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NOTICE
The information contained in this document is believed to be accurate in all respects but is
not warranted by Mitel Networks Corporation (MITEL®). The information is subject to change
without notice and should not be construed in any way as a commitment by Mitel or any of its
affiliates or subsidiaries. Mitel and its affiliates and subsidiaries assume no responsibility for
any errors or omissions in this document. Revisions of this document or new editions of it
may be issued to incorporate such changes.
Mitel Networks is a trademark of Mitel Networks Corporation.
Internet Explorer is a trademark of Microsoft Corporation.
Netscape Navigator is a trademark of Netscape Communications Corporation.
Other product names mentioned in this document may be trademarks of their respective
companies and are hereby acknowledged.
For Firmware 2.0 Revision E
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Table of Contents
TABLE OF CONTENTS
ABOUT YOUR SIP PHONE ...................................................................................... 1
Welcome..............................................................................................................................1
About This User Guide.................................................................................................................... 1
The 5055 SIP Phone ...........................................................................................................2
Elements of the SIP Phone ............................................................................................................. 2
Features of the SIP Phone.............................................................................................................. 4
Accessories for the SIP Phone ....................................................................................................... 5
Important Notes ...................................................................................................................5
Passwords....................................................................................................................................... 5
Tips for Your Comfort and Safety.................................................................................................... 6
5055 SIP PHONE FEATURES .................................................................................. 7
Accessing the SIP Phone Features ..................................................................................... 7
The SIP Phone Menu Interface....................................................................................................... 7
The Web Configuration Tool ........................................................................................................... 7
User Profiles ...................................................................................................................... 10
Logging In and Out........................................................................................................................10
Activating a Profile......................................................................................................................... 12
Temporary Registration................................................................................................................. 12
Making and Answering Calls .............................................................................................13
Making Calls.................................................................................................................................. 14
Answering Calls............................................................................................................................. 16
Switching between Handset, Headset and Handsfree Modes...................................................... 16
Using the SIP Phone Features .......................................................................................... 17
Advisory Messages ....................................................................................................................... 17
Call Forward .................................................................................................................................. 18
Call Transfer.................................................................................................................................. 20
Call Waiting ................................................................................................................................... 21
Calling List Logs ............................................................................................................................ 21
Conference Call (3-Way)............................................................................................................... 22
Display Contrast ............................................................................................................................ 22
Display Name ................................................................................................................................ 23
Do Not Disturb............................................................................................................................... 23
Hold ............................................................................................................................................... 24
Muting a Call ................................................................................................................................. 24
Password....................................................................................................................................... 24
Personal Keys ............................................................................................................................... 25
Phone Book................................................................................................................................... 26
Redial ............................................................................................................................................ 27
Ringer Pitch and Volume .............................................................................................................. 27
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5055 SIP Phone User Guide (Rev. A)
Speaker Volume (Handset, Headset and Handsfree)................................................................... 28
Speed Dial..................................................................................................................................... 28
Time and Date............................................................................................................................... 29
Web Dialing ................................................................................................................................... 31
ADMINISTRATOR TOOLS ..................................................................................... 32
Changing Passwords......................................................................................................... 32
Setting Up User Profiles ....................................................................................................33
Viewing a Profile’s User Name...................................................................................................... 33
Creating a User Profile.................................................................................................................. 33
Modifying a User Profile ................................................................................................................ 34
Deleting a User Profile ..................................................................................................................34
Creating/Modifying a SIP Account .....................................................................................35
Modifying the Network Configuration................................................................................. 35
Viewing the IP and MAC Addresses ............................................................................................. 35
Modifying Network Configurations ................................................................................................35
Upgrading the Firmware of the SIP Phone ........................................................................ 37
Viewing the Firmware Version ...................................................................................................... 37
Upgrading the firmware using the web-page Interface ................................................................. 37
Upgrading the firmware using the SIP Phone Menu Interface...................................................... 38
Additional Settings .............................................................................................................38
Configuration Files ........................................................................................................................38
Hot Line ......................................................................................................................................... 40
Media Configuration ...................................................................................................................... 40
Resetting to Factory Defaults........................................................................................................ 41
APPENDIX A — SIP PHONE INTERFACE ............................................................ 42
SIP Phone Menu Interface................................................................................................. 42
Menu Key Commands .......................................................................................................44
APPENDIX B — WEB CONFIGURATION TOOL ................................................... 45
Accessing the Web Configuration Tool.............................................................................. 45
Home Page........................................................................................................................45
User Configuration Page ...................................................................................................46
Feature Configuration Page............................................................................................... 47
Phone Book Page.............................................................................................................. 50
Dial by URL Page .............................................................................................................. 50
Key Programming Page..................................................................................................... 51
Ring Tone Page.................................................................................................................52
SIP Configuration Page ..................................................................................................... 52
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Table of Contents
Network Configuration Page.............................................................................................. 55
Dialing Plan........................................................................................................................ 59
Ethernet Page....................................................................................................................64
Protocols Page ..................................................................................................................65
Users & Passwords Page .................................................................................................. 66
Media Configuration Page ................................................................................................. 67
Registration Page .............................................................................................................. 68
Configuration Upload/Download Page............................................................................... 69
Upgrade ............................................................................................................................. 70
APPENDIX C — CONFIGURATION FILES ............................................................ 71
Example of a Generic Configuration File ...........................................................................71
Example of a Specific Configuration File ........................................................................... 76
Enabling Multiple User Profiles.......................................................................................... 84
APPENDIX D — WORKING WITH FIREWALLS.................................................... 85
APPENDIX E — WORKING WITH THE 3050 ICP.................................................. 86
APPENDIX F — FREQUENTLY ASKED QUESTIONS .......................................... 87
How do I access the User Profiles? ..............................................................................................87
Does my 5055 SIP Phone work behind a non-SIP compliant router? .......................................... 87
Where do I go to find latest versions of the 5055 firmware?......................................................... 87
Where can I find the latest 5055 SIP Phone documentation? ......................................................88
Upon boot, the phone displays “PPPoE Initialize” and nothing else............................................. 88
How do I find out the IP address of my 5055 SIP Phone?............................................................ 89
What version of boot and main firmware is currently installed on my phone?.............................. 89
What languages are currently available for my 5055 SIP Phone? ............................................... 89
Why does my phone show *NO REG*?........................................................................................ 89
The time and date on my phone is not correct?............................................................................ 90
GLOSSARY............................................................................................................. 91
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5055 SIP Phone User Guide (Rev. A)
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About Your SIP Phone
About Your SIP Phone
Welcome
Congratulations on your purchase of the Mitel Networks™ 5055 SIP Phone, an intelligent
Session Initiation Protocol (SIP) telephone that manages its own call states and features. The
Mitel Networks 5055 SIP Phone connects you to other SIP Phones via the Internet. You can
dial by URL, IP Address, User Name, or User Number. If you have an account with a SIP
Service Provider, you can also make calls to telephones on the “regular” phone network
(PSTN).
The 5055 SIP Phone is a multi-line set, and can have up to three user profiles, each with its
own settings.
About This User Guide
This User Guide contains information on configuring and using your 5055 SIP Phone, and is
organized as follows:
• About Your SIP Phone (this section): basic information on the SIP Phone and its
features.
• 5055 SIP Phone Features: information on how to configure and use your SIP Phone.:
information on setting up user profiles, and modifying network and SIP account
configurations.
• Appendix A — SIP Phone Interface: overview of the SIP Phone Menu Interface that
can be used to program your SIP Phone.
• Appendix B — Web Configuration Tool: overview of the web-based Configuration Tool
that can be used to program your SIP Phone as well as to make calls.
• Appendix C — Configuration Files: examples of generic and specific configuration
files.
• Appendix D — Working with Firewalls: explains how to configure the SIP phone to
work with firewalls.
• Appendix E — Working with the 3050 ICP explains some of the benefits that can be
obtained by connecting at 5055 SIP phone to Mitel Networks 3050 ICP.
• Appendix F — Frequently Asked Questions provides tips on how to solve some
frequently encountered problems
• Glossary: definition of terms and acronyms found in this User Guide.
Document Conventions
• Text on the SIP Phone display or on a page of the Web Configuration Tool is shown in
double quotes (for example, “CALLING LISTS?”).
• SIP Phone keys and commands on the SIP Phone display are shown in bold (for
example, Menu).
• Sections within this document are shown in italics (for example, Using the SIP Phone
Features).
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5055 SIP Phone User Guide (Rev. A)
•
Links in a web page are shown as underlined text (for example, User Configuration).
• ▼ and ▲ represent the Down and Up Arrow keys on the SIP Phone (located just above
the keypad).
• and represent navigating softkeys on the display.
The 5055 SIP Phone
Figure 1 5055 SIP Phone
Elements of the SIP Phone
Display Screen
Provides a high-resolution, back-lit viewing area for ease of use. In default mode (default
display), it shows the name of the active user (see Figure 1 above). In Menu mode, it shows
prompts and information on the features.
Message Waiting/Ringing Indicator Lamp
Flashes when you have an incoming call or a new message in your voice mailbox. Is on
(steady) while the SIP Phone reboots.
Softkeys
Select a command or choice listed on the bottom line of the display screen. These
commands and choices change dynamically depending on the different modes of operation.
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About Your SIP Phone
Fixed Function Keys
Give you access to the following telephone functions:
• Menu (blue): Provides access to the telephone's menus.
• Trans/Conf: Initiates a call transfer or establishes a 3-party conference call.
• Redial: Redials the last number, name or address dialed.
• Message: Provides access to your voice mailbox (optional).
• Hold (red): Puts the current call on hold.
• Cancel: Selecting Cancel during a call, ends the call. When programming the SIP phone,
cancels an input and returns to the previous menu level.
• Speaker: Initiates a handsfree call, switches between handset and handsfree mode, or
disconnects a call while in handsfree mode.
• Microphone: Toggles the microphone off and on. In handsfree mode, a red light
indicates that the microphone is ON (your party hears you). In handset and headset
mode, the microphone key acts like a Mute key, and a red light indicates that the
microphone is OFF (your party can’t hear you).
Arrow Keys
Adjust the volume of the handset, headset, or speaker, and of the ringer volume. When
entering letters, changes character input from upper or lower case or vice versa. Are also
used to change the display contrast, and to navigate through some menus when
programming the SIP Phone. In this User Guide, the arrow keys are represented by
▲.
Fixed Feature Keys
Give you access to the following telephone features:
• Calling Lists: Provides immediate access to your Phone Book, Answered Calls Log,
Missed Calls Log, and Outgoing Calls Log.
• Advisory Message: Allows you to turn your Advisory Message on or off.
• Headset: Allows you to enable and disable headset operation.
Keypad
When making a call, used to enter the number, name, URL or IP address you want to dial.
When programming the SIP Phone, used to enter information. Depending on the context, the
keypad lets you enter only numbers, or numbers, letters and some special characters.
Line Keys
Allow you to initiate, receive, and manage calls by using the four pre-assigned line keys. The
default Line key is Line 1. If a line is busy, subsequent calls are received on the next
available Line key (Line 2, Line 3, then Line 4). The Line keys are not assigned to a specific
directory number or address (multi-line operation).
▼ and
¡ Personal Keys
Provide one-touch access to programmed Speed Dial numbers.
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5055 SIP Phone User Guide (Rev. A)
Features of the SIP Phone
User Profiles
The 5055 SIP Phone can have up to three user profiles, including a default user profile. Each
user profile has its user name and password, and can be personalized to the user’s
preferences. See User Profiles on page 10 for more information.
Administrative Mode
Some settings (network information, SIP Service Provider information, etc.) can only be
modified by the system administrator, using the Administrator user name and password.
Accessing the SIP Phone’s Features
You can personalize/change settings for your SIP Phone from the SIP Phone itself (SIP
Phone Menu Interface) and from any personal computer connected to the Internet, using a
web browser (Web Configuration Tool). You can make calls using your SIP Phone or the
Web Configuration Tool. See 5055 SIP Phone Features on page 7 for more information.
Entering Numbers and Letters Using the SIP Phone Keypad
Depending on the context, the keypad lets you enter only numbers, or numbers, letters and
some special characters.
When entering letters and special characters, you rapidly press the appropriate number key
several times until the desired character is displayed. Letters correspond to those on the
keypad, and characters to the table below. A flashing cursor indicates the position of the
character you are entering; it will advance if you press a different key on the keypad, or wait
about one second.
To enter a letter in uppercase, press the ▲ key before entering the letter. Press the ▼ key to
return to lowercase mode. To delete the last entered character, press the <−−− softkey.
Table 1 Alphanumeric Character Entry
Dial Pad
Key
1
2
3
4
5
6
7
8
9
0
*
#
Once Twice 3 Times 4 Times 5 Times 6 Times 7 Times 8 Times 9 Times
1 space ? ! → |
2 a b c
3 d e f
4 g h i
5 j k l
6 m n o
7 p q r s
8 t u v
9 w x y z
0 + & % $ ¥ “
* . = : / ; , – _
# @ ( ) [ ] < >
Press
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About Your SIP Phone
Accessories for the SIP Phone
The 5055 SIP Phone supports the following accessories:
Headsets
Mitel Networks has qualified a Plantronics, Inc. headset for use with the 5055 SIP Phone’s
dedicated headset port (Mitel part number 9132-800-500-NA). This headset is available in
North America only.
The 5055 SIP Phone has a dedicated headset port (identified by the icon) at the back to
connect an approved headset. An external amplifier is not needed.
Conference Units
Mitel Networks supports two conference units for use with the 5055 SIP Phone (Mitel
Networks 5305 Conference Unit and Mitel Networks 5310 Conference Unit).
Figure 2 Mitel Networks Conference Unit
Important Notes
Passwords
When you first receive your 5055 SIP Phone, it has default user names and passwords for
the Administrator and the default user profile. You should change these passwords as soon
as possible to prevent unauthorized changes to your SIP Phone.
Table 2 Default User Names and Passwords
Default User Name Default Password
Administration
Default User Profile
Only the Administrator can change the Administrator password. The Administrator can also
change the password of all other users.
Note: The Administrator default user name cannot be changed.
admin 5055
user hello
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Tips for Your Comfort and Safety
Don’t Cradle the Handset
Prolonged use of the handset can lead to neck, shoulder, or back discomfort, especially if you
cradle the handset between your ear and shoulder. If you use your 5055 SIP Phone a lot, you
may find it more comfortable to use a headset.
Protect your Hearing
Your 5055 SIP Phone has a control for adjusting the volume of the handset receiver or
headset. Because continuous exposure to loud sounds can contribute to hearing loss, keep
the volume at a moderate level.
Adjust the Phone for Easiest Viewing
• Press the tilt-release paddle on the telephone base.
• Tilt your telephone to the desired angle.
• Release the tilt-release paddle.
Figure 3 Tilt-Release Paddle
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5055 SIP Phone Features
5055 SIP Phone Features
Once your SIP Phone is installed and configured (see the 5055 SIP Phone Installation Guide
for more details), you can start using your phone.
This section contains information on using and personalizing your 5055 SIP Phone, and is
organized as follows:
• Accessing the SIP Phone Features: information on how to use the Web Configuration
Tool and the SIP Phone Menu Interface to access the features of your SIP Phone.
• User Profiles: information on user profiles, and on how to log in, log out, and activate
your user profile.
• Making and Answering Calls: information on the basic telephony features of your SIP
Phone.
• Using the SIP Phone Features: information on using the features of your SIP Phone.
Accessing the SIP Phone Features
You can make calls and personalize your SIP Phone from the SIP Phone itself (SIP Phone
Menu Interface) or using a computer (Web Configuration Tool).
The SIP Phone Menu Interface
Most features can be directly accessed using the keys on your SIP Phone. For other
features, you must use the SIP Phone Menu Interface, which is accessed using the Menu
key (see Appendix A — SIP Phone Interface on page 42 for an overview of these features).
To scroll backwards or forwards through the main menu of the SIP Phone Menu Interface,
press the << or >> softkeys. To scroll forwards or backwards through the sub-menus, press
the or softkeys. To go back a menu level, press the Cancel key. To exit the SIP Phone
Menu Interface, press the Menu key, or go off-hook (lift handset).
In this document, procedures using the SIP Phone Menu Interface are identified by a small
phone icon (
Appendix A — SIP Phone Interface on page 42 lists all the menus and submenus available
through the Sip Phone Menu Interface.
).
The Web Configuration Tool
You can personalize/change settings for your SIP Phone from any computer connected to the
Internet using a web browser (Netscape Navigator 4 or Internet Explorer 4 (minimum), or any
other equivalent browser). You can also make calls using the Web Configuration Tool.
Note: If your network is protected by a firewall, you normally will not be able to access your
SIP phone via the Web Configuration Tool from outside the firewall.
In this document, procedures using the Web Configuration Tool are identified by a small
computer icon (
).
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Appendix B — Web Configuration Tool on page 45 shows all the pages of the Web
Configuration Tool, and the functions for each page.
Accessing the Web Configuration Tool
To access the Web Configuration Tool:
1. Get the SIP Phone’s IP address:
• Press the Menu key.
• Press the Line 1 key on the SIP Phone. The top line of the display shows the IP
address of the SIP Phone.
• Note the IP address of the SIP Phone, and press the Menu key to return to the
default display.
2. Launch your computer’s browser.
3. Enter your SIP Phone's IP address in your browser's URL or Address field. The login
screen for the Web Configuration Tool appears.
Figure 4 Web Configuration Tool Login Screen
192.168.xx.xx
4. Enter your user profile user name and password in the appropriate fields.
5. Click the OK button. The home page of the Web Configuration Tool is loaded.
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Figure 5 Web Configuration Tool Home Page
5055 SIP Phone Features
Refreshing Web Configuration Tool Pages
If you need to refresh a page after changing settings using the Web Configuration Tool (for
example, you set your advisory message to On using the Web Configuration Tool, then
turned it back to Off using the Advisory Message key on the SIP Phone), go to another page
in the Web Configuration Tool, then come back to the page you wanted to refresh.
Do not use the Refresh or Reload button of your browser to refresh a Web Configuration
Tool page after changing settings using the Web Configuration Tool. Doing so will just
reapply the change you just did (and reboot your SIP phone, if you clicked a Save and
Reboot button).
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User Profiles
NOTE: By default the User Profiles are disabled for security purposes, to enable User
Profiles browse to the Phones User Configuration screen, locate the MultiUser
Profile option and select On. Select Save & Reboot to enable MultiUser
Profiles.
To access all the features the 5055 SIP Phone has to offer, you need a user profile and must
be registered with a SIP Service Provider (you can use the SIP Phone without a user profile,
but will not be able to use all its features). A user profile is usually created and registered by
the system administrator.
Once you have a user profile, you can personalize the following:
• your user profile information (password, display name),
• your user profile feature settings (Personal keys, phone book, call answer settings, etc.).
Your SIP Phone supports up to three user profiles, including a default user profile. The
default user profile is always logged in, and cannot be deleted.
When you log in the SIP Phone, you are automatically registered with your SIP Service
Provider, and can receive calls on the SIP Phone. When you activate your user profile, the
SIP Phone uses your user profile preferences (Speed Dial keys, etc.), and you can use the
Web Configuration Tool to make calls or change your user profile settings. More than one
user can be logged in at the same time, but only one user profile can be active at a time.
Note: Only users who have a user profile defined on a SIP Phone can log in to that SIP
Phone.
You can also temporarily register with your SIP Service Provider on a SIP Phone that does
not have your user profile. While you are temporarily registered on that phone, you can make
and receive calls with the SIP Phone, but cannot use the Web Configuration Tool.
Logging In and Out
Like a personal computer, the 5055 SIP Phone allows different users to log in and access
their personal settings. Incoming calls addressed to the logged-in user's name, SIP URL or
Number will be delivered to that SIP Phone. So, for example, if you need to do some work in
the lab but still want to answer incoming calls that you would normally answer at your desk,
you can log in to the lab's 5055 SIP Phone.
Note: Logging in on your SIP Phone automatically registers you with your SIP Service
Provider (assuming you have a SIP Service Provider, and a user profile defined on
the SIP Phone). If you have a SIP Service Provider, and for any reason the
registration process fails, “*NO REG*” appears on your display.
Logging In
1. Press the Menu key.
2. Press the >> softkey. “USERS?” is displayed.
3. Press the OK softkey. “1.LOGIN?” is displayed.
4. Press the OK softkey.
5. Enter your user profile user name, and press the Submit softkey.
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5055 SIP Phone Features
6. Enter your user profile password, and press the Submit softkey (if your SIP Service
Provider does not require a password, enter any character, and delete it using the <−−−
softkey before pressing the Submit softkey).
• If you made a mistake while entering your user name or password, “LOGIN
UNSUCCESSFUL” is displayed. Press the Retry softkey to return to step 4, or the
Cancel softkey to return to step 3.
7. “ACTIVATE PROFILE?” is displayed. To log in and activate your user profile, press the
Yes softkey. To log in without activating your user profile, press the No softkey.
8. “SET LOGIN EXPIRY?” is displayed.
• If you don’t want a login expiry time, press the No softkey, and continue with step 10.
• If you want to set your user profile to automatically be logged out (de-registered) after
a given period, press the Yes softkey and continue with the next step.
9. “LOGIN EXPIRY (HR)” is displayed.
• To enter a value in hours, enter the value using the keypad, and press the Submit
softkey.
• To enter a value in minutes, press the Minute softkey, enter the value using the
keypad, and press the Submit softkey.
• To enter a value in days, press the Days softkey, enter the value using the keypad,
and press the Submit softkey.
10. Once you are logged in, “LOGIN SUCCESSFUL” is displayed. Press the OK softkey,
then the Menu key to return to the default display.
Logging Out
If you have an account with a SIP Service Provider, logging out of the SIP Phone
automatically de-registers you with your SIP Service Provider.
1. Press the Menu key.
2. Press the >> softkey. “USERS?” is displayed.
3. Press the OK softkey. “1.LOGIN?” is displayed.
4. Press the softkey. “2.LOGOUT?” is displayed. Press the OK softkey.
5. Enter your user profile user name, and press the Submit softkey.
6. Enter you user profile password, and press the Submit softkey (if your SIP Service
Provider does not require a password, enter any character, and delete it using the <−−−
softkey before pressing the Submit softkey).
7. Your user name is displayed. Press the LogOut softkey.
8. “LOGOUT CONFIRMED” is displayed. Press the OK softkey, then the Menu key to
return to the default display.
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5055 SIP Phone User Guide (Rev. A)
Activating a Profile
When you activate your user profile, the SIP Phone uses your preferences (Speed Dial keys,
Display Name, etc.), and you can access the Web Configuration Tool to make calls or
change your user profile settings.
Note: To activate your user profile, you must already be logged in the SIP Phone.
1. Press the Menu key.
2. Press the >> softkey. “USERS?” is displayed.
3. Press the OK softkey.
4. Press the softkey until “3.ACTIVATE PROFILE?” is displayed. Press the OK softkey.
5. The name of the active user profile is displayed. Press the Change softkey to activate a
different user profile.
6. Enter your user profile user name, and press the Submit softkey.
7. Enter you user profile password, and press the Submit softkey (if you’re SIP Service
Provider does not require a password, enter any character, and delete it using the <−−−
softkey before pressing the Submit softkey).
8. Your user name is displayed. Press the Yes softkey to activate your profile.
9. “PROFILE ACTIVATED” is displayed. Press the OK softkey, then the Menu key to return
to the default display.
Temporary Registration
Temporary registration tells your SIP Service Provider that you can temporarily receive calls
at that phone.
To temporarily register on a SIP Phone, you need the following information from your SIP
Service Provider:
• Registration user name and password
• SIP Service Provider server IP address
• Phone IP address
• Registration method
Registering on the SIP Phone
1. Get the IP address of the SIP Phone to which you are registering:
• Press the Menu key.
• Press the Line 1 key on the SIP Phone. The top line of the display shows the IP
address of the SIP Phone.
• Note the IP address of the SIP Phone, and press the Menu key to return to the
default display.
2. Press the Menu key.
3. Press the >> softkey. “USERS?” is displayed.
4. Press the OK softkey.
5. Press the softkey until “6.REGISTRATION?” is displayed. Press the OK softkey.
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5055 SIP Phone Features
6.
For ENTER USERID: enter your SIP Registration user name, and press the Submit
softkey.
7. For ENTER PASSWORD: enter your SIP Registration password, and press the Submit
softkey (if your SIP Service Provider does not require a password, enter any character,
and delete it using the <−−− softkey before pressing the Submit softkey).
8. For CONTACT IP ADDREESS: enter the SIP Phone’s IP address, and press the Submit
softkey.
9. For SERVER ADDRESS: enter your SIP Service Provider server address, and press the
Submit softkey.
10. For TO ADDRESS: enter <user>@<SIP server address>, and press the Submit softkey.
11. For REGISTRATION METHOD: select the registration method by pressing the
appropriate softkey:
• None: no registration authentication.
• Basic: Authentication
• Digest: Authentication
12. The display shows the registration method chosen. Press the Submit softkey to confirm
your choice, or the Cancel softkey to choose another registration method (step 11).
13. Enter the registration duration (in hours), and press the Submit softkey (after that
duration, you will automatically be de-registered).
14. The display shows “REGISTER NOW”. Press the Confirm softkey to register, or the
Cancel softkey to change your registration duration (step 13).
When registration is complete, the display shows “REGISTRATION SUCCESSFUL”.
Making and Answering Calls
This section shows you how to make and receive calls with the SIP Phone. Only basic call
making procedures are shown (that is, calls made using the keypad). For information on
other call making features see Using the SIP Phone Features on page 17.
Your SIP Phone can be used in any of the three following modes:
• Handset mode: this is when you are using the handset to talk and listen to your party.
• Headset mode: this is when you have a headset connected to your SIP Phone, and you
use it to talk and listen to your party. Headset mode is activated by pressing the Headset
key (you can have a headset connected to your SIP Phone and still use the handset to
make your calls).
• Handsfree mode: this is when you are using the SIP Phone’s handsfree speaker to talk
and listen to your party. Handsfree mode is activated by pressing the Speaker key.
This section shows how to make and receive call in any of the three modes, as well as how to
change from one mode to another. In the rest of the document, instructions are given for the
handset mode only, for clarity’s sake.
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5055 SIP Phone User Guide (Rev. A)
Making Calls
With the 5055 SIP Phone you can dial:
• by number (user number, telephone number).
• by name (user name).
• by URL (SIP URL address, SIP IP address).
Note: To dial a regular telephone number, you must have a SIP Service Provider that
provides access to the regular (PSTN) phone network.
When the connection is successful, the address of your party is displayed (truncated to 14
characters), and a counter starts at the top right of the display. If the line is busy, the display
shows “BUSY HERE”.
To end or abort a call:
• All modes:
− Press the Cancel key or the Hangup softkey to get a new dial tone.
− Press the Line key associated with the call to return to the default display.
• Handset mode only:
− Put the handset back in its cradle.
• Handsfree mode only:
− Press the Speaker key. This returns you to the default display.
Dialing by Number
To dial by number:
1. Get a dial tone:
• Handset mode: lift the handset.
• Headset mode, press a Line key. You can also start entering a number after pressing
the Headset key (you won’t get a dial tone).
• Handsfree mode: press the Speaker key or press a Line key. You can also start
entering a number after pressing the Speaker key (you won’t get a dial tone).
This selects the first free line (Line 1 if all lines are free). To select another line, press the
associated Line key. The light of the selected Line key turns red. In handsfree mode, the
Microphone key light turns red; in headset mode, and the Headset key light turns red.
2. Enter the number of the party you want to reach using the keypad. 5055 SIP. Phones that
are connected to 3050 ICPs, can use abbreviated two or three-digit numbers to reach
other 3050 ICP attached phones, or regular 7-digit numbers to reach phones on the
PSTN. Check with your phone system installer to learn if your system supports these
options.
• If you mistype a number, press the <−−− softkey to delete it, and re-enter the correct
number.
• To delete all characters entered and enter a different number, press the Cancel
softkey.
3. Press the Dial softkey. The number you entered is dialed, and the light of the selected
Line key turns green.
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Dialing by Name
To dial by name:
1. Get a dial tone:
• Handset mode: lift the handset.
• Headset mode,: press a Line key.
• Handsfree mode: press the Speaker key or press a Line key.
This selects the first free line (Line 1 if all lines are free). To select another line, press the
associated Line key. The light of the selected Line key turns red. In handsfree mode, the
Microphone key light turns red; in headset mode, and the Headset key light turns red.
2. Press the Name softkey.
3. Using the keypad, enter the name of the party you want to reach (see Entering Numbers
and Letters Using the SIP Phone Keypad on page 4 for information on entering letters
and symbols).
• If the name has more than 20 characters, the display will only show the rightmost 20
characters.
• If you mistype a character, press the <−−− softkey to delete it, and re-enter the
correct character.
• To delete all characters entered and enter a different name, press the Cancel
softkey.
4. Press the Dial softkey. The name you have entered is dialed, and the light of the selected
Line key turns green.
Dialing by SIP URL
To dial a URL:
1. Get a dial tone:
• Handset mode: lift the handset.
• Headset mode press a Line key.
• Handsfree mode: press the Speaker key or press a Line key.
This selects the first free line (Line 1 if all lines are free). To select another line, press the
associated Line key. The light of the selected Line key turns red. In handsfree mode, the
Microphone key light turns red; in headset mode, and the Headset key light turns red.
2. Press the URL softkey.
3. Using the keypad, enter the address of the party you want to reach (see Entering
Numbers and Letters Using the SIP Phone Keypad on page 4 for information on entering
letters and symbols).
• If the URL has more than 20 characters, the display will only show the rightmost 20
characters.
• If you mistype a character, press the <−−− softkey to delete it, and re-enter the
correct character.
• To delete all characters entered and enter a different address, press the Cancel
softkey.
4. Press the Dial softkey. The address you have entered is dialed, and the light of the
selected Line key turns green.
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Answering Calls
An incoming call will ring on the first available line (Line 1 if all lines are free). If all lines are
bus, and Call Forward on Busy is not enabled the caller gets a busy signal (refer to
Enabling/Disabling Call Forward for details on call forwarding). While the phone is ringing, the
Ringing indicator flashes red, the name of the caller is displayed and the associated Line key
light flashes green.
To answer an incoming call:
• Handset mode:
− Lift the handset.
• Headset mode:
− Press the Headset key and press the flashing green Line key.
• Handsfree mode:
− Press the Speaker key, or
− Press the associated Line key (this will activate the handsfree mode).
When you answer the call, a counter starts at the top right of the display.
Switching between Handset, Headset and Handsfree Modes
With the 5055 SIP Phone, you can make and receive calls using the attached handset, the
handsfree speaker, or an approved headset (see Headsets on page 5 for information on
approved headsets).
Switching Between Handset and Handsfree Modes
• To go from handset to handsfree mode:
− Press the Speaker key.
− Put the handset back in its cradle.
− When you are connected, the Microphone key light turns red. You can now talk to
your party using the handsfree speaker.
• To go from handsfree to handset mode:
− Lift the handset. The Microphone key light turns off.
− You can now talk to your party using the handset.
Switching Between Handset and Headset Modes
• To go from handset to headset mode:
− Press the Headset key. The Headset key light turns red.
− Put the handset back in its cradle.
− You can now talk to your party using the headset.
• To go from headset to handset mode:
− Lift the handset.
− Press the Headset key. The Headset key light turns off.
− You can now talk to your party using the handset.
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Switching Between Headset and Handsfree Modes
• To go from headset to handsfree mode:
− Press the Headset Key, followed by the Speaker key. The Headset key light turns
off.
− When you are connected, the Microphone key light turns red. You can now talk to
your party using the handsfree speaker.
• To go from handsfree to headset mode:
− Press the Headset key. The Headset key light turns red, and the Microphone key
light turns off.
− You can now talk to your party using the headset.
Using the SIP Phone Features
The features in this section can be accessed using the Web Configuration Tool and/or the
SIP Phone Menu Interface. You cannot change these settings using the SIP Phone Menu
Interface while on a call. You can change these settings using the Web Configuration Tool
while on a call, but the changes will not take effect until you’ve finished your current calls.
Note: Some Web Configuration Tool settings require you to reboot your SIP Phone. If you
click the Save and Reboot button while on a call, you will lose the connection when
your phone reboots.
Advisory Messages
Setting up an Advisory Message
You can set up an advisory message to alert callers to your current status (for example, when
you’re going on vacation). You set up advisory messages using the Web Configuration Tool.
Note: You cannot change your Advisory Message settings while on a call. Access the Web
Configuration Tool (see The Web Configuration Tool on page 7).
1. Click Feature Configuration.
2. Turn Advisory message On in the pull-down menu (you can use the Advisory message
key on the phone to toggle this on or off too).
3. Choose the message you want to display from the drop-down menu at the right of
“Advisory Message:”.
• If none of the choices suit you, select the message “Other reason”, and enter the
desired message beside “Other:” (any message longer than 20 characters will be
truncated on the SIP Phone display).
4. Click the Apply button at the bottom of the web page. A confirmation screen is displayed.
5. Click the OK button. Your SIP Phone is updated.
Enabling/Disabling your Advisory Message
When your Advisory Message is on, the Advisory Message indicator key turns red, and your
Advisory Message periodically replaces the time and date on your SIP Phone display.
Note: You cannot change your Advisory Message settings while on a call.
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Using the SIP Phone Interface
• Press the Advisory Message key to activate/deactivate your Advisory Message.
Using the Web Configuration Tool
1. Access the Web Configuration Tool (see The Web Configuration Tool on page 7).
2. Click Feature Configuration.
3. To enable the advisory message, select On in the drop-down menu at the far right of
“Advisory Message:”. To disable the advisory message, select Off.
4. Click the Apply button at the bottom of the web page. A confirmation screen is displayed.
5. Click the OK button. Your SIP Phone is updated.
Call Forward
Call Forward lets you redirect incoming calls to an alternate number:
• Call Forward Always redirects all incoming calls regardless of the state of your telephone.
• Call Forward No Answer redirects calls after the programmed number of rings if you don't
answer.
• Call Forward Busy redirects calls when all four lines are busy.
The default setting is Call Forward Off for all three options. You can set two or more Call
Forward options On at the same time.
Note: When Call Forward is active, “*FWD ON*” alternates with the date on the SIP Phone’s
display.
Note: You cannot change these settings while on a call.
Setting Up Call Forward
You can set your Call Forward settings using the Web Configuration Tool or the SIP Phone
Menu Interface.
Note: You cannot change your Call Forward settings while on a call.
Using the Web Configuration Tool
1. Access the Web Configuration Tool (see The Web Configuration Tool on page 7).
2. Click Feature Configuration.
3. You can enable Call forwarding by either:
• setting the Call Forwarding field to On, and entering the Forwarding Address in the
associated field. This can be the URL of another SIP phone, URL of a SIP voicemail
account or a PSTN number (provided that the SIP server supports PSTN gateway
functions).
or
• setting the Call Forwarding field to On , and leaving the Forwarding Address blank. In
this case, the phone will forward the call to voicemail automatically. This will work
provided you have the URL of your SIP voicemail account programmed into the
Voice Mail Server field of the SIP Configuration Page.
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5055 SIP Phone Features
4.
For Call Forward No Answer, enter how many times the phone will ring before the call is
forwarded.
5. Click the Apply button. A confirmation screen is displayed.
6. Click the OK button. Your SIP Phone is updated.
Using the SIP Phone Menu Interface
1. Press the Menu key.
2. Press the >> softkey until “FEATURE CONFIG?” is displayed.
3. Press the OK softkey. “CALL FORWARDING?” is displayed.
4. Press the OK softkey. “FWD ALWAYS:” is displayed, with its status beside it (“*ON*” or
“*OFF*”).
• If you do not need to change your Call Forward Always settings, go to step 10.
• To program your Call Forward Always settings, continue below.
5. Press the Review softkey. The display shows the current forwarding address.
• If the address is blank and a valid voicemail server URL has been programmed into
SIP Configuration, the SIP Phone will forward calls to your voice mailbox
6. Press the Program softkey to change the address to which the call will be forwarded.
7. To forward your calls to your voice mailbox, press the Yes softkey and continue with
step 10. To forward your calls to another address, press the No softkey and continue with
the next step.
8. Enter the address where your calls will be forwarded.
• To enter a name, press the Name softkey before entering any characters.
• To enter a URL, press the URL softkey before entering any characters.
9. Press the Submit softkey.
10. Press the Next softkey. “FWD NO ANSWER:” is displayed, with its status beside it
(“*ON*” or “*OFF*”)
• If you do not need to change your Call Forward No Answer settings, go to step 14.
11. Press the Options softkey to change the number of rings before a call is forwarded.
12. Enter the number of rings (from 0 to 9) with the keypad, and press the Save softkey.
13. To program your other Call Forward No Answer settings, repeat steps 5 to 9, then
continue below.
14. Press the Next softkey. “FWD BUSY:” is displayed, with its status beside it (“*ON*” or
“*OFF*”)
• To program your Call Forward Busy settings, repeat steps 5 to 9, then continue
below.
• If you do not need to change your Call Forward Busy settings, continue below.
15. Press the Exit softkey, then the Menu key to return to the default display.
Enabling/Disabling Call Forward
Using the Web Configuration Tool
1. Access the Web Configuration Tool (see The Web Configuration Tool on page 7).
2. Click Feature Configuration.
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3. For each type of Call Forward (Always, No Answer, Busy), change its status (On or Off)
using the drop down menu beside its name.
4. Click the Apply button at the bottom of the web page. A confirmation screen is displayed.
5. Click the OK button. Your SIP Phone is updated.
Using the SIP Phone Menu Interface
1. Press the Menu key.
2. Press the >> softkey until “FEATURE CONFIG?” appears, and press the OK softkey.
3. “1.CALL FORWARDING?” is displayed. Press the OK softkey.
4. The display shows the status of Call Forward Always at the top right (“*ON* or *OFF*).
Press the TurnOn softkey to activate call Forward Always, or the TurnOff softkey to
deactivate it.
5. Press the Next softkey.
6. The display shows the status of Call Forward No Answer at the top right (“*ON* or
*OFF*). Press the TurnOn softkey to activate Call Forward No Answer, or the TurnOff
softkey to deactivate it.
7. Press the Next softkey.
8. The display shows the status of Call Forward Busy at the top right (“*ON* or *OFF*).
Press the TurnOn softkey to activate Call Forward Busy, or the TurnOff softkey to
deactivate it.
9. Press the Exit softkey, then the Menu key to return to the default display.
Call Transfer
You can transfer an active call to another party. To do so, at least one line must be free on
the SIP Phone.
Note: The 5055 SIP phone supports 4 lines. If all lines are busy on your phone, none of your
callers will be able to transfer their call away from you to another phone. You must
first free up one of the lines to allow callers to transfer a call away from your phone.
Transferring a Call to an Unconnected Third Party
1. While on a call, press the Trans/Conf key. The call is put on hold.
2. Press a free Line key, a Speed dial key or redial.
3. Call the party to whom you want to transfer the call.
• If you want to talk to this person, wait until the connection is established then press
the Trans softkey to transfer the held call (attended call transfer).
• If you don’t need to talk to this person, press the Trans softkey immediately, and then
hang up. The held call will be transferred to the call in progress, even if it has not yet
been picked up (blind or unattended call transfer).
• If you want to cancel the transfer, press the Cancel softkey. You are returned to the
held call.
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5055 SIP Phone Features
Transferring a Call to a Third Party Already on Hold
1. While on a call, press the Trans/Conf key. The call is put on hold.
2. Press the Line key of the call on hold to which you want to transfer the call, and press the
Trans softkey. You can then hang up.
Call Waiting
You can have up to four active calls on your SIP Phone. Any new call goes to the next free
line; if all lines are busy, the caller gets a busy signal.
When a new call comes in, you hear a call waiting tone, the name of the new caller is
displayed, and the next available Line key light flashes green.
• To answer the incoming call while already connected to another call, press the Line key
of the incoming call. The current call will be put on hold, and you are connected to the
new caller (see Putting a Call on Hold on page 24 for information on dealing with calls on
Hold).
Calling List Logs
The Calling Lists Logs keeps a record of your answered, missed and outgoing calls. It
records the five most recent calls for each of the three types of calls. For example, the five
most recent incoming calls are logged while the five most recent missed calls are logged. The
most recent call appears at the top of the each log.
The call information recorded includes the party’s number, name or URL address, the call
duration, and the time and date of each call.
When you have missed incoming calls, the number of calls missed replaces the date on the
SIP Phone display.
Note: The calling lists log information is stored directly in the SIP Phone. Your user profile
must be logged in and active to use your Calling List Logs.
Viewing the Calling List Logs
To view information on an incoming, missed, or outgoing Calling List entry:
1. Press the Calling Lists key (you can also get to the Calling List by pressing the Menu
key, then the >> softkey until you reach “CALLING LISTS?”, then the OK softkey).
2. Scroll using the and softkeys to the desired log (Missed Calls, Answered Calls or
Outgoing Calls), and press the OK softkey.
3. The display shows how many calls are in that log. Use the ▼ and ▲ keys to view the
calls in the log.
4. For each call, you can:
• View the information about that call (press the Detail softkey, then the Done softkey
to return).
• Delete the call from the Calling List (press the Delete softkey and follow the
prompts).
• Dial the caller’s address (press the Dial softkey. This exits the call log, and the SIP
Phone returns to the default display at the end of the call).
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5. To view calls from another log, press the Cancel key, then the OK softkey, and repeat
steps 2 to 4. When you are finished, press the Menu key to return to the default display.
Conference Call (3-Way)
The 5055 SIP Phone supports three-party conferences.
Note: Conference call is not available when your SIP Phone is set for G.729 audio codec
(the Conf softkey is replaced by NA to indicate that the feature is not available).
Adding a Third Party to a Call in Progress
1. Press the Trans/Conf key. The call is put on hold.
2. Press a free Line key.
3. Enter the address of the new party and press the Dial softkey.
4. Once you have connected with the new party, press the Conf softkey. The call on hold is
connected to the call in progress.
Note: If the new party does not answer, press the Cancel key twice to return to the
held party.
Adding a Party on Hold to a Call in Progress
1. Press the Trans/Conf key. The call is put on hold.
2. Press the Line key of the party already on hold.
3. Once you have connected with the new party, press the Conf softkey. The call put on
hold in step 1 is connected to the call in progress.
Leaving a Conference Call
To leave a Conference Call, hang up the handset, press the Hangup softkey, or press the
Cancel key.
Note: If the originator of the conference call hangs up, then the other two parties do not
remain connected. If either of the called parties hangs up, the call will remain
connected.
Display Contrast
You can adjust the contrast of the display to suit your preference.
Changing the Display Contrast
• While the phone is idle, use the ▼ or ▲ key to adjust the display contrast to the desired
level (press repeatedly to change by more than one level).
Note: When changing the display contrast this way, the setting is first stored in temporary
memory. The temporary memory is saved to permanent (flash) memory at regular
intervals during the day. If your SIP Phone loses power or reboots between the time
you changed the setting and a flash memory update, the new setting will be lost.
Changing the Display Contrast—Immediate Save
1. Press the Menu key.
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2. Press the >> softkey until “PHONE SETTINGS?” appears, and press the OK softkey.
3. Press the softkey until “3.LCD CONTRAST?” appears, and press the OK softkey.
4. Use the ▼ or ▲ key to adjust the display contrast to the desired level (press repeatedly
to change by more than one level), and press the OK softkey. The new setting is saved in
the permanent (flash) memory.
5. You are returned to the Phone Settings menu. To return to the default display, press the
Menu key.
Display Name
The Display Name is the name displayed on your SIP Phone when you are logged in and
your user profile is active. That name also appears on the display of a call’s recipient. You
can change your display name using the Web Configuration Tool.
Note: You cannot change your Display Name while on a call.
1. Access the Web Configuration Tool (see The Web Configuration Tool on page 7).
2. Click User Configuration.
3. Beside “User Display name”, enter the name you want to appear on the display.
4. Click the Save and Reboot button. A confirmation screen is displayed.
5. Click the OK button. Your SIP Phone is updated, and then reboots.
Do Not Disturb
Do Not Disturb forwards all your calls directly to your voice mailbox, so you are not disturbed
by a ringing phone. If you do not have a voice mailbox setup, the callers will get a busy
signal.
Note: When Do Not Disturb is active, “*DND ON*” alternates with the date on the SIP
Phone’s display (if both Call Forward and Do Not Disturb are on, “*DND ON*”
alternates with the time on the display).
Note: You cannot change your Do Not Disturb settings while on a call.
Activating/Deactivating Do Not Disturb
Using the Web Configuration Tool
1. Access the Web Configuration Tool (see The Web Configuration Tool on page 7).
2. Click Feature Configuration.
3. Select On or Off from the drop down menu beside “Do Not Disturb”.
4. Click the Apply button. A confirmation screen is displayed.
5. Click the OK button. Your SIP Phone is updated.
Using the Phone Menu Interface
1. Press the Menu key.
2. Press the >> softkey until “FEATURE CONFIG?” is displayed, and press the OK softkey.
3. Press the softkey. “2.DO NOT DISTURB?” is displayed.
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4. Press the OK softkey. The current status of Do Not Disturb is displayed at the top right
(“*ON*” or “*OFF*”).
5. Press the TurnOn softkey to activate Do Not Disturb, or the TurnOff softkey to
deactivate it.
6. Press the Exit softkey, then the Menu key to return to the default display.
Hold
You can have up to four calls on hold at the same time on your SIP Phone.
Putting a Call on Hold
• To place a call on Hold, press the Hold key; the associated Line key flashes red while its
call is on hold. To retrieve a call from Hold, press the associated Line key.
Changing On Hold Settings
When you place a call on hold, you will get a regular beep after a programmed delay to
remind you that you have a call on hold (if the handset is in its cradle, you will hear the beep
through the handsfree speaker). When another party puts you on hold, you hear a regular
beep to remind you that you are on hold; you can turn off this beep if desired.
Note: You cannot change your Hold settings while on a call.
1. Access the Web Configuration Tool (see The Web Configuration Tool on page 7).
2. Click Feature Configuration.
3. To remove the regular beep you hear when you are on hold, select Off in the drop-down
menu beside “Beep on Hold”.
4. To define the delay before your SIP Phone reminds you that you have a caller on hold,
enter a value in seconds beside “Held call will ring back after:”.
5. Click the Apply button. A confirmation screen is displayed.
6. Click the OK button. Your SIP Phone is updated.
Muting a Call
To mute your SIP Phone so the person on the other end of the line cannot hear you, press
the Microphone key. To turn off the Mute function, press the Microphone key once more.
• In handset and headset modes, the Microphone key light is red while the call is muted.
• In handsfree mode, the Microphone key light is off while the call is muted.
Password
You can change your user profile password using the SIP Phone Menu Interface, or the Web
Configuration Tool.
Note: If you have an account with a SIP Service Provider, use the password given to you by
the SIP Service Provider.
Note: You cannot change your password while on a call.
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Changing Your Password
Using the Web Configuration Tool
1. Access the Web Configuration Tool (see The Web Configuration Tool on page 7).
2. Click User Configuration.
3. Change the password beside “Password:”.
4. Click the Save and Reboot button. A confirmation screen is displayed.
5. Click the OK button. Your SIP Phone is updated, and then reboots.
Using the SIP Phone Menu Interface
1. Press the Menu key.
2. Press the >> softkey. “USERS?” is displayed.
3. Press the OK softkey.
4. Press the softkey until “4.CHANGE PASSWORD?” is displayed, and press the OK
softkey.
5. Enter your user profile user name, and press the Submit softkey.
6. Enter your current password, and press the Submit softkey (if your existing password is
blank, enter any character, and delete it using the <−−− softkey before pressing the
Submit softkey).
7. Enter your new password, and press the Submit softkey.
8. Enter your new password again, and press the Submit softkey.
• If you have entered both instances of the new password correctly, “NEW
PASSWORD CREATED” is displayed. Press the OK softkey.
• If you have made a mistake, “PASSWORD MISMATCH” is displayed. Press the
Retry softkey to go back to step 7.
9. Press the Menu key to return to the default menu.
Personal Keys
Using the SIP Phone Menu Interface, you can program a Personal key as a Speed Dial key.
You change the feature of a Personal key by deleting the existing programming and applying
a new program (see the entry for the actual feature for instructions on programming and
using Personal Keys with that feature).
Note: You cannot change your Personal Keys settings while on a call.
Verifying a Personal Key’s Program
1. Press the Menu key.
2. Press the >> softkey until “PROGRAM MEMORY KEYS?” appears, and press the OK
softkey.
3. Press the Personal key you want to check. The key’s light turns red.
• If the key is not yet programmed, the display reads “UNUSED KEY”.
• If the key is already programmed, its associated program is displayed.
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4. Press the Menu key to return to the default display.
Deleting a Personal Key’s Program
1. Press the Menu key.
2. Press the >> softkey until “PROGRAM MEMORY KEYS?” appears, and press the OK
softkey.
3. Press the Personal key you want to clear. The key’s light turns red, and its associated
programming is displayed.
4. Press the Delete softkey.
5. “DELETE ITEM?” is displayed. Press the YES softkey to delete it.
6. “UNUSED KEY” is displayed. To return to the default menu, press the Menu key.
Phone Book
Your user profile’s Phone Book can hold up to five contacts.
Note: You cannot change your Phone Book settings while on a call.
Creating/Modifying a Phone Book
1. Access the Web Configuration Tool (see The Web Configuration Tool on page 7).
2. Click Dial by Phone Book.
3. For each contact, enter a nickname, and the SIP address for this contact (name, number
or URL).
• To change a contact, simply type over an existing one.
Note: When entering a telephone number, enter it without any separators.
4. Click the Apply button to save your contacts. A confirmation screen is displayed.
5. Click the OK button. Your SIP Phone is updated.
Making Calls With the Phone Book
Using the SIP Phone:
To make a call to a contact on your Phone Book:
1. Get a dial tone (see Making Calls on page 14).
2. Press the Calling Lists key.
3. “1.PHONE BOOK?” is displayed, Press the OK softkey.
4. Use the ▼ and ▲ keys to go to the contact you want to call, and press the Dial softkey.
The contact’s address is dialed.
Using the Web Configuration Tool:
1. Access the Web Configuration Tool (see The Web Configuration Tool on page 7).
2. Click Dial by Phone Book.
3. Select the contact you want to reach in the drop-down menu beside “Select Contact:”.
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4. Click Dial. The address is dialed (in handsfree mode) on the next available Line key
(Line 1 if all lines are free).
• If you want to use the handset or headset, lift the handset or press the Headset key
before clicking “Dial” in the Web Configuration Tool.
• If you want to use another line than Line 1, press the desired line key on the SIP
Phone before clicking “Dial” in the Web Configuration Tool.
Redial
Redial calls back the last party you dialed using the SIP Phone or the Web Configuration
Tool.
Note: Your SIP Phone will not remember the last call dialed if it loses power.
1. Get a dial tone (see Making Calls on page 14).
2. Press the Redial key. The last number/name/address you called (or tried calling) is
dialed.
Note: Pressing the Redial key without lifting the handset will automatically put you in
handsfree (speaker) mode.
Ringer Pitch and Volume
You can change the ringer volume using the SIP Phone Menu Interface, or by pressing the
▼ or ▲ key on the SIP Phone while the phone is ringing (one key press per level). You can
also change the pitch of the ringer by using the SIP Phone Menu Interface.
Note: The ringer settings are specific to the SIP Phone, not to user profiles. You cannot
change your ringer settings while on a call.
1. Press the Menu key.
2. Press the >> softkey until “PHONE SETTINGS?” appears, and press the OK softkey.
3. Press until “2.RINGER SOUNDS?” appears, and press the OK softkey.
4. “SET RINGER VOLUME?” is displayed.
• If you don’t want to change the ringer volume, press the No softkey and go to step 7.
• If you want to change the ringer volume, press the Yes softkey and continue below.
5. The phone starts ringing. Use the ▼ or ▲ key to adjust the volume to the desired level
(one key press per level), and press the Submit softkey.
6. “SET RINGER VOLUME?” is displayed. Press the No softkey.
7. “SET RINGER PITCH?” is displayed.
• If you don’t want to change the ringer pitch, press the No softkey to return to the
Phone Settings menu, or the Menu key to return to the default display.
• If you want to change the ringer pitch, press the Yes softkey and continue below.
8. The phone starts ringing. Use the ▼ or ▲ key to adjust the pitch to the desired level
(one key press per level), and press the Submit softkey.
9. “SET RINGER PITCH?” is displayed. Press the No softkey to return to the Phone
Settings menu, or the Menu key to return to the default display.
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5055 SIP Phone User Guide (Rev. A)
Speaker Volume (Handset, Headset and Handsfree)
To change the volume of the handset, headset or handsfree speaker volume:
Note: The time and date are specific to the SIP Phone, not to user profiles.
1. Get a dial tone (see Making Calls on page 14).
2. Press the ▼ key to decrease the volume, or the ▲ key to increase the volume (one key
press per level).
3. Put the SIP Phone on-hook (replace handset on cradle, press the Headset key, or press
the Speaker key).
The new setting will stay in effect until you change it again (if the SIP Phone loses power, the
settings will return to the factory default settings).
Speed Dial
You can program a Personal Key with Speed Dial, so you call someone with one key press.
Programming a Speed Dial Key
1. Press the Menu key.
2. Press the >> softkey until “PROGRAM MEMORY KEYS?” appears, and press the OK
softkey.
3. Press the Personal key you want to program. The key’s light turns red.
• If the key is not yet programmed, the display reads “UNUSED KEY”.
• If the key is already programmed, its associated feature is displayed. You must
delete a key’s programming before you can apply a new one (press the Delete
softkey and follow the prompts).
4. Press the AddNew softkey.
5. “ENTER NUMBER” is displayed.
• If you want to enter a name address, press the Name softkey.
• If you want to enter a URL address, press the URL softkey.
6. Enter the number (or name, or URL address), and press the Save softkey.
7. “KEY SAVED” is displayed. Press the OK softkey.
8. “PROGRAM MEMORY KEYS?” is displayed. Press the OK softkey to program more
function keys, or the Menu key to return to the default display.
9. To add a label beside a Personal key you programmed:
• Lift the plastic protector using the tab at the bottom.
• Write the information on the card below the plastic protector.
• Put the card and plastic protector back on the phone (insert top first).
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5055 SIP Phone Features
Editing a Speed Dial Key
1. Press the Menu key.
2. Press the >> softkey until “PROGRAM MEMORY KEYS?” appears, and press the OK
softkey.
3. Press the Personal key you want to edit. The key’s light turns red, and its associated
feature is displayed.
4. Press the Edit softkey.
5. The current number, name or URL is displayed. Press the <−−− softkey to delete the
characters, starting from the rightmost character, and type in the new number, name or
address.
6. Press the Save softkey.
7. “KEY SAVED” is displayed. Press the OK softkey.
8. “PROGRAM MEMORY KEYS?” is displayed. Press the OK softkey to program more
function keys, or the Menu key to return to the default display.
Making Calls Using Speed Dial
To make a call using a personal key programmed with Speed Dial:
1. Get a dial tone (see Making Calls on page 14).
2. Press the Personal key programmed with the desired Speed Dial number/name/address.
The key’s number/name/address is dialed.
Time and Date
You can change the date and time using the SIP Phone Menu Interface, or the Web
Configuration Tool. Usually, your SIP Phone gets its time and date from an SNTP server (see
Modifying the Network Configuration in the section), and all you need to do is adjust your time
zone twice a year if your area uses Daylight Savings Time. If you don’t have an SNTP server,
you will need to set the time and date manually.
Note: The time and date are specific to the SIP Phone, not to user profiles. You cannot
change your time and date settings while on a call.
Adjusting your Time Zone
SNTP servers usually provide Greenwich Mean Time data. To adjust the time and date for
your area, you need to specify your time zone (if your area uses daylight Savings Time, you
will need to adjust this twice a year):
1. Access the Web Configuration Tool (see The Web Configuration Tool on page 7).
2. Click Network Configuration.
3. In the “Additional Servers” section, beside “Time Zone:”, enter the difference between
your time zone and the GMT, adjusting for Daylight Savings Time as needed (see Time
Zones on page 58 for a table of world time zones versus GMT).
4. Click the Save an Reboot button. A confirmation screen is displayed.
5. Click the OK button. Your SIP Phone is updated, and then reboots.
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Changing the Time and Date
Use this procedure only if no SNTP server is provided. You will need to reprogram these
settings every time the phone reboots.
With the Web Configuration Tool:
1. Access the Web Configuration Tool (see The Web Configuration Tool on page 7).
2. Click Feature Configuration.
3. Enter your date and time. Use the day-month-year format for the date and the 24-hour
clock format for the time.
4. Click the Apply button. A confirmation screen is displayed.
5. Click the OK button. Your SIP Phone is updated.
With the SIP Phone Menu Interface:
1. Press the Menu key.
2. Press the >> softkey until “PHONE SETTINGS?” appears.
3. Press the OK softkey. “1.TIME/DATE” is displayed.
4. Press the OK softkey. “SET TIME?” is displayed, with the currently programmed time.
• If you don’t need to change the time, press the No softkey and go to step 9 to change
the date.
• If you need to change the time, press the Yes softkey and continue below.
5. “12 or 24 HR FORMAT?” is displayed. Press the 12 softkey if you want to enter the time
in am/pm, or the 24 softkey to enter the time using the 24-hour clock format.
6. Enter the time (for example, 1236 for 12:36, or 220 for 2:20), and press the Submit
softkey.
7. If you are entering the time using am/pm, press the AM or the PM softkey.
8. “SET TIME?” is displayed, with the new time. Press the No softkey to set the date.
9. “SET DATE?” is displayed.
• If you don’t need to change the date, press the No softkey to return to the Phone
Settings menu, or the Menu key to return to the default display.
• If you need to change the date, press the Yes softkey and continue below.
10. Enter the date (for example, enter 161202 for 16 December 2002), and press the Submit
softkey.
11. “SET DATE?” is displayed. Press the No softkey to return to the Phone Settings menu, or
the Menu key to return to the default display.
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5055 SIP Phone Features
Web Dialing
You can make calls using the Web Configuration Tool. You can dial by URL, or using the
Phone Book (see Making Calls With the Phone Book on page 26 for information on the
latter).
To dial by URL:
1. Access the Web Configuration Tool (see The Web Configuration Tool on page 7).
2. Click Dial by URL.
3. Enter the URL of the party you want to reach.
4. To make a call:
• Handset mode: lift the handset, and click the Dial button on the Web Configuration
Tool.
• Headset mode: press the Headset key, and click the Dial button on the Web
Configuration Tool.
• Handsfree mode: click the Dial button on the Web Configuration Tool.
The call is made on the first available line (Line 1 if all lines are free). To use a specific
Line key, press the desired Line key before clicking the Dial button on the web page.
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5055 SIP Phone User Guide (Rev. A)
Administrator Tools
This section contains the following information on configuring the administrative settings of
the 5055 SIP Phone:
• Changing Passwords
• Setting Up User Profiles
• Creating/Modifying a SIP Account
• Modifying the Network Configuration
• Upgrading the Firmware of the SIP Phone
• Additional Settings
These settings are changed using the Web Configuration Tool or the SIP Phone Menu
Interface.
Note: You cannot change these settings while on a call.
Changing Passwords
The Administrator can change the password for all defined users using the Web
Configuration Tool or the SIP Phone Menu Interface.
Note: The Administrator password should be changed as soon as possible to prevent
unauthorized access to the Administrator functions of the SIP Phone.
Using the Web Configuration Tool
1. Access the Web Configuration Tool using the Administrator user name and password
(see The Web Configuration Tool on page 7).
2. Click Security Config.
3. Change the passwords as required (note: you cannot change user names using this
screen).
4. Click the Apply button. A confirmation screen is displayed.
5. Click the OK button. The SIP Phone is updated.
Using the SIP Phone Menu Interface
You must repeat this procedure for each password that needs to be changed.
1. Press the Menu key.
2. Press the >> softkey. “USERS?” is displayed.
3. Press the OK softkey.
4. Press the softkey until “4.CHANGE PASSWORD?” is displayed, and press the OK
softkey.
5. Enter the Administrator user name, or the user name for the user profile whose password
you are changing, and press the Submit softkey.
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6. Enter the current password, and press the Submit softkey (if the existing password is
blank, enter any character, and delete it using the <−−− softkey before pressing the
Submit softkey).
7. Enter the new password, and press the Submit softkey.
8. Enter the new password again, and press the Submit softkey.
• If you have entered both instances of the new password correctly, “NEW
PASSWORD CREATED” is displayed. Press the OK softkey.
• If you have made a mistake, “PASSWORD MISMATCH” is displayed. Press Retry to
go back to step 7.
9. Press the Menu key to return to the default menu.
Setting Up User Profiles
Your SIP Phone can have two personalized user profiles, in addition to the default profile.
Each profile stores information about the associated user as well as personalized
configurations. Users access their profile with a user name and password.
Viewing a Profile’s User Name
To view the active user profile’s user name:
1. Press the Menu key.
2. Press the Line 3 key. The display shows the user profile display name (top) and user
name (bottom).
3. Press the Menu key to return to the default display.
Creating a User Profile
1. Press the Menu key.
2. Press the >> softkey. “USER?” is displayed.
3. Press the OK softkey.
4. Press the softkey until “5.MANAGE PROFILES?” is displayed.
5. Press the OK softkey.
6. Enter the Administrator user name, and press the Submit softkey.
7. Enter the Administrator password, and press the Submit softkey.
8. Press the ▼ key until your reach a vacant user profile (profile 1 is the default user
profile).
9. Press the AddNew softkey.
10. Enter a user name for this new user profile, and press the Submit softkey.
11. Enter a password for this new user profile, and press the Submit softkey (if the password
is blank, enter any character, and delete it using the <−−− softkey before pressing the
Submit softkey).
12. Enter the password again, and press the Submit softkey.
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5055 SIP Phone User Guide (Rev. A)
13. Enter the name that will appear on the phone display when the new user profile is active,
and press the Submit softkey.
14. Enter the user’s SIP Server Authentication name, and press the Submit softkey.
15. “NEW PROFILE CREATED” is displayed. Press the OK softkey.
16. To create another user profile, press Exit, and repeat this procedure from Step 4. To exit
the procedure and return to the default display, press the Menu key.
Modifying a User Profile
The following user profile information can be added/modified using the Web Configuration
Tool:
• User ID/Extension
• SIP Authentication User Name
• SIP Authentication Password
• Public (PSTN) Phone Number
• E-Mail Address
Note: The user profile to be modified must be logged in and active.
To enter/change that information:
1. Access the Web Configuration Tool (see The Web Configuration Tool on page 7).
2. Click User Configuration.
3. Enter/change the information as needed.
4. Click the Save and Reboot button. A confirmation screen is displayed.
5. Click the OK button. The SIP Phone is updated and reboots.
Deleting a User Profile
1. Press the Menu key.
2. Press the >> softkey. “USER?” is displayed.
3. Press the OK softkey.
4. Press the softkey until “5.MANAGE PROFILES?” is displayed.
5. Press the OK softkey.
6. Enter the Administrator user name, and press the Submit softkey.
7. Enter the Administrator password, and press the Submit softkey.
8. Press the ▼ key until your reach the user profile you want to delete (profile 1 is the
default user profile, and cannot be deleted).
9. Press the Remove softkey.
10. “REMOVE USER PROFILE?” is displayed. Press the Confirm softkey to delete the user
profile.
11. Press AddNew to create a new user profile (see Creating a User Profile above), the▼ or
▲ key to go to another user profile, or the Menu key to return to the default display.
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Creating/Modifying a SIP Account
You can modify the SIP account information of the SIP Phone using the Web Configuration
Tool.
1. Access the Web Configuration Tool.
2. Click SIP Configuration.
3. Enter/change the SIP Account information as needed (see Table 11 on page 54 for more
information on these settings).
4. Click the Save and Reboot button. A confirmation screen is displayed.
5. Click the OK button. The SIP Phone is updated and reboots.
6. For each user profile, enter/change the associated SIP Authentication user name and
password (see Modifying a User Profile on page 34 for more information).
Modifying the Network Configuration
Viewing the IP and MAC Addresses
To view the Internet Protocol (IP) address and the Media Access Control (MAC) address of
the SIP Phone:
1. Press the Menu key.
2. Press the Line 1 key. The IP and MAC addresses are displayed.
3. Press the Menu key to return to the default display.
Modifying Network Configurations
You can modify the following network configuration settings of the SIP Phone using the Web
Configuration Tool or the SIP Phone Menu Interface (see Table 12 on page 57 for more
information on these settings).
• Basic Settings:
− SIP Phone host and domain names (web tool only).
− DHCP status.
− Address type.
− SIP Phone IP address and subnet mask (supplied automatically by ISP/LAN if DHCP
is on).
− Default gateway (supplied automatically by ISP/LAN if DHCP is on).
− Primary and secondary DNS addresses (supplied automatically by ISP/LAN if DHCP
is on).
• Additional Servers Settings:
− TFTP server.
− SNTP server and time zone of SIP Phone.
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5055 SIP Phone User Guide (Rev. A)
• Advanced Settings:
− TFTP configuration (allows configuration of SIP Phones using configuration files; see
Configuration Files on page 38 for more information).
− Type of Service and 802.1 Priority (Quality of Service parameters).
− Virtual LAN ID.
− PPPoE status and PPPoE login name and password.
Using the Web Configuration Tool
1. Access the Web Configuration Tool (see The Web Configuration Tool on page 7).
2. Click Network Configuration.
3. Add/update the information as needed.
4. Click the Save and Reboot button. A confirmation screen is displayed.
5. Click the OK button. The SIP Phone is updated and reboots.
Using the SIP Phone Menu Interface
1. Press the Menu key.
2. Press the >> softkey until “PHONE SETTINGS?” is displayed, and press the OK softkey.
3. Press the softkey until “4.NETWORK SETTINGS?” is displayed.
4. Press the OK softkey.
5. “DHCP” is displayed with its current status (“*ON*” or “*OFF*”).
6. If needed, press the TurnOff softkey to disable DHCP, or the TurnOn softkey to enable
it.
7. Press the Next softkey until the next parameter you want to change is displayed.
8. Press the Review softkey to view its current setting.
• If you need to change the value, press the Change softkey, enter the new value,
then press the Submit softkey (when the entry can only be an IP address, pressing *
twice, rapidly - enters a period).
• To leave the value as it is, press the Exit softkey.
9. Repeat steps 7 and 8 until all the desired changes have been made.
10. Press the Exit softkey, then the Menu key to return to the default display.
11. For the settings to take effect, you must restart your SIP Phone:
• When you are back in the default display, press the Menu key.
• Press *, then 0 on the keypad. The SIP Phone restarts.
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Upgrading the Firmware of the SIP Phone
The phone uses TFTP to download firmware upgrades from a TFTP server. There are two
methods that can be used to do this: one uses the SIP phone’s softkey menu system to
perform the upgrade, the other uses the upgrade button on the main phone configuration
web-page. The methods function differently:
• With web-page upgrade, the phone’s original configuration is preserved, so that it will
function as it did prior to upgrade, without the need to reconfigure settings.
• With soft-key menu upgrade, the phone’s previous configuration may not be saved and
restored – which would necessitate manual re-load of a previously saved configuration
file or manual re-configuration of phone parameters (refer to Configuration
Upload/Download Page).
Viewing the Firmware Version
Using the Web Configuration Tool
1. Access the Web Configuration Tool (see The Web Configuration Tool on page 7). If the
Web Configuration Tool is currently opened, click Home.
2. The main and boot firmware versions are displayed near the top of the home page.
Using the SIP Phone Menu Interface
1. Press the Menu key.
2. Press the Line 2 key. The main and boot firmware versions are displayed.
3. Press the Menu key to return to the default display.
Upgrading the firmware using the web-page Interface
You upgrade the SIP Phone by downloading the new firmware from the appropriate TFTP
server (the TFTP server is programmed with the Network Configuration).
CAUTION: During this procedure, DO NOT remove power from the SIP Phone while
firmware is downloading or the phone is rebooting. This may result in severe damage
to your SIP Phone.
1. Log into the phone configuration web page.
2. Click the Upgrade button.
3. The Firmware Upgrade page is displayed. If the TFTP server URL is OK as is, click either
the Upgrade or the Upgrade & Set Factory Defaults button. If you chose to reset to
factory defaults, you may have to reset the phone’s settings manually, or if a recent copy
of the phone configuration file exists, you may be able to restore all but the user
passwords in one easy, step. (refer to Configuration Upload/Download Page).
4. If you selected Upgrade & Set Factory Defaults in step 3, you will have to re-enter the
SIP Authorization password, and possibly the PPPoE and User Profile passwords using
the softkey menus or the web page, as these parameter are lost.
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5055 SIP Phone User Guide (Rev. A)
Upgrading the firmware using the SIP Phone Menu Interface
You upgrade the SIP Phone by downloading the new firmware from the appropriate TFTP
server (the TFTP server is programmed with the Network Configuration).
CAUTION: During this procedure, DO NOT remove power from the SIP Phone while
firmware is downloading or the phone is rebooting. This may result in severe damage
to your SIP Phone.
1. To restart the phone, press the Menu, ∗ and 0 keys in sequence.
2. When the display shows “Booting…”, press and hold the 2 key on the keypad. This
upgrades the boot firmware of the SIP Phone.
3. When “UPGRADE FIRMWARE?” appears, release the key and press the YES softkey.
4. The phone now gives you the option of using the displayed TFTP IP address (this IP
address is set in the Network Configuration Page), or entering one of your own.
5. Press the ▼ key. The firmware starts downloading.
6. When the new firmware has finished downloading, the SIP Phone reboots. This process
may take a minute or two. It’s complete when the display shows a time and date on the
top line of the display (default display). In some cases, upgrade of the boot firmware will
automatically trigger upgrade of the Main firmware. If you need to force the upgrade of
the Main firmware, press the Menu, ∗ and 0 keys in sequence, then hold the 1 key (the
phone then requires you to select the TFTP IP address, and press the
process may take a minute or two, and is complete when the display shows a time and
date on the top line of the display (default display).
▼ key). This
7. In some cases it may be necessary to reset the settings of the SIP Phone to factory
defaults (note: this procedure will erase all your settings then replace them by the factory
defaults). To do this
• Press the Menu key.
• Press * on the keypad.
• Press # on the keypad.
• Press and hold the 3 key on the keypad until “USE FACTORY DEFAULTS?” is
displayed.
• Press the YES softkey. The settings are reset, and the phone reboots.
When the time and date are displayed, you must reprogram the SIP Phone or reload a
saved configuration file (see Changing Passwords, Setting Up User Profiles,
Creating/Modifying a SIP Account and Modifying the Network Configuration in this
section). Saved configuration files do not restore passwords, so you will have to re-enter
your passwords manually.
Additional Settings
Configuration Files
The 5055 SIP Phone supports configuration files for automatic programming of the phones.
There are two types of configuration files:
• Generic: a generic configuration file applies the settings defined in it to all the SIP
Phones.
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•
Specific: a specific configuration file applies the settings defined in it to a specific SIP
Phone.
Configuration files are stored on the TFTP server, and are downloaded by the SIP Phones
connecting to that server every time the phones reboot. The generic configuration file is
loaded first, then the specific configuration file. If both files contain settings for the same
parameter, the specific configuration file will overwrite the information from the generic
configuration file.
Before a SIP Phone can automatically download configuration files from a TFTP server, it
must have the following Network Configuration settings configured via the web configuration
tool:
• TFTP Server address (in Additional Servers section)
• TFTP Configuration = Yes (in Advanced section)
When a SIP Phone with these settings reboots, it starts by requesting the generic
configuration file from the TFTP Server. If the file exists, it is downloaded and all of the
parameters in it overwrite the existing settings for these parameters on the SIP Phone. Then,
the SIP Phone requests its specific configuration file from the TFTP Server. If the file exists, it
is downloaded and all of the parameters in it overwrite the existing settings for these
parameters on the SIP Phone (including those from the specific configuration file, if
applicable). Only the parameters defined in the configuration files are overwritten on the SIP
Phone. If the SIP Phone requests a configuration file that is not on the TFTP server, no
settings are changed on the phone.
Note: When a SIP Phone uses configuration files, you can still change settings manually,
but if these settings are also defined in the configuration files, the configuration file will
overwrite the manual settings the next time the SIP Phone reboots.
Generic Configuration File (SIPGeneric.cfg)
Used to change global settings such as Media Configuration, Voice Mail server, etc. The
generic configuration file is a text file saved as “SIPGeneric.cfg” on the TFTP server. You can
create a generic configuration file by using a text application such as Notepad or SimpleText,
or by using your favorite word processing application and saving the file as a text file.
Example of a Generic Configuration File in Appendix C shows all the possible settings you
can have in a generic configuration file.
Specific Configuration File (SXXXXXXXXXXXX.cfg)
Used to change phone-specific settings such as user profiles, Hot Line configuration, etc.
Each specific configuration file (one per SIP Phone) is a text file saved as
“SXXXXXXXXXXXX.cfg” on the TFTP server, where the Xs are the 12-character
hexadecimal MAC address of the SIP Phone. You can create a generic configuration file by
using a text application such as Notepad or SimpleText, or by using your favorite word
processing application and saving the file as a text file.
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5055 SIP Phone User Guide (Rev. A)
Example of a Specific Configuration File in Appendix C shows all the possible settings you
can have in a specific configuration file.
Hot Line
When a hot line number/address is set up, the SIP phone automatically dials that
number/address when it goes off-hook (handset lifted, etc.).
The Hot Line number is programmed using the Web Configuration Tool:
1. Access the Web Configuration Tool (see The Web Configuration Tool on page 7).
2. Click Feature Configuration.
3. To activate the Hot Line, select On in the drop down menu beside “Hot Line Mode”.
4. You can enter a number or URL address for the Hot Line.
• To enter a number, select NUM_MODE in the drop down menu beside “Address
Type:”.
• To enter a URL, select URL_MODE in the drop down menu beside “Address Type:”.
5. Enter the number/URL of the Hot Line beside “Destination Address:”.
6. Click the Apply button. A confirmation screen is displayed.
7. Click the OK button. The SIP phone is updated.
The programming steps listed above let the caller over-ride the pre-programmed hot line
number by dialing their own. If you need to program the phone to block all other outgoing
calls, other than those to the Hot line number, you need to add this rule to the Dialing Plan:
Dialed Digits Digits to follow Digits to Remove Prefix to Add Suffix/Route Comments
xx 2
This rule recognizes, and blocks all outgoing manual dial attempts, permitting only hot line
dialing. You could create a plan to allow “911” calls, but block all other outgoing calls:
Dialed Digits Digits to follow Digits to Remove Prefix to Add Suffix/Route Comments
911 00
xxx 0 3
Media Configuration
You can change the following Media Configuration settings using the Web Configuration
Tool:
• Audio codec type and frame size.
• DTMF type and payload type.
To change media configuration settings:
1. Access the Web Configuration Tool (see The Web Configuration Tool on page 7).
2. Click Media Configuration.
3. Change the information as needed. See Table 19 on page 67 for more information on
these settings.
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Note: If you set the audio codec type to G.729, the users will not be able to use the
Conference Call feature.
4. Click the Apply button. A confirmation screen is displayed.
5. Click the OK button. The SIP Phone is updated.
Resetting to Factory Defaults
If needed, you can erase all the settings of a SIP Phone using the SIP Phone Interface:
1. Press the Menu key.
2. Press the * key on the keypad.
3. Press the # key on the keypad.
4. Press and hold the 3 key on the keypad until “USE FACTORY DEFAULTS?” is displayed.
5. Press the YES softkey. The settings are reset, and the phone reboots.
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Appendix A — SIP Phone Interface
The SIP Phone Menu Interface is used to view or program a number of the SIP Phone and
user profile settings.
SIP Phone Menu Interface
The SIP Phone Menu Interface is accessed by pressing the Menu key. To navigate between
items of the main menu, press the >> and << softkeys. To navigate between the items of the
sub-menus, use the and softkeys.
Table 3 SIP Phone Menu Interface Settings
Main Menu Sub-Menu Notes
USERS?
CALLING LISTS?
FEATURE
CONFIG?
PROGRAM
MEMORY KEYS?
PHONE
SETTINGS?
1.LOGIN? To log in a user profile.
2.LOGOUT? To log out a user profile.
3.ACTIVATE PROFILE? To activate a user profile.
4.CHANGE PASSWORD? To change a password.
5.MANAGE PROFILES? To add/delete a user profile.
6.REGISTRATION?
Enter UserID
Enter Password
Contact IP Address
Server Address
To Address
Registration Method
Reg’n Expiry (HR)
1.PHONE BOOK?
2.MISSED CALLS? Log of missed calls. You can make calls
3.ANSWERED CALLS? Log of answered calls. You can make
4.OUTGOING CALLS?
1.CALL FORWARDING?
Always, No Answer, Busy
2. DO NOT DISTURB? To enable/disable Do Not Disturb
Speed Dial
1. TIME/DATE? To change the time and date.
2. RINGER SOUNDS?
Volume, Pitch
3. LCD CONTRAST To change the LCD contrast (immediate
Temporarily registers with a SIP Service
Provider.
SIP Authorization. User Name
SIP Authorization. Password
Typically: user_name@server
SIP registration server IP Address
None, Basic or Digest
To make calls using phone book entries
(as programmed with the Web
Configuration Tool).
from this log.
calls from this log.
Log of outgoing calls made from your
5055 SIP Phone. You can make calls
from this log.
To program/enable/disable call forward.
feature.
To program Personal keys.
To change the ringer pitch and volume.
save).
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Table 3 SIP Phone Menu Interface Settings (continued)
Main Menu Sub-Menu Notes
PHONE
SETTINGS?
(con’t)
4. DEVICE
PARAMETERS?
Software Version
MAC Address
5. PROTCOL CONFIG?
HTTP
TFTP
TELNET
6. MULTI USER CONFIG? Turns the MultiUser profiles On or Off. User
7. LANGUAGE?
en_CA
fr_CA
fr_FR
en_US
en_GB
en_AU
es_MX
es_US
8. RING TONES?
CA
US
GB
DE
NL
AU
NZ
MX
FR
9. NETWORK SETTINGS?
DHCP
Phone IP Address
Phone Subnet Mask
Default Gateway
Outbound Server, Port
SIP Proxy Server, Port
SIP Proxy Port Num
Voice Mail Server, Port
Primary/Secondary DNS
Servers
TFTP, SNTP Servers
Eth.Autoneg
To view the software version and MAC
address of the SIP Phone.
To enable or disable protocol settings.
profiles work best with stand-alone installations
(not directly connected to a 3050 ICP). Turn
this feature on, only if your SIP phone is standalone, and you need this feature.
Changes the phone display language
English Canadian
French Canadian
French France
English USA
English Great Britain
English Australian
Spanish Mexican
Spanish USA
Changes the phone tone plan.
Canada
USA
Great Britain
Germany
Netherlands
Australia
New Zealand
Mexico
France
To change Network settings (see Table 12 on
page 57 for more information on these
settings).
Enables/disables Ethernet auto-negotiation
Appendix A — SIP Phone Interface
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5055 SIP Phone User Guide (Rev. A)
Menu Key Commands
There are a few other settings you can access using the SIP Phone’s Menu key.
Menu + Line Key Commands
1. These commands give you access to information about your SIP Phone. To access
them, press the Menu key, then press the appropriate Line key. Press the Menu key to
return to the default display.
Table 4 Menu + Line Key Commands
Display Menu + Line 1 Menu + Line 2 Menu + Line 3
Top Line SIP Phone IP address
Bottom
Line
SIP Phone MAC
address
Main software
version
Boot software
version
Display name
User name
Menu + Keypad Commands
These commands perform actions on the SIP Phone. To access them, press the Menu key,
then each of the keypad keys in succession.
• To restart the SIP Phone: Menu, *, 0.
• To reset the SIP Phone to factory defaults: Menu, *, #, hold 3.
Caution: Resetting the SIP Phone to factory defaults will erase all the programming on
the SIP Phone, and replace it by the factory default settings.
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Appendix B — Web Configuration Tool
Appendix B — Web Configuration
Tool
This appendix provides details on the settings and features available through the Web
Configuration Tool.
Accessing the Web Configuration Tool
The Web Configuration Tool is accessed from any computer using a web browser. See The
Web Configuration Tool on page 7 for instructions on accessing the Web Configuration Tool.
Home Page
Figure 6 Web Configuration Tool: Home Page
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5055 SIP Phone User Guide (Rev. A)
The Home Page shows the software version installed on the phone.
The Upgrade button displays the Firmware Update page.
User Configuration Page
The User Configuration page lets you change your user profile’s basic parameters. After
making the changes, click the Save and Reboot button; this will reboot your phone.
Figure 7 Web Configuration Tool: User Configuration Page
Table 5 Web Configuration Tool: User Configuration Settings
Setting Name Values (bold = default) Notes
Basic
Unique name or number assigned to you.
User ID or Extension <text>
User Display name <text>
SIP Authentication
User Name
<text>
Limit of 32 characters. Default is user for
default user profile.
The name displayed on your phone when
your user profile is active. Limit of 20
characters. Default is display username for
default user profile.
Your SIP Account name provided by your
SIP Service Provider. Required only if you
have a SIP Service Provider. Default is
user for default user profile.
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Appendix B — Web Configuration Tool
Setting Name Values (bold = default) Notes
Your SIP account password provided by
your SIP Service Provider. If you do not
Password <text>
have a SIP Service Provider, password
used to log in and activate your user profile.
Public Phone Number
(PSTN)
Email Address <text>
MultiUser Profile
Language Code
<numbers>
off
en_CA
fr_CA
fr_FR
en_US
en_GB
en_AU
es_MX
es_US
Public phone number used as a possible
alternative contact. Default is blank.
E-mail address used as a possible
alternative contact. Default is blank.
MultiUser profiles work best with SIP
phones that are being used in stand-alone
installations (not directly connected to a
3050 ICP). Turn this feature on, only if your
SIP phone is stand-alone, and you need
this feature.
Changes the phone display language
English Canadian
French Canadian
French France
English USA
English Great Britain
English Australian
Spanish Mexican
Spanish USA
Feature Configuration Page
The Feature Configuration page lets you program a number of settings attached to your user
profile. After making the changes, click the Apply button.
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5055 SIP Phone User Guide (Rev. A)
Figure 8 Web Configuration Tool: Feature Configuration Page
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Appendix B — Web Configuration Tool
Table 6 Web Configuration Tool: Feature Configuration Settings
Setting Name Values (bold = default) Notes
Features
Call Forwarding Always
Forwarding Address SIP address.
Call Forwarding No
Answer
Number of Rings < number>
Forwarding Address SIP address.
Call Forwarding When
Busy
Forwarding Address SIP address.
Do Not Disturb
Advisory Message
Message
Other Reason <message>
Hold
Beep on Hold
Held calls will ring
back after:
Hot Line Configuration
Hot Line Mode
Address Type
Destination Address <address>
Date/Time
Date
Time <time in 24-hour format>
On | Off
On | Off
On | Off
On | Off
On | Off
In a Meeting | Out of town,
At lunch | On vacation | In a
Conference | Back in 5
minutes | Gone Home | Off
Sick | Other reason
On | Off
<delay in seconds>
On | Off
Num_Mode, URL_Mode
<date with format daymonth-year>
Default is blank. If left blank calls will be
forwarded to voice mailbox as programmed
in SIP configuration page.
Default value is 10.
Default is blank. If left blank calls will be
forwarded to voice mailbox as programmed
in SIP configuration page.
Default is blank. If left blank calls will be
forwarded to voice mailbox as programmed
in SIP configuration page.
To enter a personalized message, select
“Other reason”.
Fill in if “Other reason” is selected above.
Limit of 20 characters.
Heard by user when on hold.
Heard by user to remind a call is on hold.
Default value is 60 seconds
When On, dials the Destination Address
automatically when the SIP Phone goes
off-hook.
Num_Mode: number address.
URL_Mode: URL address.
Must correspond to the type chosen in
“Address Type” above. Default is
operator@example.com.
Modify only if there is no SNTP server (see
Network Configuration page). These
settings will be lost if the SIP Phone
reboots.
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5055 SIP Phone User Guide (Rev. A)
Phone Book Page
The Phone Book page lets you define up to five contacts for your user profile’s Phone Book.
You can also dial any of the contacts from this page. Click the Apply button after making any
changes to your contacts. To dial a contact, select it in the drop down menu, and click the
Dial button.
Figure 9 Web Configuration Tool: Phone Book Page
Table 7 Web Configuration Tool: Phone Book Settings
Setting Name Values (bold = default) Notes
Contact n
Name
Address
Select Contact
<name or nickname of the
contact>
<number> | <name> |
sip:<URL>
Contact 1 | Contact 2 |
Contact 3 | Contact 4 |
Contact 5
Limit of 20 characters.
Limit of 128 characters.
Dial by URL Page
The Dial by URL page lets you dial a URL or IP address from the Web Configuration Tool. Click
the Dial button to dial the URL. To save the SIP URL so that you need not re-enter it in the future,
click the Apply button.
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Appendix B — Web Configuration Tool
Figure 10 Web Configuration Tool: Dial by URL Page
Table 8 Web Configuration Tool: Dial by URL Settings
Setting Name Values (bold = default) Notes
Enter SIP URL
<number> | <name> |
sip:<URL>
A URL must be preceded by “sip:”. Limit of
128 characters.
For examples of SIP URL syntax, refer to the table on the last page of this guide.
Key Programming Page
The Key Programming page lets you assign an address to one of the seven programmable
speed-call keys. An address can be a name, number or URL.
Figure 11 Web Configuration Tool: Key Programming Page
Table 9 Web Configuration Tool: Key Programming Settings
Setting Name Notes
Select a key Lets you select the key to be programmed (key 1, is the bottom-left key)
Action Mode
Address Type You must select the address type before entering it.
Address Enter the address here.
Display lists the key assignment. Update lets you change it.
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5055 SIP Phone User Guide (Rev. A)
Ring Tone Page
The Ring Tones configuration page lets you define up to three “rules” to control how the
phone treats calls originating from individual callers (specified by SIP URL) or from groups of
callers (specified by domain name). The phone can:
• Associate one of 12 different ring pitches to the call
• Automatically forward the call to voicemail
• Reject the call
Figure 12 Web Configuration Tool: Ring Tone Page
Table 10 Web Configuration Tool: Ring Tones Settings
Setting Name Values (bold = default) Notes
Key Word sip:<URL> | domain name
Ring Pitch
Forward to Voicemail
Block
0 - 12
Off/On
Off/On
A URL must be preceded by “sip:”. Domain
names Limit of 128 characters.
Selects the pitch that you will hear
Forwards the caller automatically to your
voicemail
Prevents the caller from reaching you.
SIP Configuration Page
The SIP Configuration page lets you change the SIP Service Provider configurations of your
SIP Phone. These settings are specific to the SIP Phone. After making the changes, click the
Save and Reboot button; this will reboot your phone.
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Appendix B — Web Configuration Tool
Figure 13 Web Configuration Tool: SIP Configuration Page
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5055 SIP Phone User Guide (Rev. A)
Table 11 Web Configuration Tool: SIP Configuration Settings
Setting Name Values (bold = default) Notes
Basic
SIP Proxy Server
Port <number>
SIP Registry Server
Port <number>
Authenticate Method
Registry Duration <duration in seconds>
Phone Listening Port <number>
Transport Protocol
Symmetric UDP Port
Additional Servers
Outbound Server
Outbound Server
URL
Outbound Server Port <blank>
Voice Mail Server
Number of rings
Port <number>
Backup Server Timeout
Emergency
Emergency Number <number>
Emergency Server IP <IP address>
Port <number>
<IP address> | <domain
name>
<IP address> | <domain
name>
None | Basic | Digest
UDP | TCP
symmetric Symmetric is recommended
On | Off
<blank>
<IP address> | <domain
name>
4
4
Appended to dialed name or number (for
example, name@proxy). Limit of 128
characters. Default is blank.
The SIP Proxy Server port number. Default
is 5060.
Used if SIP Proxy and Registry Servers are
not the same. Limit of 128 characters.
Default is blank.
The SIP Registry Server port number.
Default is 5060.
None: no registration authentication.
Basic: authentication without encryption.
Digest: authentication with encryption.
Time after which you are automatically
deregistered. Default value is 7200 seconds
(2 hours).
Receive port used by the SIP Phone for SIP
signaling. Default is 5060.
Default type of packets for transmitted SIP
signaling. UDP = User Datagram Protocol.
TCP = Transmission Control Protocol.
If on, all SIP request and responses are
sent to the outbound server. Default is
192.168.0.1.
Server address of external voice mail
server. Default is blank. If this field is
configured, the phone will connect to server
using the default user name during boot-up.
Port number of Voice Mail Server. Default
value is 5060.
Some ISPs offer backup servers that can be
used when the primary server is
unavailable. This field lets you configure
how long the 5055 SIP phone will wait
before it tries the backup server.
Emergency number for area, if applicable
(for example, most of North America uses
911 as an emergency number). Default is
blank.
Address of server used by emergency
number dialed. Default is 0.0.0.0.
Port number of Emergency Server. Default
is 5060.
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Appendix B — Web Configuration Tool
Setting Name Values (bold = default) Notes
Firewall
This enables or disables firewall NAT
Bypass Firewall NAT
Mode
WAN IP Discovery URL <URL> WAN IP Discovery address
WAN IP Address <IP address> Static WAN IP Address
On | Off
static | dynamic
bypass operation. When enabled, this
features lets the 5055 SIP phone function
behind a firewall which is not SIP-aware.
Determines how the 5055 SIP phone will
obtain the IP address of the firewall. A static
address is
Network Configuration Page
The Network Configuration page lets you change the network configurations of your SIP
Phone. These settings are specific to the SIP Phone. After making the changes, click the
Save and Reboot button; this will reboot your phone.
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5055 SIP Phone User Guide (Rev. A)
Figure 14 Web Configuration Page: Network Configuration Page
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Appendix B — Web Configuration Tool
Table 12 Web Configuration Tool: Network Configuration Settings
Setting Name Values (bold = default) Notes
Basic
SIP Phone Host Name <host name>
Domain Name <domain name>
DHCP
Address Type
On | Off
IPv4 | Fqdn
SIP Phone IP Address <IP address>
Subnet Mask <IP address>
Default Gateway <IP address>
Primary DNS <IP address>
Secondary DNS <IP address>
Additional Servers
TFTP Server
sipdnld.mitel.com
Required for cable access. Default is sip1.
Optional. Limit of 128 characters. Default is
–example.com.
If On, allows your ISP or LAN to allocate
you a dynamic IP address.
IPv4: outgoing SIP requests use dotted
format of IP address.
Fqdn: outgoing SIP requests use
“sip:host_name.domain” format for “contact”
SIP header.
Required. Provided automatically by ISP or
LAN when DHCP is On
Required. Provided automatically by ISP or
LAN when DHCP is On
255.255.255.0.
Required. Provided automatically by ISP or
LAN when DHCP is On
Required. Provided automatically by ISP or
LAN when DHCP is On
Optional. Provided automatically by ISP or
LAN when DHCP is On
Optional. The server where updates to the
firmware and languages can be
downloaded using the TFTP protocol.
1
.
1
. Default is
1
.
1
.
1
.
Optional. The address of the HTTP
server where updates to the firmware
and languages can be downloaded.
HTTP is an alternative protocol to TFTP.
HTTP download URL <URL>
If you leave this filed blank, the phone
will attempt to use TFTP to update
software. If HTTP update fails, then the
phone does not automatically “fall back”
to try TFTP.
SNTP Server <IP address>
Time Zone <number>
Optional. Server used for date/time
synchronization. Default is 192.53.103.103.
Optional. Difference between GMT and
local time. Default is –5 (Eastern Standard
Time)
1
If you change this value manually with DHCP On, it will be overwritten by the ISP/LAN the next
time the phone is rebooted.
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5055 SIP Phone User Guide (Rev. A)
Setting Name Values (bold = default) Notes
Advanced
Changes the phone tone plan.
CA
US
GB
Tone Code
TFTP Configuration
ToS value
802.1 Priority
VLAN ID <0–4095>
PPPoE On | Off
PPPoE Login <user/login name> Required for DSL. Provided by DSL ISP.
PPPoE Password <password> Required for DSL. Provided by DSL ISP.
DE
NL
AU
NZ
MX
FR
Yes | No
0 - 1e (even numbers only)
Off | 0 | 1 | 2 | 3 | 4 | 5 | 6 | 7
Canada
USA
Great Britain
Germany
Netherlands
Australia
New Zealand
Mexico
France
If set to yes, the SIP Phone looks for
configuration files on the TFTP server when
it boots up. The settings in the configuration
files will overwrite any manually entered
settings. Default is 192.168.0.1.
Type of Service. QOS parameter used to
define packet priority for the IP layer. You
can select values from the pull-down menu
in hexadecimal notation.
QOS parameter used to define packet
priority for Ethernet layer.
Virtual LAN Id. Used by network
administrators. Default is 1.
Point-to-Point Protocol over Ethernet.
Enabled when using a DSL network.
Time Zones
The table below is provided for your convenience. Mitel Network does not guarantee its
accuracy. For Daylight Savings Time, add +1 (for example, if your standard time is –5, your
daylight savings time is –4).
Table 13 Major Time Zones
Time Zone Diff. from GMT
Fiji, New Zealand, Marshall Islands (US)
Samoa, Midway Islands -11
Cook Island, Hawaiian Standard Time (US) -10
Alaska Standard Time (US) -9
Pacific Standard Time (US/Canada) -8
Mountain Standard Time (US/Canada) -7
Central Standard Time (US/Canada), Mexico, Central America -6
Eastern Standard Time (US/Canada), Caribbean, Colombia,
Ecuador, Peru
Atlantic Standard Time (US/Canada), Dominican Republic,
Bolivia, Paraguay, Chile, Venezuela
Newfoundland (Canada) -3.5
±12
-5
-4
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Appendix B — Web Configuration Tool
Time Zone Diff. from GMT
Mid-Atlantic -2
Azores, Cape Verde Islands -1
GMT, Western Europe Time, Morocco, Mali, Burkina Faso, Togo 0
Central Europe Time, Algeria, Chad, Angola +1
Eastern Europe Time, Libya, Sudan, Mozambique, South Africa,
Russia (East)
Moscow Time, Ethiopia , Tanzania, Madagascar +3
Iran +3.5
Russia Center, Armenia, Georgia, Oman, United Arab Republic +4
Afghanistan +4.5
Pakistan, Turkmenistan +5
India, Nepal, Sri Lanka +5.5
Bangladesh, Bhutan, Tajikistan, Kazakhstan +6
Myanmar +6.5
Cambodia, Laos, Thailand, Vietnam +7
Australian Western Standard Time, China, Indonesia, Mongolia,
Philippines
Korea, Japan, +9
Australian Center Standard Time +9.5
Australian Eastern Standard Time, Papua New Guinea +10
Russia (East), Solomon Islands, Vanuatu +11
+2
+8
Dialing Plan
The Dialing Plan table lets you define Dialed Digit matching patterns that are used to test
digits as they are entered. When a match is found, optional procedures such as removing
leading digits, adding prefix or suffix digits may be performed before the resultant string of
digits is sent to your SIP registration server, or a possible alternate server for dialing.
Effective use of dial plan rules can eliminate the need for you to terminate many common
calls by pressing the Dial soft-key, and can simplify calls to special services such as discount
long distance carriers.
The SIP phone tests digit strings using the top-most row first, then if no match is found here,
it proceeds down the table until either a match is found, or all the rules are exhausted. If no
match is found, the SIP phone will send the digits exactly as they were dialed to your SIP
registration server for dialing.
Whether or not you have to press the Dial soft-key before the digits are sent, is dependent on
the setting of the Global timer parameter
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5055 SIP Phone User Guide (Rev. A)
Figure 15 Dialing Plan Page
Note: Due to memory limitation in the phone, the contents of all Dialing Plan fields cannot
exceed a total of 120 characters. If you exceed this limit, new characters will not be
saved (you will notice this the next time you display the Dialing Plan table). Blank
fields consume two “characters”.
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Appendix B — Web Configuration Tool
Table 14 Web Configuration Tool: Dialing Plan
Setting Name Values (bold = default) Notes
When activated, this optional feature forces
all dialed digits to use the inter-digit timer
Global timer Off | On
Timer
Dialed Digits <blank>
1 | 2 | 3 | 4| 5 | 6 | 7 | 8 | 9
seconds
specified in Timer.
The Dial soft-key will be disabled as the
dialed digits will be dialed automatically
after the timer has expired.
Optional parameters sets the duration of the
inter-digit timer. When invoked using the
“.T” parameter or the Global timer setting,
the timer monitors the keypad for a pause in
digit entry. When a pause is detected, the
digits are optionally modified, then sent to
the SIP registration server or alternate
server.
Enter a partial or complete digit string (of a
minimum 2-digit length) for use as a
template to test dialed digits. Matching digit
strings are optionally manipulated, then sent
to a SIP server for dialing
If the value of one or more digits is likely to
be variable, then you can enter the wildcard
character “x” in place of each variable digit.
The square brackets [ ] can be used to
specify a set of digits, any one of which can
match a dialed digit. For example [123]
would yield a positive match if the user dials
“1”, “2” or “3”, but would fail to match “4”.
The “.T” timer parameter (which must be
placed at the end of a digit string) causes
the phone to accept an arbitrary number of
digits form the user. Once the user has
finished entering digits (signaled by no new
digits for a period equal to the timer value),
the digit string:
• may have digit manipulation
performed on it
• is sent to the SIP registration server
without requiring the user to press
the Dial soft key.
Note: The digits to follow field must be
Maximum 16 digits.
Optionally, enter the number of digits
Digits to follow 0
Digits to remove 0
expected to follow the partial number
specified under Dialed Digits here.
Optionally, enter the number of leading
dialed digits to be deleted from the dialed
number here.
set to 0 for the .T parameter to
work.
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5055 SIP Phone User Guide (Rev. A)
Table 14 Web Configuration Tool: Dialing Plan Continued
Setting Name Values (bold = default) Notes
Optionally, enter prefix digits here.
Prefix to add <blank>
Suffix/Route <blank>
Comments <blank>
Telephony digits 0 through 9 are valid. For
DTMF trunks, 0 through 9, are valid.
Maximum 16 digits.
Can be used to specify suffix digits that are
added to the digit string before it’s sent to
the SIP registration server for dialing.
Alternatively, this field can be used to
specify the URL of another SIP registration
server, (an alternate route)
If left blank, the digits will be routed to the
default SIP registration server.
Maximum 64 alphanumeric characters
Use this optional field to label a digit plan
entry. Maximum 8 characters.
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Appendix B — Web Configuration Tool
Figure 15 Example Dial Plan Rules
1) Dialed 4-digit extensions beginning with the digits 3, 4 or 5 will be matched by this rule which
makes use of wildcards to accommodate any three digits. The use of the Global time with this
rule is optional.
Dialed Digits Digits to follow Digits to Remove Prefix to Add Suffix/Route
Comments
[345]xxx 0 0
2) Dialed 7-digit (local calls in North America) numbers will be matched by this rule which strips
off the leading “9” before sending it to the SIP server for dialing. The use of the Global time with
this rule is optional.
Dialed Digits Digits to follow Digits to Remove Prefix to Add Suffix/Route
Comments
9[23456789] 0 1 Lcl Call
3) Dialed 10-digit (long distance in North America) numbers will be matched by this rule which
strips off the leading “1” before sending it to the SIP server for dialing. The use of the Global time
with this rule is optional.
Dialed Digits Digits to follow Digits to Remove Prefix to Add Suffix/Route
Comments
91 10 1 Lng Dist
4) This example illustrates the use of a Timer triggered by the Dialed Digit pattern "9011" (long
distance overseas call), with an arbitrary number of digits following it. The Global Timer should be
off in this case since the Timer parameter is used.
Dialed Digits Digits to follow Digits to Remove Prefix to Add Suffix/Route
Comments
9011.T 0 1 Lng Dist
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5055 SIP Phone User Guide (Rev. A)
Ethernet Page
For two Ethernet devices to connect, they must share the same speed setting. Most current
devices are capable of negotiating compatible settings automatically, as soon as you connect
them. Some older equipment does not support auto negotiation, for this equipment, you will
have to use this page to select the Ethernet operating parameters manually
Figure 16 Web Configuration Tool: Ethernet Page
Table 16 Web Configuration Tool: Ethernet Settings
Setting Name Values (bold = default) Notes
PC port
LAN port
Auto
10 Half
10 Full
100 Half
100 Full
Auto
10 Half
10 Full
100 Half
100 Full
The Auto setting lets the 5055 SIP phone
and the PC negotiate the Ethernet speed
and duplex automatically. If the PC does not
support auto-negotiation, you must select a
compatible setting manually.
The Auto setting lets the 5055 SIP phone
and the Ethernet device (hub, router, layer 2
switch, broadband modem etc.) negotiate
the Ethernet speed and duplex. If the device
does not support auto-negotiation, you must
select a compatible setting manually.
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Appendix B — Web Configuration Tool
Protocols Page
The Protocols Configuration page lets you activate and deactivate protocol support, if
required for security reasons.
Figure 17 Web Configuration Tool: Protocols Page
Table 17 Web Configuration Tool: Protocols Page
Setting Name Values (bold = default) Notes
Deactivating this protocol prevents future
HTTP
TFTP
Telnet
On, Off
On, Off
On, Off
web-based sessions with the phone. To
reactivate this protocol, you will have to use
the SIP Phone interface.
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5055 SIP Phone User Guide (Rev. A)
Users & Passwords Page
The Security Configuration Page lets the administrator change the passwords for the user
profiles and the administrator. This page can only be accessed using the Administrator user
name and password. After making the changes, click the Apply button.
Figure 18 Web Configuration Tool: Users & Passwords Page
Table 18 Web Configuration Tool: Users & Passwords Settings
Setting Name Values (bold = default) Notes
Cannot change this value here. Defaults are
User ID As programmed.
Password Change as needed.
admin for Administrator, and user for
default user profile.
Default is 5055 for Administrator.
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Appendix B — Web Configuration Tool
Media Configuration Page
The Media Configuration page lets you change the media configurations of your SIP Phone.
These settings are specific to the SIP Phone. After making the changes, click the Apply
button.
Figure 19 Web Configuration Tool: Media Configuration Page
Table 19 Web Configuration Tool: Media Configuration Settings
Setting Name Values (bold = default) Notes
Basic
Audio Codec Type
Frame Size
DTMF Type
DTMF Payload Type
(96-127)
Media Start Port
Media End Port
G711uLaw
G711Alaw
G729A
G729A & G711uLaw
G729A & G711uLaw &
G711ALaw
10 ms
20 ms
30 ms
Automatic
Outband & Inband
Outband
Inband
96
8000
19998
G711 µ-Law: used in North America.
G711 A-Law: used in the U.K.
G729A: 8:1 compression codec requiring
both ends to support this standard
(otherwise, reverts to G.711).
Frame size used by G711 or G729 codecs
for packet size. This can be set to 10, 20,
30 …. 100)
Used for DTMF tone generation (Outband
used for RTP DTMF).
These fields specify the UDP port range
used by the 5066 SIP phone. Change this if
the default range conflicts with other
devices in your network.
2
2
If you set the audio codec type to G.729, the users will not be able to use the Conference Call
feature (the Conf softkey is replaced by NA to indicate the feature is not available).
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5055 SIP Phone User Guide (Rev. A)
Registration Page
The Registration Status page lets you:
• determine if your 5055 SIP phone is registered with a SIP server (and how long its been
registered)
• manually initiate a registration request
Under normal use, registration is entirely automatic. This page is provided for troubleshooting
purposes
Figure 20 Web Configuration Tool: Registration Page
Table 20 Web Configuration Tool: Registration
Setting Name Notes
Registration Status
Display
Registration Status
Control
Indicates if the phone is registered and with which server.
Manually initiates a registration request.
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Appendix B — Web Configuration Tool
Configuration Upload/Download Page
The Configuration Upload/Download page lets you save the phone configuration file on your
PC, or load a previously stored phone configuration file into your 5055 SIP phone.
Figure 21 Web Configuration Tool: Configuration Upload/Download Page
Table 21 Web Configuration Tool: Configuration Upload/Download
Setting Name Values (bold = default) Notes
Configuration Upload File name
Download Saves a configuration file on disk
Use the Browse Button to locate a saved
configuration file.
Note: The phone configuration restore command does not restore passwords. If you have
changed you passwords from their system-default values, and saved the phone
configuration, then restoring this configuration to either a different phone, or the
original phone one after it’s had its factory default values restored, will not restore
these passwords to their original state.
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5055 SIP Phone User Guide (Rev. A)
Upgrade
The Firmware Update page lets you enter a source URL for a software upgrade to the phone.
The factory default configuration is URL: sipdnld.mitel.com which is the Mitel Networks
firmware upgrade site.
Note: Though you can upgrade the phone’s firmware using the manual key-method, it is
preferable to use the Upgrade button on this page, as it saves and restores your
current phone settings.
Clicking the “Upgrade” button preserves the current phone configuration, while upgrading the
software (this is the preferred option). In some cases, the firmware release notes will instruct
you to choose the second option “Upgrade & Set factory Defaults”. This second option is
required when differences between the software versions are so great, that the existing
configuration cannot be preserved. When using the “Upgrade & Set factory Defaults”
option, refer to the firmware release notes for instructions on how to restore you
configuration from saved files.
Figure 22 Web Configuration Tool: Firmware Upgrade Page
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Appendix C — Configuration Files
Appendix C — Configuration Files
This appendix contains:
• Example of a Generic Configuration File
• Example of a Specific Configuration File
• Enabling Multiple User Profiles
Note: 1) The 5055 supports up to two HTTP Clients at the same time.
2) The CFG file must have the following as the first line:
image_name= NULL
3) Maximum CFG Upload capability to the phone is 3000 characters. The examples CFG
files in this Appendix contain extensive comments. Working CFG files may not have room
within the 3000 character limit to support detailed comments.
Example of a Generic Configuration File
# Mitel 5055 SIP Phone Generic Configuration File
# this file name= SIPGeneric.cfg
# Grammar=
# comment lines are leading with a character '#'
# no escape '\' continue lines are allowed. No escape character is allowed.
# the string length of a parameter must be less than 128 characters
# parameter template=
# token = parameter ; comments
########################################################################
# System Configuration Begins Here
########################################################################
# image version
image_name= NULL
# configure the phone by the tftpserver
# choice [0-no, 1- yes, 2-always]
# if the option "always" is being chosen, every time the phone boots up, it will download
# the configuration parameters from the TFTP server which will overwrite any static
# values of these parameters. This mode is useful for administrators to control the
# phone's settings. User can't select this option from the web interface.
# If the option "Yes" is being chosen, the phone will boot up and download the
configuration
# file from the TFTP server. Therefore, the statically configured parameters, if any,
# will be overwritten by the parameters in the configuration file. After boot up, user
can
# change this parameter to "No" from web interface.
# If the option "No" is being chosen, the phone, when booting up, will
# not download the configuration file from the tftp server. The phone may only
# do a version check for the main image.
tftp_config= 1
# address type
# choice[0-IPv4 or 2-FQDN],
# when the option IPv4 is being chosen, the outgoing sip requests will use the dotted
# format of the IP address
# when the choice is FQDN, the outgoing sip requests will use the "sip:host_name.domain"
format
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5055 SIP Phone User Guide (Rev. A)
# for the "contact" sip header. The FQDN address must be a resolvable entry in the DNS
server.
addr_type= 0
# domain name
domain= -example.com; domain name
# tftp server ip address
# the sip phone will download the boot image, the main image, and the configuration
parameters
# from this ip address
tftp= 192.168.0.1
# sntp server URL address
# the sip phone will update its date and time from this server
sntp= ntp.cpsc.ucalgary.ca
# time zone
# integer [-12, 0, 12]
time_zone= -5; EST time
#time_zone= 8; China
# tos
# integer [0, 1e] (even numbers only);
tos= 0
# IEEE 802.1 priority
# integer [-1, 0, 7]
# -1 means OFF
802_priority= -1
# VLAN ID
# integer[0, 4095]
vlan_id= 0
##################################################################################
# SIP configuration
##################################################################################
# sip phone will listen for the SIP packets at this port
# when the other phone calls this phone, the sip packets must be sent to this port
local_sip_port= 5060
# transport protocol for sip
# this parameter can be overwriten by dialing a url with this parameter: transport= udp
# or transport = tcp
# choice [1-tcp, 2- udp]
trans_protocol= 2
# sip proxy address
# When dialing a number or user_id only, the proxy address will automatically be appended
# to the number (or id) as number@proxy or user_id@proxy
# The proxy address can be in IP address or domain.com format
proxy_addr=
# sip proxy port
proxy_port= 5060
# outbound_state
# choice [0-NO, 1-YES]
# if YES, all the sip requests and responses will always be sent to the outbound_ip.
# Otherwise, the sip responses will be sent to the "via" address. The requests will be
sent to
# "route" or "contact", according to the rule defined by the sip specs.
outbound_state= 0
# outbound proxy ip address
# 1) if, for some reason, the sip phone must send request to a local sip proxy first,
# config this outbound sip proxy ip address
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Appendix C — Configuration Files
# 2) if the above proxy_addr (domain.com) is not resolvable from the DNS server,
# this ip address will be used in place of the proxy address
outbound_ip= 192.168.0.1
#outbound server proxy port
outbound_port= 5060
# sip registrar address
# could be same as or different from the proxy_addr above
registrar=
# sip registrar port
registrar_port= 5060
# Registration duration in seconds for each register request
# the server may respond with a different duration
register_expire= 7200; in seconds
# Registration authentication method
# choice [0-NONE, 1-BASIC, 2-DIGEST]
auth_method= 0
# sip voice mail server addresssip phone willsend the "subsrib" request for messagesummary to this address
voice_mail_srv=
# auto forward to voice mail server after num of rings
voicemail_ringnum=4
# sip voice mail server port
voice_srv_port= 5060
# emergency number
# integer string
# when user dials this string, the phone will send the sip request to e911_ip
emerg_number= 911
# emergency ip address
# must be an ip address
emerg_ip= 192.168.0.1
# e911 port number
emerg_port= 5060
##################################################################################
# Media configuration
##################################################################################
# audio codec to offer
# the codec(s) you choose here, will be listed in the INVITE or OPTION's SDP
# choice 0-g711 uLaw
# 1-g711 ALaw
# 2-g729A
# 3-g729A and g711 uLaw
# 4-all of the above codecs
audio_codec=0
# audio codec packet size
# currently this parameter is only applicable to the g711 codec
# choice [10, 20, 30] ms
audio_pkt_size= 20; ms
# dtmf type
# defines the way the DTMF digits will be sent across
# choice [0-automatic, 1-outband & inband, 2-outband only, 3-inband only
# automatic means when a "telephone-event" is being received from the peer party, send
the
# DTMF digit in the outband-only mode
#
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dtmf_type= 0
#dtmf payload(96-127)
dtmf_payload= 96
#Media port configuration start port
start_port= 8000
#Media port configuration start port
end_port= 19998
##################################################################################
# feature configuration
##################################################################################
# auto answer mode
# choice [0-disabled, 1-enabled]
auto_answer= 0
# auto answer reason code
# 0- in a meeting
# 1- out of town
# 2- at lunch
# 3- on vacation
# 4- in a conference
# 5- in lab
# 6- back in 5 minutes
# 7- gone home
# 8- on a course
# 9- off sick
# 10- other reason
reasons= 0
# other reason string
# when reasons = 10, the sip will copy this string to the "reject reason" field.
other_reason= i am busy now!
# do not disturb
# choice [0 - disable, 1-enable]
# when enabled, all incoming calls will be rejected, or forwarded to the voicemail
do_not_disturb= 0
# call forwarding no answer mode
# choice [0-disabled, 1- enabled]
# when enabled, the incoming call will be forwarded to the "noans_fwd_addr"
# the condition defined by the "try_ring_nums"
noans_fwd_mode= 0
# call forwarding no answer after defined number of rings
# integer [1, 20]
try_ring_nums= 4
# call forwarding no answer address
# sip url for forwarding the call
# in the format of user_id, or user_id@domain.com
# if only user_id is configured, the call will be forwarded to the user_id@proxy
noans_fwd_addr= 1002
# call forwarding always mode
# choice [0-disabled, 1- enabled]
# when enabled, the incoming call will be forwarded to the "always_fwd_addr"
always_fwd_mode= 0
# call forwarding always address
# sip url for forwarding the call
# in the format of user_id, or user_id@domain.com
# if only user_id is configured, the call will be forwarded to the user_id@proxy
always_fwd_addr= 1002
# call forwarding when busy mode
# choice [0-disabled, 1- enabled]
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# when enabled, the incoming call will be forwarded to the "forward_addr"
busy_fwd_mode= 0
# call forwarding when busy forward address
# sip url for forwarding the call
# in the format of user_id, or user_id@domain.com
# if only user_id is configured, the call will be forwarded to user_id@proxy
busy_fwd_addr= 1002
# beep on hold
# choice [0- disable, 1- enable]
# If enabled, the SIP phone, when being held by the peer party, will generate beeps in
the receiver.
# if the sip proxy or back-2-back UA supports music on hold, this feature should be
disabled
beep_on_hold= 1
# on hold ringback timer
# if the sip phone puts the peer party on hold, and the handset is being put down on the
cradle, the
# sip phone will play a ringback signal after a period defined by this parameter
# to alert the user that there is a call on hold.
on_hold_alert= 300; seconds
# hot line
# choice [0-disable, 1-enable]
# when enabled, whenever the user pickup the phone handset, the call is automatically
# made to the hot_address.
hot_line= 0
# hot line address type
# choice [0-number or id mode, 1-sip url]
hot_addr_type= 1
# hot line address
# defines the address with the format defined by the hot_addr_type
hot_address= operator@-example.com
#adminId
adminId= admin
#admin password
admin_passwd=be6ad8761fe4eb9bb85934a2d21686bb
#admin displayname
admin_displayname=admin
#symetric SIP UDP
#choice[0-symetric SIP UDP, other-no-symetric SIP UDP]
#default 0
sym_udp= 0
#configuration change SIP notify
#choice[0-disabled,1-enabled] enable this only when phone is behind Mitel 3050 server
#default 0
ntfcfg= 0
#SIP backup server timeout period
#choice[2-2 seconds,3-4 seconds, 4-8 seconds,5-6 seconds]
#default 3
backupsvr_tout= 3
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Example of a Specific Configuration File
# Mitel 5055 SIP Phone Configuration File
# this file name= SXXXXXXXXXXXX.cfg
# where xxxxxxxxxxxx is the MAC address padded with 0s
# comment lines are leading with a character '#'
# no escape '\' continue lines are allowed
# the string length of the parameter must be less than 128 characters
# parameter template=
# token = parameter ; comments
########################################################################
# Configuration Begins Here
########################################################################
# System Configuration Begins Here
########################################################################
# image version
image_name= NULL
# configure the phone by the tftpserver
# choice [0-no, 1- yes, 2-always]
# if the option "always" is being chosen, everytime the phone boots up, it will download
# the configuration parameters from the TFTP server which will overwrite any static
# values of these parameters. This mode is useful for administrators to control the
# phone's settings. User can't select this option from the web interface.
# If the option "Yes" is being chosen, the phone will boot up and download the
configuration
# file from the TFTP server. Therefore, the statically configured parameters, if any,
# will be overwriten by the parameters in the configuration file. After boot up, user
can
# change this parameter to "No" from web interface.
# If the option "No" is being chosen, the phone, when booting up, will
# not download the configuration file from the tftp server. The phone may only
# do a version check for the main image.
tftp_config= 0
# address type
# choice[0-IPv4 or 2-FQDN],
# when the option IPv4 is being chosen, the outgoing sip requests will use the dotted
# format of the IP address
# when the choice is FQDN, the outgoing sip requests will use the "sip:host_name.domain"
format
# for the "contact" sip header. The FQDN address must be a resolvable entry in the DNS
server.
addr_type= 0
# host name
# defines the host name of the sip phone.
# This parameter is used when addr_type is set to FQDN
host_name= sip1
# domain name
domain= -example.com; domain name
# tftp server ip address
# the sip phone will download the boot image, the main image, and the configuration
parameters
# from this ip address
tftp= 192.168.0.1
# sntp server URL address
# the sip phone will update its date and time from this server
sntp= ntp.cpsc.ucalgary.ca
# time zone
# integer [-12, 0, 12]
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time_zone= -5; EST time
#time_zone= 8; China
# tos
# integer [0, 1e] (even numbers only);
tos= 0
# IEEE 802.1 priority
# integer [-1, 0, 7]
# -1 means OFF
802_priority= -1
# VLAN ID
# integer[0, 4095]
vlan_id= 0
##################################################################################
# IP network configuration
##################################################################################
#dhcp (0 = disable, 1 = enable)
dhcpenable= 1
#ip address
ipadr= 192.168.0.1
#network mask
ipmask= 255.255.255.0
#network gateway
ipgateway= 192.168.0.1
#primary dns server
ipdns= 192.168.0.1
#secondary dns server
ipscddns= 192.168.0.1
#pppoe(0 = disable, 1 = enable)
pppoe_enable= 0
#pppoe login
pppoe_login= NULL
#pppoe password
pppoe_passwd= NULL
##################################################################################
# user profile configuration
##################################################################################
#multiple user profile(0 = disable, 1 = enable)
multi_user_enable=0
# user id.
# used to register to the sip proxy as user_id@domain
# this id must be one sting, no space is allowed
user_id= user
# user display name.
# used for display the user's human readable name in sip "from" header
# from= "display name" sip= user_id@domain
disp_name= disp username
# user_name
# used as the user identify for authentication purpose. It could be same as user_id.
# but it could also be different. such as in the format of user_id@domain
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# no white space is allowed in this string
user_name= user@-example.com
# password
# As a pair with user_name for authentication purpose.
password= hello
# User's other pstn phone number
# this parameter is used in SDP packets to show the user can also be reached by this
# phone number.
# this is an option.
phone_num=
# email address of the user
# this parameter is used in SDP packets to show the user can also be reached by this
# email address. It is an option
email= user_name@-example.com
###################################################################################
## Additional users, up to 2 for release 1
###################################################################################
# user id.
# used to register to the sip proxy as user_id@domain
# this id must be one sting, no space is allowed
#user_id2= 1002
# user display name.
# used for display the user's human readable name in sip "from" header
# from= "display name" sip= user_id@domain
#disp_name2= user 1002
# user_name
# used as the user identify for authentication purpose. It could be same as user_id.
# but it could also be different. such as in the format of user_id@domain
# no white space is allowed in this string
#user_name2= 1002@-example.com
# password
# As a pair with user_name for authentication purpose.
#password2= 1002
# user id.
# used to register to the sip proxy as user_id@domain
# this id must be one sting, no space is allowed
#user_id3= 1003
# user display name.
# used for display the user's human readable name in sip "from" header
# from= "display name" sip= user_id@domain
#disp_name3= user 1003
# user_name
# used as the user identify for authentication purpose. It could be same as user_id.
# but it could also be different. such as in the format of user_id@domain
# no white space is allowed in this string
#user_name3= 1003@-example.com
# password
# As a pair with user_name for authentication purpose.
#password3= 1003
##################################################################################
# SIP configuration
##################################################################################
# sip phone will listen for the SIP packets at this port
# when the other phone calls this phone, the sip packets must be sent to this port
local_sip_port= 5060
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Appendix C — Configuration Files
# transport protocol for sip
# this parameter can be overwriten by dialing a url with this parameter: transport= udp
# or transport = tcp
# choice [1-tcp, 2- udp]
trans_protocol= 2
# sip proxy address
# When dialing a number or user_id only, the proxy address will automatically be appended
# to the number (or id) as number@proxy or user_id@proxy
# The proxy address can be in IP address or domain.com format
proxy_addr=
# sip proxy port
proxy_port= 5060
# outbound_state
# choice [0-NO, 1-YES]
# if YES, all the sip requests and responses will always be sent to the outbound_ip.
# Otherwise, the sip responses will be sent to the "via" address. The requests will be
sent to
# "route" or "contact", according to the rule defined by the sip specs.
outbound_state= 0
# outbound proxy ip address
# 1) if, for some reason, the sip phone must send request to a local sip proxy first,
# config this outbound sip proxy ip address
# 2) if the above proxy_addr (domain.com) is not resolvable from the DNS server,
# this ip address will be used in place of the proxy address
outbound_ip= 192.168.0.1
#outbound server proxy port
outbound_port= 5060
# sip registrar address
# could be same as or different from the proxy_addr above
registrar=
# sip registrar port
registrar_port= 5060
# Registration duration in seconds for each register request
# the server may respond with a different duration
register_expire= 7200; in seconds
# Registration authentication method
# choice [0-NONE, 1-BASIC, 2-DIGEST]
auth_method= 0
# sip voice mail server addresssip phone willsend the "subsrib" request for messagesummary to this address
voice_mail_srv=
# auto forward to voice mail server after num of rings
voicemail_ringnum=4
# sip voice mail server port
voice_srv_port= 5060
# emergency number
# integer string
# when user dials this string, the phone will send the sip request to e911_ip
emerg_number= 911
# emergency ip address
# must be an ip address
emerg_ip= 192.168.0.1
# e911 port number
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emerg_port= 5060
##################################################################################
# Media configuration
##################################################################################
# audio codec to offer
# the codec(s) you choose here, will be listed in the INVITE or OPTION's SDP
# choice 0-g711 uLaw
# 1-g711 ALaw
# 2-g729A
# 3-g729A and g711 uLaw
# 4-all of the above codecs
audio_codec=0
# audio codec packet size
# currently this parameter is only applicable to the g711 codec
# choice [10, 20, 30] ms
audio_pkt_size= 20; ms
# dtmf type
# defines the way the DTMF digits will be sent across
# choice [0-automatic, 1-outband & inband, 2-outband only, 3-inband only
# automatic means when a "telephone-event" is being received from the peer party, send
the
# DTMF digit in the outband-only mode
#
dtmf_type= 0
#dtmf payload(96-127)
dtmf_payload= 96
#Media port configuration start port
start_port= 8000
#Media port configuration start port
end_port= 19998
##################################################################################
# feature configuration
##################################################################################
# auto answer mode
# choice [0-disabled, 1-enabled]
auto_answer= 0
# auto answer reason code
# 0- in a meeting
# 1- out of town
# 2- at lunch
# 3- on vacation
# 4- in a conference
# 5- in lab
# 6- back in 5 minutes
# 7- gone home
# 8- on a course
# 9- off sick
# 10- other reason
reasons= 0
# other reason string
# when reasons = 10, the sip will copy this string to the "reject reason" field.
other_reason= i am busy now!
# do not disturb
# choice [0 - disable, 1-enable]
# when enabled, all incoming calls will be rejected, or forwarded to the voicemail
do_not_disturb= 0
# call forwarding no answer mode
# choice [0-disabled, 1- enabled]
# when enabled, the incoming call will be forwarded to the "noans_fwd_addr"
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# the condition defined by the "try_ring_nums"
noans_fwd_mode= 0
# call forwarding no answer after defined number of rings
# integer [1, 20]
try_ring_nums= 4
# call forwarding no answer address
# sip url for forwarding the call
# in the format of user_id, or user_id@domain.com
# if only user_id is configured, the call will be forwarded to the user_id@proxy
noans_fwd_addr= 1002
# call forwarding always mode
# choice [0-disabled, 1- enabled]
# when enabled, the incoming call will be forwarded to the "always_fwd_addr"
always_fwd_mode= 0
# call forwarding always address
# sip url for forwarding the call
# in the format of user_id, or user_id@domain.com
# if only user_id is configured, the call will be forwarded to the user_id@proxy
always_fwd_addr= 1002
# call forwarding when busy mode
# choice [0-disabled, 1- enabled]
# when enabled, the incoming call will be forwarded to the "forward_addr"
busy_fwd_mode= 0
# call forwarding when busy forward address
# sip url for forwarding the call
# in the format of user_id, or user_id@domain.com
# if only user_id is configured, the call will be forwarded to user_id@proxy
busy_fwd_addr= 1002
# beep on hold
# choice [0- disable, 1- enable]
# If enabled, the SIP phone, when being held by the peer party, will generate beeps in
the receiver.
# if the sip proxy or back-2-back UA supports music on hold, this feature should be
disabled
beep_on_hold= 1
# on hold ringback timer
# if the sip phone puts the peer party on hold, and the handset is being put down on the
cradle, the
# sip phone will play a ringback signal after a period defined by this parameter
# to alert the user that there is a call on hold.
on_hold_alert= 300; seconds
# hot line
# choice [0-disable, 1-enable]
# when enabled, whenever the user pickup the phone handset, the call is automatically
# made to the hot_address.
hot_line= 0
# hot line address type
# choice [0-number or id mode, 1-sip url]
hot_addr_type= 1
# hot line address
# defines the address with the format defined by the hot_addr_type
hot_address= operator@-example.com
#adminId
adminId= admin
#admin password
admin_passwd=be6ad8761fe4eb9bb85934a2d21686bb
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#admin displayname
admin_displayname=admin
#http protocol enable
# choice [0-disable, 1-enable]
http_task_enable= 1
#tftp protocol enable
# choice [0-disable, 1-enable]
tftp_task_enable= 1
#telnet protocol enable
# choice [0-disable, 1-enable]
telnet_task_enable= 1
#symetric SIP UDP
#choice[0-symetric SIP UDP, other-no-symetric SIP UDP]
#default 0
sym_udp= 0
#program key
pk1=
pk2=
pk3=
pk4=
pk5=
pk6=
pk7=
#configuration change SIP notify
#choice[0-disabled,1-enabled] enable this only when phone is behind Mitel 3050 server
#default 0
ntfcfg= 0
#SIP backup server timeout period
#choice[2-2 seconds,3-4 seconds, 4-8 seconds,5-6 seconds]
#default 3
backupsvr_tout= 3
#firmware upgrade http download url
http_download=
# language code
# fr_CA French Canadian, fr_FR French France, en_CA English Canadian, en_US English USA
# en_GB English Great Britian, en_AU English Australian, es_MX Spanish Mexican, es_US
Spanish USA
lancode= en_CA
#Ring tone code
# CA Canada, US USA, GB Great Britian, DE Germany, NL Netherlands, AU Australia, NZ New
Zealand,
# MX Mexico, FR France
tonecode= CA
#Dialing plan auto dialing global timer(0 disalbed, 1 enabled)
gtEnable= 0
#Dial plan auto dialing timer(1-9 secondes)
dtimer= 4
#dialing plan string( max len 256)
dialpl=
#firmware TFTP upgrade url
upgurl= sipdnld.mitel.com
##################################################################################
# fire wall configuration
##################################################################################
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#firmware tranversal (0 disalbed, 1 enabled)
fwEnable= 0
#fire wall WAN address discovery mode(0 static, 1 dynamic)
fwMode= 0
#fire wall WAN address discovery url
fwWanDurl=
#fire wall WAN address
fwWanurl=
###################################################################################
# phone book configuration
###################################################################################
#phone book index(0-4)
pbIndex= 0
#phone book entry 1 name
pbName1=
#phone book entry 1 address
pbAddr1=
#phone book entry 2 name
pbName2=
#phone book entry 2 address
pbAddr2=
#phone book entry 3 name
pbName3=
#phone book entry 3 address
pbAddr3=
#phone book entry 4 name
pbName4=
#phone book entry 4 address
pbAddr4=
#phone book entry 5 name
pbName5=
#phone book entry 5 address
pbAddr5=
######################################################################################
# Distinctive Ring configuration
#####################################################################################
#ring tone entery 1 key word
rdkw1=
#ring tone entry 1 type(0-16)
rdringtype1= 0
#ring tone entry 1 forward to voice mail(0 disalbed, 1 enabled)
rdvmail1= 0
#ring tone entry 1 block the call(0 diabled, 1 enabled)
rdblock1= 0
#ring tone entery 2 key word
rdkw2=
#ring tone entry 2 type(0-16)
rdringtype2= 0
#ring tone entry 2 forward to voice mail(0 disalbed, 1 enabled)
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rdvmail2= 0
#ring tone entry 2 block the call(0 diabled, 1 enabled)
rdblock2= 0
#ring tone entry 3 key word
rdkw3=
#ring tone entry 3 type(0-16)
rdringtype3= 0
#ring tone entry 3 forward to voice mail(0 disalbed, 1 enabled)
rdvmail3= 0
#ring tone entry 3 block the call(0 diabled, 1 enabled)
rdblock3= 0
Note: You can define the same parameters defined in SIPGeneric.cfg here. When defined
here, the parameters overwrite the values in the SIPGeneric.cfg file.
Enabling Multiple User Profiles
Use this procedure to enable Multi-User Profiles:
1. Navigate to "Configure Upload/Download Page"
2. Click on "Download" to save configuration parameters to download.txt file
3. Modify download.txt file entry from:
multi_user_enable= 0
to
multi_user_enable= 1
4. From "Configure Upload/Download Page" browse to download.txt file and click on "Upload"
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Appendix D — Working with Firewalls
Appendix D — Working with Firewalls
The 5055 SIP phone can be configured to work behind Network Address Translation (NAT)
firewalls which are not SIP aware by enabling the SIP configuration Bypass Firewall NAT
feature and configuring the firewall correctly. To do this:
1. Locate the documentation that came with your NAT firewall and look for instructions on how
to configure a Demilitarized Zone (DMZ) server. You must configure the 5055 SIP phone to
function as a DMZ server to the firewall.
2. Use the 5055 SIP phone configuration web page to:
A. Login to the phone using the web interface and select Network Configuration
Configure a static IP address, Subnet Mask, Gateway, and DNS server address.
Turn DHCP off.
B. Select SIP Configuration. Set Bypass Firewall NAT On.
C. In the "Mode" box select static if the IP of the WAN port of the router never changes.
Select dynamic if the IP changes as the result of DHCP or PPPOE.
D. If you selected static IP in step C., enter this IP in the WAN IP Address field. If you
selected dynamic, enter the URL of the service provider in the "WAN IP Discovery
URL" field.
E. Click the Apply button and verify if Bypass Firewall NAT is still set to On. If dynamic
IP address is used, also verify that WAN IP Address field has been filled in with a
valid address. If Bypass Firewall NAT resets to off, or there is no IP address in the
WAN IP Address field, then there is likely a problem with the WAN IP Discovery
URL that is preventing the phone from obtaining the router’s WAN IP address.
Tip: The 5055 SIP phone must be in a factory-default state (this configuration will not work
with the phone registered or trying to register). To do this power-cycle the phone while
holding down the 3 key and answer “yes” when asked if you want to “use factory
default.”
Tip: There are some service providers that provide free Dynamic IP services. If your Service
provider does not provide this service, you can try either of these:
www.sdforlaget.se/remoteip.asp or
http://www.changeip.com/ip.asp
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5055 SIP Phone User Guide (Rev. A)
Appendix E — Working with the 3050
ICP
The 3050 ICP adds features to your 5055 SIP phone that provide greater ease of use within a
small office or retail establishment, and opportunities to improve how you interact with your
customers.
Convenience
Mitel networks SIP phones connected to 3050 ICPs share a fast two, three or four-digit
dialing for station-to-station calls.
Cost Savings
The 3050 ICP can help you to reduce or avoid many long-distance charges by using fixedrate broadband IP to carry much of the voice traffic that would otherwise travel over the
PSTN.
Customer Interaction
The 3050 ICP has an autoattendant that can be used to direct your callers to specific
individuals or functions in your organization. For instance, if someone in your office usually
handles inquires about customer orders, then your automatic attendant can be programmed
to direct callers to this individual.
The 3050 ICP lets you assign voicemail accounts to each user for those occasions when they
cannot answer the phone in person. Users record their personalized greeting using the 5055
SIP phone. When a caller leaves a message, it is delivered to the user’s email inbox for later
retrieval using their PC or a phone.
For further details on the Mitel Networks 3050 ICP contact your local Mitel representative or visit
Mitel online at http://www.mitel.com
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Appendix F — Frequently Asked Questions
Appendix F — Frequently Asked
Questions
How do I access the User Profiles?
User Profiles must be enabled on the phone, before they will work. Refer to User Profiles to
learn how to do this.
Does my 5055 SIP Phone work behind a non-SIP compliant router?
It can - if you follow the steps outlined in Appendix D — Working with Firewalls
Where do I go to find latest versions of the 5055 firmware?
The latest version of the 5055 firmware is available on the TFTP server at: sipdnld.mitel.com.
For detailed instructions on firmware upgrade, refer to
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5055 SIP Phone User Guide (Rev. A)
Upgrading the Firmware of the SIP Phone
Where can I find the latest 5055 SIP Phone documentation?
You can use a copy of Netscape Navigator or Internet Explorer to view and download an
Adobe Acrobat compatible user guide by:
1. Point you browser to http://edocs.mitel.com.
2. Click the User Guides link to display the user guides page.
3. Locate the Other section – its has links for 5055 user and installation guides
Upon boot, the phone displays “PPPoE Initialize” and nothing
else
The phone is configured to work with a DSL connection using PPPoE but this connection
cannot be established.
1. Check the PPPoE login name and password in the Network Configuration Page
2. Make sure the DSL modem is plugged in, and powered up.
3. Verify that the SIP phone has got a valid IP address from the modem by pressing the
Menu key, followed by the Line 1 key.
The phone indicates a valid PPPoE connection by clearing the “PPPoE Initialize” message
and proceeding with the rest of the boot-up process.
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Appendix F — Frequently Asked Questions
How do I find out the IP address of my 5055 SIP Phone?
1. Press the Menu key.
2. Press the Line 1 key. The phone’s IP and MAC address are displayed.
What version of boot and main firmware is currently installed on my phone?
1. Press the Menu key.
2. Press the Line 2 key. The Main and Boot software versions are displayed.
What languages are currently available for my 5055 SIP Phone?
As of release 2.0:
• North American English
• North American French
• Latin American Spanish
You can select the display language using the User Configuration web page, or with the
softkey menu system (Phone Settings).
Why does my phone show *NO REG*?
This message indicates that your phone has failed to register with a SIP registration server.
Connection to a SIP registration server is necessary for your phone to be able to make and
receive SIP calls.
For registration to be successful:
1. The SIP registration Server must be up and running.
2. You must have correctly entered the SIP Registration Server’s URL (provided to you
by your SIP service provider) in the SIP Registry Server field of the SIP
Configuration Page.
3. If you SIP registration server is located on the Internet, then you must have a working
connection to the Internet.
You must have correctly entered the SIP Authentication user name and password (provided
to you by your SIP service provider) in the corresponding fields of the
4. User Configuration Page.
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5055 SIP Phone User Guide (Rev. A)
The time and date on my phone is not correct?
The phone can be configured to obtain its time automatically by consulting a Simple Network
Time Protocol Server (SNTP), or it can be manually set to a specific time.
• If you are using SNTP to set the time automatically, then you must have entered the URL
of a functioning SNTP server in the Additional Servers section of the Network
Configuration Page. As many SNTP servers base their clocks on Coordinated Universal
Time (Greenwich Mean Time), you may have to enter an offset value into the Time Zone
field to accurately reflect your local time.
• If you set your phone’s time manually, then you will have to reset it using the softkey
menus (Phone settings), or the appropriate fields in the Feature Configuration Page.
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Glossary
Glossary
Term Definition
3050 Mitel Networks(tm) 3050 Integrated Communications Platform (ICP)
DHCP Dynamic Host Configuration Protocol
DNS Domain Name Server
DSL Digital Subscriber Loop
DTMF Dual Tone Multiple Frequency
GMT Greenwich Mean Time (the time at Meridian 0, which goes through Greenwich, UK)
HTTP Hypertext Transfer Protocol
ICP Integrated Communications Platform
ID Identification
IP Internet Protocol
LAN Local Area Network
MAC Media Access Control
NAT Network Address Translation
PPPoE Point-to-Point Protocol over Ethernet
PSTN Public Switched Telephone Network
QOS Quality of Service
RTP Real-time Transport Protocol
SIP Session Initiation Protocol
SNTP Simple Network Time Protocol
TCP Transmission Control Protocol
TFTP Trivial File Transfer Protocol
ToS Type of Service
UDP User Datagram Protocol
URL Uniform Resource Locator
VLAN Virtual LAN
WAN Wide Area Network
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