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Mitel RFP 12 System Guide

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0% found this document useful (0 votes)
734 views

Mitel RFP 12 System Guide

Terminal vozip mitel rfp 12

Uploaded by

Carlos Benitez
Copyright
© © All Rights Reserved
Available Formats
Download as PDF, TXT or read online on Scribd
You are on page 1/ 100

Mitel RFP 12 Single Cell DECT

VOIP SYSTEM GUIDE


Release 1.0
VoIP System Guide

NOTICE
The information contained in this document is believed to be accurate in all respects but is not
warranted by Mitel Networks™ Corporation (MITEL®). The information is subject to change without
notice and should not be construed in any way as a commitment by Mitel or any of its affiliates or
subsidiaries. Mitel and its affiliates and subsidiaries assume no responsibility for any errors or
omissions in this document. Revisions of this document or new editions of it may be issued to
incorporate such changes.
No part of this document can be reproduced or transmitted in any form or by any means - electronic
or mechanical - for any purpose without written permission from Mitel Networks Corporation.

TRADEMARKS
The trademarks, service marks, logos and graphics (collectively "Trademarks") appearing on Mitel's
Internet sites or in its publications are registered and unregistered trademarks of Mitel Networks
Corporation (MNC) or its subsidiaries (collectively "Mitel") or others. Use of the Trademarks is
prohibited without the express consent from Mitel. Please contact our legal department at
[email protected] for additional information. For a list of the worldwide Mitel Networks Corporation
registered trademarks, please refer to the website: http://www.mitel.com/trademarks.

VoIP System Guide


Release 1.0 - January

®,™ Trademark of Mitel Networks Corporation


© Copyright 2016 Mitel Networks Corporation
All rights reserved

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VoIP System Guide

ABOUT THIS DOCUMENT ........................................................................................ 7


Audience .............................................................................................................................. 7
When Should I Read This Guide ......................................................................................... 7
Important Assumptions ........................................................................................................ 7
COntents of this Guide ........................................................................................................ 2
Abbreviations ....................................................................................................................... 2
References/Related Documentation .................................................................................... 3

INTRODUCTION – SYSTEM OVERVIEW ................................................................ 4


Hardware Setup ................................................................................................................... 4
Components of MITEL RFP 12 Single Cell DECT System .................................................. 4
MITEL 112 DECT Phone ..................................................................................................... 4
Base Station ........................................................................................................................ 4
Repeater .............................................................................................................................. 4
VoIP Administration Interface .............................................................................................. 5
Wireless Bands .................................................................................................................... 5
System Capacity .................................................................................................................. 5

MAKE HANDSET READY ......................................................................................... 6


Package Inspection ............................................................................................................. 6
Contents .............................................................................................................................. 6
Before Using the Phone ....................................................................................................... 7
Open Back Cover ................................................................................................................ 7
Record Handset Serial Number (IPEI Number) ................................................................... 8
Install the Battery ................................................................................................................. 8
Charge the Battery ............................................................................................................... 8
Using the Handset ............................................................................................................... 9

INSTALL BASE STATION/REPEATER ..................................................................... 9


Package Inspection ............................................................................................................. 9
Package Contents ............................................................................................................... 9
Base Station Mechanics .................................................................................................... 10
Base Station – Reset Feature ............................................................................................ 10
Installing the Base Station ................................................................................................. 10
Determine IP Address of Base Station .............................................................................. 11

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VoIP System Guide

CONFIGURE COMMUNICATIONS PLATFORM ..................................................... 11


Program MiVoice Business Phones .................................................................................. 11
Configure a PRG with Call Handoff (Optional) .................................................................. 12
Configure for Suite Services (Optional) ............................................................................. 12
MiVoice Office 250 SIP Phone Programming .................................................................... 12
Configure Dynamic Extension Express (Optional)............................................................. 13

CONFIGURE VOIP SYSTEM .................................................................................. 13


Login to VoIP System Administration Interface ................................................................. 13
Configure System Parameters........................................................................................... 15
Add Handsets And Extensions .......................................................................................... 21
Register the Handsets ....................................................................................................... 23

VOIP ADMINISTRATION INTERFACE .................................................................... 26


Web Navigation ................................................................................................................. 26
Home/Status ...................................................................................................................... 27
Extensions ......................................................................................................................... 29
Add Extension.................................................................................................................... 29
Group Call.......................................................................................................................... 31
Extensions List................................................................................................................... 32
Handset List ....................................................................................................................... 33
Edit Extension .................................................................................................................... 34
Servers .............................................................................................................................. 35
Network.............................................................................................................................. 39
IP Settings ......................................................................................................................... 39
VLAN Settings ................................................................................................................... 40
DHCP Options ................................................................................................................... 41
NAT Settings...................................................................................................................... 41
SIP/RTP Settings ............................................................................................................... 42
Management Settings Definitions ...................................................................................... 44
Firmware Update Definitions ............................................................................................. 48
Time Server ....................................................................................................................... 49
Country .............................................................................................................................. 52
Security .............................................................................................................................. 53
Certificates ......................................................................................................................... 53
SIP Client Certificates ........................................................................................................ 54

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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

FIRMWARE UPGRADES ........................................................................................ 69


Download Firmware Files .................................................................................................. 69
Upgrade the Firmware ....................................................................................................... 69
Verification of Firmware Upgrade ...................................................................................... 71

FUNCTIONALITY OVERVIEW ................................................................................ 72


Base Station Interfaces ...................................................................................................... 72
Software Features ............................................................................................................. 73
Call Features ..................................................................................................................... 75

APPENDIX A: BASIC NETWORK SERVER(S) CONFIGURATION ........................ 78


Server Setup ...................................................................................................................... 78
Requirements .................................................................................................................... 78
DNS Server Installation/Setup ........................................................................................... 78
DHCP Server Setup ........................................................................................................... 79
DHCP Server tRoubleshooting .......................................................................................... 79
TFTP Server Setup ............................................................................................................ 81
TFTP Server Settings ........................................................................................................ 81

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APPENDIX B: USING BASE WITH VLAN NETWORK ............................................ 83


Introduction ........................................................................................................................ 83
Backbone/ VLAN Aware Switches ..................................................................................... 84
How VLAN Switch Work: VLAN Tagging ........................................................................... 85
Implementation Cases ....................................................................................................... 85
Base station Setup............................................................................................................. 86
Configure Time Server....................................................................................................... 87
VLAN Setup: Base Station................................................................................................. 88

APPENDIX C: LOCAL CENTRAL DIRECTORY FILE HANDLING .......................... 89


Central Directory Contact List Structure ............................................................................ 89
Central Directory Contact List Filename Format ................................................................ 90
Import Contact List to Central Directory............................................................................. 90
Central Directory using Server........................................................................................... 91
Verification of Contact List Import to Central Directory ...................................................... 93

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VOIP System Guide

ABOUT THIS DOCUMENT


This document describes the configuration, customization, management, operation,
maintenance and trouble shooting of the Mitel RFP 12 Single Cell DECT system (Mitel 112
DECT handset, base station, and repeaters).

AUDIENCE

This guide is intended for


• networking professionals responsible for designing and implementing the wireless
networks, and
• network administrators and IT support personnel that need to install, configure, maintain
and monitor components of the system.

WHEN SHOULD I READ THIS GUIDE

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

This document was written with the following assumptions:


1. You have understanding of network deployment in general.
2. You have working knowledge of basic TCP/IP/SIP protocols, Network Address
Translation, and so forth.
3. A proper site survey has been performed, and the administrator has access to the plans.

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VoIP System Guide

CONTENTS OF THIS GUIDE

The contents of this document are summarized in the table below:


SECTION PURPOSE

System Overview Describes the different elements in a typical VoIP


Network

Make Handset Ready Provides instructions on how to assemble handsets


for use in the system

Install Base Provides instructions for installing base units and


Station/Repeater repeaters

Configure Communication Provides an overview of the configuration required


Platform on the MiVoice Business or MiVoice Office 250
platforms to support the handsets.

Configure VoIP System Lists steps required to configure the system

VoIP Administration Describes the configuration interface and defines the


Interface parameters that are used to set up the system.

Firmware Upgrades Provides the procedure of how to upgrade firmware


to base stations and/or handsets and/or repeaters

Functionality Overview Describes system functionality and features.

Basic Network Servers Describes how to set up network servers.


Configuration

VLAN Setup Management Explains how to set up VLAN in the network

Local Central Directory File Describes the central directory file format and
Handling provides instructions on how to upload it.

ABBREVIATIONS

For the purpose of this document, the following abbreviations apply:


DHCP: Dynamic Host Configuration Protocol
DNS: Domain Name Server
HTTP(S): Hyper Text Transfer Protocol (Secure)
(T)FTP: (Trivial) File Transfer Protocol
IOS: Internetworking Operating System
IPEI International Portable Equipment Identity
PCMA: A-law Pulse Code Modulation
PCMU: mu-law Pulse Code Modulation
PoE: Power over Ethernet

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VOIP System Guide

RTP: Real-time Transport Protocol


RPORT: Response Port (Refer to RFC3581 for details)
SIP: Session Initiation Protocol
VLAN: Virtual Local Access Network
TOS: Type of Service (policy based routing)
URL: Uniform Resource Locator
UA: User Agent

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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VoIP System Guide

INTRODUCTION – SYSTEM OVERVIEW


The MITEL RFP 12 Single Cell DECT system is a VoIP solution with support for up to 20
registered handsets and three repeaters.

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.

COMPONENTS OF MITEL RFP 12 SINGLE CELL DECT SYSTEM

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

MITEL 112 DECT PHONE

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.

The RYX4024 provides the following features:


• Up to three repeaters are supported per base station
• Wide band audio

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VOIP System Guide

• DECT encryption
• Automatic registration
• Maximum of three repeaters in daisy chain.

VOIP ADMINISTRATION INTERFACE

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

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

SYSTEM CAPACITY

The network capacity of relevant components can be summarized as follows:


DESCRIPTION CAPACITY

Single Cell Setup 1

Maximum number of repeaters per base station 3

Maximum number of handsets (SIP registrations) per base station 20

Single Cell Setup: Maximum number of simultaneous calls 5

Repeater: Max number of calls (narrow band) 5

Repeater: Maximum number of calls (G.722) 2

Note: Each base station supports up to 20 handsets and three repeaters.

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VoIP System Guide

MAKE HANDSET READY


This section describes how to prepare the handset for use.

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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VOIP System Guide

BEFORE USING THE PHONE

OPEN BACK COVER


1. Press down the back cover and slide it towards the bottom of the handset.
2. Remove back cover from handset.

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VoIP System Guide

RECORD HANDSET SERIAL NUMBER (IPEI NUMBER)

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.

INSTALL THE BATTERY


1. Never dispose of a battery in a fire; otherwise it will explode.
2. Never replace the batteries in potentially explosive environments, for example close to
flammable liquids or gases.
3. ONLY use approved batteries and chargers from the vendor or operator.
4. Do not disassemble, customize, or short circuit the battery.

CHARGE THE BATTERY

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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VOIP System Guide

The battery displayed in the top right of the screen indicates the charging status.

USING THE HANDSET

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.

INSTALL BASE STATION/REPEATER


The following sections how to install the base station.

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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VoIP System Guide

BASE STATION MECHANICS

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:

LED STATE STATUS

OFF No power

FLASHING GREEN Initialization in progress

SOLID GREEN Ethernet connection is available (Normal operation)

FLASHING ORANGE No IP address

SOLID ORANGE Reset required

FLASHING RED Factory setting in progress


OR
Ethernet connection not available OR
Handset registration/deregistration failed.

SOLID RED Factory reset warning after a long press (10


seconds or more) of the Reset button
OR
Error condition. Replace base station if error condition persists.

BASE STATION – RESET FEATURE

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.

INSTALLING THE BASE STATION


1. Record the MAC address of the base station. The MAC address is listed on the bottom
panel of the base.
2. Determine the best location that will provide an optimal coverage taking account the
construction of the building, architecture, and building materials.
3. Mount the base station on a wall to cover a range of 50 meters (164 feet) for indoor
installations or 300 meters (984 feet) for outdoor installations. We recommend the base
station be mounted an angle on concrete, wood, or plaster pillars and walls for optimal
radio coverage. Do not mount the base units upside down because it significantly
reduces radio coverage.
4. Mount the base unit as high as possible to clear all nearby objects (for example: office
cubicles and cabinets). If necessary, extend coverage to remote offices or halls with
fewer telephony users by installing repeaters.
5. When you fasten the base stations to the pillar or wall, ensure that the screws do not
touch the PC board in the unit. Secondly, avoid all contact with any high voltage lines.

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VOIP System Guide

DETERMINE IP ADDRESS OF BASE STATION

To identify the IP address of the base station:


1. On the handset press the round “Menu” button to access the main menu:

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.

CONFIGURE COMMUNICATIONS PLATFORM


PROGRAM MIVOICE BUSINESS PHONES

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.

• Replace System based with Device based In-Call Features

• Enable Digit Collection In Busy Or Alerting State

• Allow Display Updates

• Enable Distinctive Ringing

• Prevent the Use of IP Address 0.0.0.0 in SDP Messages.


4. Use the Search field in the “User and Device Configuration” form to locate the directory
numbers that will be assigned to the handsets. Click the Service Details tab and assign
the SIP Device Capabilities index number to each handset.

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VoIP System Guide

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 in a Personal Ring Group, or

• Mitel desktop phone and a handset for Suite Services (typically, used in a hospitality
environment).

CONFIGURE A PRG WITH CALL HANDOFF (OPTIONAL)

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.

CONFIGURE FOR SUITE SERVICES (OPTIONAL)

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.

MIVOICE OFFICE 250 SIP PHONE PROGRAMMING

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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VOIP System Guide

CONFIGURE DYNAMIC EXTENSION EXPRESS (OPTIONAL)

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.

CONFIGURE VOIP SYSTEM


This section describes basic configuration of the system. See VoIP Administration Interface
on page 26 for descriptions of the system parameter settings.

LOGIN TO VOIP SYSTEM ADMINISTRATION INTERFACE


1. Connect the base station to a private network via standard Ethernet cable (CAT-5).
2. Use the IP Search function on the handset to determine the IP address of the base
station:

• Press the center Menu button on the handset to access the main menu:

• 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.

• 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.

• Record the IP address.


3. Open a standard internet browser (for example, FireFox)
4. In the browser address bar enter http//<IP Address of Base Station>.
Username: admin (default)
Password: admin (default)
5. Click OK.

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VoIP System Guide

6. The browser displays the Welcome page of the VoIP Administration interface. It lists the
base station information.

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VOIP System Guide

CONFIGURE SYSTEM PARAMETERS

From the VoIP Administration interface, perform the following configuration:


1. Click Servers.

• 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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VoIP System Guide

2. Click Network:

• Set DHCP/Static field to “Static” (recommended).

• Enter the IP address of the base station.

• Enter the IP address of the Default Gateway (if required).

• Click Save.

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VOIP System Guide

3. Click Management.

• If you are deploying the handsets in a hospitality (Hotel/Motel) environment, enable


Hotel Mode. Note that when this option is enabled, it changes default handset PIN
from 0000 to 9351 and the PIN is required to access the Settings menu.

• Click Save.

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VoIP System Guide

4. Click Time.

• Set the system time.

• Click Save.

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VOIP System Guide

5. Click Country.

• Set the country settings.

• Click Save.

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VoIP System Guide

6. Click Security:

• Under Password, change the administrator password (default admin) used to


access this interface.

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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VOIP System Guide

ADD HANDSETS AND EXTENSIONS


1. Click Extensions and add the handsets.

• Click Handset.

• Click Add 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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VoIP System Guide

2. Click Extensions and add the extensions.

• Click Add extension

• Enter the Extension number.

• Enter the user’s Display Name.

• Select the Server (MiVoice communications platform).

• Under Select Handset(s), check the box to associate the extension with a handset.

• Click Save.

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VOIP System Guide

REGISTER THE HANDSETS


1. Open base station to handset registration:
• Click Extensions.
• Optionally, change the AC (Access code). You enter the AC on the handset to initiate
registration.
• Check the boxes of the handsets that you want to register.
• Click Register Handset(s).

2. The parameters are saved.

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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VoIP System Guide

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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VOIP System Guide

5. The following screen shows an example of the Extensions page after you have registered
several handsets.

6. Initial system configuration is now complete.


Note: After you have configured the handsets on a base station, ensure that you have
changed the administration interface username and password, and the handset AC codes
from the default values to prevent unauthorized access.

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VoIP System Guide

VOIP ADMINISTRATION INTERFACE


You manage and troubleshoot the system through the VoIP Administration Interface. The
interface is an HTTP Web Server service that resides in each base station.

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.

WEB PAGE DESCRIPTION

Home/Status This “Welcome” page displays the system information and base station status.

Extensions Manage the system handsets and extensions

Servers Define which SIP/NAT server the network should connect to.

Network Configure the Network settings:


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: specify either dynamic (DHCP) or static IP address for your network. Only
complete the IP address if you using a static IP address, Otherwise, leave it blank.
Subnet mask: Leave blank if using DHCP. Complete if assigning a static IP address.
DNS server: Specify if using DHCP; otherwise, leave it blank. Enter the DNS server
address of your Internet service provider. If you are using a static IP address the 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, 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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VOIP System Guide

WEB PAGE DESCRIPTION


Note: LDAP and central directory cannot operate at the same time.

Repeaters Administration and configuration of repeaters of the system

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.

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.

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

This section describes the Home/Status page.

PARAMETER DESCRIPTION

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VoIP System Guide

System information This base current multi-cell state

Phone Type Always IPDECT

System Type This base customer configuration

RF Band This base RF band setting

Current local time This base local time

Operation time Time from last boot of base

RFPI-Address This base RFPI address

MAC-Address This base MAC address

IP-Address This base IP address

Firmware version This base firmware version

Firmware URL Firmware update server address and firmware path on server

Base Station Status “Idle” : When no calls on base


“In use” : When active calls on base

SIP Identity Status on List of extensions present at this base station.


this Base Station Format: “extension”@“this base IP address” followed by status to the right.
Below is listed possible status:
OK: Handset is registered
SIP Error: SIP registration error

Reboot Reboot after all connections is stopped on base. Connections are active
call, directory access, firmware update active

Forced Reboot Reboot immediately even active calls are ongoing.

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VOIP System Guide

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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VoIP System Guide

PARAMETER DEFAULT DESCRIPTION


VALUE(S)

Extension Empty Handset phone number depending on the setup.


Possible value(s): 8-bit string length
Example: 1024
Note: The Extension must also be configured in SIP server in
order for this feature to function.

Authentication Empty Username: SIP authentication username


User Name Permitted value(s): 8-bit string length

Authentication Empty Password: SIP authentication password.


Password Permitted value(s): 8-bit string length

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.

Server Server 1 IP FQDN or IP address of SIP server.


Drop down menu to select between the defined Servers of VoIP
Service provider.

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.

Forwarding Empty Number to which incoming calls must be re-routed, regardless of


Unconditional the current state of the handset.
Number Forwarding Unconditional must be enabled to function.
Disabled 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

Empty Number to which incoming calls must be re-routed to when there is


Forwarding No no response from the SIP end node.
Answer Number Forwarding No Answer Number must be enabled to function.
Disabled Note: Feature must be enabled in the SIP server before it can
function in the network.
Specify delay from call to forward in seconds.
90 Note: Feature is automatically disabled if the handset or extension
is part of a group.

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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PARAMETER DEFAULT DESCRIPTION


VALUE(S)

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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VoIP System Guide

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.

PARAMETER DESCRIPTION

Idx 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 Server IP or URL

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.

IPEI 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.

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VOIP System Guide

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.

PARAMETER DESCRIPTION

Idx 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 Handset type and firmware version of handset


FW info

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.

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VoIP System Guide

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

In the below table each command is described.

ACTIONS DESCRIPTION

Add extension Access to the “Add extension” sub menu

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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VOIP System Guide

SERVERS

In this section, we describe the different parameters available in the Servers configurations
menu. A maximum of 10 servers can be configured.

PARAMETER DEFAULT VALUE DESCRIPTION

Server Alias Empty Parameter for server alias

NAT Adaption Disabled To ensure all SIP messages goes directly to the NAT
gateway in the SIP aware router.

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VoIP System Guide

PARAMETER DEFAULT VALUE DESCRIPTION

Registrar Empty 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.

Outbound Proxy Empty 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
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”.

Conference Server Empty 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.

Call Log Server Empty 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

Re-registration time 600 The “expires” value 36nalyse36n 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

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

SIP Transport UDP Select UDP, TCP, TLS 1.0

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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VOIP System Guide

PARAMETER DEFAULT VALUE DESCRIPTION

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
0) will not be used. The “Signal TCP Source Port”
parameter specifies if the used source port shall be
signaled explicitly in the SIP messages.

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.

Show Extension on Enabled If enabled extension will be shown on handset idle


Handset Idle Screen screen.

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

Use Own Codec Disabled Default disabled.


Priority By enable the system codec priority during incoming

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VoIP System Guide

PARAMETER DEFAULT VALUE DESCRIPTION

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.

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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VOIP System Guide

PARAMETER DEFAULT VALUE DESCRIPTION

_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

PARAMETER DEFAULT DESCRIPTION


VALUES

DHCP/Static IP DHCP 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.

IP Address NA 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

Subnet Mask NA 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.

Default Gateway NA Device’s default network router/gateway (32-bit).

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VoIP System Guide

PARAMETER DEFAULT DESCRIPTION


VALUES

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.

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.

DNS (Secondary) NA This is an alternate DNS server.

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.

PARAMETER DEFAULT DESCRIPTION


VALUES

VLAN id 0 Is a 12 bit identification of the 802.1Q VLAN.


Permitted value(s): 0 to 4094 (only decimal values are
accepted)
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.

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)

For further help on VLAN configuration refer to Appendix.

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VOIP System Guide

DHCP OPTIONS

PARAMETER DEFAULT DESCRIPTION


VALUES

Plug-n-Play Disabled Enabled: DHCP option 43 to automatically provide PBX IP


address to base.

NAT SETTINGS

We define some options available when NAT aware routers are enabled in the network.

PARAMETER DEFAULT DESCRIPTION


VALUES

Enable STUN Disabled Enable to use STUN

STUN Server NA Permitted value(s): AAA.BBB.CCC.DDD (Currently only Ipv4 are


supported) or url (e.g.: firmware.rtx.net).

STUN Bindtime Enabled


Determine

STUN Bindtime 80 Permitted values: Positive integer default is 90, unit is in seconds
Guard

Enable RPORT Disabled Enable to use RPORT in SIP messages.

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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VoIP System Guide

SIP/RTP SETTINGS

These are some definitions of SIP/RTP settings:

PARAMETER DEFAULT DESCRIPTION


VALUES

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.

RTP Collision Enabled


Detection

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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VOIP System Guide

PARAMETER DEFAULT DESCRIPTION


VALUES

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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VoIP System Guide

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 DEFAULT VALUE DESCRIPTION

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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VOIP System Guide

PARAMETER DEFAULT VALUE DESCRIPTION

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

Enable Automatic Disabled


Prefix

Set Maximum 0
Digits of Internal
Numbers

Set Prefix for Blank


Outgoing Calls

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 Empty Server/device that provides configuration file to base


server address station.
Type: DNS or IP address
Permitted value(s): AAA.BBB.CCC.DDD or <URL>

Base Specific File Empty Base configuration file

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 Controlled Disabled Provisioning server options.


Config Server 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.

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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VoIP System Guide

PARAMETER DEFAULT VALUE DESCRIPTION

DHCP Custom Empty URL: URL of server with path.


Option Typr 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.

Text Messaging Disabled 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

Text Messaging & Empty Permitted value(s): AAA.BBB.CCC.DDD or <URL>


Alarm server

Text Messaging 1300 Port number of message server.


Port

Text Messaging 30 This defines the frequency of how keep-alive are sent
Keep Alive (m) Permitted values: Positive integer, unit is in minutes

Text Messaging 30 This defines the frequency of how response timeout


Response (s) Permitted values: Positive integer, unit is in seconds

Text Messaging 0 This defines the text messaging time to live


TTL Permitted values: Positive integer, unit is in seconds

SIP Log Server Empty Permitted value(s): AAA.BBB.CCC.DDD or <URL>


Address Requires a predefined folder named: \SIP

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

Syslog Server IP- NA Permitted value(s): AAA.BBB.CCC.DDD or <URL>


Address

Syslog Server Port NA Port number of syslog server.

Syslog Level 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 46nalyse level also contains
the messages from normal operation.
Debug: Used by Design for debug. Should not be enabled
during normal operation.

Enable Automatic Disabled Disabled: Feature off.


Prefix Enabled: The base will add the leading digit defined in “Set

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VOIP System Guide

PARAMETER DEFAULT VALUE DESCRIPTION

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

Set Maximum 0 Used to detect internal numbers. In case of internal


Digits of Internal numbers no prefix number will be added to the dialled
Numbers number.

Set Prefix for Empty Prefix number for the enabled automatic prefix feature.
Outgoing Calls
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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VoIP System Guide

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.

PARAMETER DEFAULT VALUE(S) DESCRIPTION

Firmware update Empty IP address or DNS of firmware update files source


server address Valid Inputs: AAA.BBB.CCC.DDD or <URL>
Example: firmware.rtx.net or 10.10.104.41

Firmware path Empty 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

Required Version Empty Version of firmware to be upgraded (or downgraded) on


Type 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.

Required Version Empty Version of firmware to be upgraded (or downgraded) on


Base 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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VOIP System Guide

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.

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VoIP System Guide

PARAMETER DEFAULT DESCRIPTION


VALUES

Time Server 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.

Allow broadcast NTP Checked

Refresh time (h) Empty The window time in hours within which time server
refreshes.
Valid Inputs: positive integer

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VOIP System Guide

PARAMETER DEFAULT DESCRIPTION


VALUES

Set timezone by Checked By checked country setting is used (refer to country web
country/region page).

Time Zone 0 Refers to local time in GMT or UTC format.


Min: -12:00
Max: +13:00

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 Month March Month that DST begins


Valid Input(s): Gregorian months (e.g. January, February,
etc.)

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 Time 3 DST start time in the day


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 Time 2 The time of day DST stops


Valid Inputs: positive integer (1 to 12)

DST Stop Day of Sunday The day of week DST stops


Week

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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VoIP System Guide

COUNTRY

The country setting controls the in-band tones used by the system. To select web interface
language go to the management page.

PARAMETER DEFAULT VALUES DESCRIPTION

Select Country 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 NA Only shown by country selection US/Canada, Autralia,


Brasil

Select Language English Web interface language. Number of available languages:


English, Dansk, Italiano, Tyrkie, Deutsch, Portuguese,
Hrvatski, Srpski, Slovenian, Nederlands, Francaise,
Espanol, Russian, Polski.

Set timezone by checked When checked timezone will follow country/region


country/region

Set DST by checked When checked DST will follow country/region


country/region

Notes 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.

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VOIP System Guide

The following types of in-band tones are supported:


1. Dial tone
2. Busy tone
3. Ring Back tone
4. Call Waiting tone
5. Re-order tone

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.

The certificate format supported is DER encoded binary X.509 (.cer).

Certificates list
PARAMETER DEFAULT VALUES DESCRIPTION

Idx Fixed indexes Index number

Issued To Empty IP address – which is part of the certificate file

Issued To Empty Organization, Company – which is part of the certificate file

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VoIP System Guide

PARAMETER DEFAULT VALUES DESCRIPTION

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.

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.

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VOIP System Guide

PASSWORD

In the below the password parameters are defined.

PARAMETER DEFAULT VALUES DESCRIPTION

Username Admin Can be modified to any supported character and number

Current Password Admin Can be modified to any supported character and number

New Password Empty Change to new password

Confirm Password Empty Confirm password to reduce accidently wrong changes of


passwords

Password valid special signs: @/|<>-_:.!?*+#

Password valid numbers: 0-9

Password valid letters: a-z and A-Z

CENTRAL DIRECTORY AND LDAP

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 CENTRAL DIRECTORY

Select local and save for local central directory.


Screenshot

PARAMETER DEFAULT VALUES DESCRIPTION

Local Local Drop down menu to select between local central directory
and LDAP based central directory

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VoIP System Guide

Server Empty 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.

Filename Empty The parameter is used if directory file is located on server.


Refer to appendix for further details

Phonebook reload 0 The parameter is controlling the reload interface of


interval (s) 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.

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.

LDAP

In the Location field, select LDAP Server and click Save.

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VOIP System Guide

PARAMETER DEFAULT VALUES DESCRIPTION

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.

Server Empty IP address of the LDAP server.


Valid Inputs: AAA.BBB.CCC.DDD or <URL>

Port Empty The server port number that is open for LDAP connections.

Sbase Empty 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 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

Password Empty Password is the password for the LDAP Server

Name Empty The name can be used to specify if sn+givenName or cn


(common name) is return in the LDAP search results

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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VoIP System Guide

PARAMETER DEFAULT VALUES DESCRIPTION

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

From repeaters web select “Add Repeater”


Screenshot

Then select “DECT Sync mode”


Screenshot

PARAMETERS DESCRIPTION

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VOIP System Guide

Name Repeater name. If no name specified the field will be empty

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

Idx System counter

RPN SINGLE CELL SYSTEM:


The base has always RPN00, first repeater will then be RPN01, second repeater
RPN02 and third RPN03 (3 repeaters maximum per base)

MULTI CELL SYSTEM:


Bases are increment by 2^2 in hex, means first base RPN00 second base RPN04
etc., in between RPN01, 02, 03 addressed for repeaters at Primary base and 05,
06, 07 addressed for Secondary base (3 repeaters maximum per base)

DECT sync source Select the base or repeater the repeater has to be synchronized to.

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VoIP System Guide

Local Automatical

Repeater controlled by auto detects best base signal and auto assign RPN. The RPN and
DECT sync source are greyed out.

The repeater RPN is dynamic assigned in base RPN range.

With local automatical mode, repeater on repeater (chain) is not supported.

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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VOIP System Guide

PARAMETERS DESCRIPTION

IDx Repeater unit identity in the chained network.


Permitted Output: Positive Integers

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

State Present@unit means connected to unit with RPN yy

FW info Firmware version

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
”Conn. term. wait” (Repeater): All FWU is complete and is now waiting for
connections to stop before repeater restart.
Complete HS/repeater: FWU complete
Error: Not able to fwu e.g. file not found, file not valid etc

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VoIP System Guide

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 system data web is access by http://ip/SystemStatistics.html and data is 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 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 Failed SIP registrations count the number of times a SIP registration has
failed

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VOIP System Guide

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.

Count Counts number of calls on a base.

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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VoIP System Guide

PARAMETERS DESCRIPTION

Codecs Logging and count of used codec types on each call.

Handover Success Counts the number of successful handovers.

Handover Failed Counts the number of failed handovers.

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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VOIP System Guide

DECT Source Changed Number of time a repeater has changed sync source

Wide Band Number of wideband calls on repeaters

Narrow Band Number of narrow band calls on repeaters

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.

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.

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VoIP System Guide

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
....
....

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VOIP System Guide

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 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
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 logs, simply copy and paste the full contents.

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VoIP System Guide

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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VOIP System Guide

FIRMWARE UPGRADES
DOWNLOAD FIRMWARE FILES

STEP 1 Log into Mitel Connect (https://connect.mitel.com/connect/).

STEP 2 Access Mitel Online.

STEP 3 Under Support, click Software Downloads, and then click IP DECT.

STEP 4 Download the ZIP file of the firmware files.

UPGRADE THE FIRMWARE

This procedure describes how to upgrade the base station and handset firmware. You can
also use this procedure to upgrade the repeater firmware.

Note: In the following example, the TFTP server is running on a PC.

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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VoIP System Guide

STEP 3 Login to the Mitel 112 DECT base station management interface.

STEP 4 Click Firmware Update.


 In the Firmware update server address field, enter the IP address of the
TFTP server.
 Leave the Firmware path blank.
 Leave the Image path field blank.
 Set the Required version field to the last three digits of the file version. For
example, for firmware file 9430_v0355_b0004.fwu, enter 355.
 Set the Required branch field. For example, for firmware file
9430_v0355_b0004.fwu, enter 004.

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VOIP System Guide

STEP 5 Click Save/Start Update.

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.

VERIFICATION OF FIRMWARE UPGRADE

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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VoIP System Guide

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.

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

1: Reset key, Page and Default

LED indicator

One Status LED (multicolour, red, green, orange)

1
With active security 4 channels is supported

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VOIP System Guide

RF

Frequency Bands 1880 – 1900 MHz (EMEA)


1910 – 1930 MHz (Latam)
1920 – 1930 MHz (USA)
Factory setting which can’t be modified after production

Output Power 250 mW or 140mW depending on country version

Antenna Two antennas for diversity

Software upgrade

Downloadable Remote firmware update using HTTP, HTTPS or TFTP

Temperatures

Operation 0˚C to 40˚C

SOFTWARE FEATURES
CODEC’s

G.711 PCM A-law & U-law Yes

G.722 Yes

G.726 Yes

G.729 A/AB (including VAD), max 4 coders


G729 licence not included

SIP

RFC2327 SDP: Session Description Protocol

RFC2396 Uniform Resource Identifiers (URI): Generic Syntax

RFC2833 In-Band DTMF/Out of band DTMF support

RFC2976 The SIP INFO method

RFC3261 SIP 2.0

RFC3262 Reliability of Provisional Responses in the Session Initiation Protocol


(PRACK)

RFC3263 Locating SIP Servers (DNS SRV, redundant server support)

RFC3264 Offer/Answer Model with SDP

RFC3265 Specific Event Notification

RFC3326 The Reason Header Field for the Session Initiation Protocol

RFC3311 The Session Initiation Protocol UPDATE Method

RFC3325 P-Asserted Identity

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RFC3326 The Reason Header Field for the Session Initiation Protocol (SIP)

RFC3489 STUN

RFC3515 REFER: Call Transfer

RFC3550 RTP: A Transport Protocol for Real-Time Application

RFC3581 Rport

RFC3842 Message Waiting Indication

RFC3891 Replace header support

RFC3892 The Session Initiation Protocol (SIP) Referred-By Mechanism

RFC3960 Early Media and Ringing Tone Generation in the Session Initiation
Protocol (SIP)

RFC4475 Session Initiation Protocol (SIP) Torture Test Messages

SIPS Secure SIP

In-band DTMF No

SRTP Yes, packet authentication will limit the number of calls to 4

SIP registrations max 20

RTP streams max 10

SIP transport UDP, TCP or TLS

Web server

Embedded web server, accessed using HTTP

Other features

IP quality Warning – Network outage, VoIP service outage

Jitter buffer Yes, adaptive

Automatic DST Yes

Tone Scheme Country Depend Tone Scheme

Provisioning Yes

Re-direct server Yes

SIP configuration Yes, from web page or configuration file

Call groups Yes

IP features

IPv4 Yes

IPv6 Hardware ready, software not included

TCP/IP/UDP Yes

DHCP Support Yes

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DHCP option 66, 120

Static IP Yes

DNS srv Yes

VLAN Yes, 802.1p/q

Quality of service Type of Service (ToS) including DiffServ Tagging, and QoS per IEEE
802.1p/q

TLS Yes, 1.0

Certificates Yes, X.509 (certificate not included)

TFTP Yes, for firmware and configuration file download

HTTP server Yes

HTTP client Yes, for firmware and configuration file download

HTTPS Yes, for firmware and configuration file download

SNTP Yes, For internet clock synchronization

DECT

DECT handover Yes, inter-cell handover for repeater support

CAT-IQ v1.0 HD audio or NB audio support

Repeater support Yes

Intercom No

DECT encryption Yes

DECT Authentication Yes

Group TPUI support Yes, for call groups

GAP compliant No

CAT-IQ compliant No

Handset registrations 20

CALL FEATURES
Call supported 5 simultaneous call supported

Simultaneous calls/base 5 Wideband calls (g.722). 5 narrowband calls (PCMA, PCMU,


G.726) or 4 when using G729

Simultaneous calls/handset 2

Call features Codec Negotiation

Codec Switching

Missed call notification

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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

Multiple calls

Call identification

Calling Name Identification Presentation (CNIP)

Calling Line Identification Presentation (CLIP)

Call Completed Elsewhere

Distinctive Ringing

Central Phone Book:

- LDAP Yes

- XML Yes, remote or file load from web interface

- CSV Yes, file load through web interface

DND: Yes

Call Forward: Configurable from base or handset (Not with Call Group active))

- CFU Yes

- CFNA Yes

- CFB Yes

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Call groups: Yes, 1-20 handsets/SIP account

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APPENDIX A: BASIC NETWORK SERVER(S)


CONFIGURATION
In this chapter we describe how to setup the various server elements in the system.

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 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.

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.

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.

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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.

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.

DHCP SERVER TROUBLESHOOTING


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.

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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:

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. NAT/Firewall rules that block the DHCP bootp protocol on UDP ports 67 and 68;
3. 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.
4. Finally restart the dhcpd service daemon

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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.

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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APPENDIX B: USING BASE WITH VLAN NETWORK


In this chapter we describe how to setup a typical VLAN in the network.

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.

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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.

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:

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-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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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.

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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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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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APPENDIX C: LOCAL CENTRAL DIRECTORY FILE


HANDLING
This appendix the Local Central Directory file format, import and configuration is described.

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

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CENTRAL DIRECTORY CONTACT LIST FILENAME FORMAT

The Contact list is saved as file format: .txt .csv or .xml

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.

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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.

The figure below shows the import procedure is in process.

CENTRAL DIRECTORY USING SERVER

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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VERIFICATION OF CONTACT LIST IMPORT TO CENTRAL


DIRECTORY

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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