->Block Out.
2. Enter the desired value in the BlockOut Number field.
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User Guide for the SIP-T42G IP Phone
Block out number is configurable via web user interface only.
Contact your local phone service provider for available emergency numbers in your
area.
3. Enter the desired line ID in the Account field or leave it blank.
For more information on the valid value of the Account field, refer to Replace Rule
on page 59.
Note
Note
4. Click Confirm to add the block out number.
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.
To specify emergency numbers via web user interface:
1. Click on Features->Phone Lock.
64
Customizing Your Phone
Emergency number is configurable via web user interface only.
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 110, 911 and 120.
3. Click Confirm to accept the change.
Note
You can enable live dialpad on the SIP-T42G IP phone, which enables the IP phone to
automatically dial out the phone number without pressing any other key. You can also
configure a delay, where the phone will dial out the phone number automatically after
the specified period of time.
To enable live dialpad via web user interface:
1. Click on Settings->Preference.
2. Select Enabled from the pull-down list of Live Dialpad.
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User Guide for the SIP-T42G IP Phone
Live dialpad is configurable via web user interface only.
3. Enter the desired delay time in the Inter Digit Time (1~14s) field.
The default delay time is 4s.
4. Click Confirm to accept the change.
Note
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 specified 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.
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
AnsweringCalls
Ending Calls
Redialing Numbers
Recent Call In Dialing
Auto Answer
Auto Redial
Call Completion
Call Return
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 also dial using the SIP URI or IP address. To obtain the IP address of a phone,
press the OK key. 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.
You can place a call in 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 29and Call History
Management on page 43.
During a call, you can alternate between Speakerphone, Headset, or Handset mode by
pressing the Speakerphone key, the Headset key, or picking up the handset.
The call duration of active calls is visible on the LCD screen. In the figure below, the call
to the number”1007” has lasted 15 seconds.
Note
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.
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 , or the Send soft key.
68
Note
To permanently use the headset mode, refer to Headset Prior on page 45.
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 , , or the Send soft key.
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 , , or the Send soft key.
You can press or to switch between calls, and then press the Resume soft key
to retrieve the desired call.
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User Guide for the SIP-T42G IP Phone
You can ignore incoming calls by pressing the X key, the Reject soft key or the Silence soft
key. You can also activate Do Not Disturb mode to ignore the incoming calls without ring
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 Fwd soft key. For more
information, refer to Call Forward on page 80.
When you are not in another call, you can answer a call in three ways:
Using the handset
Using the speakerphone
Using the headset
Note
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).
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.
70
Note
To end a call placed on hold, you can press the Cancel soft key to end the call directly, or
press the Resume soft key to resume the call before ending it.
Basic Call Features
- 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 Cancel soft key or hang up the handset.
- If you are using the headset, press the Cancel soft key.
- If you are using the speakerphone, press or the Cancel soft key.
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 when the phone is idle.
2. Press or to select the desired entry from the placed calls list, and then
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.
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User Guide for the SIP-T42G IP Phone
Recent call in dialing is configurable via web user interface only.
2. Select Enabled from the pull-down list of Recent Call In Dialing.
Note
3. Click Confirm to accept the change.
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.
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
field.
72
Note
Auto answer is only applicable when there is no other call in progress on the phone.
Basic Call Features
3. Press or , or the Switch soft key to select Enable from the Status field.
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.
You can enable auto redial to redial the phone number automatically when the called
party is busy. You can also configure the times settings for auto redial 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 Enable 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 times 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.
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:
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User Guide for the SIP-T42G IP Phone
1. Press the OK soft key to activate auto redial.
The LCD screen prompts the following:
2. Wait for a 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 callee
when the callee becomes available to receive a call.
To configure call completion via web user interface:
1. Click on Features->General Information.
2. Select Enable from the pull-down list of Call Completion.
3. Click Confirm to accept the change.
74
Basic Call Features
Call completion is not available on all servers. For more information, contact your system
administrator.
Call completion is configurable via web user interface only.
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. Press the OK soft key to redial the number.
Note
You can press a call return key to place a call back to the last incoming call.
To configure a call return 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 Call Return 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.
Call return 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
The phone will beep softly every 30 seconds to remind you that you still have a call on
hold.
You can mute the microphone of the active audio device during an active call, and then
the other party cannot hear you.
To mute a call:
1. Press during an active call.
The mute key LED illuminates solid red.
The LCD screen indicates that the call is on mute.
To un-mute a call:
1. Press again to un-mute the call.
Note
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 on hold 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.
To resume a held call:
1. Press the Resume soft key.
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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. Callers will
receive a busy message.
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.
2. In the DND block, mark the desired radio box in the Mode field.
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User Guide for the SIP-T42G IP Phone
DND mode is configurable via web user interface only.
3. Click Confirm to accept the change.
Note
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 Missed Call(s)" ("n" indicates
the number of the missed calls) will prompt on the LCD screen.
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 Enable from the DND Enables
field.
4. Press the Save soft key to accept the change.
The icon appears on the idle screen, and the associated line icon changes
to .
Incoming calls on the specific line will be rejected automatically, and the LCD
screen prompts "n Missed Call(s)" ("n" indicates the number of the missed calls).
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Basic Call Features
The prompt message displays 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.
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.
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 Missed Call(s)" ("n" indicates
the number of the missed calls) will be prompted on the LCD screen.
Note
To configure the DND authorized numbers via web user interface:
1. Click on Features->Forward & DND.
2. Select Enable from the pull-down list of DND Emergency.
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User Guide for the SIP-T42G IP Phone
DND authorized number is configurable via web user interface only.
3. Enter the numbers in the DND Authorized Numbers field.
For multiple numbers, enter a comma between every two numbers.
Note
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.
You can configure your phone to forward incoming calls to another party by the 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.
80
Basic Call Features
Call forward mode is configurable via web user interface only.
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:
1. Click on Features->Forward & DND.
2. In the Forward block, mark the desired radio box in the Mode field.
Note
3. Click Confirm to accept the change.
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 Enable from the Always
Forward field.
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User Guide for the SIP-T42G IP Phone
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 Enable 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 Enable 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.
4. Press the Save soft key to accept the change or the Back soft key to cancel.
The icon on the idle screen indicates 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.
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Basic Call Features
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 Enable 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.
b.) If you select Busy Forward, you can enable it for a specific account.
1) Press or , or the Switch soft key to select Enable 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.
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User Guide for the SIP-T42G IP Phone
You can also enter the SIP URL or IP address in the Forward to field. For more information
on using the SIP URL 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.
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 Enable 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 or to highlight the No Answer 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.
5. Press the Save soft key to accept the change or the Back soft key to cancel.
The icon appears on the idle screen, and the associated line icon changes to .
Call forward is configurable via web user interface at the path Features->Forward &
DND.
Note
84
Basic Call Features
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
soft key.
3. Press or , or the Switch soft key to select Disable to disable the call
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
soft key.
4. Press or , or the Switch soft key toselect Disable to disable the call
forward.
5. Press the Save soft key to accept the change.
Dynamic Forwarding
To forward an incoming call to another party:
1. When the phone is ringing, press the Fwd 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.
You can transfer a call to another party in one of the 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 Tran soft key during a call.
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User Guide for the SIP-T42G IP Phone
2. Enter the number you want to transfer the call to.
3. Press the Tran 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:
1. Press the Tran 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 should be configured in advance.
Refer to Directory on page 29 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 29 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 29 and Remote
Phone Book on page 40 for more information).
3. Press or to dial out.
4. Press the Tran soft key to complete the transfer when receiving ringback.
To perform an attended transfer:
1. Press the Tran 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 29 for more information).
86
- 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 29 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 29 and Remote
Basic Call Features
Network conference is not available on all servers. For more information, contact your
system administrator.
Phone Book on page 40 for more information).
3. Enter the number you want to transfer the call to.
4. Press or to dial out.
5. After the party answers the call, press the Tran soft key to complete the transfer.
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 when there is 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.
To configure call waiting via phone user interface:
1. Press Menu->Features->Call Waiting.
2. Press or , or the Switch soft key to select Enable from the Call Waiting field.
3. Press or , or the Switch soft key to select Enable from the Play Tone field.
4. (Optional.) Enter the call waiting on code or off code respectively in the CW On
Code or CW 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 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
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User Guide for the SIP-T42G IP Phone
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. 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 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 make sure 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:
- 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.
- 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.
- 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.
88
- Press to mute the conference call.
- Press the Cancel 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.
Basic Call Features
Network conference is configurable via web user interface only.
This feature allows you to perform the following:
Join two calls together into a conference call.
Invite another party into an active conference call.
Remove a specific conference party.
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.
3. Enter the conference URI (e.g., conference@example.com) in the Conference URI
field.
4. Click Confirm to accept the change.
Note
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.
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User Guide for the SIP-T42G IP Phone
Call park is not available on all servers. Contact your system administrator for more
information.
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.
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
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, then
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