Mitel RFP 12 System Guide
Mitel RFP 12 System Guide
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Password ........................................................................................................................... 55
Central Directory and LDAP .............................................................................................. 55
Local Central Directory ...................................................................................................... 55
LDAP ................................................................................................................................. 56
Repeaters .......................................................................................................................... 58
Add Repeater ..................................................................................................................... 58
Register Repeater .............................................................................................................. 60
Repeaters List.................................................................................................................... 60
Statistics ............................................................................................................................ 62
System Data ...................................................................................................................... 62
Call Data ............................................................................................................................ 63
Repeater Data ................................................................................................................... 64
DECT Data ........................................................................................................................ 65
Settings – Configuration File Setup ................................................................................... 65
Sys Log .............................................................................................................................. 67
SIP Logs ............................................................................................................................ 68
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AUDIENCE
Read this guide before you install the system components and when you are ready to setup
or configure SIP server, NAT aware router, advanced VLAN settings, base stations, and
multi-cell setup. This guide describes how to deploy a fully functionally system.
IMPORTANT ASSUMPTIONS
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Local Central Directory File Describes the central directory file format and
Handling provides instructions on how to upload it.
ABBREVIATIONS
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REFERENCES/RELATED DOCUMENTATION
[1]: 112 DECT Phone (Universal) and RFP 12 Single Cell Base Station Installation
Guide (part number 57011091): provides instructions on how to make the required
cable and power connections for the base station and charging cradle. It also provides
instructions for installing the handset batteries.
[2]: Mitel 112 DECT Phone (Universal) User Guide: describes the features and
functionalities provided by the Mitel 112 DECT Phone
[3]: 112 DECT Phone Quick Reference Guide for MiVoice Business: provides
instructions on how to use the features of the handset when it is connected to a MiVoice
Business communications platform.
[4]: 112 DECT Phone Quick Reference Guide for MiVoice Office 250: provides
instructions on how to use the features of the handset when it is connected to a MiVoice
Office 250 communications platform.
[5]: MiVoice Business System Administration Help: Refer to this online help system
for instructions on how to program
• Mitel 112 DECT Phone as a “SIP generic device type” on the MiVoice Business
system
• Mitel 112 DECT Phones into personal ring groups
• Support for Suite Services.
[6]: MiVoice Office 250 Features and Programming Guide and Database
Programming Online Help: provides instructions on how to program the Mitel 112
DECT phone as a “SIP Phone” on the MiVoice Office 250.
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HARDWARE SETUP
The base-stations are mounted on walls or poles so that each base-station is separated from
each other by up to 10 meters (for indoor installation). Radio coverage can be extended using
repeaters. Repeaters are range extenders only and cannot be used to increase local
capacity.
The base-station antenna mechanism is based on a space diversity feature which improves
coverage. The base-stations use the complete DECT MAC protocol layer and IP media
stream audio encoding feature to provide up to five simultaneous calls.
The system is made up of (but not limited to) the following components:
• Mitel 112 DECT Phone and charging cradle.
• Base station connected over an IP network and using DECT as air-core interface
• Repeater (optional)
• VoIP Administration Interface
The phone is a lightweight, ergonomically and portable handset compatible with Wideband
Audio (G.722), DECT, GAP standard, CAT-iq audio compliant.
The handset includes a color display with graphical user interface. It can also provide the
subscriber with most of the features available for a wired phone, in addition to its roaming and
handover capabilities.
BASE STATION
The Base Station converts IP protocol to DECT protocol and transmits the traffic to and from
the wireless handsets over a channel. The base station has five available channels.
REPEATER
The base supports the IP DECT CAT-IQ repeater RTX4024. A repeater can be deployed to
extend the range of a DECT handset. The repeater can also be utilized wherever there is a
need to increase limited coverage or improve reception in remote areas.
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• DECT encryption
• Automatic registration
• Maximum of three repeaters in daisy chain.
The VoIP Configuration Interface is a web based administration that you use
• configure the base station and relevant network end-nodes. For example, handsets can
be registered or de-registered from the system using this interface.
• install software or firmware downloads onto base stations, repeaters and handsets.
• access system logs that can be used to troubleshoot the system.
WIRELESS BANDS
SYSTEM CAPACITY
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PACKAGE INSPECTION
Before you open the package, examine it for evidence of physical damage or mishandling. If
the package appears damaged, report it to the relevant support centre of the regional
representative or operator.
The following are the recommended procedure for you to use for inspection:
1. Examine all relevant components for damage.
2. If damage is detected, make a “defective on arrival – DOA” report to Mitel Customer
Service. The Mitel Customer Service representative will initiate the necessary procedure
to process the return. 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.
CONTENTS
Ensure that the following components where provided in the handset package before
proceeding with the installation:
• 1 x handset and battery cover
• 2 AAA batteries
• 1 x charging cradle with wired A/C adapter
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The International Portable Equipment Identity (IPEI) of each handset is printed either on a
label located behind the battery or on the packaging label. Remove the handset back cover,
take out the battery (if installed) and record the IPEI number.
You need this number to enable service to the handset. You must program it into the system
database via the VoIP Administration interface.
Each handset is charged using a handset charger. The charger is a compact desktop unit
that automatically maintains the correct battery charge levels and voltage.
The handset charger is powered by AC power adapter that supplies 5VDC at 1000mA. The
AC power adapter is supplied from 110-240 VAC.
When charging the batteries for the first time, it is necessary to leave the handset in the
charger for at least 10 hours before they are fully charged and the handset is ready for use.
For correct charging, ensure that the room temperature is between 0°C and 25°C (32°F and
77°F). Do not place the handset in direct sunlight.
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The battery displayed in the top right of the screen indicates the charging status.
For instructions on how to use the handset features, refer to the Mitel 112 DECT Phone
(Universal) User Guide available on the Mitel Customer Documentation site.
PACKAGE INSPECTION
Before you open the package, examine it for evidence of physical damage or mishandling. If
the package appears damaged, report it to the relevant support centre of the regional
representative or operator.
PACKAGE CONTENTS
Ensure that the following components where provided in the base unit package before
proceeding with the installation:
• 2 x mounting screws and 2 x Anchors
• 1 x Category 5 cable (Ethernet cable)
• Base unit
• Power supply adapter
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The base station front panel has an LED indicator that signals the 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 summarizes the various LED states:
OFF No power
To reset the base station unit, press the small Reset button on the back of the unit. You can
also reset the base station from the VoIP Administration Interface.
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2. Dial *47*. “Searching” is displayed. Depending on the number of active base stations
and the distance to the base it can take up to 5 minutes to find a base.
3. If there are multiple base stations available, use the down/up cursor to select the MAC
address of the desired base. The base IP address is displayed.
4. Record the IP address.
5. Configure 112 DECT Phone on Communication Platform.
Before you register the handset with the base station, complete the following MiVoice
Business programming tasks. Refer to the MiVoice Business System Administration Tool
online help for instructions:
1. License the Mitel 112 DECT Phone (handset) as a SIP device.
2. Program a user and handset extension in the “User and Services Configuration” form as
a “Generic SIP Phone”.
3. Access the SIP Device Capabilities form. Program a SIP Device Capabilities index
number using the standard defaults with the exception of the following options. In the SIP
Device Capabilities form tabs, set the following options to Yes.
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5. In the “Multiline Set Keys” form of the MiVoice Business System Administration tool,
configure the handset with a second multi-call appearance of the prime line with the Ring
Type set to “Ring”. Refer to the System Administration Tool online help for instructions.
6. You can optionally configure a
• Mitel desktop phone and a handset for Suite Services (typically, used in a hospitality
environment).
Personal Ring Groups (PRGs) allow you to associate two or more devices for a single user
under a common, prime directory number (DN). The devices ring simultaneously (Ring All)
when the prime directory number is called. You can use PRGs to twin a person's desktop
phone and his or her Mitel 112 DECT Phone together. The desk phone is considered the
prime extension, which is referred to as the pilot number or prime member of the group. The
cordless handset is programmed as a non-prime member of the group.
You can also program and label a Handoff key on the user’s desk phone. Users can press
the Handoff feature key to
• push a call that is in progress from their desktop phone to their Mitel 112 DECT
Phone, or
• pull a call that is in progress from their Mitel 112 DECT Phone to their desktop phone.
The Handoff key is only supported on Mitel desktop phones. It is not supported on SIP
devices and you cannot program it on a Mitel 112 DECT Phone.
Refer to the “Ring Groups Personal” and “Handoff (Personal Ring Groups)” topics in the
“Features” book of the MiVoice Business System Administration Tool online help for
programming instructions.
Suite Service provides the ability to group a number of telephone lines through
interconnected hotel/motel rooms, or suites, for the purposes of billing and shared telephone
service. Refer to the following online book in the MiVoice Business System Administration
Tool online help for a detailed description of Suite Services and programming instructions:
System Applications > Hospitality > Suite Services.
Before you register a handset with base station, complete the following MiVoice Office 250
Database Programming tasks. Refer to the MiVoice Office 250 Features and Programming
Guide and Database Programming online help for detailed instructions:
1. Ensure that you have a valid Category F license available for each handset that will be
connected to the base station.
2. Program each handset as a “SIP Phone” (or part of a SIP Phone Group).
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Dynamic Extension Express (DEE) allows you to associate two or more devices for a single
user under a common main extension number. You can use DEE to “twin” a person's desktop
phone and his or her handset together. The desk phone is considered the main extension,
while the cordless handset is programmed as a secondary destination.
You can also program and label a DEE Handoff key (default feature code is 388) on the
user’s desk phone. Users can press the DEE Handoff feature key to push a call that is in
progress from their desk phone to their handset.
For programming instructions, refer to the DEE topics in the latest MiVoice Office 250
Features and Programming Guide and Database Programming Online Help.
• Press the center Menu button on the handset to access the main menu:
• If multiple base stations are available, use the up/down Menu button to highlight the
MAC address of the desired base. Press Select. The base IP address is displayed.
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6. The browser displays the Welcome page of the VoIP Administration interface. It lists the
base station information.
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• Enter the name of the MiVoice communications platform in the “Server Alias” field.
• Enter the IP address of the MiVoice communications platform in the “Registrar” and
“Outbound Proxy” fields.
• Click Save.
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2. Click Network:
• Click Save.
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3. Click Management.
• Click Save.
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4. Click Time.
• Click Save.
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5. Click Country.
• Click Save.
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6. Click Security:
CAUTION: Ensure that you record the new password. If you forget the
administrator password, you must reset the base station to the default
configuration values and reconfigure the system.
• Click Save.
Note: You can reset the stand to the default configuration values (including the username
and password) using the RESET button on the base station. Press and hold the RESET
button for greater than 10 seconds to reset the base station configuration to the default
values.
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• Click Handset.
• Enter the IEPI of the handset. The IEPI is printed on a label located under the
handset batteries.
• Click Save.
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• Under Select Handset(s), check the box to associate the extension with a handset.
• Click Save.
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3. The base station is now open (in the ready state) for handset registration for the next 5
minutes. You must register the selected handsets with the base station using the
following procedure in the next 5 minutes.
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4. Next, register each handset with the base station. Start the registration procedure on the
handset by following step “a” to “d” below.
a) Select main menu “Connectivity” b) Select menu ”Register”
c) Type in the “AC code” and press “OK” to start d) After a while the handset is registered, and the
the registration. The default AC code is “0000”. idle display is shown.
Note: The unique handset IPEI is displayed on sheet “Extensions” when the handset is
successfully registered. The web page must be manually updated by pressing “F5” to see that
the handset is registered; otherwise the handset IPEI (International Portable Equipment
Identity) isn’t displayed on the web page.
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5. The following screen shows an example of the Extensions page after you have registered
several handsets.
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Note: Enabling secure web is not possible. For secure configuration use secure
provisioning.
This section defines the variables and parameters for configuration in the network.
WEB NAVIGATION
This section describes the left menu of the VoIP Administration Interface.
Home/Status This “Welcome” page displays the system information and base station status.
Servers Define which SIP/NAT server the network should connect to.
Management Defines the Configuration server address, Management transfer protocol, and the
sizes of logs/traces that should be catalogued in the system.
Firmware Update Remote firmware updates (HTTP(s)/TFTP) settings of base stations and handsets.
Time Configures a time server for the system. Use a time server that applies to the country
of installation. The time server must deliver the time in Network Time Protocol (NTP).
The base station and handsets clocks are synchronized to the time server.
Country Specify the country/territory where the network is located to ensure that your phone
functions properly.
Note: The base language and country setting are independent of each other.
Security Allows users to administrate certificates and create account credentials with which
they can log in or log out of the embedded HTTP web server.
Central Directory Interface to a common directory. You can import up to 3000 entries using *csv format
file or configure a connection to an LDAP directory.
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Alarm 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.
Configuration This shows detail and complete network settings for base station(s),
HTTP/DNS/DHCP/TFTP server, SIP server, etc.
Syslog Overall network related events or logs are displayed here (only live feed is shown).
SIP Log SIP related logs can be retrieved from url link. It is also possible to clear logs from this
feature.
HOME/STATUS
PARAMETER DESCRIPTION
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Firmware URL Firmware update server address and firmware path on server
Reboot Reboot after all connections is stopped on base. Connections are active
call, directory access, firmware update active
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EXTENSIONS
This section describes the different parameters available whenever the administrator is
creating extensions for handsets. Note, you cannot add extensions unless servers are
defined. This section also describes the group call feature.
The system supports a maximum of 20 extensions with 20 associated handsets which can be
divided between servers. Once 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.
ADD EXTENSION
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Display Name Empty Name displayed on the handset for the extension
Permitted value(s): 8-bit string length
Mailbox Name Empty Name of centralized system that is 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
Mailbox Number Empty 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.
Call waiting Enabled Used to enable/disable Call Waiting feature. When disabled a
feature second incoming call will be rejected. If enabled, a second call will
be presented as call waiting.
Empty Number to which incoming calls must be re-routed when SIP node
Forwarding On is busy.
Busy Number Forwarding On Busy Number must be enabled to function.
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Disabled Note: Feature must be enabled in the SIP server before it can
function in the network
Note: Feature is automatically disabled if the handset or extension
is part of a group.
GROUP CALL
When you add or edit an extension, you can subscribe handsets to the extension by selecting
them in the Selected Handset(s) table, and make them part of a group.
Group Call is when a SIP extension is associated with multiple handsets. All handsets that
are assigned with the extension can receive incoming calls and initiate outgoing calls from
that extension. When assigned with Group Call, a handset supports all normal call features
such as Hold, Transfer and so forth.
When an incoming call arrives to a group, all of the handsets assigned to the group are
alerted. For example, if a 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 2nd 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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EXTENSIONS LIST
The list can be sorted by any of the top headlines, by mouse click on the headline link.
PARAMETER DESCRIPTION
Display Name Given display name is displayed. If no name given this field will be empty
Server Alias Given server alias is displayed. If no alias given this field will be empty.
State SIP registration state – if empty the handset is not SIP registered.
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HANDSET LIST
PARAMETER DESCRIPTION
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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
ACTIONS DESCRIPTION
Stop Registration Manually stop DECT registration mode of the system. This prevents any
handset from registering to the system
Delete Handset(s) 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 Handset(s) Deregister the selected handset(s) and delete the extension(s).
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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SERVERS
In this section, we describe the different parameters available in the Servers configurations
menu. A maximum of 10 servers can be configured.
NAT Adaption Disabled To ensure all SIP messages goes directly to the NAT
gateway in the SIP aware router.
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SIP Session Timers: Disabled 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.
Session Timer Values 1800 Default value is 1800s according to the RFC.
(s): If disabled session timers will not be used.
Permitted value(s): Minimum value 90, Maximum
65636
Signal TCP Source Disabled When SIP Transport is set to TCP or TLS, a TCP (or
Port TLS) connection will be established for each SIP
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Use One TCP/TLS Disabled When using TCP or TLS as SIP transport, choose if a
Connection per SIP TCL/TLS connection
Extension: 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
extensions)
1: Enabled. (Use one TCP/TLS connection per SIP
extensions).
Keep Alive Enabled This directive defines the window period (30 sec.) to
keep opening the port of relevant NAT-aware router(s),
etc.
Hold Behaviour RFC 3264 Specify the hold behaviour by handset hold feature.
RFC 3264: Hold is 37nalyse37n 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 37nalyse37ng 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
Attended Transfer Hold 2nd Call 1. When we have two calls, and one call is on hold,
Behaviour 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.
2. The "Attended Transfer Behaviour" feature
defines whether or not the 2nd call shall be put
on hold before the REFER is sent.
3. 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
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DTMF Signalling RFC 2833 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
packets in different internet layer as the voice stream)
Both: Enables SIP INFO and RFC 2833 modes.
DTMF Payload Type 101 This feature enables the user to specify a value for the
DTMF payload type / telephone event (RFC2833).
Codec Priority G.711U Defines the codec priority that base stations uses for
G.711A audio compression and transmission.
G.726 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 Disabled With enable RTP will be encrypted (AES-128) using the
key negotiated via the SDP protocol at call setup.
Secure RTP Auth Disabled 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 Field list of supported SRTP Crypto Suites. The device
_SHA1_32 is born with two suites.
AES_CM_128_HMAX
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_SHA1_80
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.
NETWORK
In this section, we describe the different parameters available in the network configurations
menu.
IP SETTINGS
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DNS (Primary) NA 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.
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.
VLAN User 0 This is a 3 bit value that defines the user priority.
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)
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DHCP OPTIONS
NAT SETTINGS
We define some options available when NAT aware routers are enabled in the network.
STUN Bindtime 80 Permitted values: Positive integer default is 90, unit is in seconds
Guard
Keep alive time 90 This defines the frequency of how keep-alive are sent to maintain
NAT bindings.
Permitted values: Positive integer default is 90, unit is in seconds
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SIP/RTP SETTINGS
Use Different Disabled If disabled, the Local SIP port parameter specifies the source
SIP Ports 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.
Local SIP port 5060 The source port used for SIP signalling
Permitted values: Port number default 5060.
SIP ToS/QoS 0x68 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
RTP port 50004 The first RTP port to use for RTP audio streaming.
Permitted values: Port number default 50004 (depending on
the setup).
RTP port range 40 The number of ports that can be used for RTP audio streaming.
Permitted values: Positive integers, default is 40
RTP TOS/QoS 0xB8 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.
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Base Station VoIP It indicates the title that appears at the top window of the
Name: browser and is used in the multicell page.
Management TFTP The protocol assigned for configuration file and central
Transfer Protocol directory
Valid Input(s): TFTP, HTTP, HTTPs
HTTP Management Empty The folder location or directory path that contains the
upload script 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
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be used.
HTTP Management Empty Password that should be entered in order to have access to
password the configuration server.
Permitted value(s): 8-bit string length
Set Maximum 0
Digits of Internal
Numbers
Hotel Mode Disabled For hospitality (Hotel/Motel) environments, enable the Hotel
Mode setting to
• Black out the handset display when placed in cradle
(after 65 seconds)
• Protect the handset Settings menu (changes default
handset PIN from 0000 to 9351; PIN is required to
access the Settings menu)
• Enable silent upgrades and resets
• Disable call logging
• Prevent phonebook modification.
Configuration File Disabled Base Specific file: Used when configuring a single cell base
Download 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.
DHCP Custom Empty By default option 160, but custom option can be defined.
Option An option 160 URL defines the protocol and path
information by using a fully qualified domain name for
clients that can use DNS.
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Text Messaging 30 This defines the frequency of how keep-alive are sent
Keep Alive (m) Permitted values: Positive integer, unit is in minutes
Upload of SIP Log Disabled 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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Set Prefix for Empty Prefix number for the enabled automatic prefix feature.
Outgoing Calls
Permitted value(s): 1 to 9999
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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.
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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.
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Refresh time (h) Empty The window time in hours within which time server
refreshes.
Valid Inputs: positive integer
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Set timezone by Checked By checked country setting is used (refer to country web
country/region page).
Set DST by Checked By checked country setting is used (refer to country web
country/region page).
Daylight Saving Time Disabled The system administrator can Enable or Disable DST
(DST) manually.
Automatic: Enter the start and stop dates if you select
Automatic.
DST Fixed By Day Use Month and You determine when DST actually changes. Choose the
Date relevant date or day of the week, etc. from the drop down
menu.
DST Start Date 25 Numerical day of month DST comes to effect when DST is
fixed to a specific date
Valid Inputs: positive integer
DST Start Day of Monday Day within the week DST begins
Week
DST Start Day of Last in Month Specify the week that DST will actually start.
Week, Last in Month
DST Stop Month October The month that DST actually stops.
DST Stop Date 1 The numerical day of month that DST turns off.
Valid Inputs: positive integer (1 to 12)
DST Stop Day of First in Month The week within the month that DST will turn off.
Week Last in Month
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COUNTRY
The country setting controls the in-band tones used by the system. To select web interface
language go to the management page.
Note: By checked timezone and DST the parameters in web page Time will be
discarded.
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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).
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.
Certificates list
PARAMETER DEFAULT VALUES DESCRIPTION
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Valid Until Empty 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.
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.
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.
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PASSWORD
Current Password Admin Can be modified to any supported character and number
The system supports 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.
Local Local Drop down menu to select between local central directory
and LDAP based central directory
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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.
LDAP
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LDAP Server 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.
Port Empty The server port number that is open for LDAP connections.
LDAP filter Empty LDAP Filter is used to as a search filter, e.g. setting LDAP
filter to (|(givenName=%*)(sn=%*)) the IP-DECT will use this
filter 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 Empty Bind is the username that will be used when the IP-DECT
phone
connects to the server
Work Number Empty Work number is used to specify that LDAP attribute that will
be mapped to the handset work number
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Home Number Empty Home number is used to specify that LDAP attribute that will
be mapped to the handset home number
Mobile Number Empty Mobile number is used to specify that LDAP attribute that
will be mapped to the handset mobile number
REPEATERS
Within this section we describe the repeater parameter, and how to operate the repeater.
ADD REPEATER
PARAMETERS DESCRIPTION
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DECT sync mode Manually: User controlled by manually assign “Repeater RPN” and “DECT sync
source RPN”
Local Automatical: Repeater controlled by auto detects best base signal and
auto assign RPN.
Manually
User controlled by manually assign “Repeater RPN” and “DECT sync source RPN”. The
parameters are selected from the drop down menu.
Screenshot
PARAMETERS DESCRIPTION
DECT sync source Select the base or repeater the repeater has to be synchronized to.
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Local Automatical
Repeater controlled by auto detects best base signal and auto assign RPN. The RPN and
DECT sync source are greyed out.
REGISTER REPEATER
Adding a repeater makes it possible to register the repeater. Registration is made by select
the repeater and pressing register repeater. The base window for repeater registration will be
open until the registration is stopped. By stopping the registration all registration on the
system will be stopped inclusive handset registration.
REPEATERS LIST
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PARAMETERS DESCRIPTION
RPN The Radio Fixed Part Number is an 8-bit DECT cell identity allocated by the
installer. The allocated RPN within the must be geographically unique.
Permitted Output: 0 to 255 (DEC) OR 0x00 to 0xFF (HEX)
Name/IPEI Contains the name and the unique DECT serial number of the repeater. If name
is given the field will be empty.
DECT sync Source The “multi cell chain” connection to the specific Base/repeater unit. Maximum
number of chain levels is 12.
Sync. source format: “RPNyy (-zz dBm)”
yy: RPN of source
zz: RSSI level seen from the actual repeater
DECT sync Mode Manually: User controlled by manually assign “Repeater RPN” and “DECT sync
source RPN”
Local Automatical: Repeater controlled by auto detects best base signal and
auto assign RPN.
Chaining Automatical: Base controlled by auto detects best base or repeater
signal and auto assign RPN. This feature will be supported in a future version
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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.
SYSTEM DATA
The table is organized with headline row, data pr. base rows and with last row containing the
sum of all base parameters.
PARAMETERS DESCRIPTION
Base Station Name Base IP address and base station name from management settings
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 Failed SIP registrations count the number of times a SIP registration has
failed
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Handset Removed Handset removed count is the number of times a handset has been marked
as removed
Searching 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 Changed Number of time a base has changed sync source
CALL DATA
The call data web is access by http://ip/CallStatistics.html and data are organized in a table
as shown in below example.
Screenshot
The table is organized with headline row, data pr. base rows and with last row containing the
sum of all base parameters.
PARAMETERS DESCRIPTION
Base Station Name Base IP address and base station name from management settings
Operation Total operation time for the base since last reboot or reset
time/Duration Duration is the time from data was cleared or system has been firmware
upgraded.
Dropped 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 Call duration is total time that calls are active on the base.
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.
Max Active Maximum active calls are the maximum number of calls that has been active at
the same time.
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PARAMETERS DESCRIPTION
REPEATER DATA
The table is organized with headline row, data pr. base rows and with last row containing the
sum of all base parameters.
PARAMETERS DESCRIPTION
Idx Base IP address and base station name from management settings
Operation time/Duration Total operation time for the repeater since last reboot or reset
Duration is the time from data was cleared or system has been firmware
upgraded.
Busy Busy Count is the number of times the repeater has been busy.
Busy Duration Busy duration is the total time a repeater has been busy for speech (5 or
more calls active).
Max Active Maximum active calls are the maximum number of calls that has been
active at the same time.
Searching Repeater searching is the number of times a repeater has been searching
for it’s sync source
Recovery In case the sync source is not present anymore the repeater will go into lock
on another base or repeater and show recovery mode
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DECT Source Changed Number of time a repeater has changed sync source
DECT DATA
The DECT data web is access by http://ip/DectStatistics.html and data is organized in a table
as shown in below example.
Screenshot
Please note that frequencies 0, 1 and 2 were manually removed in the example above.
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.
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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.
To dump the logs, simply copy and paste the full contents.
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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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FIRMWARE UPGRADES
DOWNLOAD FIRMWARE FILES
STEP 3 Under Support, click Software Downloads, and then click IP DECT.
This procedure describes how to upgrade the base station and handset firmware. You can
also use this procedure to upgrade the repeater firmware.
STEP 1 Create a folder on the tftp server for the firmware files. For example:
C:\TFTP\9430\
C:\TFTP\8430\
STEP 2 Copy the firmware files into their respective folders. The 9430 folder is for the
base station and the 8430 folder is for the handset:
C:\TFTP\9430\9430_v0355_b0004.fwu
C:\TFTP\8430\8430_v0355_b0004.fwu
STEP 3 In the TFTP server settings, enter C:\TFTP in the Base Directory field and
change the Timeout to 20 seconds.
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STEP 3 Login to the Mitel 112 DECT base station management interface.
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STEP 6 Monitor the log on the TFTP server to confirm that the file transfer is taking place.
The Base LED starts flashing (orange, then red, then solid green). The Base
station performs its upgrade first. Then, the phone firmware is transferred and the
handset is upgraded.
The firmware upgrade is confirmed by the FWU Progress status in the second and first right
column on the handset extension list or repeater list. The “FWU info” column contains the
software version and the “FWU Progress” column contains the status. In case status is
“Complete”, the unit is firmware upgraded.
Alternatively the handset firmware can be verified from the Handset Menu by navigate to
Settings > Scroll down to Status this will list information regarding Base station and Handset
firmware versions.
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FUNCTIONALITY OVERVIEW
So far we have set up our system. Next, in this chapter we list what features and
functionalities are available in the system. The System supports all traditional and advanced
features of most telephony networks. In addition, 3rd 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 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 RTX System also supports robust security functionalities for Base-station.
Most security 1 functionality is intrinsically woven into the VOIP network structure so that
network connections can be encrypted and terminal authentication can be performed.
• Hospitality: For Hotel/Motel environments you can apply the following system behaviors
by enabling the Hotel Mode setting in Management Settings page of the of the IP
DECT web configuration interface:
o Black out the handset display when placed in cradle (after 65 seconds)
o Protect the handset Settings menu (changes default handset PIN from 0000 to 9351 and the
PIN is required to access the Settings menu)
o Enable silent upgrades and resets
o Disable call logging
o Prevent phonebook modification.
Keys
LED indicator
1
With active security 4 channels is supported
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RF
Software upgrade
Temperatures
SOFTWARE FEATURES
CODEC’s
G.722 Yes
G.726 Yes
SIP
RFC3326 The Reason Header Field for the Session Initiation Protocol
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RFC3326 The Reason Header Field for the Session Initiation Protocol (SIP)
RFC3489 STUN
RFC3581 Rport
RFC3960 Early Media and Ringing Tone Generation in the Session Initiation
Protocol (SIP)
In-band DTMF No
Web server
Other features
Provisioning Yes
IP features
IPv4 Yes
TCP/IP/UDP Yes
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Static IP Yes
Quality of service Type of Service (ToS) including DiffServ Tagging, and QoS per IEEE
802.1p/q
DECT
Intercom No
GAP compliant No
CAT-IQ compliant No
Handset registrations 20
CALL FEATURES
Call supported 5 simultaneous call supported
Simultaneous calls/handset 2
Codec Switching
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Parallel calls
Call Hold
Call Retrieve
Conference (3PTY)
Conference, Network
Outgoing call
Call Toggle/Swap
Incoming call
Line identification
Multiple Lines
Multiple calls
Call identification
Distinctive Ringing
- LDAP Yes
DND: Yes
Call Forward: Configurable from base or handset (Not with Call Group active))
- CFU Yes
- CFNA Yes
- CFB Yes
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SERVER SETUP
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 Base-station 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.
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.
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.
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.
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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.
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.
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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:
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There are several TFTP servers in the market place; in this section we describe how to setup
a commonly used TFTP 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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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.
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.
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.
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• All switch ports can communicate with one another at the Data Link Layer, if they become
members of the same VLAN.
• 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 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.
2. 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.
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.
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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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-tag-
aware 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.
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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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 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.
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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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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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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
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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.
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Next, click on the Load button. This will import the contents of contacts in the selected file
into the relevant Base station.
Alternative way to import a Contact List is to get it from a server. Click Management to
access the Management Settings page, then select the protocol of your server
(TFTP/HTTP/HTTPS) in Management Transfer Protocol, then save the setting by clicking
Save.
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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.
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On the Handset, navigate to Central Directory. The contact list should be populated with the
list of contacts that you uploaded to the base station.
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