RTX9431 VOIP Phone

Product Information

Specifications

• Product Name: SME VoIP System

• Model: RTX9431 / D200 / 8328 SIP-DECT SINGLE BASE STATION / RFP
  14 Base Station NA series

• Version: 4.9

About This Document

This document provides detailed information on the
configuration, customization, management, operation, maintenance,
and troubleshooting of the SME VoIP System. It covers the RTX9431
base, RTX8630 handset, RTX8430 handset, RTX8830 ruggedized handset,
and RTX4024 Repeater in RTX generic mode. For customer-specific
modes, please refer to specific customer agreements.

Audience

This guide is intended for:

• Networking professionals responsible for designing and
  implementing RTX based enterprise networks.

• Network administrators and IT support personnel involved in the
  installation, configuration, maintenance, and monitoring of
  elements in a live SME VoIP network.

• Individuals seeking knowledge on fundamental features in the
  Beatus system.

When Should I Read This Guide

You should read this guide:

• Before installing the core network devices of VoIP SME
  System.

• When you are ready to set up or configure SIP server, NAT aware
  router, advanced VLAN settings, base stations, and multi-cell
  setup.

Important Assumptions

This document assumes:

1. You have a general understanding of network deployment.

2. You have a working knowledge of basic TCP/IP/SIP protocols,
   Network Address Translation, etc.

3. A proper site survey has been performed, and the administrator
   has access to these plans.

Contents

different elements in a typical SME
VoIP Network
3| Making Handsets Ready| To determine precautions to take in preparing handsets for use
in the system
4| SME VoIP Administration Interface| To learn about the Configuration Interface and define the full
meaning of various parameters needed to be set up in the
system
5| Multi-Cell Setup & Management| To learn how to add servers and set up multiple bases into a
multi-cell network. Also, learn how to register handsets and
extensions to base stations

Product Usage Instructions

Chapter 2: Installation of Base station/Repeater

This chapter provides instructions on how to install the base
stations and repeaters in your SME VoIP network.

Chapter 3: Making Handsets Ready

This chapter guides you through the process of preparing
handsets for use in the SME VoIP system. It covers the necessary
precautions and steps to ensure the handsets are ready for
registration.

Chapter 4: SME VoIP Administration Interface

In this chapter, you will learn about the Configuration
Interface of the SME VoIP System. It provides a comprehensive
explanation of various parameters that need to be set up in the
system. This includes adding servers and defining their
settings.

Chapter 5: Multi-Cell Setup & Management

This chapter focuses on setting up a multi-cell network in the
SME VoIP System. You will learn how to add multiple bases to the
network, register handsets and extensions to base stations, and
manage the multi-cell setup effectively.

FAQ

Q: What is the purpose of this document?

A: This document provides detailed information on the
installation, configuration, management, operation, maintenance,
and troubleshooting of the SME VoIP System.

Q: Who should read this guide?

A: This guide is intended for networking professionals, network
administrators, IT support personnel, and anyone interested in
gaining knowledge about the SME VoIP System.

Q: When should I read this guide?

A: You should read this guide before installing the core network
devices of VoIP SME System and when you are ready to set up or
configure various components of the system.

Q: What assumptions does this document make?

A: This document assumes that you have a general understanding
of network deployment, working knowledge of basic TCP/IP/SIP
protocols, and have performed a proper site survey.

SME VoIP System Guide for RTX9431 / D200 / 8328 SIP-DECT SINGLE BASE STATION /
RFP 14 Base Station NA series
Installation & Configuration Network Deployment Operation & Management

Trademarks SME VOIP SYSTEM GUIDE 4.6 Proprietary and Confidential

Technical Reference Document Version 4.9
© Dec-2019 RTX A/S, Denmark

RTX and the combinations of its logo thereof are trademarks of RTX A/S, Denmark. Other product names used in this publication are for identification purposes and maybe the trademarks of their respective companies. Disclaimer The contents of this document are provided about RTX products. RTX makes no representations with respect to completeness or accuracy of the contents of this publication and reserves the right to make changes to product descriptions, usage, etc., at any time without notice. No license, whether express, implied, to any intellectual property rights are granted by this publication. Confidentiality This document should be regarded as confidential, unauthorized copying is not allowed. © Dec-2019 RTX A/S, Denmark, All rights reserved http://www.rtx.dk
SME VOIP SYSTEM GUIDE 4.6 Proprietary and Confidential

Table of Contents
1 About This Document ………………………………………………………………………………………………………………………………. 7 1.1 Audience ……………………………………………………………………………………………………………………………………….. 7 1.2 When Should I Read This Guide ………………………………………………………………………………………………………… 7 1.3 Important Assumptions……………………………………………………………………………………………………………………. 7 1.4 What’s Inside This Guide………………………………………………………………………………………………………………….. 7 1.5 What’s Not in This guide ………………………………………………………………………………………………………………….. 8 1.6 Abbreviations …………………………………………………………………………………………………………………………………. 8 1.7 References/Related Documentation………………………………………………………………………………………………….. 8 1.8 Document History …………………………………………………………………………………………………………………………… 9 1.9 What is new……………………………………………………………………………………………………………………………………. 9 1.10 Documentation Feedback ………………………………………………………………………………………………………………… 9
2 Introduction ­ System Overview ……………………………………………………………………………………………………………… 10 2.1 Hardware Setup ……………………………………………………………………………………………………………………………. 10 2.2 Components of SME VoIP System ……………………………………………………………………………………………………. 11 2.2.1 RTX Base Stations………………………………………………………………………………………………………………………. 11 2.2.2 SME VoIP Administration Server/Software……………………………………………………………………………………. 11 2.2.3 RTX Wireless Handset ………………………………………………………………………………………………………………… 11 2.3 Wireless Bands ……………………………………………………………………………………………………………………………… 11 2.4 System Capacity (in Summary)………………………………………………………………………………………………………… 11 2.5 Advantages of SME VoIP System……………………………………………………………………………………………………… 12
3 Installation of Base Stations/Repeater ……………………………………………………………………………………………………… 13 3.1 Package ­ Contents/Damage Inspection…………………………………………………………………………………………… 13 3.2 RTX Base Station Mechanics …………………………………………………………………………………………………………… 14 3.3 RTX Base Unit ­ Reset feature…………………………………………………………………………………………………………. 15 3.4 Installing the Base Station………………………………………………………………………………………………………………. 15 3.4.1 Mounting the Base Stations/Repeaters:……………………………………………………………………………………….. 15 3.5 Find IP of Base Station……………………………………………………………………………………………………………………. 16 3.5.1 Using handset Find IP feature……………………………………………………………………………………………………… 16 3.5.2 Using browser IPDECT………………………………………………………………………………………………………………… 16 3.6 Login to Base SME Configuration Interface……………………………………………………………………………………….. 16
4 Making Handset Ready …………………………………………………………………………………………………………………………… 17 4.1 Package ­ Contents/Damage Inspection…………………………………………………………………………………………… 18 4.2 Before Using the Phone …………………………………………………………………………………………………………………. 18 4.3 Using the Handset …………………………………………………………………………………………………………………………. 20
5 SME VoIP Administration Interface ………………………………………………………………………………………………………….. 20 5.1 Web navigation …………………………………………………………………………………………………………………………….. 20 5.2 Home/Status ………………………………………………………………………………………………………………………………… 22 5.3 Extensions ……………………………………………………………………………………………………………………………………. 23
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5.3.1 Group call…………………………………………………………………………………………………………………………………. 23 5.3.2 Add extension …………………………………………………………………………………………………………………………… 24 5.3.3 Edit Extension …………………………………………………………………………………………………………………………… 28 5.3.4 Edit Handset……………………………………………………………………………………………………………………………… 29 5.4 Servers…………………………………………………………………………………………………………………………………………. 31 5.5 Network……………………………………………………………………………………………………………………………………….. 35 5.5.1 IP Settings ………………………………………………………………………………………………………………………………… 36 5.5.2 VLAN Settings……………………………………………………………………………………………………………………………. 37 5.5.3 DHCP Options……………………………………………………………………………………………………………………………. 37 5.5.4 Static IP settings ………………………………………………………………………………………………………………………… 37 5.5.5 NAT Settings……………………………………………………………………………………………………………………………… 38 5.5.6 SIP/RTP Settings ………………………………………………………………………………………………………………………… 39 5.5.7 TCP Options………………………………………………………………………………………………………………………………. 40 5.5.8 Discovery………………………………………………………………………………………………………………………………….. 40 5.6 Management Settings Definitions……………………………………………………………………………………………………. 41 5.6.1 Settings: …………………………………………………………………………………………………………………………………… 41 5.6.2 Configuration: …………………………………………………………………………………………………………………………… 42 5.6.3 Text messaging: ………………………………………………………………………………………………………………………… 43 5.6.4 Terminal: ………………………………………………………………………………………………………………………………….. 43 5.6.5 Syslog/SIP Log: ………………………………………………………………………………………………………………………….. 43 5.6.6 Location Gateway ……………………………………………………………………………………………………………………… 44 5.6.7 License: ……………………………………………………………………………………………………………………………………. 44 5.7 Firmware Update ………………………………………………………………………………………………………………………….. 44 5.7.1 Warning message when firmware upgrading ………………………………………………………………………………… 46 5.8 Location Gateways ………………………………………………………………………………………………………………………… 46 5.8.1 Register Location gateway ………………………………………………………………………………………………………….. 46 5.9 Country/Time Settings …………………………………………………………………………………………………………………… 47 5.10 Security………………………………………………………………………………………………………………………………………… 49 5.10.1 Certificates……………………………………………………………………………………………………………………………. 50 5.10.2 Certificates list ………………………………………………………………………………………………………………………. 51 5.10.3 SIP Client Certificates……………………………………………………………………………………………………………… 51 5.10.4 Device identity………………………………………………………………………………………………………………………. 52 5.10.5 Trusted Server Certificates ……………………………………………………………………………………………………… 53 5.10.6 Trusted Root Certificates………………………………………………………………………………………………………… 53 5.10.7 Password ……………………………………………………………………………………………………………………………… 53 5.10.8 Secure Web Server ………………………………………………………………………………………………………………… 54 5.11 Central Directory and LDAP…………………………………………………………………………………………………………….. 54 5.11.1 Local Central Directory …………………………………………………………………………………………………………… 54 5.11.2 LDAP ……………………………………………………………………………………………………………………………………. 55 SME VOIP SYSTEM GUIDE 4.6 Proprietary and Confidential

5.11.3 Characters supported …………………………………………………………………………………………………………….. 57 5.12 Dual-cell Parameter Definitions ………………………………………………………………………………………………………. 57
5.12.1 Settings for Base Unit …………………………………………………………………………………………………………….. 57 5.12.2 DECT System Settings …………………………………………………………………………………………………………….. 59 5.12.3 Base System Settings ……………………………………………………………………………………………………………… 60 5.12.4 Base Station Group ………………………………………………………………………………………………………………… 61 5.12.5 DECT Chain …………………………………………………………………………………………………………………………… 62 5.12.6 RTX8660 -RTX8663 Mixed mode ……………………………………………………………………………………………… 63 5.13 LAN SYNC……………………………………………………………………………………………………………………………………… 64 5.13.1 Settings for Base Unit …………………………………………………………………………………………………………….. 64 5.13.2 Base station group…………………………………………………………………………………………………………………. 65 5.13.3 This unit debug ……………………………………………………………………………………………………………………… 65 5.14 Repeaters …………………………………………………………………………………………………………………………………….. 66 5.14.1 Add repeater ………………………………………………………………………………………………………………………… 66 5.14.2 Register Repeater ………………………………………………………………………………………………………………….. 68 5.14.3 Repeaters list ………………………………………………………………………………………………………………………… 68 5.15 Alarm …………………………………………………………………………………………………………………………………………… 69 5.15.1 Use of Emergency Alarms……………………………………………………………………………………………………….. 72 5.16 Statistics ………………………………………………………………………………………………………………………………………. 73 5.16.1 System data ………………………………………………………………………………………………………………………….. 73 5.16.2 Free Running explained ………………………………………………………………………………………………………….. 73 5.16.3 Call data ……………………………………………………………………………………………………………………………….. 74 5.16.4 Repeater data ……………………………………………………………………………………………………………………….. 75 5.16.5 DECT data …………………………………………………………………………………………………………………………….. 76 5.16.6 Call quality ……………………………………………………………………………………………………………………………. 77 5.17 Generic Statistics…………………………………………………………………………………………………………………………… 78 5.17.1 DECT Synchronization Statistics ………………………………………………………………………………………………. 80 5.17.2 RTP Statistics…………………………………………………………………………………………………………………………. 81 5.17.3 IP ­ Stack statistics…………………………………………………………………………………………………………………. 82 5.17.4 System Statistics ……………………………………………………………………………………………………………………. 82 5.18 Diagnostics …………………………………………………………………………………………………………………………………… 83 5.18.1 Base Stations ………………………………………………………………………………………………………………………… 83 5.18.2 Extensions…………………………………………………………………………………………………………………………….. 83 5.18.3 Logging ………………………………………………………………………………………………………………………………… 84 5.19 Configuration………………………………………………………………………………………………………………………………… 86 5.20 Sys log………………………………………………………………………………………………………………………………………….. 87 5.21 SIP Logs………………………………………………………………………………………………………………………………………… 87 Appendix ­ How-To setup a Dual-Cell System…………………………………………………………………………………………………… 88 Adding Base stations………………………………………………………………………………………………………………………………….. 88 SME VOIP SYSTEM GUIDE 4.6 Proprietary and Confidential

Country and Time Server Setup ……………………………………………………………………………………………………………….. 89 SIP Server (or PBX Server) Setup………………………………………………………………………………………………………………. 90 Add an extension and handset ………………………………………………………………………………………………………………… 91 Appendix ­ Adding Extensions………………………………………………………………………………………………………………………… 94 Appendix ­ Firmware Upgrade Procedure ……………………………………………………………………………………………………….. 97 Network Dimensioning ………………………………………………………………………………………………………………………………. 97 TFTP Configuration ……………………………………………………………………………………………………………………………………. 98 Create Firmware Directories……………………………………………………………………………………………………………………….. 99 Base: ……………………………………………………………………………………………………………………………………………………. 99 Handsets/Repeaters: ……………………………………………………………………………………………………………………………. 100 Handset Firmware Update Settings……………………………………………………………………………………………………………. 100 Handset(s) and Repeater Firmware Upgrade ………………………………………………………………………………………………. 101 Monitor handset firmware upgrade ……………………………………………………………………………………………………….. 101 Monitor Repeater firmware upgrade ……………………………………………………………………………………………………… 102 Verification of Firmware Upgrade ………………………………………………………………………………………………………….. 102 Base Station(s) Firmware Upgrade …………………………………………………………………………………………………………….. 102 Base firmware confirmation ………………………………………………………………………………………………………………….. 103 Verification of Firmware Upgrade ………………………………………………………………………………………………………….. 103 Appendix ­ Multiline Feature ……………………………………………………………………………………………………………………….. 104 How to setup Multiline. ……………………………………………………………………………………………………………………………. 104 Appendix ­ Functionality Overview ……………………………………………………………………………………………………………….. 106 Gateway Interface …………………………………………………………………………………………………………………………………… 106 Detail Feature List ……………………………………………………………………………………………………………………………………. 107
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1 About This Document
This document describes the configuration, customization, management, operation, maintenance and troubleshooting of the SME VoIP System (RTX9431 base, RTX8630 handset, RTX8430 handset, RTX8830 ruggedized handset and RTX4024 Repeater) in RTX generic mode. For customer, specific modes refer to specific customer agreements, which describe the software operational deviations from this document.
1.1 Audience
Who should read this guide? First, this guide is intended for networking professionals responsible for designing and implementing RTX based enterprise networks. Second, network administrators and IT support personnel that need to install, configure, maintain, and monitor elements in a “live” SME VoIP network will find this document helpful. Furthermore, anyone who wishes to gain knowledge on fundamental features in the Beatus system can also benefit from this material.

1.2 When Should I Read This Guide
Read this guide before you install the core network devices of VoIP SME System and when you are ready to setup or configure SIP server, NAT aware router, advanced VLAN settings, base stations, and multi cell setup.
This manual will enable you to set up components in your network to communicate with each other and deploy a fully functionally VoIP SME System.
1.3 Important Assumptions
This document was written with the following assumptions in mind: 1) You understand network deployment in general. 2) You have working knowledge of basic TCP/IP/SIP protocols, Network Address Translation, etc… 3) A proper site survey has been performed, and the administrator have access to these plans.

1.4 What’s Inside This Guide
We summarize the contents of this document in the table below:

WHERE IS IT? CHAPTER 2
CHAPTER 3
CHAPTER 4
CHAPTER 5
APPENDIX ­ HOW-TO SETUP A DUAL-CELL SYSTEM APPENDIX ­ ADDING EXTENSIONS

CONTENT Introduction ­ System Overview
Installation of Base station/Repeater Making Handsets Ready
SME VoIP Administration Interface
Multi-Cell Setup & Management
Registration Management ­ Handsets

PURPOSE To gain knowledge about the different elements in a typical SME VoIP Network Considerations to remember before unwrapping and installing base units and repeaters To determine precautions to take in preparing handsets for use in the system To learn about the Configuration Interface and define full meaning of various parameters needed to be setup in the system. Learn how to add servers and setup multiple bases into a multi-cell network
Learn how to register handset and extensions to base stations

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APPENDIX ­ FIRMWARE UPGRADE APPENDIX ­ MULTILINE FEATURE APPENDIX ­ FUNCTIONALITY OVERVIEW

Firmware Upgrade/Downgrade Provides the procedure of how to upgrade firmware to base

Management

stations and/or handsets and/or repeaters

Multiline

Allows the same handset to have more then one number/line

System Functionality Overview To gain detail knowledge about the system features.

1.5 What’s Not in This guide
This guide provides overview material on network deployment, how-to procedures, and configuration examples that will enable you to begin configuring your VoIP SME System.

It is not intended as a comprehensive reference to all detail and specific steps on how to configure other vendor specific components/devices needed to make the SME VoIP System functional. For such a reference to vendor specific devices, please contact the respective vendor for documentation.

1.6 Abbreviations
For this document, the following abbreviations hold:

DHCP: DNS: DLC: HTTP(S): (T)FTP: IOS: PCMA: PCMU: PoE: RTP: RPORT: SIP: SME: VLAN: TOS: URL: UA:

Dynamic Host Configuration Protocol Domain Name Server Data Link Control Hyper Text Transfer Protocol (Secure) (Trivial) File Transfer Protocol Internetworking Operating System A-law Pulse Code Modulation mu-law Pulse Code Modulation Power over Ethernet Real-time Transport Protocol Response Port (Refer to RFC3581 for details) Session Initiation Protocol Small and Medium scale Enterprise Virtual Local Access Network Type of Service (policy-based routing) Uniform Resource Locator User Agent

1.7 References/Related Documentation
RTX8430 Handset_Manual_Operations_v4.6 RTX8630 Handset_Manual_Operations_v4.6 RTX8631_Handset_Manual_Operations_v4.6 RTX8632_Handset_Manual_Operations_v4.6 RTX8633_Handset_Manual_Operations_v4.6 RTX8830_Handset_Manual_Operations_v4.6 RTX8663 SME VoIP System Guide_SIP_V4.6 How to Deploy SME VOIP System v1.4 Provisioning of SME VoIP System (23)

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1.8 Document History

REVISION 1.0 1.1 1.2 1.3 1.4

AUTHOR DKO TWL TWL QCC QCC

ISSUE DATE 14-08-2019 7-Nov-2019 11-Dec-2019 16-Jun-2021 30-May-2023

COMMENTS
Add the FCC and ISEDC warning message Add Avaya model D200 in model variant. Add Mitel model RFP 14 Base Station NA Add Alcatel model 8328 SIP-DECT SINGLE BASE STATION

1.9 What is new
What new features have been added.

VERSION V420
V430 V440 V450
V460

FEATURE uaCSTA LDAP over SSL SME VoIP handset ­ login(for GDPR) TLS 1.2 Secure Syslog LLDP Support Firmware update warning New Generic statistics 8660 ­ 8663 Mixed mode Diagnostics Logging RTX BTLE Beacon support

1.10 Documentation Feedback
We always strive to produce the best and we also value your comments and suggestions about our documentation. If you have any comments about this guide, please enter them through the Feedback link on the RTX website. We will use your feedback to improve the documentation.

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2 Introduction ­ System Overview
In a typical telephony system, the network setup is the interconnection between Base-stations, “fat” routers, repeaters, portable parts, etc. The backbone of the network depends on the deployment scenario, but a ring or hub topology is used. The network has centralized monitoring, and maintenance system.
The model variant is included RTX9431 D200 (Avaya model), RFP 14 Base Station NA (Mitel model) and 8328 SIP-DECT SINGLE BASE STATION (Alcatel model).
The system is easy to scale up and supports from 1 to 249 bases in the same network. Further it can support up to 20 registered handsets (RTX8630, RTX8830 and RTX8430). The Small and Medium Scale Enterprise (SME) VoIP system setup is illustrated below. Based on PoE interface each base station is easy to install without additional wires other than the LAN cable. The system supports the IP DECT CAT-IQ repeater RTX4024 with support up to 5 channels simultaneous call sessions.
The following figure gives a graphical overview of the architecture of the SME VoIP System:

2.1 Hardware Setup
SME network hardware setup can be deployed as follows: Base-station(s) are connected via Layer 3 and/or VLAN Aware Router depending on the deployment requirements. The Layer 3 router implements the switching function. The base- stations are mounted on walls or lamp poles so that each base-station is separated from each other by up to 50m indoor1 (300m outdoor). Radio coverage can be extended using repeaters that are installed with same distance to basestation(s). Repeaters are range extenders and cannot be used to solve local call capacity issues. In this case additional bases must be used. The base-station antenna mechanism is based on space diversity feature which improves coverage. The base-stations uses complete DECT MAC protocol layer and IP media stream audio encoding feature to provide up to 10 simultaneous calls.

1 Measured with European DECT radio and depends on local building layout and material. SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential

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2.2 Components of SME VoIP System
RTX SME VoIP system is made up of (but not limited to) the following components: · At least one RTX Base Station is connected over an IP network and using DECT as air-core interface. · RTX IP DECT wireless Handset. · RTX SME VoIP Configuration Interface; is a management interface for SME VoIP Wireless Solution. It runs on all IP DECT Base
stations. Each Base station has its own unique settings.
2.2.1 RTX Base Stations
The Base Station converts IP protocol to DECT protocol and transmits the traffic to and from the end-nodes (i.e. wireless handsets) over a channel. It has 12 available channels. In a dual-cell setup, each base station has: · 8 channels that have associated DSP resources for media streams. · The remaining 4 channels are reserved for control signaling between IP Base Stations and the SIP/DECT end nodes (or
phones). If two Base Stations are used, they are grouped into a cluster. Within the Cluster, Base Stations are synchronized to enable a seamless handover when a user moves from one base station coverage to the other. It is necessary for Base Stations to communicate directly with each other in the system in order to guarantee synchronization in the situation that one of them fails.
The 4 control signaling channels are used to carry bearer signals that enable a handset to initiate a handover process.
2.2.2 SME VoIP Administration Server/Software
This server is referred to as SME VoIP Configuration Interface. The SME VoIP Configuration Interface is a web-based administration page used for configuration and programming of the base station and relevant network end- nodes. E.g. handsets can be registered or de-registered from the system using this interface. The configuration interface can be used as a setup tool for software or firmware download to base stations, repeaters and handsets. Further, it is used to check relevant system logs that can be useful to administrator. These logs can be used to troubleshoot the system when the system faces unforeseen operational issues.
2.2.3 RTX Wireless Handset
The handset is a lightweight, ergonomically, and portable unit compatible with Wideband Audio (G.722), DECT, GAP standard, CAT-iq audio compliant. The handset includes 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. Refer to the relevant handset manuals for full details handset features.
2.3 Wireless Bands
The bands supported in the SME 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
Transmit Power: 23.7 dBm in Europe mode.

2.4 System Capacity (in Summary)
SME network capacity of relevant components can be summarized as follows: SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential

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DESCRIPTION Min ## of Bases Single Cell Setup Max ## of Bases in Dual-cell Setup (configurable) Single/Dual-Cell Setup: Max ## of Repeaters Dual-Cell Setup: Total Max ## of Repeaters Max ## of Users (SIP registrations) per Base Max ## of Users per SME VoIP System Dual-cell Setup: Max ## of Synchronization levels Single Cell Setup: Max ## Simultaneous Calls Dual-Cell Setup: Max ## of Calls Total Max ## Simultaneous Calls (Dual-cell Setup) Repeater: Max ## of Calls (Narrow band) Repeater: Max ## of Calls (G722)

CAPACITY 1 2 1 base and 6 repeaters per base 12 30 limited to 1000 24 10 per Base station 20 per system Limited to 1000 10 4

Quick Definitions Single Cell Setup: Dual-cell Setup: Synchronization Level:

SME telephony network composed of one base station Telephony network that consists of two base stations Is the air core interface between two base stations.

2.5 Advantages of SME VoIP System
They include (but not limited to):
1. Simplicity. Integrating functionalities leads to reduced maintenance and troubleshooting, and significant cost reductions.
2. Flexibility. Single network architecture can be employed and managed. Furthermore, the architecture is amenable to different deployment scenarios, including Isolated buildings for in-building coverage, location with co- located partners, and large to medium scale enterprises deployment for wide coverage.
3. Scalability. SME network architecture can easily be scaled to the required size depending on customer requirement.
4. Performance. The integration of different network functionalities leads to the collapse of the protocol stack in a single network element and thereby eliminates transmission delays between network elements and reduces the call setup time and packet fragmentation and aggregation delays.

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3 Installation of Base Stations/Repeater
After planning the network, next is to determine the proper places or location the relevant base stations will be installed. Therefore, we briefly describe the how to install the base station in this chapter.

3.1 Package ­ Contents/Damage Inspection
Before Package Is Opened: Examine the shipping package for evidence of physical damage or mishandling prior to opening. If there is a proof of mishandling prior to opening, you must report it to the relevant support center of the regional representative or operator.

Contents of Package: Make sure all relevant components are available in the package before proceeding to the next step. Every shipped base unit package/box contains the following items:

x Box for Base station (DC+PoE) unit + PSU o 1 x Base Station unit o 1 x Ethernet cable 1m o 1 x Power supply single plug o 1 x Quick guide o 1 x Safety sheet

Depending on the manufacturer P/N, the DC adaptor type may vary as listed below:

Manufacturer P/N
S008ACM0500200 S010WB0500200 S010WV0500200 S010WU0500200 S010WS0500200

DC adaptor plug type by countries
Multi-plug UK EU US AU

x Box for PoE only Base station unit o 1 x Base Station unit o 1 x Ethernet cable 1m o 1 x Quick guide o 1 x Safety sheet
x Spare accessories o PSU single plug o PSU multi plug

Please note that mounting screws and anchors are not added in the packaging.

Damage Inspection: The following are the recommended procedure for you to use for inspection:
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1. Examine all relevant components for damage. 2. Make a “defective on arrival ­ DOA” report or RMA to the operator. Do not move the shipping carton until the operator
has examined it. If possible, send pictures of the damage. The operator/regional representative will initiate the necessary procedure to process this RMA. They will guide the network administrator on how to return the damaged package if necessary. 3. If no damage is found, then unwrap all the components and dispose of empty package/carton(s) in accordance with country specific environmental regulations.
3.2 RTX Base Station Mechanics
RTX9431 can operate on a maximum temperature of 50. With such small dimensions as 109mm (height) and 93mm (width), it allows the user to mount the device on the wall or easily leave it standing on any furniture. (please see image below for more details).
Alternative mechanics casing.

The base station front end shows an LED indicator that signals different functional states of the base unit and occasionally of the overall network. The indicator is off when the base unit is not powered. The table below summarizes the various LED states:

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LED STATE UNLIT UNLIT/SOLID RED BLINKING GREEN SOLID RED BLINKING RED SOLID GREEN BLINKING RED SOLID RED
ORANGE BLINKING ORANGE

STATE No power in unit Error condition Initialization Factory reset warning or long press in BS reset button Factory setting in progress Ethernet connection available (Normal operation) Ethernet connect not available OR handset de/registration failed Critical error (can only be identified by RTX Engineers). Symptoms include no system/SIP debug logs are logged, etc. Press reset button of base station. No IP address received

3.3 RTX Base Unit ­ Reset feature
It is possible to restart or reset the base station unit by pressing a knob at the bottom side of the unit (see image below). Alternatively, it can be reset from the SME Configuration Interface. We do not recommend this; but unplugging and plugging the Ethernet cable back to the PoE port of the base station also resets the base unit.

3.4 Installing the Base Station
First determine the best location that will provide an optimal coverage taking account the construction of the building, architecture, and choice of building materials. Next, mount the Base Station on a wall to cover range between 50 ­ 300 meters (i.e. 164 to 984 feet), depending whether it’s an indoor or outdoor installation.
3.4.1 Mounting the Base Stations/Repeaters:
We recommend the base station to be mounted an angle other than vertical on both concrete/wood/plaster pillars and walls for optimal radio coverage. Avoid mounting the base unit’s upside down as it significantly reduces radio coverage.
As mentioned before, the screws and anchors are not included in the packaging. Therefore, you will have to provide your own two pieces of screws M3.5 x 31mm. The distance between them is 70mm (please see the images below). The height of wall mount is suggested to be less than or equal to 2 meters.

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Mount the base unit as high as possible (not more than 2m) to clear all nearby objects (e.g. office cubicles and cabinets, etc.). Occasionally extend coverage to remote offices/halls with lower telephony users by installing Repeaters. Make sure that when you fix the base stations with screws, the screws do not touch the PCB on the unit. Secondly, avoid all contacts with any high voltage lines.

3.5 Find IP of Base Station
To find IP of the installed base station two methods can be used; Using handset Find IP feature or browser IPDECT feature.

3.5.1 Using handset Find IP feature
On the handset press “Menu” key followed by the keys: 47 to get the handset into find bases menu. The handset will now scan for 8660 / 9431 bases. Depending on the amount of powered on bases with active radios and the distance to the base it can take up to minutes to find a base.

– Use the cursor down/up to select the base MAC address for the base that you want to connect to – The base IP address will be shown in the display below the MAC address of the device

The feature is also used for deployment.

3.5.2 Using browser IPDECT
Open any standard browser and enter the address: http://ipdect<MAC-Address- Base-Station> for e.g. http://ipdect00087B00AA10. This will retrieve the HTTP Web Server page from the base station with hardware address 00087B00AA10. This feature requires an available DNS server.

3.6 Login to Base SME Configuration Interface
1. Connect the Base station to a private network via standard Ethernet cable (CAT-5).

2. Use the IP find menu in the handset (Menu 4 7 ) to determine the IP- address of the base station by matching the MAC address on the back of the base station with the MAC address list in the handset

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3. On the Login page, enter your authenticating credentials (i.e. username and password). By default, the username and password are admin. Click OK button.
4. Once you have authenticated, the browser will display front end of the SME Configuration Interface. The front end will show relevant information of the base station. Screenshot:

4 Making Handset Ready
In this chapter, we briefly describe how to prepare the handset for use, install, insert and charge new batteries. Please refer to an accompanying Handset User Guide for more information of the features available in the Handset.

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4.1 Package ­ Contents/Damage Inspection
Before Package Is Opened: Examine the shipping package for evidence of physical damage or mishandling prior to opening. If there is a proof of mishandling prior to opening, you must report it to the relevant support center of the regional representative or operator.
Contents of Package: Make sure all relevant components are available in the package before proceeding to the next step. Every shipped base unit package/box contains the following items:
x 2 x mounting screws and 2 x Anchors x 1 x Handset hook x 1 x A/C Adaptor x 1 x Battery x 1 x charger x 1 x Handset Unit, 1 x Battery cover

Damage Inspection: The following are the recommended procedure for you to use for inspection:
1. Examine all relevant components for damage. 2. Make a “defective on arrival ­ DOA” report or RMA to the operator. Do not move the shipping carton until the operator
has examined it. The operator/regional representative will initiate the necessary procedure to process this RMA. They will guide the network administrator on how to return the damaged package if necessary. 3. If no damage is found, then unwrap all the components and dispose of empty package/carton(s) in accordance with country specific environmental regulations.
4.2 Before Using the Phone
Here are the pre-cautions users should read before using the Handset:
Installing the Battery 1. Never dispose battery in fires, otherwise it will explode. 2. Never replace the batteries in potentially explosive environments, e.g. close to inflammable liquids/ gases. 3. ONLY use approved batteries and chargers from the vendor or operator. 4. Do not disassemble, customize, or short circuit the battery
Using the Charger Each handset is charged using a handset charger. The charger is a compact desktop unit designed to charge and automatically maintain the correct battery charge levels and voltage. The charger Handset is powered by AC supply from 110-240VAC that supplies 5.5VDC at 600mA. When charging the battery for the first time, it is necessary to leave the handset in the charger for at least 10 hours before the battery is fully charged and the handset ready for use.

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Handset in the Charger For correct charging, ensure that the room temperature is between 5 and 25/41°F and 77°F. Do not place the handset in direct sunlight. The battery has a built-in heat sensor which will stop charging if the battery temperature is too high. If the handset is turned off when placed in charger, only the LED indicates the charging. When handset is turned off, the LED flashes at a low frequency while charging and lights constantly when the charging is finished. There will be response for incoming calls. If the handset is turned on when charging, the display shows the charging status.
Open Back Cover 1. Press down the back cover and slide it towards the bottom of the handset. 2. Remove Back Cover from Handset

– Handset Serial Number The serial number (IPEI/IPUI number) of each handset is found either on a label, which is placed behind the battery, or on the packaging label. First, lift off handset back cover and lift the battery and read the serial number. The serial number is needed to enable service to the handset. It must be programmed into the system database via the SME VoIP Configuration interface.

– Replace Battery Remove Back Cover from Handset. Remove the old battery and replace with a new one.

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4.3 Using the Handset
Please refer to the handset manual for detailed description of how to use the handset feature.
5 SME VoIP Administration Interface
The SME VoIP Administration Interface is also known as SME VoIP Configuration. It is the main interface through which the system is managed and debugged. The SME VoIP Configuration Interface is an in-built HTTP Web Server service residing in each base station. This interface is a user-friendly interface and easy to handle even to a first-time user. NOTE: Enabling secure web is not possible. For secure configuration use, secure provisioning. This chapter seeks to define various variables/parameters available for configuration in the network.
5.1 Web navigation
We describe the left menu in the front end of the SME VoIP Administration Interface. For detailed overview of each parameter from the menu bar, please see the next chapters. Screenshot

FEATURE HOME/STATUS
EXTENSIONS SERVERS

DESCRIPTION This is the front end of the Base station’s HTTP web interface. This page shows the summary of current operating condition and settings of the Base station and Handset(s). Administration of extensions and handsets in the system On this page, the user can define which SIP/NAT server the network should connect to.

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NETWORK MANAGEMENT FIRMWARE UPDATE LOCATION GATEWAY COUNTRY
SECURITY CENTRAL DIRECTORY DUAL CELL LAN SYNC REPEATERS ALARM
STATISTICS GENERAL STATISTICS DIAGNOSTICS CONFIGURATION SYSLOG SIP LOG LOGOUT

Network settings can be configured in this menu such as IP settings, NAT, SIP, VLAN,etc. Defines the Configuration server address, Management transfer protocol, sizes of logs/traces that should be catalogued in the system. Remote firmware updates (HTTP(s)/TFTP) settings of Base stations and handsets.
If Location Gateway is connected, this parameter will be added to the menu bar, serving for administration of Location Gateways Specifying the country/territory where the SME network is located ensures that your phone connection functions properly. Note: The base language and country setting are independent of each other. Time settings: Here the user can configure the Time server. It should be used as time server in relevant country for exact time. The time servers must deliver the time to conform to the Network Time Protocol (NTP). Handsets are synchronised to this time. Base units synchronise to the master using the Time server. The users can administrate certificates and create account credentials with which they can log in or log out of the embedded HTTP web server. Interface to common directory load of up to 3000 entries using *csv format or configuration of LDAP directory. Note: LDAP and central directory cannot operate at the same time. Specify to connect up to two base stations to the network. Make sure the system ID for the relevant base stations are the same otherwise the dual-cell feature will not work. Allows base stations to connect over LAN PTP Sync, this makes it possible to have greater distance between the base stations, compared to Air Sync. Administration and configuration of repeaters of the system Administration and configuration of the alarm settings on the system. This controls the settings for alarms that can be sent to the handsets. This feature is only available on certain types of handsets. Overview of system and call statistics for a system. Overview of general parameter statistics of the system
Overview of Base stations and Extensions diagnostics This shows detail and complete SME network settings for base station(s), HTTP/DNS/DHCP/TFTP server, SIP server, etc. Overall network related events or logs are displayed here (only live feed is shown). SIP related logs can be retrieved from URL link. It is also possible to clear logs from this feature. Logout of the web interface.

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5.2 Home/Status
We describe the parameters found in the Welcome front-end home/status of the SME VoIP Administration Interface.
Screenshot:

PARAMETER SYSTEM INFORMATION PHONE TYPE SYSTEM TYPE RF BAND
CURRENT LOCAL TIME OPERATION TIME RFPI-ADDRESS MAC-ADDRESS IP-ADDRESS FIRMWARE VERSION FIRMWARE URL REBOOT BASE STATION STATUS
SIP IDENTITY STATUS SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential

DESCRIPTION Status of the base (Single cell as the Dual cell is not activated) Always IPDECT Customer configuration of the base RF band setting of the base The parameter is defined in production and relates to the radio approvals shown on the label of the base. Local Time of the base Operation is operation time for the base since last reboot RFPI address of the base MAC address of the base IP address of the base Firmware version of the base Firmware update server address and firmware path on server Shows the last reboots of the base station and the reason for reboot “Idle”: When no calls on base “In use”: When active calls on base Shows list of extensions present at this base station.
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REBOOT FORCED REBOOT

Format: “extension”@”this base IP address”(“server name”) followed by status to the right. Below is listed possible status: OK: Handset is ok Error: SIP registration error Reboot after all connections are stopped on base. Connections are active calls, directory access, firmware update active Reboot immediately.

5.3 Extensions
In this section, we describe the different parameters available whenever the administrator is creating extensions for handsets. Note, it is not possible to add extensions if no servers are defined. As well the section describes the administration of extensions and handsets using the extension list and the extension list menu.

The system can handle maximum 1000 extensions matching 1000 handsets which can be divided between servers. When 1000 handsets are registered it is not possible to add more extensions. With active multiline feature, the system can handle maximum 1000 extensions. With 4 active lines in multiline maximum 200 handsets can be active in the system.

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.

5.3.1 Group call
Call Group is a SIP extension where multiple handsets are associated. All handsets that subscribe to a given extension (and hence Call Group) can receive incoming calls and initiate outgoing calls on the given extension. It is possible for any handset to perform any call action which is possible without the Call Group feature. That is, call actions as Hold, transfer etc. are possible if the PBX supports them.
When an incoming call arrives to a given Call Group, all Call Group subscribed handsets will alert. Thus, if a Call Group contains 20 handsets, all 20 handsets 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.
NOTE: Group call, does not work with paired headset.

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5.3.2 Add extension
1. Click add extension Screenshot:
2. Fill in the required information Screenshot:

PARAMETER EXTENSION

DEFAULT VALUE(S) Empty

AUTHENTICATION USER NAME AUTHENTICATION PASSWORD DISPLAY NAME

Empty Empty Empty

XSI USERNAME Empty

XSI PASSWORD Empty

DESCRIPTION
Handset phone number depending on the setup. Possible value(s): 8-bit string length Example: 1024, etc. Note: The Extension must also be configured in SIP server in order for this feature to function. Username: SIP authentication username Permitted value(s): 8-bit string length Password: SIP authentication password. Permitted value(s): 8-bit string length Human readable name used for the given extension Permitted value(s): 8-bit string length Username: SIP authentication username Permitted value(s): 8-bit string length Password: SIP authentication password. Permitted value(s): 8-bit string length

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MAILBOX NAME Empty

MAILBOX NUMBER

Empty

SERVER

Server 1 IP

CALL WAITING FEATURE

Enabled

BROADWORKS FEATURE EVENT PACKAGE UACSTA FORWARDING UNCONDITIONAL NUMBER

Disable
Disabled Empty

FORWARDING NO Empty ANSWER NUMBER

FORWARDING ON Empty BUSY NUMBER

REJECT ANONYMOUS CALLS

Disabled

Name of centralized system used to store phone voice messages that can be retrieved by recipient later. Valid Input(s): 8-bit string Latin characters for the Name Dialed mail box number by long key press on key 1. Valid Input(s): 0 ­ 9, *, # Note: Mailbox Number parameter is available only when it’s enabled from SIP server. FQDN or IP address of SIP server. Drop down menu to select between the defined Servers of Service provider. Used to enable/disable Call Waiting feature. When disabled a second incoming call will be rejected. If enabled a second call will be presented as call waiting. Enable/Disable Broadworks features
Enable/Disable uaCSTA support Number to which incoming calls must be re-routed to irrespective of the current state of the handset. Forwarding Unconditional must be enabled to function. Note: Feature must be enabled in the SIP server before it can function in the network Note: Feature will be automatically disabled in case the handset or extension is part of a group Number to which incoming calls must be re-routed to when there is no response from the SIP end node. Forwarding No Answer Number must be enabled to function. Note: Feature must be enabled in the SIP server before it can function in the network Specify delay from call to forward in seconds. Note: Feature will be automatically disabled in case the handset or extension is part of a group Number to which incoming calls must be re-routed to when SIP node is busy. Forwarding On Busy Number must be enabled to function. Note: Feature must be enabled in the SIP server before it can function in the network Note: Feature will be automatically disabled in case the handset or extension is part of a group Calls from anonymous numbers will automatically be rejected. Enable to rejects anonymous calls

NOTE: Call forwarding can as well be configured from the handset by the user (for operation refer to the handset guide).

When an extension is added (or edited) it can be selected (right side check box) which handsets shall subscribe to the given extension, and hence be a part of this call group, see above figure. It is also possible to choose to add a new handset entry at this point, and if this is done, DECT registration for the new entry can be enabled afterwards on the handsets subpage.

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5.3.2.1 Extension list The added extensions will be shown in the extension lists. The list can be sorted by any of the top headlines (Extensions / Handset), by mouse click on the headline link.
Screenshot

PARAMETER IDX EXTENSION DISPLAY NAME SERVER SERVER ALIAS STATE IPEI

DESCRIPTION
Index of handsets ; Select / deselect for delete, register and deregister handsets Given extension is displayed. Given display name is displayed. If no name given this field will be empty Server IP or URL Given server alias is displayed. If no alias given this field will be empty. SIP registration state ­ if empty the handset is not SIP registered. Handset IPEI. IPEI is a unique DECT identification number. Group call: One extension can be associated to up to 20 IPEI’s. The IPEI’s will be listed in this cell.

5.3.2.2 Handset list The added handsets will be shown in the handset lists. The list can be sorted by any of the top headlines (Extensions / Handset), by mouse click on the headline link.
Screenshot

PARAMETER IDX

DESCRIPTION Index of handsets ; Select / deselect for delete, register and deregister handsets

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IPEI HANDSET STATE
HANDSET TYPE FW INFO FWU PROGRESS
EXTENSION

Handset IPEI. IPEI is unique DECT identification number. 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 and firmware version of handset
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)): All FWU is complete and is now waiting for handset restart. Complete HS: FWU complete Error: Not able to fwu e.g. file not found, file not valid etc Given extension is displayed. Group call: The cell will show all the extensions associated with this handset and IPEI.

5.3.2.3 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/deregistrations) 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.
Screenshot

In the below table, each command is described.

ACTIONS ADD EXTENSION / ADD HANDSET STOP REGISTRATION
DELETE HANDSET(S) REGISTER HANDSET(S)
DEREGISTER HANDSET(S) START SIP REGISTRATION(S) DELETE SIP EXTENSION(S)

DESCRIPTION Access to the “Add extension” or “Add Handset” sub menu
Manually stop DECT registration mode of the system. This prevents any handset from registering to the system Deregister selected handset(s), but do not delete the extension(s). Enable registration mode for the system making it possible to register at a specific extension (selected by checkbox) Deregister the selected handset(s) and delete the extension(s). Manually start SIP registration for selected handset(s).
Deregister the selected handset(s) and delete the extension(s).

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NOTE: By powering off the handset, the handset will SIP deregister from the PBX.
5.3.3 Edit Extension
To edit an extension simply click the extension number that you want to edit. Screenshot:
Editing the extension will open the same configuration possibilities as “Add extension”. Refer to the previous chapter (5.3.2) for more details.

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5.3.4 Edit Handset
Use the mouse to click the handset IPEI link to open the handset editor window.
Screenshot

PARAMETER IPEI
AC

DEFAULT VALUE(S) Handset IPEI
Handset AC code

ALARM LINE

No Alarm Line

Selected

ALARM NUMBER Empty

RECEIVE MODE Disabled

DESCRIPTION Shows the handset IPEI. For an already registered handset changing the IPEI will deregister the handset at next handset location update. Shows the handset AC code. AC code is used at handset registration. Changing the AC code for an already registered handset will have no effect. The line of multiline to be used for alarm call feature
Number to be dialed in case of handset alarm key is pressed (Long keypress > 3 seconds on navigation center key) NOTE: This feature is only shown if Handset has BTLE. (RTX8630 and RTX8430 is not supported)

TRANSMIT INTERVAL

Disabled

Enter Proximity: Leave Proximity: Enter or Leave Proximity: NOTE: This feature is only shown if Handset has BTLE. (RTX8630 and RTX8430 is not supported)

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ALARM PROFILES IMPORT LOCAL PHONEBOOK EXPORT LOCAL PHONEBOOK

Not configured

Short: Step1: Step2: Step3: Step4: Step5: Long: Check the wanted alarm profiles for the particular handset. Import phonebook from csv file to this specific extension
Exports this extensions phonebook as csv file NB: Home is not exported as this is considered private data.

5.3.4.1 Import local phonebook The import local phonebook feature is using a browse file approach. After file selection press the load button to load the file. The system supports only the original *.csv format. Please note that some excel csv formats are not the original csv format.
Screenshot

NOTE: The local phonebook can have 100 entries for RTX863x and RTX8830 and 50 entries for RTX8430. 5.3.4.2 Export local phonebook The Export local phonebook feature makes it possible to retrieve all contracts from a specific phone to a .CSV file. Screenshot
Press the export button and save the .CSV file on you PC or Server.

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5.4 Servers
In this section, we describe the different parameters available in the Servers configurations menu. Maximum 10 servers can be configured.
Screenshot

PARAMETER SERVER ALIAS NAT ADAPTION

DEFAULT VALUE Empty Disabled

REGISTRAR Empty
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DESCRIPTION Parameter for server alias To ensure all SIP messages go directly to the NAT gateway in the SIP aware router. If the system receives a SIP response to a REGISTER request with a “Via” header that includes the “received” parameter (ex: “Via: SIP/2.0/UDP 10.1.1.1:4540;received=68.44.20.1”), the base will adapt its contact information to the IP address from the “received” parameter. Thus, the base will issue another REGISTER request with the updated contact information. If NAT Adaption is disabled, the “received” parameter is ignored. SIP Server proxy DNS or IP address
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OUTBOUND PROXY

Empty

CONFERENCE Empty SERVER

CALL LOG SERVER

Empty

MUSIC ON HOLD SERVER REREGISTRATION TIME

Empty 600

SIP SESSION TIMERS:

Disabled

SESSION TIMER VALUES (S): SIP TRANSPORT SIGNAL TCP SOURCE PORT

1800 UDP Disabled

USE ONE

Disabled

TCP/TLS

CONNECTION

PER SIP

EXTENSION:

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Permitted value(s): AAA.BBB.CCC.DDD: or : Note: Specifying the Port Number is optional. This is a Session Border Controller DNS or IP address (OR SIP server outbound proxy address) Set the Outbound proxy to the address and port of private NAT gateway so that SIP messages sent via the NAT gateway. Permitted value(s): AAA.BBB.CCC.DDD or or <URL

: Examples: “192.168.0.1”, “192.168.0.1:5062”, “nat.company.com” and “sip:nat@company.com:5065”. If empty call is made via Registrar. 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 on 8630 is used. Broadsoft call log feature. Set the IP address of the XSI call log server. In case an IP is specified pressing handset will use the call log server. If the field is empty, the local call log is used Add the address of a server for ensuring music is on when call is on hold The “expires” value in SIP REGISTER requests. This value indicates how long the current SIP registration is valid, and hence is specifies the maximum time between SIP registrations for the given SIP account. Permitted value(s): A value below 60 sec is not recommended, Maximum value 65636 RFC 4028. A “keep- alive” mechanism for calls. The session timer value specifies the maximum time between “keepalive” or more correctly session refresh signals. If no session refresh is received when the timer expires the call will be terminated. Default value is 1800 s according to the RFC. Min: 90 s. Max: 65636. If disabled session timers will not be used. Default value is 1800s according to the RFC. If disabled session timers will not be used. Permitted value(s): Minimum value 90, Maximum 65636 Select UDP, TCP, TLS
When SIP Transport is set to TCP or TLS, a TCP (or TLS) connection will be established for each SIP extension. The source port of the connection will be chosen by the TCP stack, and hence the local SIP port parameter, specified within the SIP/RTP Settings (see 5.5.6) will not be used. The “Signal TCP Source Port” parameter specifies if the used source port shall be signaled explicitly in the SIP messages. When using TCP or TLS as SIP transport, choose if a TCL/TLS connection shall be established for each SIP extension or if the base station shall establish one connection which all SIP extensions use. Please note that if TLS is used and SIP server
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RTP FROM OWN BASE STATION:

Disabled

KEEP ALIVE

Enabled

SHOW EXTENSION ON HANDSET IDLE SCREEN HOLD BEHAVIOUR

Enabled RFC 3264

LOCAL RING BACK TONE REMOTE RING TONE CONTROL ATTENDED TRANSFER BEHAVIOUR

Enabled Enabled
Hold 2nd Call

DIRECT CALL PICKUP DIRECT CALL PICKUP CODE GROUP CALL PICKUP GROUP CALL PICKUP CODE USE OWN CODEC PRIORITY

Disabled Empty Disabled Empty Disabled

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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). If disabled RTP stream will be send from the base, where the handset is located. By enable the RTP stream will always be send from the base, where the SIP registration is made. This setting is typically enabled for operation with Cisco. This directive defines the window period (30 sec.) to keep opening the port of relevant NAT-aware router(s), etc. If enabled extension will be shown on handset idle screen.
Specify the hold behavior by handset hold feature. RFC 3264: Hold is signaled 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 sent only, recvonly or inactive dependent of the context RFC 2543: The “old” way of signaling HOLD. The connection information part of the SDP is set to 0.0.0.0, and the direction attribute is sent only, recvonly or inactive dependent of the context In case the server doesn’t play local ring back tone the handset will do it. Sometimes call distinguished ringing. It enables the server to control what ring tone that is used on the handsets. When we have two calls, and one call is on hold, it is possible to perform attended transfer. When the transfer soft key is pressed in this situation, we have traditionally also put the active call on hold before the SIP REFER request is sent. However, we have experienced that some PBX’s do not expect that the 2nd call is put on hold, and therefore attended transfer fails on these PBX’s. The “Attended Transfer Behavior” feature defines whether the 2nd call shall be put on hold before the REFER is sent. If “Hold 2nd Call” is selected, the 2nd call will be held before REFER is sent. If “Do Not Hold 2nd Call” is selected, the 2nd call will not be held before the REFER is sent This is Part of BroadWorks SCA feature. Enabled a direct call pickup code is sent to the Handsets Code used to direct call pick up
Enable for a call group pickup
Code used to pick up a group call
Default disabled. By enablbling the system codec, priority during incoming call is used instead of the calling party priority.
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DTMF SIGNALLING

RFC 2833

DTMF PAYLOAD TYPE REMOTE CALLER ID SOURCE PRIORITY CODEC PRIORITY

101
FROM
G.711U G.711A G.726

G729 Annex B

USE PTIME RTP PACKET SIZE

Enabled 20ms

RTCP SEND SDP CAPABILITIES IN OFFER (RFC 5939) SECURE RTP

Enabled Disabled
Disabled

SECURE RTP AUTH

Disabled

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E.g. If base has G722 as top codec and the calling party has Alaw on top and G722 further down the list, the G722 will be chosen as codec for the call. Conversion of decimal digits (and *’ and#’) into sounds that share similar characteristics with voice to easily traverse networks designed for voice SIP INFO: Carries application level data along SIP signaling 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. This feature enables the user to specify a value for the DTMF payload type / telephone event (RFC2833).
SIP information field used for Caller ID source: PAI – FROM FROM ALERT_INFO – PAI – FROM Defines the codec priority that base stations use for audio compression and transmission. Possible Option(s): G.711U,G.711A, G.726, G.729, G.722. Note: Modifications of the codec list must be followed by a “reset codes” and “Reboot chain” on the multipage 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. To use G.729, add on DSP module must be installed in all base stations. Contact your local dealer for price information. Enable/Disable Annex B of codec G729 Note: Both parts have to support it in order to avoid noise and any other kind of voice interruption Use the RTP Packet size, chosen in the below setting. The packet size offered as preferred RTP packet size by 8630 when RTP packet size negotiation. Selections available: 20ms, 40ms, 60ms, 80ms Enable/Disable RTCP Enable to support RFC 5939
With enable RTP will be encrypted (AES-128) using the key negotiated via the SDP protocol at call setup. With enable secure RTP is using authentication of the RTP packages. Note: with enabled SRTP authentication maximum 4 concurrent calls are possible per base in a single or multicell system.
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SRTP CRYPTO AES_CM_128_HMAX_SHA1_32

SUITES

AES_CM_128_HMAX_SHA1_80

Field list of supported SRTP Crypto Suites. The device is born with two suites.

Note: Within servers or even with multi servers, extensions must always be unique. This means same extension number on server 1 cannot be re-used on server 2.

5.5 Network
In this section, we describe the different parameters available in the network configurations menu.

Screenshot

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5.5.1 IP Settings
Screenshot

PARAMETER DHCP/STATIC IP

DEFAULT VALUES DHCP

IP ADDRESS

N/A

SUBNET MASK

N/A

DEFAULT

N/A

GATEWAY

DNS (PRIMARY)

N/A

DNS (SECONDARY) N/A

MDNS

Disabled

DESCRIPTION
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: the network administrator manually assigns IP addresses. If the user chooses DHCP option, the other IP settings or options are not available. 32-bit IP address of device (e.g. base station). 64-bit IP address will be supported in the future. Permitted value(s): AAA.BBB.CCC.DDD Is device subnet mask. Permitted value(s): AAA.BBB.CCC.DDD This is a 32-bit combination used to describe which portion an IP address refers to the subnet and which part refers to the host. A network mask helps users know which portion of the address identifies the network and which portion of the address identifies the node. Device’s default network router/gateway (32-bit). Permitted value(s): AAA.BBB.CCC.DDD e.g. 192.168.50.0 IP address of network router that acts as entrance to another network. This device provides a default route for TCP/IP hosts to use when communicating with other hosts on hosts networks. Main server to which a device directs Domain Name System (DNS) queries. Permitted value(s): AAA.BBB.CCC.DDD or 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. This is an alternate DNS server. Enable to allow Multicast Domain Name system (MDNS)

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5.5.2 VLAN Settings
Enable users to define devices (e.g. Base station, etc.) with different physical connection to communicate as if they are connected on a single network segment.
The VLAN settings can be used on a managed network with separate Virtual LANs (VLANs) for sending voice and data traffic. To work on these networks, the base stations can tag voice traffic it generates on a specific “voice VLAN” using the IEEE 802.1q specification.
Screenshot

PARAMETER VLAN ID
VLAN USER PRIORITY
VLAN SYNCHRONIZATION

DEFAULT VALUES 0
0
Disabled

DESCRIPTION
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. This is a 3-bit value that defines the user priority. Values are from 0 (best effort) to 7 (highest); 1 represents the lowest priority. These values can be used to prioritize different classes of traffic (voice, video, data, etc.). Permitted value(s): 8 priority levels (i.e. 0 to 7) Default disabled. By enabled the VLAN ID is automatic synchronized between the bases in the chain. Bases will be automatic rebooted during the synchronization.

For further help on VLAN configuration refer to Appendix.

5.5.3 DHCP Options
Screenshot

PARAMETER PLUG-N-PLAY

DEFAULT VALUES DESCRIPTION

Enabled

Enabled: DHCP option 66 to automatically provide PBX IP address to base.

5.5.4 Static IP settings
If there is no DHCP server present you need to set a static IP. When you plug- in the LAN cable and the Base station don’t get IP from a DHCP server it uses RFC3927 Static IP fall back.
Static IP address fall back – RFC3927 If a base station boots without a DHCP server on the network, it boots up using static IP as defined in RFC3927. Base continuously request IP and after 3min. the base enters static IP address in range: 169.254.x.x

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To find base static IP use “Find IP” handset feature. To access the web interface, set the PC to static IP in the same subnet as the base station and you can now access the web interface. NOTE: “Find IP” go to menu and press 47, then the handset with start searching for base stations.
5.5.5 NAT Settings
We define some options available when NAT aware routers are enabled in the network.
Screenshot

PARAMETER ENABLE STUN STUN SERVER
STUN BINDTIME DETERMINE STUN BINDTIME GUARD ENABLE RPORT
KEEP ALIVE TIME

DEFAULT VALUES Disabled N/A
Enabled

DESCRIPTION Enable to use STUN Permitted value(s): AAA.BBB.CCC.DDD (Currently only Ipv4 is supported) or URL (e.g.: firmware.rtx.net).

80

Permitted values: Positive integer default is 80, unit is in seconds

Disabled

Enable to use RPORT in SIP messages.

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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5.5.6 SIP/RTP Settings
These are some definitions of SIP/RTP settings:
Screenshot

PARAMETER USE DIFFERENT SIP PORTS

DEFAULT VALUES Disabled

RTP COLLISION DETECTION ALWAYS REBOOT ON CHECK-SYNC OUTBOUND PROXY MODE
FAILOVER SIP TIMER B
FAILOVER SIP TIMER F
LOCAL SIP PORT

Enabled Disabled
Use Always 5 5 5060

SIP TOS/QOS 0x68

RTP PORT
RTP PORT RANGE RTP TOS/QOS

50004 254 0xB8

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DESCRIPTION If disabled, the Local SIP port parameter specifies the source port used for SIP signaling 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. Enable: If two sources with same SSRC, the following RTX is discarded. Disabled: No check ­ device will accept all sources. Reboot base station when new configuration I loaded.
Use Always: All outbound calls are sent to outbound proxy Only Initial request: Only use outbound proxy for initial SIP requests When the time expires and the corresponding SIP transaction fails, failover will be triggered When the time expires and the corresponding SIP transaction fails, failover will be triggered The source port used for SIP signaling Permitted values: Port number default 5060. Priority of call control signaling traffic based on both IP Layers of Type of Service (ToS) byte. ToS is referred to as Quality of Service (QoS) in packetbased networks. Permitted values: Positive integer, default is 0x68 The first RTP port to use for RTP audio streaming. Permitted values: Port number default 50004 (depending on the setup). The number of ports that can be used for RTP audio streaming. Permitted values: Positive integers, default is 254 Priority of RTP traffic based on the IP layer ToS (Type of Service) byte. ToS is referred to as Quality of Service (QoS) in packet-based networks. See RFC 1349 for details. “cost bit” is not supported.
o Bit 7..5 defines precedence. o Bit 4..2 defines Type of Service. o Bit 1..0 are ignored. Setting all three of bit 4..2 will be ignored.
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REJECT ANONYMOUS CALLS

Disabled

5.5.7 TCP Options
Screenshot

Permitted values: Positive integer, default is 0xB8 If disabled, all calls will be received. If enabled, calls not registered will be automatically rejected

PARAMETER TCP KEEP ALIVE INTERVAL

DEFAULT VALUES 120s

DESCRIPTION Specifies the interval the client waits before sending a keep- alive message on a TCP connection.

5.5.8 Discovery

The following parameters of the “Discovery” section are explained

PARAMETER LLDP-MED SEND LLDP-MED SEND DELAY

DEFAULT VALUES Disabled
30

VLAN VIA LLDP- Disabled MED

DESCRIPTION If “Enabled”, the BS will send 5 LLDP-MED messages when started.
Sends messages every 30 seconds to inform the network about its LLDPMED data Note: This option works only if the first parameter is enabled (LLDP-MED SEND) If “Enabled”, the BS will try to retrieve a VLAN ID from the received LLDPMED from a switch Note: This feature is available only if the first parameter is enabled (LLDPMED SEND)

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5.6 Management Settings Definitions
The administrator can configure base stations to perform some specific functions such as configuration of file transfers, firmware up/downgrades, password management, and SIP/debug logs.
Screenshot

5.6.1 Settings:

PARAMETER BASE STATION NAME:

Default value SME VoIP

MANAGEMENT TFTP TRANSFER PROTOCOL

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Description It indicates the title that appears at the top window of the browser and is used in the dualcellpage. Maximum characters: 35 The protocol assigned for configuration file and central directory Valid Input(s): TFTP, HTTP, HTTPs
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HTTP MANAGEMENT UPLOAD SCRIPT

Empty

HTTP MANAGEMENT USERNAME HTTP MANAGEMENT PASSWORD FACTORY RESET FROM BUTTON ENABLE AUTOMATIC PREFIX

Empty Empty Enabled Disabled

SET MAXIMUM DIGITS FOR INTERNAL NUMBERS SET PREFIX FOR OUTGOING CALLS

0 Empty

The folder location or directory path that contains the configuration files of the Configuration server. The configuration upload script is a file located in e.g. TFTP server or Apache Server which is also the configuration server. Permitted value(s): / Example: /CfgUpload Note: Must begin with (/) slash character. Either / or can be used. Username that should be entered in order to have access to the configuration server. Permitted value(s): 8-bit string length Password that should be entered in order to have access to the configuration server. Permitted value(s): 8-bit string length If enabled a factory reset will be possible by pressing the button on the BS If disabled, no action will be present by pressing the button on the BS Disabled: Feature off. Enabled: The base will add the leading digit defined in “Set Prefix for Outgoing Calls”. Enabled + fall through on and #: Will enable detection of or # at the first digit of a dialed number. In case of detection the base will not complete the dialed number with a leading 0. Examples: 1: dialed number on handset 1234 – > dialed number to the pabx 1234 2: dialed number on handset #1234 – > dialed number to the pabx #1234 3: dialed number on handset 1234 – > dialed number to the pabx 01234 Used to detect internal numbers. In case of internal numbers, no prefix number will be added to the dialed number.
Set the prefix for outgoing calls. Users need to dial this prefix to get an outside line.

5.6.2 Configuration:

PARAMETER CONFIGURATION FILE DOWNLOAD
CONFIGURATION SERVER ADDRESS
BASE SPECIFIC FILE MULTI CELL SPECIFIC FILE
AUTO RESYNC POLLING AUTO RESYNC TIME AUTO RESYNC DAYS AUTO RESYNC PERIODIC (MIN)

Default value Base Specific File
Empty
Empty Empty
Disabled 00:00 0 0

Description Base Specific file: Used when configuring a single cell base Base and Multicell Specific File: Used on out of factory bases to specify VLAN and settings. Server/device that provides configuration file to base station. Type: DNS or IP address Permitted value(s): AAA.BBB.CCC.DDD or Base configuration file
The file name must be the chain id of the system. E.g. 00087b0a00b3.cfg Permitted value(s): Format of file is chain ID.cfg Enable to have the base station look for new configuration file, with a predefined time interval Time when the base station shall load the configuration file 24 hour setting Number of days between Auto Resync
Number of minutes between Auto Resync

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AUTO RESYNC DELAY DHCP CONTROLLED CONFIG SERVER
DHCP CUSTOM OPTION
DHCP CUSTOM OPTION TYPE

15 DHCP Option 66
Empty Empty

Delay time in sec, to prevent all base station asking for configuration fin at the same time. Provisioning server options. DHCP Option 66: Look for provision file by TFTP boot up server. DHCP Custom Option: Look for provision file by custom option DHCP Custom Option & Option 66: Look for provision file by first custom option and then option 66. By default, option 160, but custom option can be defined. An option 160 URL defines the protocol and path information by using a fully qualified domain name for clients that can use DNS. URL: URL of server with path. Example of URL: http://myconfigs.com:5060/configs Default configuration file on server must follow the name: MAC.cfg IP Address: IP of server with path.

5.6.3 Text messaging:

PARAMETER

DEFAULT VALUE

TEXT MESSAGING Disabled

TEXT MESSAGING & ALARM SERVER TEXT MESSAGING PORT TEXT MESSAGING KEEP ALIVE (M) TEXT MESSAGING RESPONSE (S) TEXT MESSAGING TTL

Empty 1300 30 30 0

DESCRIPTION Disable/enable messaging using a Message/Alarm server Enable Without Server. With this setting handset can send messages to other handsets, which support messaging. Permitted value(s): AAA.BBB.CCC.DDD or
Port number of message server.
This defines the frequency of how keep-alive are sent Permitted values: Positive integer, unit is in minutes This defines the frequency of how response timeout Permitted values: Positive integer, unit is in seconds This defines the text messaging time to live Permitted values: Positive integer, unit is in seconds

5.6.4 Terminal:

PARAMETER KEEP ALIVE (M)
AUTO STOP ALARM AUTO STOP ALARM DELAY (S)

DEFAULT VALUE 0
Disabled 30

DESCRIPTION If different from “0” the handset sends a (emergencyLocationMsg) containing the RSSI measurements with interval “x” that is set. Permitted values: Positive integer, unit is in minutes Enable to activate “AUTO STOP ALARM DELAY”
Handset automatically stops alarm announcement (emergencySms) after “x” sec.

5.6.5 Syslog/SIP Log:

PARAMETER UPLOAD OF SIP LOG

DEFAULT VALUE Disabled

DESCRIPTION Enable this option to save low level SIP debug messages to the server. The SIP logs are saved in the file format: