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The Konftel 300Wx User Guide
ENGLISH
OmniSound
®
HD – Optimal sound
By incorporating full duplex audio, the patented audio technology OmniSound ® guarantees the audio conference an impressive and crystal clear sound in both directions through a hypersensitive microphone with 360° sound pickup and speakers that provide powerful sound distribution to all participants.
OmniSound
®
HD offers HD audio transmission for VoIP calls. In addition features like echo cancellation, background noise suppression which minimizes distracting background noise and an equalizer for personal adjustment of the sound’s various characteristics are also included. Move freely around the room, talk, discuss and debate while retaining great sound quality that is entirely free from irritating sound clipping, damping or echoes. It’s as close as you can get to natural sound.
DECT conforms to the GAP standard
DECT – Digital Enhanced Cordless Telephone. DECT incorporates encryption for maximum call security. GAP (Global Access Profile) ensures the interoperability of DECT equipment of different manufacturers.
product is in compliance with Annex V of the Directive 1999/5/EC (R&TTE) and complies with the requirements set out in the Council Directives 2004/108/EC, 2006/95/EC and with Directive 2011/65/EU (ROHS).
See www.konftel.com/doc for a complete declaration.
Warning!
Do not expose the Konftel 300Wx, charging stand or batteries to water or moisture.
Warning!
Do not open the casing of the Konftel 300Wx.
Warning!
Only use the power adapter provided to connect the unit to the mains.
Warning!
Only use the battery provided or a Konftel-recommended replacement battery.
Warning!
Do not attempt to open or modify the battery.
Warning!
Extreme heat, short-circuiting or any attempt to modify the battery may cause it to ignite or explode.
B
TABLE OF CONTENTS
Description 2
General .................................................. 2
Maintenance .......................................... 2
Keypad .................................................. 4
Display screen ........................................ 5
Navigation and selection in menus ........... 6
General 7
Unpacking ............................................. 7
Connecting and installing the unit ............ 8
Select connectivity option ........................ 9
Using a computer ................................... 9
Switching of Konftel 300Wx ...................10
Recharging batteries ..............................11
Connect to a DECT system 12
Registration ..........................................12
Answering and making calls 14
Answering calls......................................14
Making calls ..........................................14
Making three-way calls (DECT) ................15
VoIP calls via a computer ........................15
Calls via mobile phones ..........................15
During a call 17
Adjusting the speaker volume ..................17
Muting the microphone ..........................17
Placing a call on temporary hold .............17
Ending calls ..........................................17
Switching connections during calls ..........18
Conference guide 19
Creating conference groups.....................19
Making conference group calls ................19
Making a temporary conference group call
(multi-party call) using the Instant Guide 20
Displaying conference group participants 20
Editing conference groups ......................21
Deleting conference groups .....................21
Delete all conference groups ...................21
Conference guide settings ..................... 22
Displaying conference guide status ......... 22
Phonebook 23
Searching for contacts .......................... 23
Adding contacts ................................... 23
Editing contacts ................................... 23
Deleting contacts ...................................24
Deleting all contacts ..............................24
Displaying phonebook status ...................24
Recording 25
Starting a recording .............................. 25
Mute function ....................................... 25
Hold .................................................... 25
Ending a recording ................................ 25
Listening to a recording ......................... 26
Managing recordings ............................. 27
Adjusting the recording indicator ............ 27
Displaying recording feature status ......... 27
Settings 28
DECT................................................... 28
Apperance ........................................... 29
Date and time ...................................... 29
Sound ................................................. 30
Sleep mode ...........................................31
Software ...............................................31
Checking status .....................................31
Upgrading software 32
Upgrading from a PC............................. 32
Technical specifications 33
Konftel DECT base station (accessories) . 34
Service and guarantee 35
1
DESCRIPTION
GENERAL
The Konftel 300Wx is a battery-driven wireless conference phone that can be connected to DECT systems, mobile phones or PCs. It can be connected to a Konftel DECT base station or any GAP-compatible DECT system.
The conference phone has a host of innovative features:
• A fully wireless unit that provides talk time of up to 48 hours without having to recharge.
• OmniSound
®
HD audio technology.
• USB connectivity allows the phone to be used to make VoIP calls.
• The connectivity menu enables you to combine and switch easily between connectivity technologies – DECT, USB, mobile.
• Conference guide for easy set-up of multiparty calls.
• Record your meetings with an SD memory card so that you can listen to them later on.
• Expansion microphones can be added to increase the voice pickup range. (Optional.)
• Can be upgraded with smart features, making it future-proof.
• Two-year guarantee.
MAINTENANCE
Clean the equipment with a soft, dry cloth. Never use liquids.
2
DESCRIPTION
Speaker Microphone Display screen Keypad LEDs
SD memory card port
LEDs
Flashing blue Incoming call
Steady blue light Call in progress
Flashing red Hold, microphone and speaker turned off
Steady red light Mute, microphone turned off
Security lock port Expansion microphone port
Battery
Mini USB port for
PC connection
Expansion microphone port
Power supply port
Mobile phone connection
3
DESCRIPTION
KEYPAD
No/end/back
Start/stop recording
Increase volume
Decrease volume
Mute, the caller cannot hear you
Hold, puts a call on hold
Navigation in menus
Display of call list
Menu
Settings
Alphanumerical buttons
Hold down a button for 2 seconds to open the phonebook
Navigation in menus
Display of call list
OK
Yes/confirm choice
Answer/connect calls
During a call: Press to call a new person
End call
On/off
Conference
Automatic dialling of conference groups
One press of this button will always connect all parties to a conference call
Connectivity menu
Writing text
Each button contains letters and characters – more than those shown on the button (see illustration below). Press the same button repeatedly to change to another character. If there are two letters under the same button that you want to enter one after the other, you need to pause slightly before entering the second letter.
Press the button to delete the last character you entered.
(space) . - 1
G H I 4
P Q R S 7
A B C Å Ä Á À 2
J K L 5
T U V Ü Ú 8
0
D E F É È 3
M N O Ö Ø Ñ 6
W X Y Z 9
4
DESCRIPTION
DISPLAY SCREEN
On Hook
Signal strength DECT
Date
Time
Battery charging
Mains connected
Fully charged
< 1 hr talk time
Off Hook
Call duration
Recording indicator
Connectivity menu
Press to switch to and from this menu.
Note that USB is automatically activated when connected. The alternative USB is only shown during a DECT call when it normaly is disabled.
Menu
Press to switch to and from a menu.
Current menu
Submenu
Marked option – open by pressing OK button
Scrolling list
(indication of where the marked option is in the list or menu)
List of setting options:
Existing settings
Marked option – select by pressing OK button
List of names:
Marked name – select by pressing OK button
5
DESCRIPTION
NAVIGATION AND SELECTION IN MENUS
Press .
Select the option you want from the menu using the arrow buttons.
Confirm by pressing OK to select the marked option.
Cancel the setting or go back one level in the menu by pressing .
Quit the menu by pressing again.
Note that after you have made changes to a setting, you must press OK to activate the setting.
It is possible to open a menu option directly by pressing the number button that corresponds to the position of the option in the menu (e.g. 2 to open PHONE BOOK and then 3 to select EDIT CONTACT).
Menu tree
CALL LIST
OUTGOING
INCOMING
MISSED
ERASE LISTS
PHONE BOOK
SEARCH CONTACT
ADD CONTACT
EDIT CONTACT
ERASE CONTACT
ERASE ALL
STATUS
CONF GUIDE
SEARCH GROUP
ADD GROUP
EDIT GROUP
ERASE GROUP
ERASE ALL
SETTINGS
STATUS
RECORDING
PLAYBACK FILE
RENAME FILE
ERASE FILE
ERASE ALL
SETTINGS
STATUS
SETTINGS
DECT
REGISTER
DE-REGISTER
BASE
PARK
ACTIVATE DTMF
MULTIKEY DIALING
G722 CODEC
APPERANCE
LANGUAGE
SCREEN TEXT
LCD CONTRAST
TIME DATE
TIME
TIME FORMAT
DATE
DATE FORMAT
SOUND
RING LEVEL
KEY TONE
EQUALIZER
MOBILE LEVEL
SLEEP MODE
UPGRADE
FACTORY RESET
STATUS
6
GENERAL
UNPACKING
Check that all parts are included in the package. If any parts are missing, please contact your retailer.
• The Konftel 300Wx conference phone
• Battery
• Charging cradle
• 7.5 metre power cable for charging cradle/Konftel 300Wx
• AC adapter
• 1.5 metre USB cable
• Quick reference guides in different languages
Packages with base station also include:
• Konftel DECT Base station
• AC adapter with 1.5 metre cable
• 1.8 metre cable for line connection
• Intermediate plugs for analogue connection
• Quick reference guides in different languages
7
GENERAL
CONNECTING AND INSTALLING THE UNIT
Open the battery cover and insert the battery as illustrated.
(An optional phone cable is required for connection to mobile phones.)
Mini USB port for
PC connection
Place the conference phone at least 20 cm from the users.
8
GENERAL
SELECT CONNECTIVITY OPTION
The unit is default set for a DECT connection but it is possible to switch between and combine all available connectivity options.
OK to confirm.
The box in front of the selected option is filled in which indicates that this connectivity option can be used for calls.
Remove a connectivity option in the same way.
If you have selected MOBILE, you connect the call on the mobile phone and via USB use a software program in the PC (see pages 15 and 16).
You can add another connection or turn off a connection during a call.
The USB connection is automatically activated when the cable is connected. You can manually switch off USB via this menu in offhook.
USING A COMPUTER
When connected to the PC, the Konftel 300Wx acts as a loudspeaker and microphone when making VoIP calls. To make a VoIP call, use Skype
®
, Microsoft Lync
®
, an online meeting service or some other VoIP (Voice over IP) software.
Connect the Konftel 300Wx to a computer using the USB cable provided.
Check that
Konftel 300Wx USB Audio is set as the sound device in the computer’s settings.
The computer’s operating system normally automatically selects the Konftel 300Wx as a speaker and microphone. If the Konftel 300Wx is not automatically selected as the audio device, you may have to open the computer’s settings and select
Konftel 300Wx
USB Audio.
If you have an active application in the computer that is using the computer’s built-in speakers, it may be necessary to restart the program to switch to the new speaker
(Konftel 300Wx USB Audio).
The computer only activates the microphone in the Konftel 300Wx if there is a program running that uses the microphone. This means that the Konftel 300Wx usually acts as a normal computer speaker and that the audio processing in OmniSound
®
HD only is turned on when needed.
9
GENERAL
Using the Konftel as a conference unit
When communication software or audio recording software is started on the computer, the
Konftel 300Wx microphone is also activated. OmniSound ® is now handling full duplex and echo cancellation. The blue LEDS on the Konftel 300Wx also light up at the same time.
Note that you might need to turn off the automatic audio processing in some VoIP software (eg. Skype ® ). The audio processing shall be done by Konftel 300Wx for best audio quality.
Disabling USB
The USB connection is automatically activated when the USB cable is connected.
If you choose to make a DECT call or answer an incoming DECT call, the USB connection is automatically put on hold. During the DECT call, you can manually choose to enable and disable the USB connection in the connectivity menu.
Press the key.
Select USB and press OK to confirm.
SWITCHING OF KONFTEL 300WX
It is possible to switch off and on the unit.
Press the key for 2 seconds.
Press OK to confirm.
The unit is turned off.
It is possible to switch on the unit by pressing the same button again.
10
GENERAL
RECHARGING BATTERIES
The Konftel 300Wx lets you know when the battery’s remaining capacity is less than one hour of talk time. An audible signal is emitted and the LEDs flash three times. The battery recharging symbol on the display screen will also start flashing.
The battery can also be recharged by plugging the power cable into the Konftel
300Wx.
Batteries
The battery has a button to display the battery’s remaining capacity.
The Konftel 300Wx comes with a charged battery, but it may need recharging to full capacity depending on how long it has been in storage.
A fully-charged battery provides talk time of about 48 hours or 20 days of standby time.
The Konftel 300Wx automatically switches to sleep mode after a 40 minutes of inactivity.
Warning!
• Do not attempt to open or modify the battery.
• Do not expose the battery to moisture or extreme heat.
• Extreme heat, short-circuiting or any attempt to modify the battery may cause it to ignite or explode.
• Only use the battery provided or a Konftel-recommended replacement battery.
Battery charger (optional)
The optional battery charger allows for external recharging of one or two batteries. Charge status is denoted by the LED for each charging channel on the charger. The LED is red during charging and green when the battery is fully charged. A red flashing LED indicates that the battery is damaged and must be replaced.
11
CONNECT TO A DECT SYSTEM
The Konftel 300Wx connects to different makes of GAP-compatible DECT systems. The
Konftel 300Wx plus Konftel DECT base station package provides a solution for offices without a compatible DECT system.
REGISTRATION
The Konftel 300Wx has to be registered with a base station before it can be used. Note that both the base station and the Konftel 300Wx have to be put into registration mode.
Here we explain how to register in a
Konftel DECT base station. To register your Konftel
300Wx in other makes of base station, simply follow the same procedure. Please refer to your base station instruction manual or contact your company’s switchboard administrator if you are unsure.
If you have purchased a Konftel 300Wx with a Konftel base station, the conference phone will already be registered with the base station. You will only need to register if you have purchased the base station separately or are using a different make of base station.
Put the Konftel DECT base station in registration mode
Hold down the key/LED on the base station for at least 3 seconds.
The LED flashes rapidly.
The base station is now in registration mode. Once registration has been completed, the LED will stop flashing. If the base station does not identify a DECT phone within 10 minutes, the registration mode will switch off automatically.
A Konftel DECT Base station is automatically set to registration mode when it is connected to the power supply.
Key and LED
12
CONNECT TO A DECT SYSTEM
Registering the Konftel 300Wx with a DECT base station
Select >
SETTINGS > DECT > REGISTER to start the registration guide in the
Konftel 300Wx.
Please note that you must confirm all your choices by pressing OK.
Select base ( 1–4).
Four base stations can be registered in the Konftel 300Wx. If no base station has been programmed, choose no. 1. Numbers being used are indicated by a filled box.
Enter the base station’s security code (PIN code).
The code is 0000 for the Konftel DECT base station. Check with your company’s switchboard administrator if a different DECT system is being used.
The Konftel 300Wx is now in registration mode and is searching for a base station in registration mode. Wait until you receive confirmation that registration has been completed. (The base station has to be in registration mode.)
Registration can be interrupted by pressing . If registration was successful, the word SUCCESS will appear in the display.
The Konftel 300Wx is now ready to use.
Please see page 23 for other menu options and settings.
If registration was not successful
If registration was not successful, the word FAILURE will appear in the display. Possible reasons for this:
• Incorrect security code (check with your company’s switchboard administrator).
• A PARK code might be needed (check with your company’s switchboard administrator). Activate PARK ( > SETTINGS > DECT > PARK) and try to register again using your PARK code.
• No base station within range or not activated in registration mode.
13
ANSWERING AND MAKING CALLS
ANSWERING CALLS
A ring tone is heard and the blue LEDs start flashing.
Press to answer.
The LEDs show a steady blue light. The duration of the call is shown on the display screen.
MAKING CALLS
Dial the number.
Press .
The number is called and the LEDs show a steady blue light. The duration of the call is shown on the display screen.
Or:
Press .
Dial the number.
The number is called and the LEDs show a steady blue light. The duration of the call is shown on the display screen.
Redialling
Press an arrow button.
The last number that you dialled appears on the display screen.
Use the arrow buttons to look through previously dialled numbers.
Press .
The number you select is dialled.
Calling from the phonebook
Please turn to page 23 for instructions on adding and updating contacts in the phonebook.
Press and hold down a number button for two seconds.
The phonebook appears on the display screen. The pre-programmed names appear in alphabetical order, starting with the number button’s first letter.
Use the arrow buttons to look through the phonebook.
Press .
The contact you select is dialled.
14
ANSWERING AND MAKING CALLS
MAKING THREE-WAY CALLS (DECT)
The simplest way to set up a multi-party call is to use the conference guide (see page 19).
You can connect another party to an on-going call as follows. During a call, the button acts as Flash/R-pulse, which means it transmits commands to the PBX.
This description applies for DECT calls only.
Press during a call.
The calls is put on hold and a second dialling tone is heard.
Dial the second participant’s number and wait for an answer.
Press the button to connect all the meeting participants.
VOIP CALLS VIA A COMPUTER
For these instructions, it is assumed that the computer is turned on, the VoIP software
(e.g. web meeting software or Skype
®
) is running, that the Konftel 300Wx is connected via the USB cable to the computer and that Konftel 300Wx USB Audio is selected as the sound device in the computer’s settings.
Making VoIP calls
Make the call (or answer an incoming call) using the computer’s VoIP software.
The microphone is automatically activated and the LEDs show a steady blue light.
Ending a VoIP call
End the call in the computer’s VoIP software.
The Konftel 300Wx now only functions as a speaker for the computer.
CALLS VIA MOBILE PHONES
For these instructions, it is assumed that the Konftel 300Wx is connected to a mobile phone via a cable (optional accessory).
Select mobile as the telephone connection
15
ANSWERING AND MAKING CALLS
The box in front of MOBILE is filled in which indicates that this connection can be used for call.
Deselect the DECT connection in the same way if it is not going to be used.
You can combine all available connectivity options.
Making calls using a mobile phone
Press to activate Konftel 300Wx.
Dial the number on the mobile phone as you normally would.
The Konftel 300Wx now functions as a speaker and microphone for the mobile phone. The LEDs show a steady blue light.
Ending calls
End the call on the mobile phone.
Press to inactivate the Konftel 300Wx.
16
DURING A CALL
ADJUSTING THE SPEAKER VOLUME
Adjust the speaker volume using the buttons and .
Adjusting the speaker volume when multiple connections are connected
If you have selected multiple connections, you can adjust the volume level for each connection. This is useful if the connections have different volume levels.
or key to display the current volume setting.
The current main volume appears on the display screen. Above it are two or three smaller volume indicators which can be used to adjust the different connections.
the and keys.
and .
MUTING THE MICROPHONE
Press to turn off the microphone.
The LEDs change from blue to red. The other party cannot hear what you are saying.
Press to reconnect the call.
PLACING A CALL ON TEMPORARY HOLD
Press to put the call on hold.
The LEDs change from blue to flashing red. Both the microphone and the speaker are turned off and the call is on hold.
Press to reconnect the call.
ENDING CALLS
End the call by pressing .
17
DURING A CALL
SWITCHING CONNECTIONS DURING CALLS
Adding a new connection during a call
You can add another connection or switch between connection during a call.
Press the button during the call.
An indicator appears on the display screen in front of the selected connection.
Use the arrow buttons to mark the connection you want to activate and confirm your choice by pressing
OK.
If you selected USB or MOBILE, this connection will open. You then connect the call via the computer or mobile phone.
If you selected DECT, you will hear a dialling tone. You then dial the number on the Konftel
300Wx or use the Phonebook.
Putting a connection on hold during a call
Press the button during the call.
Use the arrow buttons to mark the connection you want to put on hold and press the
button.
The indicator in front of the selected connection starts to flash.
Repeat the above to reconnect the connection.
Disconnecting a connection during a call
Press the button during the call.
Use the arrow buttons to mark the connection you want to disconnect and confirm your choice by pressing .
The indicator in front of the selected connection goes out and the connection is disconnected.
The other calls can continue.
18
CONFERENCE GUIDE
The conference guide allows you to create 20 conference groups with up to six people in each. You can now call the participants in a meeting by simply pressing a few buttons.
You may need to adjust some settings in your Konftel 300Wx to make it compatible with your PBX. Please turn to page 22 in this user guide for more details.
nected via the PC and calls via MOBILE from the mobile phone.
Entering text with the keypad
Please turn to page 4 for instructions on using the keypad to enter text.
CREATING CONFERENCE GROUPS
Select > CONF GUIDE > ADD GROUP.
Enter a group name and press
OK to confirm.
Press OK to enter the first person.
Write the name of the first person and press
OK to confirm. Enter the number and press OK to confirm.
Press OK to add another person and repeat the above steps.
Conclude by pressing .
MAKING CONFERENCE GROUP CALLS
Press the button.
Select the conference group you want to call.
You can use the arrow buttons to press an alphanumeric button to jump to the first name that starts with that letter in the list.
Press OK if you want to see who is in the group before you make the call.
(Then press OK again or to call.)
Press directly if you want to make the call without first checking who is in the group.
The Konftel 300Wx dials the first person and asks you to confirm when the person answers.
Press OK when the first person has answered.
Continue in the same way for each person in the group.
19
CONFERENCE GUIDE
If you don’t get an answer
Press if one of the persons you are trying to call does not answer.
The conference guide will ask you if you want to redial that person.
Either press OK or .
If you decide not to redial ( ) you will be asked if you want to end the conference guide or continue to the next person.
Press OK to continue and to end the guide.
When the guide has ended, you will be connected with up to six people and ready to start your conference.
Please note that you can also call a conference group via SEARCH GROUP in the
CONF GUIDE menu.
MAKING A TEMPORARY CONFERENCE GROUP CALL (MULTI-
PARTY CALL) USING THE INSTANT GUIDE
Press the button.
Select INSTANT with the arrow buttons. Press OK to confirm.
The Instant guide asks you to dial the first number.
Dial the first person’s telephone number. Press OK to confirm.
Press OK when the first person has answered.
Press OK to call another person and repeat the above steps. Press when you have called all the persons you want for your meeting.
If one of the persons does not answer, follow the steps for calling a saved conference group (see above).
DISPLAYING CONFERENCE GROUP PARTICIPANTS
Select > CONF GUIDE > SEARCH GROUP.
Mark the group you want and press
OK.
The names of the conference group participants are displayed.
Press to return to the CONFERENCE GUIDE menu or to exit the menu. Press
OK or to dial the group.
20
CONFERENCE GUIDE
EDITING CONFERENCE GROUPS
Select > CONF GUIDE > EDIT GROUP.
Here, you can change the name of a conference group, add or delete a person from a group and change a person’s name and phone number.
Mark your choice of option in the menu and press OK to confirm.
The names of the conference groups are displayed.
Mark the group you want to change and press
OK to confirm.
Mark a person you want to change and press OK to confirm.
Make the change and press
OK to confirm.
Delete a number or a letter in a name by pressing . If you want to cancel the change you are making and leave the name or number as they were, press until the CONFERENCE GUIDE menu appears.
DELETING CONFERENCE GROUPS
Select > CONF GUIDE > ERASE GROUP.
Mark the group you want to delete and press OK.
Confirm that you really want to delete the group by pressing OK or cancel the deletion of a group by pressing .
DELETE ALL CONFERENCE GROUPS
Select >
CONF GUIDE > ERASE ALL.
Confirm that you really want to delete all groups by pressing OK or cancel the deletion of all groups by pressing .
21
CONFERENCE GUIDE
CONFERENCE GUIDE SETTINGS
When a multi-party call is going to be connected, a special signal (called flash or R-pulse) is produced which transmits the command (enquiry, conference and return) to the PBX. It may be necessary to change the type of signal if calls are connected via a PBX. Different
PBXs may require different signals. These requirements should be specified in the technical data for the PBX. Settings for the most common PBXs are listed on Konftel’s website.
Select > CONF GUIDE > SETTINGS.
Enter the command for a enquiry for a new dialling tone and press OK to confirm. (“F” is the default setting.)
Enter the command to connect everyone to a multi-party call and press
OK to confirm.
(“F3” is the default setting.)
Enter the command to return to your original call and press OK to confirm. (“F1” is the default setting.)
If you experience problems making multi-party calls
Your PBX may not be programmed to admit multi-party calls or the procedure may be different. Please contact your telephone administrator or service provider.
DISPLAYING CONFERENCE GUIDE STATUS
Select > CONF GUIDE > STATUS.
The display screen indicates the number of conference groups already stored plus how many more groups can be added.
22
PHONEBOOK
The Konftel 300Wx has a phonebook that holds up to 200 entries. See page 14 for making calls via the phonebook.
Entering text with the keypad
Please turn to page 4 for instructions on using the keypad to enter text.
SEARCHING FOR CONTACTS
The simplest way to search for a contact in the address book is to hold down a number button for two seconds (see page 14). You can also search for a contact via the PHONEBOOK menu.
Select >
PHONE BOOK > SEARCH CONTACT.
Mark the contact you want.
You can use the arrow buttons to press an alphanumeric button to jump to the first name that starts with that letter in the list.
Press OK or to dial the selected contact, to return to the PHONEBOOK menu or to exit the menu.
ADDING CONTACTS
Select > PHONE BOOK > ADD CONTACT.
Enter the name of the contact and press
OK to confirm.
Write the number and press OK to confirm.
Press
OK to add another contact or to exit the menu.
EDITING CONTACTS
Select > PHONE BOOK > EDIT CONTACT.
Mark the contact you want to change. Press OK to confirm.
Make the change and press OK to confirm.
Delete a number or a letter in a name by pressing . If you want to cancel the change you are making and leave the name or number as they were, press until the PHONEBOOK menu appears.
23
PHONEBOOK
DELETING CONTACTS
Select > PHONE BOOK > ERASE CONTACT.
Mark the contact you want to delete and press OK.
Confirm that you really want to delete the contact by pressing OK or cancel the deletion of a contact by pressing .
DELETING ALL CONTACTS
Select > PHONE BOOK > ERASE ALL.
Confirm that you really want to delete all contacts by pressing OK or cancel the deletion of a contact by pressing .
DISPLAYING PHONEBOOK STATUS
Select >
PHONE BOOK > STATUS.
The display screen indicates the number of contacts already stored plus how many more contacts can be added.
24
RECORDING
It is possible to record a phone conference on an SD (Secure Digital) memory card so you can listen to it later on. It is possible to record a call in progress and use the Konftel
300Wx as a dictaphone.
A flashing symbol appears on the display screen during recording. A beep is heard every
10 seconds so that other parties in the call know it is being recorded. The beep can be turned off (see page 27).
The recorded call is saved on the memory card. The document is named after the time at which the recording began (e.g. 14-35-01.wav) and is saved in a folder with the current date. This name can be changed after the recording has ended (see page 27).
STARTING A RECORDING
Remember to notify all parties in the call before you start recording.
Press the button and press OK to confirm.
The recording symbol flashes on the display screen.
MUTE FUNCTION
Press the button during a call to turn the microphone off.
The LEDs change from blue to red.
The other party cannot hear what you are saying and only the called party is recorded.
HOLD
Press during the call to put the call on hold.
The LEDs change from blue to flashing red.
Both the microphone and the speaker are turned off and the call is on hold. None of the parties will now be recorded.
ENDING A RECORDING
Press the button press OK to confirm.
The recording symbol is no longer displayed on the screen.
If you are recording a call and the call is ended, you will be asked if you want to stop recording.
25
RECORDING
LISTENING TO A RECORDING
You can listen to a recorded phone conference on your Konftel 300Wx or a computer using an SD memory card reader. Recordings are saved in .wav format and can be played back on your choice of media player.
Listening to recordings on the Konftel 300Wx:
Select > RECORDING > PLAYBACK FILE.
Mark the folder you want (date) and press OK to confirm.
A list of all recordings for the specified date is displayed. Names, date, time and length are displayed for each recording.
Mark the recording you want and start the playback by pressing OK.
Adjusting the speaker volume
Increase the volume using the button and lower it using .
Pause playback
Pause the playback by pressing OK.
Restart playback by pressing
OK again.
Move forward in the recording during playback
Move forward in the recording by pressing arrow down or backward by pressing arrow up.
The recording advances about 10 seconds every time you press the button.
You can also jump further forward in the recording:
Key 3 – forward 1 minute
Key 6 – forward 10 minutes
Key 1 – backward 1 minute
Key 4 – backward 10 minutes
Stop playback
Stop the playback by pressing .
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RECORDING
MANAGING RECORDINGS
Editing the name of a recording
The recording is named by default with the time at which the recording started.
Select > RECORDING > EDIT FILE.
Mark the folder you want and press OK to confirm.
Mark the recording to be edited and press OK.
Make the change and press OK to confirm.
Delete a number or a letter in a name by pressing .
If you want to cancel the change you are making and leave the name or number as they were, press several times until the RECORDING menu appears.
Deleting a recording
Select >
RECORDING > DELETE FILE.
Mark the folder you want and press OK.
Mark the recording to be deleted and press
OK.
Confirm that you really want to delete the recording by pressing OK again, or cancel the deletion of a recording by pressing .
Deleting all recordings
Select > RECORDING > DELETE ALL.
Confirm that you really want to delete all recordings by pressing OK again, or cancel the deletion of a recording by pressing .
ADJUSTING THE RECORDING INDICATOR
A short beep is heard every 10 seconds so that all the far-end parties in the call know it is being recorded. This feature can be turned off.
Select > RECORDING > SETTINGS.
Select your choice and press
OK to confirm.
DISPLAYING RECORDING FEATURE STATUS
Select > RECORDING > STATUS.
The display screen indicates the number of recordings, the total length of recorded calls and how many hours of recording time are left before the memory card is full.
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SETTINGS
DECT
Registering with a base station
Select > SETTINGS > DECT > REGISTER.
The Konftel 300Wx registration guide starts.
See page 12 for a detailed description of the registration process.
Deregistering from a base station
This option only works with a Konftel DECT base station.
Select > SETTINGS > DECT > DE-REGISTER.
The Konftel 300Wx retrieves a list of the handsets that have been registered with that particular
Konftel DECT base station. There is a filled box in front of the name of the handset that you are using (Konftel 300Wx).
Mark the handset you want to de-register and press OK to remove it.
Enter the base station’s PIN code ( 0000) and confirm your choice by pressing OK.
Select a base station
Select > SETTINGS > DECT > BASE.
The AUTO option and the base stations that are registered are displayed as options in the menu.
AUTO (default) means that the Konftel 300Wx automatically selects the registered base station that has the greatest signal strength.
Enabling PARK code
Select >
SETTINGS > DECT > PARK.
Disabled is defalult. PARK is a function that makes it possible to lock a handset/conference phone to a specific base station using a security code.
Select Inband DTMF generation
Select > SETTINGS > DECT > ACTIVATE DTMF.
DTMF (Dual-tone multi-frequency signaling) is used for telecommunication signaling over analog telephone connections, eg. to a PBX or to log on to a phone bank or a conferencing service.
The tones can be generated by the phone (inband – which is default) or by the PBX it is connected to.
Select Inband DTMF generation
Select > SETTINGS > DECT > MULTIKEY DIALING.
Multikey dialing means that all DTMF tones are sent as one command using speed-dial
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SETTINGS
instead of one tone for each pressed button.
Disable G722 audio codec
Select > SETTINGS > DECT > G722.
G722 is an audio codec (compression standard) with high quality. Some PBXs don’t support this.
APPERANCE
Selecting languages
Select >
SETTINGS > APPERANCE > LANGUAGE.
Select the language you want and press OK to confirm.
Changing text on the display screen
The text on the display screen can be changed when the Konftel 300Wx is on hook.
Select >
SETTINGS > APPERANCE > SCREEN TEXT.
Delete existing text using and enter your new text. Press OK to confirm.
Adjusting LCD contrast
Select > SETTINGS > APPERANCE > LCD CONTRAST.
Select your choice and press
OK to confirm.
DATE AND TIME
Changing the time
Select >
SETTINGS > TIME DATE > TIME.
Change the flashing figures for the hour using the arrow buttons or by entering the figures on the keypad. Press OK to confirm and to continue to minutes.
Changing the time format
Select > SETTINGS > TIME DATE > TIME FORMAT.
Select either a 12-hour or a 24-hour format using the arrow buttons and press OK to confirm your choice.
Changing the date
Select > SETTINGS > TIME DATE > DATE.
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SETTINGS
Change the flashing figures for the year using the arrow buttons or by entering the figures on the keypad. Press
OK to confirm and to continue to the month, and then proceed in the same way for the day.
Changing the date format
Select > SETTINGS > TIME DATE > DATE FORMAT.
Select the preferred date format and press OK to confirm your choice.
SOUND
Adjusting ring level volume
Select > SETTINGS > SOUND > RING LEVEL.
Select your choice and press OK to confirm.
There are six volume levels plus a silent mode. You will hear the ring tone for each level you select. If you select silent mode, an LED on the phone flashes when an incoming call is received.
Deactivating keypad tones
You can select whether or not you want a tone to be heard when you press a button. Note that this setting does not affect the tone you hear when you dial a number in call mode.
Select >
SETTINGS > SOUND > KEY TONE.
Select your choice and press OK to confirm.
Adjusting sound (equalizer)
The sound reproduction can be adjusted to the required pitch (SOFT, NEUTRAL and
BRIGHT).
Select > SETTINGS > SOUND > EQUALIZER.
Select your choice and press OK to confirm.
Adjusting microphone volume with mobile phone connected
The Konftel 300Wx microphone has seven volume settings, so if the volume of calls via
Mobile phones is too soft or too loud, you can easily adjust it to the right level.
Select > SETTINGS > SOUND > MOBILE LEVEL.
Select your choice and press
OK to confirm.
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SETTINGS
SLEEP MODE
It is possible to deactivate the sleep mode.
Välj > SETTINGS > SLEEP MODE.
SOFTWARE
Upgrading
This menu option restarts the Konftel 300Wx in upgrade mode. See page 32.
Resetting
This resets all settings to the default. Please note that this also deletes all contacts in the phonebook and all groups saved in the conference guide.
Select > SETTINGS > FACTORY RESET.
Confirm that you really want to reset the settings to default by pressing OK again, or cancel the resetting procedure by pressing .
CHECKING STATUS
If you need to check the software version or serial number.
Select >
STATUS.
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UPGRADING SOFTWARE
The Konftel 300Wx is continually being developed with smart features and the software can be upgraded to ensure the unit is always running the latest software version.
UPGRADING FROM A PC
Installing Konftel Upgrade utility
Follow the instructions given by the installation program.
We recommend that you open and read the installation instructions document (PDF).
Restart the Konftel 300Wx in upgrade mode
Select > SETTINGS > UPGRADE MODE.
Confirm that you want to restart in upgrade mode by pressing
OK or cancel the restart procedure by pressing .
You can also restart the unit in upgrade mode by holding down the key while the
Konftel 300Wx is being started – i.e. the battery is inserted.
Updating software
Follow the instructions that the software guide gives you.
We recommend that you open and read the installation instructions document (PDF).
We recommend that you let Konftel Upgrade utility automatically download and install the latest version from Konftel’s website. You can also install a Konftel firmware file that has already been downloaded to the PC’s hard drive.
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TECHNICAL SPECIFICATIONS
Size
Weight
Colour
Display screen
Keypad
Anti-theft protection
Memory
Upgrades
Diameter 240 mm, height 77 mm
1 kg
Liquorice black
Illuminated graphics (LCD), 128x64
Alphanumerical 0–9, *, off, mute, hold, volume up, volume down, 5 buttons for menu navigation, connectivity menu, conference guide
Kensington security slot
Support for SD memory cards up to 2 GB
The Konftel Upgrade utility upgrades Konftel 300Wx software via the USB connection
Connectivity
Wireless
PC connection
Mobile connection
Power supply
DECT / DECT 6.0, GAP-compatible
USB 2.0 Mini B
6/6 DEC
Battery: Emerging Power Li 18S, 7.2 V 5200 mAh Lithium ion
Konftel 300Wx charging cradle
Konftel charging station
AC adapter Model S024EM1400150
100–240 V AC / 14 V DC, 1.5 A
Audio
Technology OmniSound
®
HD
Microphone
Reception area
Speakers
Volume
Equalizer
Omni-directional
Up to 30 metres 2 , >10 people
Frequency band 200–3300 Hz, analogue line (DECT)
200–7000 Hz, USB VoIP
90 dB SPL 0.5 m
Three pitches: soft, neutral, bright
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TECHNICAL SPECIFICATIONS
Environment
Temperature 5°–40°C
Relative humidity 20–80% condensation free
Recommended acoustic conditions
Reverberation period: 0.5 S Rt 60
Background noise: 45 dBA
KONFTEL DECT BASE STATION (ACCESSORIES)
Power supply AC adapter ZDA060060m-N (m=US, EU, BS)
100–240 V AC / 6 V DC, 600 mA
Cable length
Tele connection
Compatibility
Dimensions
Weight
2 metres
Analogue PBX or PSTN, RJ-11
A Konftel DECT base station is GAP-compatible and can be connected to analogue lines or any PBX equipped with an analogue extension.
Up to 7 GAP-compatible DECT units can be connected to a
Konftel DECT base station, EU version
Up to 4 GAP-compatible DECT units can be connected to a
Konftel DECT base station, US version
175x160 mm
240 grams
34
SERVICE AND GUARANTEE
If anything is wrong with your Konftel unit, please contact the place of purchase.
Guarantee
We give a two-year guarantee on our conference telephones.
Service
Service is offered after the expiration of the guarantee. Please contact your retailer and ask for a cost estimate.
Konftel support
If you have any questions about the guarantee and service, please contact your Konftel support centre
Europe: +46(0)90-706 489 (Monday–Friday 8.00–17.00 GMT+1)
E-mail: [email protected]
USA and Canada:
+1 866-606-4728 (Monday–Friday 08.00–17.00 GMT-8)
E-mail: [email protected]
35
Konftel is a leading company and the world’s fastest growing brand, within audio conferencing equipment. Since 1988, our mission has been to help people around the world to conduct meetings despite distances. Based on our success, we know that audio conferencing is a great way to save time, money and at the same time contribute to a better environment. High audio quality is essential for efficient meetings, this is why our patented audio technology OmniSound ® , is built into all Konftel Conference phones.
The products are sold globally under the Konftel brand and our head-quarter is based in Sweden. Read more about the company and products on
www.konftel.com
www.konftel.com
Konftel AB, Box 268, SE-901 06 Umeå, Sweden
Phone: +46 90 706 489 Fax: +46 90 131 435 E-mail: [email protected]
110297 Rev 1A
Konftel IP DECT 10
Manual
Installation & Configuration
Network Deployment
Operation & Management
Technical Reference Document
©
September-2015 Konftel AB, Sweden
Konftel IP DECT 10 User Guide
Proprietary and Confidential
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Trademarks
Konftel and the combinations of its logo thereof are trademarks of Konftel AB, Sweden.
Other product names used in this publication are for identification purposes and maybe the trademarks of their respective companies.
Disclaimer
The contents of this document are provided in connection with Konfel products. Konftel makes no representations with respect to completeness or accuracy of the contents of this publication and reserves the right to make changes to product descriptions, usage, etc., at any time without notice. No license, whether express, implied, to any intellectual property rights are granted by this publication
Confidentiality
This document should be regarded as confidential, unauthorized copying is not allowed
©
September-2015 Konftel AB, Sweden, All rights reserved http://www.konftel.com
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Contents
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1 About This Document
This document describes the configuration, customization, management, operation, maintenance and troubleshooting of Konftel IP DECT 10. For customer specific modes refer to specific customer agreements, which describe the software operational deviations from this document.
1.1 Audience
Who should read this guide? First, this guide is intended for networking professionals responsible for designing and implementing Konftel IP DECT 10.
Second, network administrators and IT support personnel that need to install, configure, maintain and monitor elements in a “live” VoIP network will find this document helpful.
1.2 When Should I Read This Guide
Read this guide before you install the core network devices of VoIP System and when you are ready to setup or configure SIP server, NAT aware router, advanced VLAN settings, base stations, and multi cell setup.
This manual will enable you to set up components in your network to communicate with each other and also deploy a fully functionally VoIP System.
1.3 Important Assumptions
This document was written with the following assumptions in mind:
1) You have understanding of network deployment in general
2) You have working knowledge of basic TCP/IP/SIP protocols, Network Address Translation, etc...
3) A proper site survey has been performed, and the administrator have access to these plans
1.4 What’s Inside This Guide
We summarize the contents of this document in the table below:
Where Is It?
Chapter 2
Chapter 3
Chapter 4
Chapter 5
Chapter 6
Chapter 7
Content
Introduction to the VoIP
Network
Installation of Base station/Repeater
Making Handsets Ready
VoIP Administration
Interface
Registration Management -
Handsets
Firmware
Upgrade/Downgrade
Management
Purpose
To gain knowledge about the different elements in a typical VoIP Network
Considerations to remember before unwrapping and installing base units and repeaters
To determine precautions to take in preparing handsets for use in the system
To learn about the Configuration Interface and define full meaning of various parameters needed to be setup in the system.
Learn how to register handset and extensions to base stations
Provides the procedure of how to upgrade firmware to base stations and/or handsets and/or repeaters
Chapter 8
System Functionality
Overview
To gain detail knowledge about the system features.
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9 Appendix A
Basic Network Servers
Configuration
10 Appendix B VLAN Setup Management
11 Appendix C Local central directory file handling
To learn about operating the handset and base stations including detail description of handset MMI.
Examines how to setup VLAN in the network
Detailed description of central directory file format and upload.
1.5 What’s Not in This guide
This guide provides overview material on network deployment, how-to procedures, and configuration examples that will enable you to begin configuring your VoIP System.
It is not intended as a comprehensive reference to all detail and specific steps on how to configure other vendor specific components/devices needed to make the VoIP System functional. For such a reference to vendor specific devices, please contact the respective vendor documentation.
1.6 Abbreviations
For the purpose of this document, the following abbreviations hold:
DHCP:
DNS:
HTTP(S):
(T)FTP:
IOS:
PCMA:
PCMU:
PoE:
RTP:
RPORT:
SIP:
VLAN:
TOS:
URL:
UA:
Dynamic Host Configuration Protocol
Domain Name Server
Hyper Text Transfer Protocol (Secure)
(Trivial) File Transfer Protocol
Internetworking Operating System
A-law Pulse Code Modulation mu-law Pulse Code Modulation
Power over Ethernet
Real-time Transport Protocol
Response Port (Refer to RFC3581 for details)
Session Initiation Protocol
Virtual Local Access Network
Type of Service (policy based routing)
Uniform Resource Locator
User Agent
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1.7 Document History
Revision Author Issue Date
2.4 KMR 9-Sep-2015
Comments
First version
1.8 Documentation Feedback
We always strive to produce the best and we also value your comments and suggestions about our documentation. If you have any comments about this guide, please enter them through the Feedback link on the Konftel website. We will use your feedback to improve the documentation.
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2 Introduction – System Overview
In a typical telephony system, the network setup is the interconnection between Phones, “fat” routers, repeaters, portable parts, etc. The back-bone of the network depends on the deployment scenario but a ring or hub topology is used. The network has centralized monitoring, and maintenance system.
The IP DECT 10 is a single cell VoIP solution with support of up to 20 registered handsets (eg. Konftel
300Wx).
2.1 Hardware Setup
IP DECT 10 network hardware setup can deployed as follows:
The base-stations are mounted on walls or lamp poles so that each base-station is separated from each other by up to 10m indoor.
The base-station antenna mechanism is based on space diversity feature which improves coverage. The base-stations uses complete DECT MAC protocol layer and IP media stream audio encoding feature to provide up to 6 simultaneous calls.
2.2 Components of IP DECT 10
IP DECT 10 is made up of (but not limited to) the following components:
• One IP DECT 10 is connected over an IP network and using DECT as air-core interface.
• Konftel 300Wx
• Web configuration Interface; is a management interface for IP DECT 10 Wireless Solution.
2.2.1 IP DECT 10
The Base Station converts IP protocol to DECT protocol and transmits the traffic to and from the end-nodes
(i.e. wireless handsets) over a channel. It has 6 available channels.
2.2.2 VoIP Administration Server/Software
This server is referred to as VoIP Configuration Interface.
The VoIP Configuration Interface is a web based administration page used for configuration and programming of the base station and relevant network end-nodes. E.g. handsets can be registered or deregistered from the system using this interface.
The configuration interface can be used as a setup tool for software or firmware download to base stations, repeaters and handsets. Further, it is used to check relevant system logs that can be useful to administrator. These logs can be used to troubleshoot the system when the system faces unforeseen operational issues.
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2.3 Wireless Bands
The bands supported in the VoIP are summarized as follows:
Frequency bands: 1880 – 1930 MHz (DECT)
1880 – 1900 MHz (10 carriers) Europe/ETSI
1910 – 1930 MHz (10 carriers) LATAM
1920 – 1930 MHz (5 carriers) US
2.4 System Capacity (in Summary)
network capacity of relevant components can be summarised as follows:
Capacity Description
Single Cell Setup
Max ## of Users (SIP registrations) per Base
Single Cell Setup: Max ## Simultaneous Calls
1
20
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3 Installation of Base Station
In the following we briefly describe the how to install the base station in this chapter.
3.1 Package – Contents/Damage Inspection
Before Package Is Opened:
Examine the shipping package for evidence of physical damage or mishandling prior to opening. If there is a proof of mishandling prior to opening, you must report it to the relevant support centre of the regional representative or operator.
Contents of Package:
Make sure all relevant components are available in the package before proceeding to the next step.
Every shipped base unit package/box contains the following items:
• 1 x Cat. 5 cable (Ethernet cable)
• Base unit
• Power supply
Damage Inspection:
The following are the recommended procedure for you to use for inspection:
1. Examine all relevant components for damage.
2. Make a “defective on arrival – DOA” report or RMA to the operator. Do not move the shipping carton until it has been examined by the operator. If possible send pictures of the damage. The operator/regional representative will initiate the necessary procedure to process this RMA. They will guide the network administrator on how to return the damaged package if necessary.
3. If no damage is found then unwrap all the components and dispose of empty package/carton(s) in accordance with country specific environmental regulations.
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3.2 Base station Mechanics
The base station front end shows an LED indicator that signals different functional states of the base unit and occasionally of the overall network. The indicator is off when the base unit is not powered.
The table below summarises the various LED states:
LED State
Unlit
State
No power in unit
Error condition
Unlit/Solid red
Blinking green
Solid red
Blinking red
Solid green
Blinking red
Solid red
Initialisation
Factory reset warning or long press in BS reset button
Factory setting in progress
Ethernet connection available (Normal operation)
Ethernet connect not available OR handset de/registration failed
Critical error (can only be identified by Konftel Engineers). Symptoms include no system/SIP debug logs are logged, etc.
Orange
Blinking orange
Press reset button of base station.
No IP address received
3.3 Base Station – Reset feature
It is possible to restart or reset the base station unit by pressing a knob at the rear side of the unit.
Alternatively, it can be reset from the web configuration Interface.
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3.4 Installing the Base Station
First determine the best location that will provide an optimal coverage taking account the construction of the building, architecture and choice of building materials.
Next, mount the Base Station on a wall to cover range between 50 – 300 meters (i.e. 164 to 984 feet), depending whether it’s an indoor or outdoor installation.
3.4.1 Mounting the Base Stations:
We recommend the base station be mounted an angle other than vertical on both concrete/wood/plaster pillars and walls for optimal radio coverage. Avoid mounting the base units upside down as it significantly reduces radio coverage.
Mount the base unit as high as possible to clear all nearby objects (e.g. office cubicles and cabinets, etc.).
Occasionally extend coverage to remote offices/halls with lower telephony users by installing Repeaters.
Make sure that when you fix the base stations with screws, the screws do not touch the PCB on the unit.
Secondly, avoid all contacts with any high voltage lines.
3.5 Find IP of Base Station
To find IP of the installed base station two methods can be used; Using Konftel 300Wx status page or using a web browser.
3.5.1 Using Konftel 300Wx to find IP address
On the handset press “Menu” then go to STATUS. Scroll to the bottom of the status view to see the IP address of the base. This can only be found when the handset (Konftel 300Wx) is connected to the base.
3.5.2 Using browser
Open any standard browser and enter the address: http://ipdect <MAC-Address-Base-Station> for e.g. http://ipdect 00087B00AA10. This will retrieve the HTTP Web Server page from the base station with hardware address 00087B00AA10.
This feature requires an available DNS server.
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3.6 Login to Base Configuration Interface
STEP 1
Connect the Base station to a private network via standard Ethernet cable (CAT-5).
STEP 2
Use section 3.5 to find the IP address and enter that in a web browser.
STEP 3
On the Login page, enter your authenticating credentials (i.e. username and password). By default the username and password is admin. Click OK button.
STEP 4
Once you have authenticated, the browser will display front end of the Configuration
Interface. The front end will show relevant information of the base station.
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4 VoIP Administration Interface
The VoIP Administration Interface is the main interface through which the system is managed and debugged.
The VoIP Configuration Interface is an in-built HTTP Web Server service residing in each base station. This interface is user friendly interface and easy to handle even to a first time user.
Note: Enabling secure web is not possible. For secure configuration use secure provisioning.
This chapter seeks to define various variables/parameters available for configuration in the network.
4.1 Web navigation
We describe the left menu in the front end of the VoIP Administration Interface.
Administration of extensions and handsets in the system
Feature Description
Home/Status This is the front end of the Base station’s HTTP web interface. This page shows the summary of current operating condition and settings of the Base station and
Handset(s).
Extensions
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Servers
Country
Security
On this page the user can define which SIP/NAT server the network should connect to.
Network
Typically the user configures the Network settings from here.
NAT provisioning: allows configuration of features for resolving of the NAT –
Network Address Translation. These features enable interoperability with most types of routers.
DHCP: allows changes in protocol for getting a dynamic IP address.
Virtual LAN: specifies the Virtual LAN ID and the User priority.
IP Mode: specifies using dynamic (DHCP) or static IP address for your network. IP
address: if using DHCP leave it empty. Only write in, when you use static IP address.
Subnet mask: if using DHCP, leave it empty. Only write in, when you use static IP address.
DNS server: specify if using DHCP, leave it empty. Only write in the DNS server address of your Internet service provider, when you use static IP address. (DNS =
Dynamic Name Server)
Default gateway: if using DHCP, leave it empty. Write in the IP address of your router, when you use static IP address.
Management Defines the Configuration server address, Management transfer protocol, sizes of logs/traces that should be catalogued in the system.
Firmware
Update
Time
Remote firmware updates (HTTP(s)/TFTP) settings of Base stations and handsets.
Here the user can configure the Time server. It should be used as time server in relevant country for exact time. The time servers have to deliver the time to conform to the Network Time Protocol (NTP). Handsets are synchronised to this time. Base units synchronise to the master using the Time server.
Specifying the country/territory where the network is located ensures that your phone connection functions properly.
Note: The base language and country setting are independent of each other.
The users can administrate certificates and create account credentials with which they can log in or log out of the embedded HTTP web server.
Central
Directory
Repeaters
Alarm
Syslog
SIP Log
Interface to common directory load of up to 3000 entries using *csv format or configuration of LDAP directory.
Note: LDAP and central directory cannot operate at the same time.
Administration and configuration of repeaters of the system
Administration and configuration of the alarm settings on the system. This controls the settings for alarms that can be sent to the handsets. This feature is only available on certain types of handsets.
Statistics
Overview of system and call statistics for a system.
Configuration This shows detail and complete network settings for base station(s),
HTTP/DNS/DHCP/TFTP server, SIP server, etc.
Overall network related events or logs are displayed here (only live feed is shown).
SIP related logs can be retrieved from url link. It is also possible to clear logs from this feature.
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4.2 Home/Status
We describe the parameters found in the Welcome front end home/status of the VoIP Administration
Interface.
Screenshot
Parameter
System information
Phone Type
System Type
RF Band
Current local time
Operation time
RFPI-Address
MAC-Address
IP-Address
Firmware version
Firmware URL
Base Station Status
SIP Identity Status on
this Base Station
Description
This base current multi-cell state
Always IPDECT
This base customer configuration
This base RF band setting
This base local time
Time from last boot of base
This base RFPI address
This base MAC address
This base IP address
This base firmware version
Firmware update server address and firmware path on server
“Idle” : When no calls on base
“In use” : When active calls on base
List of extensions present at this base station.
Format: “extension”@“this base IP address” followed by status to the
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Reboot right. Below is listed possible status:
OK: Handset is ok
SIP Error: SIP registration error
Reboot after all connections is stopped on base. Connections are active call, directory access, firmware update active
Reboot immediately even active calls are ongoing.
Forced Reboot
4.3 Extensions
In this section, we describe the different parameters available whenever the administrator is creating extensions for handsets. Note, it is not possible to add extensions if no servers are defined. As well the section describes the group call feature.
The system can handle maximum 20 extensions matching 20 handsets which can be divided between servers. When 20 handsets are registered it is not possible to add more extensions.
Note: Within servers or even with multi servers, extensions must always be unique. This means same extension number on server 1 cannot be re-used on server 2.
4.3.1 Group call
Call Group is a SIP extension where multiple handsets are associated. All handsets that subscribes to a given extension (and hence Call Group) can receive incoming calls and initiate outgoing calls on the given extension. It is possible for any handset to perform any call action which is possible without the Call Group feature. That is, call actions as Hold, Transfer etc. are possible if the PBX supports them.
When an incoming call arrives to a given Call Group, all Call Group subscribed handsets will alert. Thus, if a
Call Group contains 20 handsets, all 20 handset will alert.
An alerting handset cannot receive another incoming call, and therefore if a handset subscribes for multiple
Call Groups, and a call arrives for a 2 nd
Call Group while the handset is alerting, the handset will not receive this call. If DND is enabled for a given handset, it will not receive the incoming call.
For outgoing calls, it can be selected in the handset which line (i.e. Call Group) to use for the call. The maximum number of lines is 20. For any outgoing actions, the settings for the selected line (SIP extension) will be used.
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4.3.2 Add extension
Screenshot
Mailbox Name Empty
Mailbox
Number
Server
Call waiting
feature
Forwarding
Unconditional
Empty
Server 1 IP
Enabled
Empty
Parameter
Extension
Default Value(s) Description
Empty Handset phone number depending on the setup.
Possible value(s): 8-bit string length
Example: 1024, etc.
Note: The Extension must also be configured in SIP server in order for this feature to function.
Authentication
User Name
Authentication
Password
Empty
Empty
Display Name
Empty
Username: SIP authentication username
Permitted value(s): 8-bit string length
Password: SIP authentication password.
Permitted value(s): 8-bit string length
Human readable name used for the given extension
Permitted value(s): 8-bit string length
Name of centralised system used to store phone voice messages that can be retrieved by recipient at a later time.
Valid Input(s): 8-bit string Latin characters for the Name
Dialled mail box number by long key press on key 1.
Valid Input(s): 0 – 9, *, #
Note: Mailbox Number parameter is available only when it’s enabled from SIP server.
FQDN or IP address of SIP server.
Drop down menu to select between the defined Servers of VoIP
Service provider.
Used to enable/disable Call Waiting feature. When disabled a second incoming call will be rejected. If enabled a second call will be presented as call waiting.
Number to which incoming calls must be re-routed to irrespective of the current state of the handset.
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Number
Forwarding
No Answer
Number
Disabled
Empty
Disabled
90
Forwarding Unconditional must be enabled to function.
Note: Feature must be enabled in the SIP server before it can function in the network
Note: Feature will be automatically disabled in case the handset or extension is part of a group
Number to which incoming calls must be re-routed to when there is no response from the SIP end node.
Forwarding No Answer Number must be enabled to function.
Note: Feature must be enabled in the SIP server before it can function in the network
Specify delay from call to forward in seconds.
Note: Feature will be automatically disabled in case the handset or extension is part of a group
Forwarding
On Busy
Number
Empty
Disabled
Number to which incoming calls must be re-routed to when SIP node is busy.
Forwarding On Busy Number must be enabled to function.
Note: Feature must be enabled in the SIP server before it can function in the network
Note: Feature will be automatically disabled in case the handset or extension is part of a group
When an extension is added (or edited) it can be selected (right side check box) which handsets shall subscribe to the given extension, and hence be a part of this call group, see above figure. It is also possible to choose to add a new handset entry at this point, and if this is done, DECT registration for the new entry can be enabled afterwards on the handsets subpage
4.3.3 Extensions list
The added extensions will be shown in the extension lists.
The list can be sorted by any of the top headlines, by mouse click on the headline link.
Screenshot
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Parameter
Idx
Description
Select / deselect for delete, register and deregister handsets
Extension
Given extension is displayed.
Display Name
Given display name is displayed. If no name given this field will be empty
Server IP or URL
Server
Server Alias
State
IPEI
Given server alias is displayed. If no alias given this field will be empty.
SIP registration state – if empty the handset is not SIP registered.
Handset IPEI. IPEI is a unique DECT identification number.
Group call: One extension can be associated to up to 20 IPEI’s. The IPEI’s will be listed in this cell.
4.3.4 Handset list
The added handsets will be shown in the handset lists.
The list can be sorted by any of the top headlines, by mouse click on the headline link.
Screenshot
Parameter
Idx
Description
Select / deselect for delete, register and deregister handsets
IPEI
Handset IPEI. IPEI is unique DECT identification number.
Handset state The state of the given handset:
Present: The handset is DECT located at the base
Detached: The handset is detached from the system (e.g. powered off)
Removed: The handset has been out of sight for a specified amount of time (~one hour).
Handset Type
FW info
Handset type and firmware version of handset
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FWU Progress Possible FWU progress states:
Off: Means sw version is specified to 0 = fwu is off
Initializing: Means FWU is starting and progress is 0%.
X% : FWU ongoing
Verifying X%: FWU writing is done and now verifying before swap
”Waiting for charger” (HS) / ”Conn. term. wait” (Repeater): All FWU is complete and is now waiting for handset/repeater restart.
Complete HS/repeater: FWU complete
Error: Not able to fwu e.g. file not found, file not valid etc
Extension
Given extension is displayed.
Group call: The cell will show all the extensions associated with this handset and IPEI.
4.3.4.1 Handset and extension list top/sub-menus
The handset extension list menu is used to control paring or deletion of handset to the system (DECT registration/de-registrations) and to control SIP registration/de-registrations to the system.
Above and below the list are found commands for making operations on handsets/and extensions. The top menu is general operations, and the sub menu is always operating on selected handsets/extensions.
Screenshots
Deregister
Handset(s)
In the below table each command is described.
Actions
Delete Handset(s)
Description
Deregister selected handset(s), but do not delete the extension(s).
Register Handset(s) Enable registration mode for the system making it possible to register at a specific extension (selected by checkbox)
Deregister the selected handset(s) and delete the extension(s).
4.3.5 Edit Extension
To edit extension use the mouse to click the link of the extension.
Edit extension will open the same configuration possibilities as add extension. Refer to the above add extension section.
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4.4 Servers
In this section, we describe the different parameters available in the Servers configurations menu.
Maximum 10 servers can be configured.
Screenshot
Parameter Default value
Server Alias Empty
NAT
Adaption
Registrar
Outbound
Proxy
Disabled
Empty
Empty
Description
Parameter for server alias
To ensure all SIP messages goes directly to the NAT gateway in the SIP aware router.
SIP Server proxy DNS or IP address
Permitted value(s): AAA.BBB.CCC.DDD:<Port-Number>
or <URL>:<Port-Number>
Note: Specifying the Port Number is optional.
This is a Session Border Controller DNS or IP address (OR
SIP server outbound proxy address)
Set the Outbound proxy to the address and port of
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Conference
Server
Call Log
Server
Reregistration time
SIP Session
Timers:
Empty
Empty
600
Disabled private NAT gateway so that SIP messages sent via the
NAT gateway.
Permitted value(s): AAA.BBB.CCC.DDD or <URL> or
<URL>:<Port-Number>
Examples: “192.168.0.1”, “192.168.0.1:5062”,
“nat.company.com” and “sip:[email protected]:5065”.
Broadsoft conference feature.
Set the IP address of the conference server.
In case an IP is specified pressing handset conference will establish a connection to the conference server.
If the field is empty the original 3-party local conference of 8660 is used.
Broadsoft call log feature.
Set the IP address of the XSI call log server.
In case an IP is specified pressing handset will use the call log server.
If the field is empty the local call log is used
The “expires” value 24nalyse24n in SIP REGISTER requests. This value indicates how long the current SIP registration is valid, and hence is specifies the maximum time between SIP registrations for the given SIP account.
Permitted value(s): A value below 60 sec is not recommended, Maximum value 65636
RFC 4028. A “keep-alive” mechanism for calls. The session timer value specifies the maximum time between “keep-alive” or more correctly session refresh signals. If no session refresh is received when the timer expires the call will be terminated. Default value is 1800 s according to the RFC. Min: 90 s. Max: 65636.
If disabled session timers will not be used.
Default value is 1800s according to the RFC.
If disabled session timers will not be used.
Permitted value(s): Minimum value 90, Maximum 65636
Select UDP, TCP, TLS 1.0
Session
Timer
Values (s):
SIP
Transport
Signal TCP
Source Port
1800
UDP
Use One
TCP/TLS
Connection per SIP
Extension:
Disabled
Disabled
When SIP Transport is set to TCP or TLS, a TCP (or TLS) connection will be established for each SIP extension.
The source port of the connection will be chosen by the
TCP stack, and hence the local SIP port parameter,
specified within the SIP/RTP Settings (see 5.5.5) will not
be used. The “Signal TCP Source Port” parameter specifies if the used source port shall be signaled explicitly in the SIP messages.
When using TCP or TLS as SIP transport, choose if a
TCL/TLS connection shall be established for each SIP extension or if the base station shall establish one connection which all SIP extensions use. Please note that if TLS is used and SIP server requires client authentication (and requests a client certificate), this setting must be set to disabled.
0: Disabled. (Use one TCP/TLS connection for all SIP
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Keep Alive
Enabled extensions)
1: Enabled. (Use one TCP/TLS connection per SIP extensions).
This directive defines the window period (30 sec.) to keep opening the port of relevant NAT-aware router(s), etc.
If enabled extension will be shown on handset idle screen.
Show
Extension on Handset
Idle Screen
Hold
Behaviour
Attended
Transfer
Behaviour
Use Own
Codec
Priority
DTMF
Signalling
Enabled
RFC 3264
Hold 2 nd
Disabled
Call
RFC 2833
Specify the hold behaviour by handset hold feature.
RFC 3264: Hold is 25nalyse25n according to RFC 3264, i.e. the connection information part of the SDP contains the IP Address of the endpoint, and the direction attribute is sendonly, recvonly or inactive dependant of the context
RFC 2543: The ”old” way of 25nalyse25ng HOLD. The connection information part of the SDP is set to 0.0.0.0, and the direction attribute is sendonly, recvonly or inactive dependant of the context
When we have two calls, and one call is on hold, it is possible to perform attended transfer. When the transfer soft key is pressed in this situation, we have traditionally also put the active call on hold before the
SIP REFER request is sent. However, we have experienced that some PBXes do not expect that the 2nd call is put on hold, and therefore attended transfer fails on these PBXes.
The "Attended Transfer Behaviour" feature defines whether or not the 2nd call shall be put on hold before the REFER is sent.
If "Hold 2nd Call" is selected, the 2nd call will be held before REFER is sent.
If "Do Not Hold 2nd Call" is selected, the 2nd call will not be held before the REFER is sent
Default disabled.
By enable the system codec priority during incoming call is used instead of the calling party priority.
E.g. If base has G722 as top codec and the calling party has Alaw on top and G722 further down the list, the
G722 will be chosen as codec for the call.
Conversion of decimal digits (and ‘*’ and ‘#’) into sounds that share similar characteristics with voice to easily traverse networks designed for voice
SIP INFO: Carries application level data along SIP signalling path (e.g.: Carries DTMF digits generated during SIP session OR sending of DTMF tones via data packets in the same internet layer as the Voice Stream, etc.).
RFC 2833: DTMF handling for gateways, end systems and RTP trunks (e.g.: Sending DTMF tones via data
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DTMF
Payload
Type
Codec
Priority
101
G.722
G.711U
G.711A
G.726 packets in different internet layer as the voice stream)
Both: Enables SIP INFO and RFC 2833 modes.
This feature enables the user to specify a value for the
DTMF payload type / telephone event (RFC2833).
Defines the codec priority that base stations uses for audio compression and transmission.
Possible Option(s): G.711U,G.711A, G.726, G.729, G.722.
Note: Modifications of the codec list must be followed by a “reset codes” and “Reboot chain” on the multipage in order to change and update handsets.
Note:
With G.722 as first priority the number of simultaneous calls per base station will be reduced from 10 (8) to 4 calls.
With G.722 in the list the codec negotiation algorithm is active causing the handset (phone) setup time to be slightly slower than if G.722 is removed from the list.
With G.729 add on DSP module for the base is required.
RTP Packet size
20ms The packet size offered as preferred RTP packet size by
8630 when RTP packet size negotiation.
Selections available: 20ms, 40ms, 60ms, 80ms
Secure RTP
Secure RTP
Auth
Disabled
Disabled
With enable RTP will be encrypted (AES-128) using the key negotiated via the SDP protocol at call setup.
With enable secure RTP is using authentication of the
RTP packages.
Note: with enabled SRTP authentication maximum 4 concurrent calls is possible per base in a single or multicell system.
SRTP
Crypto
Suites
AES_CM_128_HMAX_SHA1_32
AES_CM_128_HMAX_SHA1_80
Field list of supported SRTP Crypto Suites. The device is born with two suites.
Note: Within servers or even with multi servers, extensions must always be unique. This means same extension number on server 1 cannot be re-used on server 2.
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4.5 Network
In this section, we describe the different parameters available in the network configurations menu.
4.5.1 IP Settings
Screenshot
Description Parameter
Default
Values
DHCP/Static IP DHCP
IP Address
Subnet Mask
Default
Gateway
NA
NA
NA
DNS (Primary) NA
If DHCP is enabled, the device automatically obtains TCP/IP parameters.
Possible value(s): Static, DHCP
DHCP: IP addresses are allocated automatically from a pool of leased address.
Static IP: IP addresses are manually assigned by the network administrator.
If the user chooses DHCP option, the other IP settings or options are not available.
32-bit IP address of device (e.g. base station). 64-bit IP address will be supported in the future.
Permitted value(s): AAA.BBB.CCC.DDD
Is device subnet mask.
Permitted value(s): AAA.BBB.CCC.DDD
This is a 32-bit combination used to describe which portion an IP address refers to the subnet and which part refers to the host.
A network mask helps users know which portion of the address identifies the network and which portion of the address identifies the node.
Device’s default network router/gateway (32-bit).
Permitted value(s): AAA.BBB.CCC.DDD e.g. 192.168.50.0
IP address of network router that acts as entrance to other network.
This device provides a default route for TCP/IP hosts to use when communicating with other hosts on hosts networks.
Main server to which a device directs Domain Name System (DNS) queries.
Permitted value(s): AAA.BBB.CCC.DDD or <URL>
This is the IP address of server that contains mappings of DNS domain names to various data, e.g. IP address, etc.
The user needs to specify this option when static IP address option is chosen.
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DNS
(Secondary)
NA This is an alternate DNS server.
4.5.2 VLAN Settings
Enable users to define devices (e.g. Base station, etc.) with different physical connection to communicate as if they are connected on a single network segment.
The VLAN settings can be used on a managed network with separate Virtual LANs (VLANs) for sending voice and data traffic. To work on these networks, the base stations can tag voice traffic it generates on a specific
“voice VLAN” using the IEEE 802.1q specification.
Screenshot
Parameter
VLAN id
Default Values Description
0 Is a 12 bit identification of the 802.1Q VLAN.
Permitted value(s): 0 to 4094 (only decimal values are accepted)
0
A VLAN ID of 0 is used to identify priority frames and ID of 4095 (i.e.
FFF) is reserved.
Null means no VLAN tagging or No VLAN discovery through DHCP.
This is a 3 bit value that defines the user priority.
VLAN User
Priority
Values are from 0 (best effort) to 7 (highest); 1 represents the lowest priority. These values can be used to prioritize different classes of traffic (voice, video, data, etc).
Permitted value(s): 8 priority levels (i.e. 0 to 7)
For further help on VLAN configuration refer to Appendix.
4.5.3 DHCP Options
Screenshot
Parameter Default Values Description
Plug-n-Play Enabled
Enabled: DHCP option 43 to automatically provide PBX IP address to base.
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4.5.4 NAT Settings
We define some options available when NAT aware routers are enabled in the network.
Screenshot
This defines the frequency of how keep-alive are sent to maintain
NAT bindings.
Parameter
Enable STUN
STUN Server
Default Values Description
Disabled Enable to use STUN
NA Permitted value(s): AAA.BBB.CCC.DDD (Currently only Ipv4 are supported) or url
STUN Bindtime
Determine
STUN Bindtime
Guard
Enabled
80
Enable RPORT Disabled
Keep alive time 90
Permitted values: Positive integer default is 90, unit is in seconds
Enable to use RPORT in SIP messages.
Permitted values: Positive integer default is 90, unit is in seconds
4.5.5 SIP/RTP Settings
These are some definitions of SIP/RTP settings:
Screenshot
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Parameter
Use Different
SIP Ports
Default Values Description
Disabled If disabled, the Local SIP port parameter specifies the source port used for SIP signalling in the system.
If enabled, the Local SIP Port parameter specifies the source port used for first user agent (UA) instance. Succeeding UA’s will get succeeding ports.
RTP
Collision
Detection
Local SIP
Enabled
5060
port
SIP ToS/QoS 0x68
RTP port
RTP port
range
RTP
TOS/QoS
50004
40
0xB8
The source port used for SIP signalling
Permitted values: Port number default 5060.
Priority of call control signalling traffic based on both IP Layers of
Type of Service (ToS) byte. ToS is referred to as Quality of Service
(QoS) in packet based networks.
Permitted values: Positive integer, default is 0x68
The first RTP port to use for RTP audio streaming.
Permitted values: Port number default 50004 (depending on the setup).
The number of ports that can be used for RTP audio streaming.
Permitted values: Positive integers, default is 40
Priority of RTP traffic based on the IP layer ToS (Type of Service) byte.
ToS is referred to as Quality of Service (QoS) in packet based networks.
See RFC 1349 for details. “cost bit” is not supported.
o Bit 7..5 defines precedence.
o Bit 4..2 defines Type of Service.
o Bit 1..0 are ignored.
Setting all three of bit 4..2 will be ignored.
Permitted values: Positive integer, default is 0xB8
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4.6 Management Settings Definitions
The administrator can configure base stations to perform some specific functions such as configuration of file transfers, firmware up/downgrades, password management, and SIP/debug logs.
Screenshot
Parameter
Base Station
Name:
Management
Transfer
Protocol
HTTP
Management
upload script
Default value Description
VoIP It indicates the title that appears at the top window of the browser and is used in the multicell page.
TFTP
Empty
The protocol assigned for configuration file and central directory
Valid Input(s): TFTP, HTTP, HTTPs
The folder location or directory path that contains the configuration files of the Configuration server. The configuration upload script is a file located in e.g. TFTP server or Apache Server which is also the configuration server.
Permitted value(s): /<configuration-file-directory>
Example: /CfgUpload
Note: Must begin with (/) slash character. Either / or \ can be used.
Empty Password that should be entered in order to have access to the
HTTP
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Management
password
Configuration server
address
Base Specific
File
Configuration
File Download
DHCP
Controlled
Config Server
DHCP Custom
Option
DHCP Custom
Option Typr
Text
Messaging
Empty
Empty
Empty
Empty
Disabled
Disabled
Disabled configuration server.
Permitted value(s): 8-bit string length
Server/device that provides configuration file to base station.
Type: DNS or IP address
Permitted value(s): AAA.BBB.CCC.DDD or <URL>
Base configuration file
Base Specific file: Used when configuring a single cell base
Multicell Specific File: Used when configuring a multicell based system
Base and Multicell Specific File: Used on out of factory bases to specify VLAN and Multicell ID and settings.
Provisioning server options.
DHCP Option 66: Look for provision file by TFTP boot up server.
DHCP Custom Option: Look for provision file by custom option
DHCP Custom Option & Option 66: Look for provision file by first custom option and then option 66.
By default option 160, but custom option can be defined.
An option 160 URL defines the protocol and path information by using a fully qualified domain name for clients that can use DNS.
URL: URL of server with path.
Example of URL: http://myconfigs.com:5060/configs
Default configuration file on server must follow the name: MAC.cfg
IP Address: IP of server with path.
Disable/enable messaging with Mobicall server
The third option is to “Enable Without Server”. With this setting handset can send messages to other handsets, which support messaging.
Note: Contact Mobicall to get the proper version and setup for
Mobicall server
Permitted value(s): AAA.BBB.CCC.DDD or <URL>
Text
Messaging &
Alarm server
Text
Messaging
Port
Text
Messaging
Keep Alive
(m)
Text
Messaging
Response (s)
Text
Messaging
TTL
SIP Log Server
Address
Upload of SIP
Log
Empty
1300
30
30
0
Empty
Disabled
Port number of message server.
This defines the frequency of how keep-alive are sent
Permitted values: Positive integer, unit is in minutes
This defines the frequency of how response timeout
Permitted values: Positive integer, unit is in seconds
This defines the text messaging time to live
Permitted values: Positive integer, unit is in seconds
Permitted value(s): AAA.BBB.CCC.DDD or <URL>
Requires a predefined folder named: \SIP
Enable this option to save low level SIP debug messages to the server.
The SIP logs are saved in the file format:
<MAC_Address><Time_Stamp>SIP.log
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Syslog Server
IP-Address
Syslog Server
Port
Syslog Level
NA
NA
Permitted value(s): AAA.BBB.CCC.DDD or <URL>
Port number of syslog server.
Off Off: No data is saved on syslog server
Normal Operation: Normal operation events are logged, incoming call, outgoing calls, handset registration, DECT location, and call lost due to busy, critical system errors, general system information.
System Analyze: Handset roaming, handset firmware updates status.
The system 33nalyse level also contains the messages from normal operation.
Debug: Used by IP DECT 10 for debug. Should not be enabled during normal operation.
Enable
Automatic
Prefix
Disabled Disabled: Feature off.
Enabled: The base will add the leading digit defined in “Set Prefix for
Outgoing Calls”.
Enabled + fall through on * and #: Will enable detection of * or # at the first digit of a dialled number. In case of detection the base will not complete the dialled number with a leading 0.
Examples:
1: dialed number on handset * 1234 - > dialed number to the pabx
*1234
2: dialed number on handset #1234 - > dialed number to the pabx
#1234
3: dialed number on handset 1234 - > dialed number to the pabx
01234
Used to detect internal numbers. In case of internal numbers no prefix number will be added to the dialled number.
Set Maximum
Digits of
Internal
Numbers
0
Set Prefix for
Outgoing Calls
Empty Prefix number for the enabled automatic prefix feature.
Permitted value(s): 1 to 9999
There are three ways of configuring the system.
1. Manual configuration by use of the Web server in the base station(s)
2. By use of configuration files that are uploaded from a disk via the “Configuration” page on the Web server.
3. By use of configuration files which the base station(s) download(s) from a configuration server.
For further details refer to doc reference [3].
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4.7 Firmware Update Definitions
In this page, the system administrator can configure how base stations and SIP nodes upgrade/downgrade to the relevant firmware. Handset firmware update status can be found in the extensions page and repeater firmware update status in the repeater page. Base firmware update status is found in the multicell page.
Screenshot
Parameter
Firmware update
server address
Default
Value(s)
Empty
Firmware path
Empty
Required Version
Type
Empty
Required Version
Base
Empty
Description
IP address or DNS of firmware update files source
Valid Inputs: AAA.BBB.CCC.DDD or <URL>
Example: 10.10.104.41
Location of firmware on server (or firmware update server path where firmware update files are located).
Example: /East_Fwu
Note: Must begin with (/) slash character
Version of firmware to be upgraded (or downgraded) on handset type or repeater.
Valid Input(s): 8-bit string length. E.g. 280
Note: Value version 0 will disable firmware upgrade for handsets and/or repeater
Note: Two handset types will be serial firmware upgraded. First type 8630 then type 8430.
Version of firmware to be upgraded (or downgraded) on Base station. Base units are referred to as gateways over here.
Valid Input(s): 8-bit string length. E.g. 280
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4.8 Time Server
In this section, we describe the different parameters available in the Time Server menu.
The Time server supplies the time used for data synchronisation in a multi-cell configuration. As such it is mandatory for a multi-cell configuration. The system will not work without a time server configured.
As well the time server is used in the debug logs and for SIP traces information pages, and used to determine when to check for new configuration and firmware files.
NOTE: It is not necessary to set the time server for standalone base stations (optional).
Press the “Time PC” button to grab the current PC time and use in the time server fields.
NOTE:
When time server parameters are modified/changed synchronisation between base stations can take up to
15 minutes before all base stations are synchronised, depending on the number of base stations in the system.
Screenshot
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Parameter
Time Server
Allow broadcast
NTP
Refresh time (h)
Set timezone by
country/region
Time Zone
DST Start Month
DST Start Date
DST Start Time
DST Start Day of
Week
DST Start Day of
Week, Last in
Month
DST Stop Month
DST Stop Date
DST Stop Time
DST Stop Day of
Week
DST Stop Day of
Week Last in
Month
Default Values Description
Empty DNS name or IP address of NTP server.
Enter the IP/DNS address of the server that distributes reference clock information to its clients including Base stations, Handsets, etc.
Valid Input(s): AAA.BBB.CCC.DDD or URL (e.g. time.server.com)
Currently only Ipv4 address (32-bit) nomenclature is supported.
Checked
Empty
Checked
0
Set DST by
country/region
Daylight Saving
Time (DST)
Checked
Disabled
DST Fixed By Day
Use Month and
Date
March
25
3
Monday
Last in Month
October
1
2
Sunday
First in Month
The window time in seconds within which time server refreshes.
Valid Inputs: positive integer
By checked country setting is used (refer to country web page).
Refers to local time in GMT or UTC format.
Min: -12:00
Max: +13:00
By checked country setting is used (refer to country web page).
The system administrator can Enable or Disable DST manually.
Automatic: Enter the start and stop dates if you select
Automatic.
You determine when DST actually changes. Choose the relevant date or day of the week, etc. from the drop down menu.
Month that DST begins
Valid Input(s): Gregorian months (e.g. January, February, etc.)
Numerical day of month DST comes to effect when DST is fixed to a specific date
Valid Inputs: positive integer
DST start time in the day
Valid Inputs: positive integer
Day within the week DST begins
Specify the week that DST will actually start.
The month that DST actually stops.
The numerical day of month that DST turns off.
Valid Inputs: positive integer (1 to 12)
The time of day DST stops
Valid Inputs: positive integer (1 to 12)
The day of week DST stops
The week within the month that DST will turn off.
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4.9 Country
The country setting controls the in-band tones used by the system. To select web interface language go to the management page.
Screenshot
Parameter
Select Country
Default Values Description
Germany Supported countries: Australia, Belgium, Brasil, Denmark,
Germany, Spain, France, Ireland, Italia, Luxembourg,
Nederland, New Zealand, Norway, Portugal, Swiss, Finland,
Sweden, Tyrkey, United Kingdom, US/Canade, Austria
State / Region
Select Language
NA
English
Only shown by country selection US/Canada, Autralia, Brasil
Web interface language. Number of available languages:
English, Dansk, Italiano, Tyrkie, Deutsch, Portuguese, Hrvatski,
Srpski, Slovenian, Nederlands, Francaise, Espanol, Russian,
Polski.
When checked timezone will follow country/region
Set timezone by
country/region
Set DST by
country/region
Notes checked checked When checked DST will follow country/region
Empty Only showing notes to time setting for countries: US/Canada,
Brasil
NOTE: By checked timezone and DST the parameters in web page Time will be discarded.
The following types of in-band tones are supported:
- Dial tone
- Busy tone
- Ring Back tone
- Call Waiting tone
- Re-order tone
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4.10 Security
The security section is used for loading of certificates and for selecting if only trusted certificates are used.
Furthermore, web password can be configured.
The Security web is divided into three sections: Certificates (trusted), SIP Client Certificates (and keys) and
Password administration.
To setup secure fwu and configuration file download select HTTPs for the Management Transfer Protocol
(refer to management web).
SIP and RTP security is server dependent and in order to configure user must use the web option Servers
(refer to servers web).
4.10.1 Certificates
The certificates list contains the list of loaded certificates for the system. Using the left column check mark it is possible to check and delete certificates. To import a new certificate use the mouse “select file” and browse to the selected file. When file is selected, use the “Load” bottom to load the certificate.
The certificate format supported is DER encoded binary X.509 (.cer).
Screenshot
Certificates list
Parameter
Idx
Issued To
Issued To
Valid Until
Screenshot
Default Values Description
Fixed indexes
Empty
Index number
IP address – which is part of the certificate file
Empty
Empty
Organisation, Company – which is part of the certificate file
Date Time Year – which is part of the certificate file
By enabling Use Only Trusted Certificates, the certificates the base will receive from the server must be valid and loaded into the system. If no valid matching certificate is found during the TLS connection establishment, the connection will fail. When Use Only Trusted Certificates is disabled, all certificates received from the server will be accepted.
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Note: It is important to use correct date and time of the system when using trusted certificates. In case of time/date not defined the certificate validation can fail
4.10.2 SIP Client Certificates
To be able to establish a TLS connection in scenarios, where the server requests a client certificate, a certificate/key pair must be loaded into the base. This is currently supported only for SIP.
To load a client certificate/key pair, both files must be selected at the same time, and it is done by pressing
“select files” under “Import SIP Client Certificate and Key Pair” and then select the certificate file as well as the key file at the same time. Afterwards, press load.
The certificate must be provided as a DER encoded binary X.509 (.cer) file, and the key must be provided as a binary PKCS#8 file.
Note: Use Chrome for loading SIP Client Certificates
Screenshot
4.10.3 Password
In the below the password parameters are defined.
Screenshot
Parameter
Username
Default Values Description
Admin Can be modified to any supported character and number
Current Password Admin
New Password
Empty
Confirm Password Empty
Can be modified to any supported character and number
Change to new password
Confirm password to reduce accidently wrong changes of passwords
Password valid special signs: @/|<>-_:.!?*+#
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Password valid numbers: 0-9
Password valid letters: a-z and A-Z
4.11 Central Directory and LDAP
The VOIP system support two types of central directories, a local central directory or LDAP directory.
For both directories caller id look up is made with match for 6 digits of the phone number.
4.11.1 Local Central Directory
Select local and save for local central directory.
Screenshot
Parameter
Local
Server
Filename
Phonebook reload
interval (s)
Default Values Description
Local Drop down menu to select between local central directory and
LDAP based central directory
Empty
Empty
0
The parameter is used if directory file is located on server.
Valid Inputs: AAA.BBB.CCC.DDD or <URL>
Refer to appendix for further details.
The parameter is used if directory file is located on server.
Refer to appendix for further details
The parameter is controlling the reload interface of phonebook in seconds. The feature is for automatic reload the base phonebook file from the server with intervals. It is recommended to specify a conservative value to avoid overload of the base station.
With default value setting 0 the reload feature is disabled.
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4.11.1.1 Import Central Directory
The import central directory feature is using a browse file approach. After file selection press the load button to load the file. The system support only the original *.csv format. Please note that some excel csv formats are not the original csv format. The central directory feature can handle up to 3000 contacts. For further details of the central directory feature refer to appendix.
4.11.2 LDAP
Select LDAP Server and save for LDAP server configuration.
Screenshot
Parameter
LDAP Server
Server
Port
Sbase
LDAP filter
Default Values Description
LDAP Server Drop down menu to select between local central directory and
LDAP based central directory. LDAP Server is displayed when
LDAP server is selected.
Empty IP address of the LDAP server.
Valid Inputs: AAA.BBB.CCC.DDD or <URL>
Empty
Empty
Empty
The server port number that is open for LDAP connections.
Search Base. The criteria depends on the configuration of the
LDAP server. Example of the setting is CN=Users, DC=umber,
DC=loc
LDAP Filter is used to as a search filter, e.g. setting LDAP filter to (|(givenName=%*)(sn=%*)) the IP-DECT will use this filter
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Bind
Password
Name
Work Number
Home Number
Mobile Number
Empty
Empty
Empty
Empty
Empty
Empty when requesting entries from the LDAP server. % will be replaced with the entered prefix e.g searching on J will give the filter (|(givenName=J*)(sn=J*)) resulting in a search for given name starting with a J or surname starting with J.
Bind is the username that will be used when the IP-DECT phone connects to the server
Password is the password for the LDAP Server
The name can be used to specify if sn+givenName or cn
(common name) is return in the LDAP search results
Work number is used to specify that LDAP attribute that will be mapped to the handset work number
Home number is used to specify that LDAP attribute that will be mapped to the handset home number
Mobile number is used to specify that LDAP attribute that will be mapped to the handset mobile number
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4.12 Statistics
The statistic feature is divided into four administrative web pages, which can be access from any base.
1. System
2. Calls
3. Repeater
4. DECT data
All four views have an embedded export function, which export all data to comma separated file.
By pressing the clear button all data in the full system is cleared.
4.12.1 System data
The system data web is access by http://ip/SystemStatistics.html
and data is organised in a table as shown in below example.
Screenshot
The table is organised with headline row, data pr. base rows and with last row containing the sum of all base parameters.
Parameters
Base Station
Description
Base IP address and base station name from management settings
Name
Operation time Total operation time for the base
Busy Count
Busy Count is the number of times the base has been busy.
Busy Duration
Busy duration is the total time a base has been busy for speech (8 or more calls active).
SIP Failed
Handset
Removed
Searching
Failed SIP registrations count the number of times a SIP registration has failed
Changed
Handset removed count is the number of times a handset has been marked as removed
Base searching is the number of times a base has been searching for it’s sync source
Free Running
Base free running is the number of times a base has been free running
DECT Source
Number of time a base has changed sync source
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4.12.2 Call data
The call data web is access by http://ip/CallStatistics.html
and data are organised in a table as shown in below example.
Screenshot
The table is organised with headline row, data pr. base rows and with last row containing the sum of all base parameters.
Parameters
Base Station
Name
Operation time/Duration
Count
Dropped
Description
Base IP address and base station name from management settings
Total operation time for the base since last reboot or reset
Duration is the time from data was cleared or system has been firmware upgraded.
Counts number of calls on a base.
Dropped calls are the number of active calls that was dropped.
E.g. if a user has an active call and walks out of range, the calls will be counted as a dropped call. An entry is stored in the syslog when a call is dropped.
No response
No response calls is the number of calls that have no response, e.g. if a external user tries to make a call to a handset that is out of range the call is counted as no response. An entry is stored in the syslog when a call is no response.
Duration
Active
Call duration is total time that calls are active on the base.
Max Active
Active call shows how many active calls that are active on the base (Not active
DECT calls, but active calls). On one base there can be up to 30 active calls.
Maximum active calls are the maximum number of calls that has been active at the same time.
Logging and count of used codec types on each call.
Codecs
Handover
Success
Handover
Failed
Counts the number of successful handovers.
Counts the number of failed handovers.
4.12.3 DECT data
The DECT data web is access by http://ip/DectStatistics.html
and data is organised in a table as shown in below example.
Screenshot
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Please note 3 frequencies are manually removed in the example system.
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4.13 Settings – Configuration File Setup
This page provides non editable information showing the native format of entire VoIP Configuration parameter settings. The settings format is exactly what is used in the configuration file. The configuration file is found in the TFTP server.
The filename for the configuration server is <MAC_Address>.cfg. The configuration file is saved in the folder /Config in the TFTP sever.
There are three ways to edit the configuration file or make changes to the settings page:
1) Using the VoIP Configuration interface to make changes. Each page of the HTTP web interface is a template for which the user can customise settings in the configuration file.
2) Retrieving the relevant configuration file from the TFTP and modify and enter new changes. This should be done with an expert network administrator.
3) Navigate to the settings page of the VoIP Configuration interface > copy the contents of settings > save them to any standard text editor e.g. notepad > modify the relevant contents, make sure you keep the formatting intact > Save the file as <Enter_MAC_Address_of_RFP>.cfg > upload it into the relevant TFTP server.
For details refer to [3].
An example of contents of settings is as follows:
~RELEASE=UMBER_FP_V0054
%GMT_TIME_ZONE%:16
%COUNTRY_VARIANT_ID%:18
%FWU_POLLING_ENABLE%:0
%FWU_POLLING_MODE%:0
%FWU_POLLING_PERIOD%:86400
%FWU_POLLING_TIME_HH%:3
%FWU_POLLING_TIME_MM%:0
%DST_ENABLE%:2
%DST_FIXED_DAY_ENABLE%:0
%DST_START_MONTH%:3
%DST_START_DATE%:1
....
....
4.14 Sys log
This page shows live feed of system level messages of the current base station. The messages the administrator see here depends on what is configured at the Management settings. The Debug logs can show only Boot Log or Everything that is all system logs including boot logs.
The Debug log is saved in the file format <Time_Stamp>b.log in a relevant location in the TFTP server as specified in the upload script.
A sample of debug logs is as follows:
0101000013 [N](01):DHCP Enabled
0101000013 [N](01):IP Address: 192.168.10.101
0101000013 [N](01):Gateway Address: 192.168.10.254
0101000013 [N](01):Subnet Mask: 255.255.255.0
0101000013 [N](01):TFTP boot server not set by DHCP. Using Static.
0101000013 [N](01):DHCP Discover completed
0101000013 [N](01):Time Server: 192.168.10.11
0101000013 [N](01):Boot server: 10.10.104.63 path: Config/ Type: TFTP
0101000013 [N](01):RemCfg: Download request of Config/00087b077cd9.cfg from
10.10.104.63 using TFTP
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0101000014 [N](01):accept called from task 7
0101000014 [N](01):TrelAccept success [4]. Listening on port 10010
0101000019 [N](01):RemCfg: Download request of Config/00087b077cd9.cfg from
10.10.104.63 using TFTP
0101000019 [W](01):Load of Config/00087b077cd9.cfg from 10.10.104.63 failed
To dump the log simply copy and page the full contents.
4.15 SIP Logs
This page shows SIP server related messages that are logged during the operation of the system. The full native format of SIP logs is saved in the TFTP server as <MAC_Address><Time_Stamp>SIP.log
These logs are saved in 2 blocks of 17Kbytes. When a specific SIP log is fully dumped to one block, the next
SIP logs are dumped to the other blocks. An example of SIP logs is shown below:
.....
Sent to udp:192.168.10.10:5080 at 12/11/2010 11:56:42 (791 bytes)
REGISTER sip:192.168.10.10:5080 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.101:5063;branch=z9hG4bKrlga4nkuhimpnj4.qx
Max-Forwards: 70
From: <sip:[email protected]:5080>;tag=3o5l314
To: <sip:[email protected]:5080>
Call-ID: p9st.zzrfff66.ah8
CSeq: 6562 REGISTER
Contact: <sip:[email protected]:5063>
Allow: INVITE, CANCEL, BYE, ACK, REGISTER, OPTIONS, REFER, SUBSCRIBE, NOTIFY,
MESSAGE, INFO, PRACK
Expires: 120
User-Agent: Generic-DPV-001-A-XX(Generic_SIPEXT2MLUA_v1)
Content-Type: application/X-Generic_SIPEXT2MLv1
Content-Length: 251
.....
To dump the log simply copy and page the full contents.
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5 Firmware Upgrade Procedure
This step-by-step chapter describes how to upgrade or downgrade base station(s) and/or handset(s) / repeater (s) to the relevant firmware provided by Konftel.
5.1 Network Dimensioning
In principle, a number of hardware and software components should be available or be satisfied before base station/handset update can be possible.
The minimum hardware and software components that are required to be able update via TFTP include the following (but not limited to):
• Handsets
• Base stations
• TFTP Server (Several Windows and Linux applications are available)
• DHCP Server (Several Windows and Linux applications are available)
• Workstation (e.g. Normal terminal or PC)
• Any standard browser (e.g. Firefox)
• Public/Private Network
5.2 TFTP Configuration
This section illustrate TFTP Server configuration using “SolarWinds” vendor TFTP Server. Create the following relevant folders as shown in the snap shots and choose defaults settings for the remaining options and save.
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NOTE: If TFTP server timeout settings are too short firmware upgrade might not complete. Recommended time out setting is more than 3 seconds.
5.3 Create Firmware Directories
The admin from the service provider’s side must create the relevant firmware directory in the server where both old and new firmware(s) can be placed in it. (See the STEP above)
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5.3.1 Base:
On the TFTP server root, create directory “9430”.
Copy Base station firmware to the named directory.
IMPORTANT:
The 9430 directory name cannot be changed.
5.4 Firmware Update Settings
Scroll down and Click on Firmware Update url link in the VoIP Configuration Interface to view the
Firmware Update Settings page.
Type IP address and firmware path followed by save.
For Http download the firmware update server settings must be entered as follows:
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5.5 Base Station Firmware Upgrade
On the Firmware Update Settings page > scroll down to the Update Gateways section > Enter the relevant firmware version of the base station to upgrade or to downgrade. Enter 202 for base version V0202.
Efter entering required version choose Start update button > select OK button from the dialog window to start the update/downgrade procedure.
The base station will automatically reboot and retrieve the firmware specified from the server and update itself accordingly.
The base firmware update behaviour is: Base will fetch the fwu file for approximately 3 minutes, then reboot and start flashing the LED - indicated by LED fast flashing for approximately 3 minutes and reboots in new version.
Note: All on-going voice calls are dropped from the base station immediately the firmware update procedure starts.
5.5.1 Base firmware confirmation
Base station firmware version status can be seen on the web Welcome page.
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5.5.2 Verification of Firmware Upgrade
Syslog information when Management Syslog level is set to “Debug”
[ FWU Downloading File tftp://10.1.24.101/FwuPath/Beatus/BeatusSw_4181_v0202.fwu]
[ Base FWU started]
[ Base FWU ended with exit code 2101 (NE_FILE_TRANSFER_EOF): End of file]
The log window of the TFTP server:
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6 Functionality Overview
So far we have setup our VoIP system. Next, in this chapter we list what features and functionalities are available in the system. The VOIP system supports all traditional and advanced features of most telephony networks. In addition, 3 rd
party components handle features like voice mail, call forward, conference calls, etc. A brief description of VOIP network functionalities are:
• Outgoing/incoming voice call management: The VOIP system can provide multiple priority user classes. Further, up to 3 repeaters can be linked to a Base-station.
• Internal handover: User locations are reported to SIP Server in order to provide differentiated services and tariff management. Within a DECT traffic area, established calls can seamlessly be handover between Base-station and repeaters using connection handover procedures.
• Security: The IP DECT 10 system also supports robust security functionalities for Base-station. Most security
functionality is intrinsically woven into the VOIP network structure so that network
connections can be encrypted and terminal authentication can be performed.
6.1 Base Station Interfaces
Interfaces
Power
Input: 100-240 VAC 50-60Hz (90 – 265 VAC)
Output Nom: 5VDC 1000mA
Type: Switch mode single or multi-plug solution
Plugs: UK, EU, US and AUS
LAN Interface
Standard : 10BASE-T(IEEE 802.3 100Mbps)
Connector: RJ45 8/8
Keys
LED indicator
RF
Frequency Bands
1: Reset key, Page and Default
One Status LED (multicolour, red, green, orange)
1880 – 1900 MHz (EMEA)
1910 – 1930 MHz (Latam)
1920 – 1930 MHz (USA)
Factory setting which can’t be modified after production
Output Power
Antenna
Software upgrade
Downloadable
Temperatures
Operation
250 mW or 140mW depending on country version
Two antennas for diversity
Remote firmware update using HTTP, HTTPS or TFTP
0˚C to 40˚C
1
With active security 4 channels is supported
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RFC4475
SIPS
In-band DTMF
SRTP
SIP registrations
RTP streams
SIP transport
Web server
Other features
IP quality
Jitter buffer
Automatic DST
Tone Scheme
Provisioning
Re-direct server
SIP configuration
Call groups
IP features
IPv4
IPv6
6.2 Software Features
CODEC’s
G.711 PCM A-law & U-law
G.722
G.726
G.729
Yes
Yes
Yes
A/AB (including VAD), max 4 coders
G729 licence not included
SIP
RFC2327
RFC2396
RFC2833
RFC2976
RFC3261
RFC3262
RFC3263
RFC3264
RFC3265
RFC3326
RFC3311
RFC3325
RFC3326
RFC3489
RFC3515
RFC3550
RFC3581
RFC3842
RFC3891
RFC3892
RFC3960
SDP: Session Description Protocol
Uniform Resource Identifiers (URI): Generic Syntax
In-Band DTMF/Out of band DTMF support
The SIP INFO method
SIP 2.0
Reliability of Provisional Responses in the Session Initiation Protocol (PRACK)
Locating SIP Servers (DNS SRV, redundant server support)
Offer/Answer Model with SDP
Specific Event Notification
The Reason Header Field for the Session Initiation Protocol
The Session Initiation Protocol UPDATE Method
P-Asserted Identity
The Reason Header Field for the Session Initiation Protocol (SIP)
STUN
REFER: Call Transfer
RTP: A Transport Protocol for Real-Time Application
Rport
Message Waiting Indication
Replace header support
The Session Initiation Protocol (SIP) Referred-By Mechanism
Early Media and Ringing Tone Generation in the Session Initiation Protocol
(SIP)
Session Initiation Protocol (SIP) Torture Test Messages
Secure SIP
No
Yes, packet authentication will limit the number of calls to 4 max 20 max 10
UDP, TCP or TLS
Embedded web server, accessed using HTTP
Warning – Network outage, VoIP service outage
Yes, adaptive
Yes
Country Depend Tone Scheme
Yes
Yes
Yes, from web page or configuration file
Yes
Yes
Hardware ready, software not included
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TCP/IP/UDP
DHCP Support
DHCP option
Static IP
DNS srv
VLAN
Quality of service
TLS
Certificates
TFTP
HTTP server
HTTP client
HTTPS
SNTP
DECT
DECT handover
CAT-IQ v1.0
Repeater support
Intercom
DECT encryption
DECT Authentication
Group TPUI support
GAP compliant
CAT-IQ compliant
Handset registrations
6.3 Call Features
Call supported
Simultaneous calls/base
Yes
Yes
66, 120
Yes
Yes
Yes, 802.1p/q
Type of Service (ToS) including DiffServ Tagging, and QoS per IEEE 802.1p/q
Yes, 1.0
Yes, X.509 (certificate not included)
Yes, for firmware and configuration file download
Yes
Yes, for firmware and configuration file download
Yes, for firmware and configuration file download
Yes, For internet clock synchronization
Yes, inter-cell handover for repeater support
HD audio or NB audio support
Yes
No
Yes
Yes
Yes, for call groups
No
No
20
Simultaneous calls/handset
Call features
5 simultaneous call supported
5 Wideband calls (g.722). 5 narrowband calls (PCMA, PCMU, G.726) or 4 when using G729
2
Codec Negotiation
Codec Switching
Missed call notification
Voice mssage waiting notification
Date and Time synchronization
Parallel calls
Call Hold
Call Retrieve
Call transfer unannounced
Call transfer announced
Conference (3PTY)
Conference, Network
Call Waiting Indication
Calling line identity
Outgoing call
Call Toggle/Swap
Incoming call
Line identification
Multiple Lines
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Central Phone Book:
-
LDAP
-
XML
-
CSV
DND:
Call Forward:
-
CFU
-
CFNA
-
CFB
Call groups:
Multiple calls
Call identification
Calling Name Identification Presentation (CNIP)
Calling Line Identification Presentation (CLIP)
Call Completed Elsewhere
Distinctive Ringing
Yes
Yes, remote or file load from web interface
Yes, file load through web interface
Yes
Configurable from base or handset (Not with Call Group active))
Yes
Yes
Yes
Yes, 1-20 handsets/SIP account
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Appendix
7 Appendix A: Basic Network Server(s) Configuration
In this chapter we describe how to setup the various server elements in the system.
7.1 Server setup
In the network, the server environment is installed as a centralized system.
The main server types hosted on the network include SIP, DNS/DHCP and HTTP/TFTP Servers. These servers can be hosted both in one or multiple windows and/or Linux Server environment.
Management servers are normally installed to monitor and manage the network in detail. Each Basestation status can be checked. Each Repeater and each Subscriber Terminal can be monitored over the air from a centralized location.
Further, new software can be uploaded to all system elements from the centralized location (typically a
TFTP server) on an individual basis. This includes Subscriber Handsets where the latest software is downloaded over the air.
7.2 Requirements
Regardless of whether or not you will be installing a centrally provisioned system, you must perform basic
TCP/IP network setup, such as IP address and subnet mask configuration, to get your organization’s phones up and running.
7.3 DNS Server Installation/Setup
Name server is a name server service installed in a server for mapping or resolution of humanly memorable domain names and hostnames into the corresponding numeric Internet Protocol (IP) addresses.
The customer should refer to the platform vendor either windows or Linux vendor for detail step-by-step guide on how to install and configure Domain Name System for internet access. In this section, we briefly describe hints on how to setup DNS behind NAT or Firewall.
7.3.1.1 Hints on how to Configure DNS behind a Firewall/NAT
Proxy and Network Address Translation (NAT) devices can restrict access to ports. Set the DNS to use UDP port 53 and TCP port 53. For windows Servers, set the RCP option on the DNS Service Management console and configure the RCP to use port 135.
These settings should be enough to resolve some of potential issues that may occur when you configure
DNS and firewalls/NAT.
7.4 DHCP Server Setup
A DHCP Server allows diskless clients to connect to a network and automatically obtain an IP address. This server is capable of supplying each network client with an IP address, subnet mask, default gateway, an IP address for a WINS server, and an IP address for a DNS server. This is very often used in enterprise networks to reduce configuration efforts. All IP addresses of all computers/routers/bases are stored in a database that resides on a server machine.
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The network administrator should contact the relevant vendors for detail information or step-by-step procedure on how to install and setup DHCP process or service on windows/Linux servers. In this section, we will provide some hints of how to resolve potential problems to be encountered you setup DHCP
Servers.
7.4.1 Hint: Getting DHCP Server to Work
Windows Server:
1) Clients are unable to obtain an IP address
If a DHCP client does not have a configured IP address; it generally means that the client has not been able to contact a DHCP server. This is either because of a network problem or because the DHCP server is unavailable. If the DHCP server has started and other clients have been able to obtain a valid address, verify that the client has a valid network connection and that all related client hardware devices
(including cables and network adapters) are working properly.
2) The DHCP server is unavailable
When a DHCP server does not provide leased addresses to clients, it is often because the DHCP service has failed to start. If this is the case, the server may not have been authorized to operate on the network. If you were previously able to start the DHCP service, but it has since stopped, use Event
Viewer to check the system log for any entries that may explain the cause.
Next, restart the DHCP service, click Start, click Run, type cmd, and then press ENTER. Type
net start dhcpserver, and then press ENTER.
Linux Platform:
Troubleshooting DHCP, check the following:
1)
Incorrect settings in the /etc/dhcpd.conf file such as not defining the networks for which the DHCP server is responsible;
2)
3)
4)
NAT/Firewall rules that block the DHCP bootp protocol on UDP ports 67 and 68;
Routers failing to forward the bootp packets to the DHCP server when the clients reside on a separate network. Always check your /var/logs/messages file for dhcpd errors.
Finally restart the dhcpd service daemon
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7.5 TFTP Server Setup
There are several TFTP servers in the market place; in this section we describe how to setup a commonly used TFTP Server.
7.5.1 TFTP Server Settings
The administrator must configure basic parameters of the TFTP application:
• Specify UDP 69 port – for TFTP incoming requests and TCP 12000 – for remote management of the server. For file transmission the server opens UDP ports with random numbers. In case the option
Enable NAT or firewall support is activated on the server, the server uses the same port for files transmission and listening to the TFTP incoming requests (UDP 69 port on default).
• Specify the interface bindings, TFTP root directory, port which the TFTP Server will listen, timeout and number of retries, and TFTP options supported by the server.
• Configure the relevant TFTP virtual folder in the server. The TFTP virtual folder is the file folder, visible for TFTP clients under a certain name. You can set security settings separately for every virtual TFTP folder. Next, set rights to access TFTP folders according to the relevant clients.
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7.6 SIP Server Setup
SIP server is one of the main components of a network, dealing with the setup of all SIP calls in the network. A SIP server is also referred to as a SIP Proxy or a Registrar.
Although the SIP server is the most important part of the SIP based phone system, some servers only handles call setup and call tear down. It does not actually transmit or receive any audio. This is done by the media server in RTP.
The IP DECT 10 is fully interoperable with the most of SIP Server applications. There are many off-the-shelf vendor and open source SIP servers. In this section, we will briefly explain settings required to take full advantage of FreePBX SIP Server feature set. The settings are similar for other SIP servers.
7.6.1 FreePBX SIP Server
FreePBX is an easy to use GUI (graphical user interface) that controls and manages Asterisk, which the most popular open source telephony engine software.
The administrator should refer to the relevant detail step-by-step procedure of how to install FreePBX SIP server. This section briefly describes SIP Server setup parameters.
1) SIP Server Setup
Settings
NAT
Description
This option determines the settings for users connecting to an asterisk server.
Possible values: Yes, No, Never, Route
NAT=route
Asterisk will send the audio to the port and IP where it’s receiving the audio from.
Instead of relying on the addresses in the SIP and SDP messages.
This will only work if the phone behind NAT send and receive audio on the same port and if they send and receive the signalling on the same port. (The signalling port does not have to be the same as the RTP audio port).
NAT=No
Asterisk will add an RPORT to the via header of the SIP messages
NAT=never
This will cause asterisk not to add an RPORT in the VIA line of the sip invite header
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Other NAT
Settings
Codecs
Video Codecs
Media & RTP
Settings
Choose the relevant option or enter the settings in IP configuration, External IP,
Local Network.
Some SIP Servers supports dynamic codec support. Codecs are algorithm used to compress or decompress speech or audio signals. The user should select the relevant Codecs and other speech compression techniques whose traffic will be routed to the network.
The user should enable this option if network supports video telephony.
This option should be enabled to provide for deliver media streams (e.g., audio and video) or out-of-band events signalling (DTMF in separate payload type).
2) Extensions
This feature allows administrators create handset profiles in the network. In other words, Extensions describes the Dial plan for the PBX SIP system. Enter the relevant parameters
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8 Appendix B: Using Base with VLAN Network
In this chapter we describe how to setup a typical VLAN in the network.
8.1 Introduction
In this chapter, we describe how to setup VLAN to typical network. There are three main stages involved in this procedure:
1) Configure a VLAN Aware Switch to a specific (un)tagged VLAN ID, so the system can process untagged frames forwarded to it.
2) Setup the Time Server (NTP Server) and other relevant network servers.
3) Configure the HTTP server in the Base station to access the features in the PBX or system.
VLAN allows administrators to separate logical network connectivity from physical connectivity analogous to traditional LAN which is limited by its physical connectivity. Normally, users in a LAN belong to a single broadcast domain and communicate with each other at the Data Link Layer or
“Layer 2”. LANs are segmented into smaller units for each IP subnets and here communication between subnets is possible at the Network Layer or “Layer 3”, using IP routers.
A VLAN can be described as a single physical network that can be logically divided into discrete
LANs that can operate independently of each other.
An Illustration of using VLANs to create independent broadcast domains across switches is shown below:
The figure above highlights several key differences between traditional LANs and VLANs.
•
•
All switches are interconnected to each other. However, there are three different VLANs or broadcast domains on the network. Physical isolation is not required to define broadcast domains. If the figure was a traditional LAN without VLAN-aware switches, all stations would belong to one broadcast domain.
All switch ports can communicate with one another at the Data Link Layer, if they become members of the same VLAN.
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•
The physical location of an end station does not define its LAN boundary.
1.
An end station can be physically moved from one switch port to another without losing
2.
its “view of the network”. That is, the set of stations it can communicate with at the
Data Link Layer remains the same, provided that its VLAN membership is also migrated from port to port.
By reconfiguring the VLAN membership of the switch port an end station is attached to, you can change the network view of the end station easily, without requiring a physical move from port to port.
8.2 Backbone/ VLAN Aware Switches
To implement a VLAN in your network, you must use VLAN-aware switches.
Before we continue, let consider two rules to remember regarding the functioning of a regular
LAN switch:
1. When the switch receives a broadcast or multicast frame from a port, it floods (or broadcasts) the frame to all other ports on the switch.
2. When the switch receives a unicast frame, it forwards it only to the port to which it is addressed.
A VLAN-aware switch changes the above two rules as follows:
1. When the switch receives a broadcast or multicast frame from a port, it floods the frame to only those ports that belong to the same VLAN as the frame.
2. When a switch receives a unicast frame, it forwards it to the port to which it is addressed, only if the port belongs to the same VLAN as the frame.
3. A unique number called the VLAN ID identifies each VLAN.
Which VLAN Does a Frame Belong To?
The previous section notes that a frame can belong to a VLAN. The next question is—how is this association made?
•
A VLAN-aware switch can make the association based on various attributes of the type of frame, destination of MAC address, IP address, TCP port, Network Layer protocol, and so on.
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An illustration of IEEE 802.1Q VLAN tag in Ethernet frame is as follows:
8.3 How VLAN Switch Work: VLAN Tagging
VLAN functionality can be implemented via explicit frame tagging by switches and end stations.
Network switches and end stations that know about VLANs are said to be VLAN aware. Network switches and end stations that can interpret VLAN tags are said to be VLAN tag aware. VLAN-tagaware switches and end stations add VLAN tags to standard Ethernet frames–a process called explicit tagging. In explicit tagging, the end station or switch determines the VLAN membership of a frame and inserts a VLAN tag in the frame header (see figure above for VLAN tagging), so that downstream link partners can examine just the tag to determine the VLAN membership.
8.4 Implementation Cases
Common types of usage scenarios for VLANs on typical VLAN switches: port-based VLANs, protocol-based VLANs, and IP subnet-based VLANs. Before figuring out which usage scenario suits your needs, you must understand what each type of usage scenario implies.
•
Port-based VLAN: All frames transmitted by a NIC are tagged using only one VLAN ID. The
NIC does not transmit or receive any untagged frames.
All protocols and applications use this virtual interface’s virtual PPA to transmit data traffic.
Therefore all frames transmitted by that NIC port are tagged with the VLAN ID of that
Virtual Interface.
•
Protocol-based VLAN: The NIC assigns a unique VLAN ID for each Layer 3 protocol (such as
IPv4, IPv6, IPX, and so on). Therefore, the VLAN ID of outbound frames is different for each protocol. An inbound frame is dropped if the protocol and VLAN ID do not match.
•
IP subnet-based VLAN: The NIC assigns a unique VLAN ID for each IP subnet it belongs to.
Therefore, the VLAN ID of outbound frames is different for different destination subnets.
An inbound frame is dropped if the IP subnet and VLAN ID do not match.
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8.5 Base station Setup
After the admin have setup the Backbone switch, next is to configure the Base station via HTTP interface.
STEP 1
Connect the Base station to a private network via standard Ethernet cable (CAT-5).
STEP 2
Use one of the two methods to find the base IP
STEP 3
On the Login page, enter your authenticating credentials (the username and password is
admin by default unless it is changed). Click OK button.
STEP 4
Once you have authenticated, the browser will display front end of the Configuration
Interface. The front end will show relevant information of the base station.
STEP 5
Create the relevant SIP server information in the system. Each service provider/customer should refer SIP server vendor on how to setup SIP servers.
8.6 Configure Time Server
STEP 6
Navigate to the Time settings and configure it. Scroll on the left column and click on Time url link to Open the Time Settings Page. Enter the relevant parameters on this page and press the Save button.
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8.7 VLAN Setup: Base station
STEP 7
Navigate to the Network url > On the network page enter the relevant settings in the VLAN section > VLAN Id should be the same as those configured into the backbone.
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9 Appendix C: Local Central directory file handling
In this appendix the Local Central Directory file format, import and configuration is described.
9.1 Central Directory Contact List Structure
The structure of Contact List is simple. The figure below shows an example of structure of Contact List in
Text format and in Xml format. Contact name must not contain more than 23 characters and contact
number must not contain more than 21 digits.
.csv or .txt
.xml
Txt file limitations:
Contact name must NOT be longer than 23 characters (name will be truncated)
Contact name must NOT contain “,”
Contact number must be limited to 21 digits (entry will be discarded, no warning)
Contact number digits must be: +0123456789
Contact number does not support SIP-URI
Spaces between name section “,” and number section is not supported
9.2 Central Directory Contact List Filename Format
The Contact list is saved as file format: .txt .csv or .xml
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9.3 Import Contact List to Central Directory
On the Central Directory page, the admin should click on Browse button and the Choose File to Load dialog window will be shown.
On the Choose File to Upload dialog window, navigate to the directory or folder that contains the right file to be imported to the base station > Click on Open button.
Next, click on the Load button. This will import the contents of contacts in the selected file into the relevant
Base station.
The figure below shows the import procedure is in process.
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9.4 Central directory using server
Alternative way to import a Contact List is to get it from a server. First click on Management url to get
Management Settings page, then select the protocol of your server (TFTP/HTTP) in Management Transfer
Protocol, then save the setting by clicking Save.
Go back to Central Directory page and enter Server IP address (inclusive the path in the end of the address) and Filename of the contact list, then save the setting by clicking Save. (See example below).
Then reboot the Base station to ensure that the changes take effect.
9.5 Verification of Contact List Import to Central Directory
On the Handset, navigate to Central Directory where the correct contact list should populate to the contacts uploaded to the Base station.
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