FCC ID: T7HX9431 VOIP Phone

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 Technical Reference Document Version 4.9 Dec-2019 RTX A/S, Denmark Trademarks SME VOIP SYSTEM GUIDE 4.6 Proprietary and Confidential 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 1.2 1.3 1.4 1.5 1.6 1.7 1.8 1.9 Audience ........................................................................................................................................................... 7 When Should I Read This Guide ........................................................................................................................ 7 Important Assumptions ..................................................................................................................................... 7 Whats Inside This Guide ................................................................................................................................... 7 Whats Not in This guide ................................................................................................................................... 8 Abbreviations .................................................................................................................................................... 8 References/Related Documentation ................................................................................................................. 8 Document History ............................................................................................................................................. 9 What is new ....................................................................................................................................................... 9 1.10 Documentation Feedback ................................................................................................................................. 9 2 Introduction System Overview .............................................................................................................................. 10 2.1 2.2 Hardware Setup .............................................................................................................................................. 10 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 2.4 2.5 Wireless Bands ................................................................................................................................................ 11 System Capacity (in Summary) ........................................................................................................................ 11 Advantages of SME VoIP System ..................................................................................................................... 12 3 Installation of Base Stations/Repeater ..................................................................................................................... 13 3.1 3.2 3.3 3.4 Package Contents/Damage Inspection ......................................................................................................... 13 RTX Base Station Mechanics ........................................................................................................................... 14 RTX Base Unit Reset feature ......................................................................................................................... 15 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 4.2 4.3 Package Contents/Damage Inspection ......................................................................................................... 18 Before Using the Phone .................................................................................................................................. 18 Using the Handset ........................................................................................................................................... 20 5 SME VoIP Administration Interface .......................................................................................................................... 20 5.1 5.2 5.3 Web navigation ............................................................................................................................................... 20 Home/Status ................................................................................................................................................... 22 Extensions ....................................................................................................................................................... 23 SME VOIP SYSTEM GUIDE 4.6 Proprietary and Confidential 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 5.5 Servers ............................................................................................................................................................. 31 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 5.10.2 5.10.3 5.10.4 5.10.5 5.10.6 5.10.7 5.10.8 Certificates .............................................................................................................................................. 50 Certificates list ........................................................................................................................................ 51 SIP Client Certificates .............................................................................................................................. 51 Device identity ........................................................................................................................................ 52 Trusted Server Certificates ..................................................................................................................... 53 Trusted Root Certificates ........................................................................................................................ 53 Password ................................................................................................................................................ 53 Secure Web Server ................................................................................................................................. 54 5.11 Central Directory and LDAP ............................................................................................................................. 54 5.11.1 5.11.2 Local Central Directory ........................................................................................................................... 54 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 5.12.2 5.12.3 5.12.4 5.12.5 5.12.6 Settings for Base Unit ............................................................................................................................. 57 DECT System Settings ............................................................................................................................. 59 Base System Settings .............................................................................................................................. 60 Base Station Group ................................................................................................................................. 61 DECT Chain ............................................................................................................................................. 62 RTX8660 -RTX8663 Mixed mode ............................................................................................................ 63 5.13 LAN SYNC ......................................................................................................................................................... 64 5.13.1 5.13.2 5.13.3 Settings for Base Unit ............................................................................................................................. 64 Base station group .................................................................................................................................. 65 This unit debug ....................................................................................................................................... 65 5.14 Repeaters ........................................................................................................................................................ 66 5.14.1 5.14.2 5.14.3 Add repeater .......................................................................................................................................... 66 Register Repeater ................................................................................................................................... 68 Repeaters list .......................................................................................................................................... 68 5.15 Alarm ............................................................................................................................................................... 69 5.15.1 Use of Emergency Alarms ....................................................................................................................... 72 5.16 Statistics .......................................................................................................................................................... 73 5.16.1 5.16.2 5.16.3 5.16.4 5.16.5 5.16.6 System data ............................................................................................................................................ 73 Free Running explained .......................................................................................................................... 73 Call data .................................................................................................................................................. 74 Repeater data ......................................................................................................................................... 75 DECT data ............................................................................................................................................... 76 Call quality .............................................................................................................................................. 77 5.17 Generic Statistics ............................................................................................................................................. 78 5.17.1 5.17.2 5.17.3 5.17.4 DECT Synchronization Statistics ............................................................................................................. 80 RTP Statistics........................................................................................................................................... 81 IP Stack statistics .................................................................................................................................. 82 System Statistics ..................................................................................................................................... 82 5.18 Diagnostics ...................................................................................................................................................... 83 5.18.1 5.18.2 5.18.3 Base Stations .......................................................................................................................................... 83 Extensions ............................................................................................................................................... 83 Logging ................................................................................................................................................... 84 5.19 5.20 5.21 Configuration................................................................................................................................................... 86 Sys log .............................................................................................................................................................. 87 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 SME VOIP SYSTEM GUIDE 4.6 Proprietary and Confidential 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 Whats Inside This Guide We summarize the contents of this document in the table below:

WWHERE IS IT?

CHAPTER 2 CONTENT Introduction System Overview CHAPTER 3 CHAPTER 4 CHAPTER 5 Installation of Base station/Repeater Making Handsets Ready SME VoIP Administration Interface Multi-Cell Setup & Management 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 Registration Management Handsets Learn how to register handset and extensions to base stations APPENDIX HOW-TO SETUP A DUAL-CELL SYSTEM APPENDIX ADDING EXTENSIONS SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 7 | P a g e APPENDIX FIRMWARE UPGRADE APPENDIX MULTILINE FEATURE APPENDIX FUNCTIONALITY OVERVIEW Firmware Upgrade/Downgrade Management Provides the procedure of how to upgrade firmware to base 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 Whats 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) SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 8 | P a g e 1.8 Document History RREVISION 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. SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 9 | P a g e 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 base-

station(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 10 | P a g e 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. 22.2.1 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 11 | P a g e DDESCRIPTION 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) CCAPACITY 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. SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 12 | P a g e 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:

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 DC adaptor plug type by countries S008ACM0500200 S010WB0500200 S010WV0500200 S010WU0500200 S010WS0500200 Multi-plug UK EU US AU

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

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:

SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 13 | P a g e 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. 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. 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:

SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 14 | P a g e 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 its an indoor or outdoor installation. 33.4.1 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 units 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. SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 15 | P a g e 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. 33.5.1 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 SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 16 | P a g e 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. SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 17 | P a g e 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:

2 x mounting screws and 2 x Anchors 1 x Handset hook 1 x A/C Adaptor 1 x Battery 1 x charger 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. 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. 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. SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 18 | P a g e Handset in the Charger For correct charging, ensure that the room temperature is between 5 and 25/41F and 77F. 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. SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 19 | P a g e 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 stations 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. SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 20 | P a g e 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. SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 21 | P a g e 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. 22 | P a g e 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. REBOOT FORCED REBOOT 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. 55.3.1 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. SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 23 | P a g e 55.3.2 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 SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 24 | P a g e 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 ANSWER NUMBER Empty FORWARDING ON BUSY NUMBER Empty 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 its 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. SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 25 | P a g e Extension list 5.3.2.1 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 IPEIs. The IPEIs will be listed in this cell. Handset list 5.3.2.2 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 SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 26 | P a g e 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. Handset and extension list top/sub-menus 5.3.2.3 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. 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) SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 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). 27 | P a g e NOTE: By powering off the handset, the handset will SIP deregister from the PBX. 55.3.3 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. SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 28 | P a g e 55.3.4 Edit Handset Use the mouse to click the handset IPEI link to open the handset editor window. Screenshot PARAMETER IPEI DEFAULT VALUE(S) Handset IPEI AC Handset AC code ALARM LINE ALARM NUMBER No Alarm Line Selected Empty RECEIVE MODE Disabled TRANSMIT INTERVAL Disabled SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 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) Enter Proximity:

Leave Proximity:

Enter or Leave Proximity:

NOTE: This feature is only shown if Handset has BTLE. (RTX8630 and RTX8430 is not supported) 29 | P a g e 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. ALARM PROFILES Not configured IMPORT LOCAL PHONEBOOK EXPORT LOCAL PHONEBOOK Import local phonebook 5.3.4.1 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 The Export local phonebook feature makes it possible to retrieve all contracts from a specific phone to a .CSV file. Export local phonebook Screenshot Press the export button and save the .CSV file on you PC or Server. SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 30 | P a g e 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 SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 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 31 | P a g e OUTBOUND PROXY Empty CONFERENCE SERVER Empty CALL LOG SERVER Empty Empty 600 MUSIC ON HOLD SERVER RE-

REGISTRATION TIME SIP SESSION TIMERS:

Disabled SESSION TIMER VALUES

(S):

SIP TRANSPORT SIGNAL TCP SOURCE PORT 1800 UDP Disabled Disabled USE ONE TCP/TLS CONNECTION PER SIP EXTENSION:

SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential Permitted value(s): AAA.BBB.CCC.DDD:<Port-Number> or

<URL>:<Port-Number>

Note: Specifying the Port Number is optional. This is a Session Border Controller DNS or IP address (OR SIP server outbound proxy address) Set the Outbound proxy to the address and port of 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: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 keep-

alive or more correctly session refresh signals. If no session refresh is received when the timer expires the call will be terminated. Default value is 1800 s according to the RFC. Min: 90 s. Max:

65636. If disabled session timers will not be used. Default value is 1800s according to the RFC. If disabled session timers will not be used. Permitted value(s): Minimum value 90, Maximum 65636 Select UDP, TCP, TLS 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 32 | P a g e Disabled RTP FROM OWN BASE STATION:

KEEP ALIVE Enabled SHOW EXTENSION ON HANDSET IDLE SCREEN HOLD BEHAVIOUR Enabled RFC 3264 Enabled Enabled Hold 2nd Call LOCAL RING BACK TONE REMOTE RING TONE CONTROL ATTENDED TRANSFER BEHAVIOUR Disabled Empty Disabled Empty Disabled DIRECT CALL PICKUP DIRECT CALL PICKUP CODE GROUP CALL PICKUP GROUP CALL PICKUP CODE USE OWN CODEC PRIORITY SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 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 doesnt 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 PBXs do not expect that the 2nd call is put on hold, and therefore attended transfer fails on these PBXs. 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. 33 | P a g e 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 RTCP SEND SDP CAPABILITIES IN OFFER (RFC 5939) SECURE RTP SECURE RTP AUTH Enabled 20ms Enabled Disabled Disabled Disabled SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 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. 34 | P a g e SRTP CRYPTO SUITES AES_CM_128_HMAX_SHA1_32 AES_CM_128_HMAX_SHA1_80 Field list of supported SRTP Crypto Suites. The device is born with two suites. Note: Within servers or even with multi servers, extensions must always be unique. This means same extension number on server 1 cannot be re-used on server 2. 5.5 Network In this section, we describe the different parameters available in the network configurations menu. Screenshot SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 35 | P a g e 55.5.1 IP Settings Screenshot PARAMETER DHCP/STATIC IP DEFAULT VALUES DHCP IP ADDRESS N/A SUBNET MASK N/A DEFAULT GATEWAY N/A 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. Devices 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 <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. This is an alternate DNS server. Enable to allow Multicast Domain Name system (MDNS) SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 36 | P a g e 55.5.2 VLAN Settings Enable users to define devices (e.g. Base station, etc.) with different physical connection to communicate as if they are connected on a single network segment. The VLAN settings can be used on a managed network with separate Virtual LANs (VLANs) for sending voice and data traffic. To work on these networks, the base stations can tag voice traffic it generates on a specific voice VLAN using the IEEE 802.1q specification. Screenshot PARAMETER VLAN ID DEFAULT VALUES DESCRIPTION 0 Is a 12-bit identification of the 802.1Q VLAN. Permitted value(s): 0 to 4094 (only decimal values are accepted) 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. VLAN USER PRIORITY 0 VLAN SYNCHRONIZATION Disabled 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 dont 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 SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 37 | P a g e 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. 55.5.5 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 DEFAULT VALUES DESCRIPTION Disabled N/A Enable to use STUN Permitted value(s): AAA.BBB.CCC.DDD (Currently only Ipv4 is supported) or URL (e.g.: firmware.rtx.net). Enabled 80 Permitted values: Positive integer default is 80, unit is in seconds 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 SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 38 | P a g e 55.5.6 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 Enabled Disabled Use Always 5 5 LOCAL SIP PORT 5060 SIP TOS/QOS 0x68 RTP PORT 50004 RTP PORT RANGE RTP TOS/QOS 254 0xB8 SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 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 UAs 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 packet-

based networks. Permitted values: Positive integer, default is 0x68 The first RTP port to use for RTP audio streaming. Permitted values: Port number default 50004 (depending on the setup). The number of ports that can be used for RTP audio streaming. Permitted values: Positive integers, default is 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. 39 | P a g e Permitted values: Positive integer, default is 0xB8 If disabled, all calls will be received. If enabled, calls not registered will be automatically rejected REJECT ANONYMOUS CALLS Disabled 55.5.7 TCP Options Screenshot 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 DEFAULT VALUES Disabled DESCRIPTION If Enabled, the BS will send 5 LLDP-MED messages when started. PARAMETER LLDP-MED SEND LLDP-MED SEND DELAY 30 VLAN VIA LLDP-

MED Disabled Sends messages every 30 seconds to inform the network about its LLDP-

MED 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 LLDP-

MED from a switch Note: This feature is available only if the first parameter is enabled (LLDP-

MED SEND) SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 40 | P a g e 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 55.6.1 Settings:

PARAMETER BASE STATION NAME:

MANAGEMENT TRANSFER PROTOCOL Default value SME VoIP TFTP 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 SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 41 | P a g e 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 55.6.2 Configuration:

The folder location or directory path that contains the configuration files of the Configuration server. The configuration upload script is a file located in e.g. TFTP server or Apache Server which is also the configuration server. Permitted value(s): /<configuration-file-directory>

Example: /CfgUpload Note: Must begin with (/) slash character. Either / or \ can be used. 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. Default value Base Specific File Base Specific file: Used when configuring a single cell base Description PARAMETER CONFIGURATION FILE DOWNLOAD CONFIGURATION SERVER ADDRESS Empty BASE SPECIFIC FILE MULTI CELL SPECIFIC FILE AUTO RESYNC POLLING AUTO RESYNC TIME AUTO RESYNC DAYS AUTO RESYNC PERIODIC (MIN) Empty Empty Disabled 00:00 0 0 SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 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 <URL>

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 42 | P a g e AUTO RESYNC DELAY DHCP CONTROLLED CONFIG SERVER 15 DHCP Option 66 DHCP CUSTOM OPTION DHCP CUSTOM OPTION TYPE Empty Empty 55.6.3 Text messaging:

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. PARAMETER TEXT MESSAGING Disabled DEFAULT VALUE DESCRIPTION TEXT MESSAGING

& ALARM SERVER TEXT MESSAGING PORT TEXT MESSAGING KEEP ALIVE (M) TEXT MESSAGING RESPONSE (S) TEXT MESSAGING TTL Empty 1300 30 30 0 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 <URL>

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) DEFAULT VALUE DESCRIPTION 0 AUTO STOP ALARM AUTO STOP ALARM DELAY (S) Disabled 30 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:

<MAC_Address><Time_Stamp>SIP.log SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 43 | P a g e SYSLOG LEVEL Normal Operation 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 analyze level also contains the messages from normal operation. Debug: Used by RTX for debug. Should not be enabled during normal operation. TLS SECURITY SYSLOG SERVER IP ADDRESS SYSLOG SERVER PORT Disabled Empty If enabled, it uses encrypted TCP, else - UDP Permitted value(s): AAA.BBB.CCC.DDD or <URL>

514 Port number of syslog server. 55.6.6 Location Gateway PARAMETER LOCATION GATEWAYS:

DEFAULT VALUE DESCRIPTION Disabled Enable to allow Location Gateways onto the system. When enabled Location Gateway menu will be shown on main menu on the left. Permitted value(s): AAA.BBB.CCC.DDD or <URL>

CONFIGURATION SERVER:

Empty 5.6.7 License:

PARAMETER LICENSE DEFAULT VALUE None DESCRIPTION This feature allows administrators to register RTX8930 genetic headsets to the system. License key must be obtained from authorized resellers and only license matching the systems provider code will work. 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 detailed information See Appendix D. 5.7 Firmware Update 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 home/status page. Screenshot SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 44 | P a g e PARAMETER FIRMWARE UPDATE SERVER ADDRESS DEFAULT VALUE(S) Empty FIRMWARE PATH Empty TERMINAL FILE PATH Empty REQUIRED VERSION Empty REQUIRED BRANCH Empty STARTUP PICTURE Empty BACKGROUND PICTURE Empty SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential DESCRIPTION IP address or DNS of firmware update files source Valid Inputs: AAA.BBB.CCC.DDD or <URL>

Example: firmware.rtx.net or 10.10.104.41 Location of firmware on server (or firmware update server path where firmware update files are located). Example: RTXFWU Location of image (folder where background and start up image are located). Example: Images Version of firmware to be upgraded (or downgraded) on handset, repeater, or base station. Valid Input(s): 8-bit string length. E.g. 400 Note: Value version 0 will disable firmware upgrade Note: Two handset types will be serial firmware upgraded. First type 8630 then type 8430. Branch of firmware to be upgraded (or downgraded) handset, repeater or base station. Valid Input(s): 8-bit string length. E.g. 01 Name of the startup picture you want on the handsets when they are powered up. NOTE: Image have same resolution as the screen on the handset(s), this can be found in the handset datasheets If the image does not have the same resolution as the screen, it will be placed in the top left corner. To small the rest of the screen will be black. To large only the left portion of the image will be shown. NOTE: Only .BMP is files are supported. NOTE: Changing startup picture is not available for new GUI

(RTX8631/RTX8632 and RTX8633) Name of the background picture you want on the handsets when they are powered up. NOTE: Images have same resolution as the screen on the handset(s), this can be found in the handset datasheets. If the image does not have the same resolution as the screen, it will be placed in the top left corner. To small the rest of the screen will be black. To large only the left portion of the image will be shown NOTE: Only .BMP is files are supported. NOTE: Changing background picture is not available for new GUI

(RTX8631/RTX8632 and RTX8633) 45 | P a g e 55.7.1 Warning message when firmware upgrading A warning message will be displayed when starting firmware upgrade. Screenshot 5.8 Location Gateways In this section we describe the different setting for Location gateways. NOTE: to activate Location gateways it must be enabled on the management page (Please see chapter 5.6 for more details) 5.8.1 Register Location gateway Once you have enabled the feature from the Management menu, please follow the steps below in order to add the Location Gateway:

Step 1: Select Add Location Gateway extension Screenshot Step 2: Press save and leave the IPEI: FFFFFFFFF Screenshot Step 3: Check the box on the Location gateways that you want to register SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 46 | P a g e Screenshot 5.9 Country/Time Settings In this section, we describe the different parameters available in the Country/Time settings menu. The country setting controls the following in-band tones used by the system:

- Dial tone

Busy tone

Ring Back tone

Call Waiting tone

Re-order tone The Time server supplies the time used for data synchronisation in a dual-cell configuration. As such it is mandatory for a dual-

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 or type the IP address of an NTP server that is closer to you (find it via Google). 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. Changing time settings will require a reboot of system. Screenshot SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 47 | P a g e PARAMETER SELECT COUNTRY DEFAULT VALUES US/Canada STATE / REGION SELECT LANGUAGE N/A English TIME SERVER Empty ALLOW BROADCAST NTP REFRESH TIME (H) SET TIME ZONE BY COUNTRY/REGION TIMEZONE Checked 24 Checked DESCRIPTION Supported countries: Australia, Belgium, Brazil, Denmark, Germany, Spain, France, Ireland, Italia, Luxembourg, Nederland, New Zealand, Norway, Portugal, Swiss, Finland, Sweden, Turkey, United Kingdom, US/Canada, Austria Only shown by country selection US/Canada, Australia, Brazil Web interface language. Number of available languages: English, Dansk, Italiano, Trke, Deutsch, Portuguese, Hrvatski, Srpski, Slovenian, Nederlands, Francaise, Espanyol, Russian, Polski. 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. By checked time server is used. The window time in hours within which time server refreshes. Valid Inputs: positive integer By checked country setting is used (refer to country web page). 0 Refers to local time in GMT or UTC format. SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 48 | P a g e SET DST BY COUNTRY/REGION DAYLIGHT SAVING TIME (DST) DST FIXED BY DAY DST START MONTH DST START DATE DST START TIME DST START DAY OF WEEK DST START DAY OF WEEK, LAST IN MONTH DST STOP MONTH DST STOP DATE Checked Automatic Use Month and Day of week March 0 2 Sunday Second First In Month October 0 DST STOP TIME 2 Sunday DST STOP DAY OF WEEK DST STOP DAY OF WEEK LAST IN MONTH Min: -12:00 Max: +13:00 By checked country setting is used (refer to country web page). The system administrator can Enable or Disable DST manually. Automatic: Enter the start and stop dates if you select Automatic. You determine when DST actually changes. Choose the relevant date or day of the week, etc. from the drop-down menu. Month that DST begins Valid Input(s): Gregorian months (e.g. January, February, etc.) Numerical day of month DST comes to effect when DST is fixed to a specific date Valid Inputs: positive integer DST start time in the day Valid Inputs: positive integer Day within the week DST begins Specify the week that DST will actually start. The month that DST actually stops. The numerical day of month that DST turns off. Valid Inputs: positive integer (1 to 12) The time of day DST stops Valid Inputs: positive integer (1 to 12) The day of week DST stops Last in Month The week within the month that DST will turn off. NOTE: By checked time zone and DST the parameters in web page Time will be discarded. 5.10 Security The security section is used for loading of certificates and for selecting if only trusted certificates are used. Furthermore, web password can be configured. The Security web is divided into three sections: Certificates (trusted), SIP Client Certificates (and keys) and Password administration. To setup secure fwu and configuration file download select HTTPs for the Management Transfer Protocol (refer to chapter 5.6). SIP and RTP security are dependent servers and in order to configure them , the user must use the web option Servers (refer to chapter 5.4) Screenshot SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 49 | P a g e 55.10.1 Certificates The certificates list contains the list of loaded certificates for the system. Using the left column check mark, it is possible to check and delete certificates. To import a new certificate, use the mouse to click on Choose file and browse to the selected file. When file is selected, use the Load button to load the certificate. The certificate format supported is DER encoded binary X.509 (.cer). SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 50 | P a g e Screenshot 55.10.2 Certificates list PARAMETER IDX ISSUED TO ISSUED TO VALID UNTIL Screenshot DEFAULT VALUES Fixed indexes Empty Empty Empty DESCRIPTION Index number IP address which is part of the certificate file Organization, Company which is part of the certificate file Date Time Year which is part of the certificate file By enabling Use Only Trusted Certificates, the certificates the base will receive from the server must be valid and loaded into the system. If no valid matching certificate is found during the TLS connection establishment, the connection will fail. When Use Only Trusted Certificates is disabled, all certificates received from the server will be accepted. 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. 5.10.3 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 Choose 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 Certificate Screenshot SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 51 | P a g e 55.10.4 Device identity The certificate and personal key used by the base when acting as server or when the server requires client authentication in the SSL handshake procedure. Screenshot SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 52 | P a g e 55.10.5 Trusted Server Certificates Intermediate certificates (non-root certificates) trusted by the base. Used to validate a received certificate chain (or a chain of trust) in scenarios where only the root certificate is sent by the server during the SSL handshake procedure Screenshot 5.10.6 Trusted Root Certificates Root certificates (self-signed) trusted by the base. Used to validate received root certificates sent by the server during the SSL handshake procedure. Screenshot 5.10.7 Password In the below settings the password parameters are defined. SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 53 | P a g e PARAMETER USERNAME Default Values Admin CURRENT PASSWORD NEW PASSWORD Admin Empty CONFIRM PASSWORD Empty Description Can be modified to any supported character and number Maximum characters: 15 Can be modified to any supported character and number Change to new password Maximum characters: 15 Confirm password to reduce accidently wrong changes of passwords Password valid special signs:

Password valid numbers:

Password valid letters:

0-9 a-z and A-Z 55.10.8 Secure Web Server This setting allows all communication with the Web Server to be encrypted. Screenshot PARAMETER HTTPS DEFAULT VALUES Disabled DESCRIPTION Enable to use HTTPS for Web Server Communication. 5.11 Central Directory and LDAP The SME VOIP 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. 5.11.1 Local Central Directory Select local and save for local central directory. Screenshot PARAMETER LOCAL DEFAULT VALUES Local DESCRIPTION Drop down menu to select between local central directory, LDAP based central directory and xml server SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 54 | P a g e SERVER Empty FILENAME Empty PHONEBOOK RELOAD INTERVAL (S) 0 The parameter is used if directory file is located on server. VValid inputs: aaa.bbb.ccc.ddd or <url>

Refer to appendix for further details. The parameter is used if directory file is located on server. Refer to appendix for further details The parameter is controlling the reload interface of phonebook in seconds. The feature is for automatic reload the base phonebook file from the server with intervals. It is recommended to specify a conservative value to avoid overload of the base station. With default value setting 0 the reload feature is disabled. Import Central Directory 5.11.1.1 The import central directory 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. The central directory feature can handle up to 3000 contacts (Max file size 100kb). For further details of the central directory feature refer to appendix. Screenshot 5.11.2 LDAP Select LDAP Server and save for LDAP server configuration. Screenshot SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 55 | P a g e Screenshot PARAMETER LDAP SERVER DEFAULT VALUES LDAP Server SERVER TLS SECURITY PORT SBASE Empty Disabled Empty Empty LDAP FILTER Empty BIND PASSWORD VIRTUEL LISTS NAME Empty Empty Enabled Empty WORK NUMBER Empty HOME NUMBER Empty MOBILE NUMBER Empty SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential DESCRIPTION Drop down menu to select between local central directory and LDAP based central directory. LDAP Server is displayed when LDAP server is selected. IP address of the LDAP server. Valid Inputs: AAA.BBB.CCC.DDD or <URL>

If enabled, it uses encrypted TCP, else - UDP The server port number that is open for LDAP connections. Search Base. The criteria depends on the configuration of the LDAP server. Example of the setting is CN=Users, DC=umber, DC=loc LDAP Filter is used to as a search filter, e.g. setting LDAP filter to

(|(givenName=%*)(sn=%*)) the IP-DECT will use this filter when requesting entries from the LDAP server. % will be replaced with the entered prefix e.g. searching on J will give the filter

(|(givenName=J*)(sn=J*)) resulting in a search for given name starting with a J or surname starting with J. Bind is the username that will be used when the IP-DECT phone connects to the server Password is the password for the LDAP Server By enable, virtual list searching is possible The name can be used to specify if sn+givenName or cn (common name) is return in the LDAP search results Work number is used to specify that LDAP attribute that will be mapped to the handset work number Home number is used to specify that LDAP attribute that will be mapped to the handset home number Mobile number is used to specify that LDAP attribute that will be mapped to the handset mobile number 56 | P a g e 55.11.3 Characters supported The below table shows which characters are supported in the communication between RTX9431 and handset. 5.12 Dual-cell Parameter Definitions NOTE: To join one Base Station in a dual-cell system, you need to have one handset added to the system. For details and Step-

by-Step guide to dual cell, please see Appendix In this section, we describe the different parameters available in the Dual-cell configurations menu. 5.12.1 Settings for Base Unit Description of Settings for Specific Base units is as follows:

Screenshot Dual-Cell status covers status of data synchronization. The status Keep-alive means normal operation, as well as Idle. SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 57 | P a g e PARAMETER DUAL CELL SYSTEM DEFAULT VALUES Disabled SYSTEM CHAIN ID 512 DATA SYNC:

Multicast PRIMARY DATA SYNC IP Empty DUAL CELL DEBUG None DESCRIPTION Enable this option to allow the Base unit to be set in dual-cell mode (can be set either as master or slave in the dual-cell chain system refer to MAC-

units in Chain section for details). Valid Inputs: Enable, Disable Must save and reboot after change from disabled to enable. This is an identifier (in string format e.g. 2275) that is unique for a specific dual-cell system. The Chain ID value MUST not be equal to a used SIP account. The Chain ID uses up a SIP account with this value. NOTE: Chain ID is used as SIP account for check Sync. Default value is 512, which means extension 512 must not be used unless the chain ID is modified. Chain ID can be modified by provisioning only. Note: There can be several dual-cell systems in SME network. Up to 24 levels of base stations chains are permitted in a setup. Valid Input: The Web site allow max 5 digits in this field. To select between multicast or Peer to Peer data synchronization mode. The multicast port range and IP addresses used is calculated from the chain id. The multicast feature uses the port range: 49200 49999 The multicast feature IP range: 224.1.0.0 225.1.0.0 Multicast uses UDP. For multi-cast operation make sure that Multicast/IGMP is enabled on your switch(es), else use Peer-to-peer mode. IP of base station data sync source the base handling the data synchronization. Using multicast this base IP is selected automatically. The data sync feature uses the port range: 49200 49999 NOTE: Using Peer to Peer mode the IP of the base used for data sync. source MUST be defined. NOTE: Using Peer to Peer mode with version below V306 limits the system automatic recovery feature as there is no automatic recovery of the data sync. source in Peer to Peer mode. Enable this feature, if you want the system to catalogue low level dual-cell debug information or traces. Options:

Data Sync: Writes header information for all packets received and sent to be used to debug any special issues. Generates LOTS of SysLog signaling and is only recommended to enable shortly when debugging. Auto Tree: Writes states and data related to the Auto Tree Configuration feature. Both: Both Data Sync and Auto Tree are enabled. NOTE: Must only be used for debug purpose and not enabled on a normal running system SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 58 | P a g e 55.12.2 DECT System Settings Description of DECT Settings for Specific Base units is as follows:

Screenshot PARAMETER DECT SYSTEM RFPI DEFAULT VALUES Not able ALLOW MULTI PRIMARY:

Disabled AUTO CREATE MULTI PRIMARY:

Disabled Enabled AUTO CONFIGURE DECT SYNC SOURCE TREE DESCRIPTION This is a radio network identity accessed by all Base units in a specific multi-cell system. It composed of 5 octets. It is actually 5 different variables combined together. RFPI Format: XX XX XX XX XX (where XX are HEX values) This feature is used for multi-location setups. Allows two or more primary in the same system. The two cells will be unsynchronized, and handover will not be possible. Auto Configure DECT sync source tree must be enabled for this feature to also be enabled By enabled the system can generate cells in case a base goes into faulty mode. Two cells will only be generated in case no radio connection between the two cells is present. In order to recover the full system after establishing of the faulty base, the system must be rebooted. Allow multi primary must be enabled for this feature to also be enabled. Enable this to allow the system to automatically synchronize the multi-cell chain/tree. NOTE: Must be enabled in order to allow a new primary to recover in case the original primary goes into faulty mode. NOTE: To run with a system with two separate primaries in two locations Allow multi primary and Auto configure DECT sync source tree must be enabled. To add the second primary the slave must manually be configured as primary. Alternatively, the Auto create multi primary must be enabled. SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 59 | P a g e 55.12.3 Base System Settings Description of SIP Settings for Specific Base units is as follows:

Screenshot Parameter NUMBER OF SIP ACCOUNTS BEFORE DISTRIBUTED LOAD Default Values 8 Disabled SIP SERVER SUPPORT FOR MULTIPLE REGISTRATIONS PER ACCOUNT 50/3 SYSTEM COMBINATION

(NUMBER OF BASE STATIONS/REPEATERS PER BASE STATION):

Description The maximum number of handsets or SIP end nodes that are permitted to perform location registration on a specific Base unit before load is distributed to other base units. The parameter can be used to optimize the handset distribution among visible base stations. Note: A maximum of 8 simultaneous calls can be routed through each Base unit in a multi-cell setup. Permitted Input: Positive Integers (e.g. 6) Disable this option so it is possible to use same extension (i.e. SIP Account) on multiple phones (SIP end nodes). These phones will ring simultaneously for all incoming calls. When a phone (from a SIP account group) initiates a handover from Base X to Base Y, this phone will de-register from Base X, and register to Base Y after a call. Permitted Input:

Disabled: No SIP de-registration will be made when a handset roams to another base station Enabled: The old SIP registration will be deleted with a SIP Deregistration, when a handset roams to another base station Select between basic base configurations. 50/3 : 50 bases and 3 repeaters 127/1 : 127 bases and 1 repeater 254/0 : 254 bases and 0 repeater The configuration cannot be modified after a system is established. The configuration must be set during first multicell configuration. SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 60 | P a g e 55.12.4 Base Station Group The Base station group list various parameter settings for base stations including chain level information. Screenshot:

PARAMETERS ID RPN VERSION MAC ADDRESS IP STATUS DECT SYNC SOURCE DECT PROPERTY DESCRIPTION Base unit identity in the chained network. Permitted Output: Positive Integers The Radio Fixed Part Number is an 8-bit DECT cell identity allocated by the installer. The allocated RPN within the SME must be geographically unique. Permitted Output: 0 to 255 (DEC) OR 0x00 to 0xFF (HEX) Base station current firmware version. Permitted Output: positive Integers with dot (e.g. 273.1) Contains the hardware Ethernet MAC address of the base station. It varies from Base station to Base stations. Current Base station behavior in the SME network. Possible Outputs Connected: The relevant Base station(s) is online in the network Connection Loss: Base station unexpectedly lost connection to network This Unit: Current Base station whose http Web Interface is currently being accessed With setting Auto configure DECT sync source tree set to Enable, this three will automatically be generated. If manual configured the administrator should choose the relevant multi cell chain level its wants a specific Base unit be placed. Maximum number of multi-cell chain levels is 24. Format of the selection: AAAAAxx: RPNyy (-zz dBm) AAAAA: indication of sync. source for the base. Can be Primary or Level xx xx: Sync. source base sync. level yy: Sync. source base RPN zz: RSSI level of sync. source base seen from the actual base

(Any) RPN: When a base is not synchronized to another base. State after reboot of chain. Base station characteristics in connection to the current multi cell network. Possible Output(s) Primary: Main Base station unto which all other nodes in the chain synchronizes to. Locked: The Base unit is currently synchronized and locked to the master Base unit. Searching: Base unit in the process of locating to a Master/slave as specified in Dect sync source Free Running: A locked Base unit that suddenly lost synchronization to the Master. Unknown: No current connection information from specific Base unit SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 61 | P a g e Assisted lock: Base has lost DECT sync. source and Ethernet are used for synchronization Sync. Lost: Handset has an active DECT connection with the base. But the base has lost DECT sync. source connection. The base will stay working as long as the call is active and will go into searching mode when call is stopped. Name from management settings. BASE STATION NAME 55.12.5 DECT Chain Below the Base Group Table is the DECT Chain tree. The DECT Chain tree is a graphical presentation of the Base Group table levels and connections. Repeaters are shown with green highlight. Screenshot: DECT Chain tree of above configuration Screenshot: Example of part of DECT Chain tree with repeaters SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 62 | P a g e Screenshot: Example of part of DECT Chain tree with units in Base Group but not in tree by various reasons. When a base or repeater has not joined the tree, it will be shown with read background below the tree. 55.12.6 RTX8660 -RTX8663 Mixed mode RTX8663 base station can be added to existing systems using RTX8660 base station. Because the RTX8663 have more powerful hardware and additional features, there will be some limitations. A system running mixed mode, is limited to RTX8660 features. NOTE: LAN SYNC will not work in mixed mode. The system will display a warning message on the Home/Status page. Screenshot:

SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 63 | P a g e 5.13 LAN SYNC NOTE: To join one Base Station in a dual-cell system, you need to have one handset added to the system. For details and Step-

by-Step guide to dual cell, please see Appendix In this section, we describe the different parameters available in the dual-cell configurations menu. 55.13.1 Settings for Base Unit Description of Settings for Specific Base units is as follows:

Screenshot:

PARAMETERS MULTICAST IP ADDRESS DEFAULT VALUES 224.0.1.129 MULTICAST PORT 319 DOMAIN NUMBER ALTERNATIVE DOMAIN NUMBER 0 64 MULTI CELL DEBUG MODE None DESCRIPTION IP address of the multicast group. The IP address must start with 224.0.xx.xx this cannot be changed. To be compliant with IEEE1588, this port must be default value. Before setup, make sure no other devise uses the given IP. NOTE: this should only be changed in case other IEEE1588 equipment is on the network and using this specific IP address. Define the port that the system must communicate on To be compliant with IEEE1588, this port must be default value. NOTE: this should only be changed in case other IEEE1588 equipment is on the network and using this specific port. Domain number is used to set what domain this specific base station belongs to. Valid input: 0-127 Alternative domain is only used in case the primary sync source from the main domain fails, this the base station will sync with the alternative domain. Must NOT have same value as domain number. Valid input: 0-127 Enable this feature, if you want the system to catalogue low level multi-cell debug information or traces. Options:

Data Sync: Writes header information for all packets received and sent to be used to debug any special issues. Generates LOTS of SysLog signaling and is only recommended to enable shortly when debugging. Auto Tree: Writes states and data related to the Auto Tree Configuration feature. Both: Both Data Sync and Auto Tree are enabled. IEEE1588 Debug: Writes IEEE1588 debug information NOTE: Must only be used for debug purpose and not enabled on a normal running system SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 64 | P a g e 55.13.2 Base station group The Base station group list various parameter settings for base stations. Screenshot:

PARAMETERS ID STATUS PREFERED ROLE CURRENT ROLE SYNC SOURCE ALT. SYNC SOURCE NWK JITTER

[MS]

(MIN/AVG/MAX) MWK DELAY

[MS]

(MIN/AVG/MAX) IP STATUS BASE STATION NAME DESCRIPTION Base unit identity in the chained network. Permitted Output: Positive Integers Base station characteristics in connection to the current multi cell network. Possible Output(s) Primary: Main Base station into which all other nodes in the chain synchronizes to. Locked: The Base unit is currently synchronized and locked to the master Base unit. Searching: Base unit in the process of locating a Master/slave as specified in DECT sync source Free Running: IEEE master is found, and is DECT synchronizing Disabled: Disable this base station from the chain Primary: The base station that is used for main sync, it is possible to select more than one base station as primary. NOTE: It is recommended to use base stations that is closest to the backbone as primary Secondary: Base stations that never will be selected as primary. Automatic: System finds primary sync source Alt. Primary: Backup for primary base station in case it fails. The current role of the base station Shows what base station this specific base station is synchronized with Alternative sync source in case main sync source fails Measures how the IEEE1588 packets are received, the lower the Jitter is the better Measures the time it takes an IEEE packet to travel from primary to Slave base station in ms. Current Base station behavior in the SME network. Possible Outputs Connected: The relevant Base station(s) is online in the network Connection Loss: Base station unexpectedly lost connection to network This Unit: Current Base station whose http Web Interface is currently being accessed Name from management settings. 5.13.3 This unit debug Screenshot:

SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 65 | P a g e Debug information is used only by RTX to debug IEEE1588 network issues. In case debug is needed, sent this information to RTX support team. 5.14 Repeaters Within this section we describe the repeater parameters, and how to operate the repeater. 55.14.1 Add repeater In order to add a repeater to the system, select Add Repeater Screenshot Thereafter the following window with the specific parameters will be visible Screenshot PARAMETERS NAME DECT SYNC MODE DEFAULT VALUES Empty Local Automatical DESCRIPTION Repeater name. If no name specified, the field will be empty Manually: User controlled by manually assign Repeater RPN and DECT sync source RPN SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 66 | P a g e Local Automatical: Repeater controlled by auto detects best base signal and auto assign RPN. 5.14.1.1 Manually If the mode is chosen to be Manually, the assigned parameters Repeater RPN and DECT sync source RPN must be selected by the drop-down menu. Screenshot After saving the configurations above, the repeater will be visible on the main Repeaters menu with the following parameters:

Screenshot PARAMETERS IDX RPN DESCRIPTION System counter 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) DUAL 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) Name and IPEI number of the repeater DECT Sync mode Manually or Automatic State of the repeater Enabled/Disabled Firmware version How many percentages of the firmware is loaded / Off if no firmware is being loaded NAME/IPEI DECT SYNC MODE STATE FW INFO FWU PROGRESS Good practice when adding repeaters to a Dual Cell system is to use manually registration, because then you can control what base station the repeater(s) connects to. 5.14.1.2 Local Automatical Repeater controlled by auto detects best base signal and auto assign RPN. The RPN and DECT sync source are greyed out. SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 67 | P a g e Screenshot The repeater RPN is dynamic assigned in base RPN range. With local automagical mode repeater on repeater (chain) is not supported. 55.14.2 Register Repeater Adding a repeater makes it possible to register the repeater. Registration is made by selecting the repeater via the checkbox 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 including handset registration. 5.14.3 Repeaters list Screenshot The number of repeaters allowed on each base station is mentioned above in 5.14.1.1. System combination: 50/3 127/1 -254/0. If the system combination is set to 127/1 or 254/0 you can still register more than one repeater, but it will not get a DECT Sync source and have no function. Example:

System combination 50/3:

Base stations are named RPN00 RPN04 RPN08. Etc. jumping 4 numbers each time (HEX numbers) Repeaters connect to base station RPN00 will be called RPN01 RPN02 RPN03 (HEX numbers) Repeaters connect to base station RPN04 will be called RPN05 RPN06 RPN07 (HEX numbers) SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 68 | P a g e Etc. System combination 127/1:

Base stations are named RPN00 RPN02 RPN04. Etc. jumping 2 numbers each time (HEX numbers) Repeaters connect to base station RPN00 will be called RPN01 (HEX numbers) Repeaters connect to base station RPN02 will be called RPN05 (HEX numbers) Etc. System combination 254/0:

Repeater registration not possible. PARAMETERS IDX RPN NAME/IPEI DECT SYNC SOURCE DESCRIPTION Repeater unit identity in the chained network. Permitted Output: Positive Integers The Radio Fixed Part Number is an 8-bit DECT cell identity allocated by the installer. The allocated RPN within the SME must be geographically unique. Permitted Output: 0 to 255 (DEC) OR 0x00 to 0xFF (HEX) Contains the name and the unique DECT serial number of the repeater. If name is not given the field will be empty. The dual 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 STATE FW INFO FWU PROGRESS 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 Present@unit means connected to unit with RPN yy Firmware version 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. For detailed description on how to operate repeaters please see Repeater HOW-TO guide. Link is found in Appendix. 5.15 Alarm In the Alarm Settings menu, it is controlled how an alarm appears on the handset. For example, if the handset detects Man Down, then it is defined in this menu what alarm signal this type of alarm will send out and if a pre-alarm shall be signaled etc. The Alarm is activated by a long press on the Alarm key (3 sec). Screenshot SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 69 | P a g e All configuration of the handset Alarm Settings is done from the base station. The concept is that on the Alarm page on the web server, eight different alarm profiles can be configured. Afterwards for each handset, it can be selected which of the configured alarm profiles, the given handset shall subscribe to. When this is done the selected alarm, profiles are sent to the handset. See section 5.3.4: Edit handset. PARAMETERS IDX PROFILE ALIAS DEFAULT VALUES Empty ALARM TYPE Disabled ALARM SIGNAL Call STOP ALARM FROM HANDSET TRIGGER DELAY Enabled 0 SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential DESCRIPTION Indicates the index number of a specific alarm. An alias or user-friendly name to help identify the different profiles when selecting which profiles to enable for the individual handsets. The type of alarm is dependent of what kind of event that has triggered the alarm on the handset. The type of alarms supported is handset related. RTX8632/RTX8633:

Alarm button RTX8830:

Alarm button Man Down No Movement Running Pull Cord Emergency Button Disabled The way the alarm is signaled as it received on the handset. Message: A text message to an alarm server. Call: An outgoing call to the specified emergency number. Beacon message: Sends a beacon to the alarm server which sends a message to the handset Enable/Disable the possibility to stop/cancel the alarm from the handset. The period from when the alarm has fired until the handset shows a pre-alarm warning. If set to 0, there will be no pre-alarm warning, and the alarm will be signaled immediately. The alarm algorithm typically needs about 6 sec. to detect e.g. man down etc. 70 | P a g e STOP PRE-

ALARM FROM HANDSET PRE-ALARM DELAY Enabled 0 HOWLING Disabled Enable/Disable the possibility to stop/cancel the pre-alarm from the handset. The period from the pre-alarm warning is shown until the actual alarm is signaled. The maximum value is 255. Enable/Disable if howling shall be started in the handset, when the alarm is signaled. If disabled, only the configured signal is sent (call or message). NOTE: The alarm feature is only available on some types of handsets (e.g. RTX8632, RTX8633 and RTX8830) After configuration, the handset must be rebooted. SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 71 | P a g e 55.15.1 Use of Emergency Alarms As described above, it can be configured if it shall be possible to stop an alarm from the handset. If the possibility to stop an alarm from the handset is disabled, it is ensured that an alarm is not stopped before someone at e.g. an emergency center has received the alarm and reacted upon it. The behavior of a handset when an alarm is sent depends on the configured Alarm Signal:

Call: When the Alarm Signal is configured as Call, the handset will make a call to the specified emergency number, and the alarm is considered stopped when the call is terminated. If it is not allowed to stop the alarm from the handset, it will not be possible to terminate the call from handset, and the alarm will be considered as stopped only when the remote end (e.g. the emergency center) terminates the call.

Message: When the Alarm Signal is configured as Message, the handset will send an alarm message to the specified alarm server, and enable auto answer mode. If Howling is enabled, the handset will also start the Howling tone. The alarm will not stop until a call is made, and since auto answer mode is enabled, the emergency center can make the call, and the person with the handset does not have to do anything to answer the call, it will answer automatically. Again, the alarm is considered stopped, when the call is terminated with the same restrictions as for the Call alarm signal. All type of alarms have the same priority. This means that once an alarm is active, it cannot be overruled by another alarm until the alarm has been stopped. However, if the alarm is not yet active, i.e. if it is in pre-alarm state and an alarm configured with no pre-alarm is fired, then the new alarm will become active and stop the pending alarm. Alarms with no pre-alarm are considered important, and there is no possibility to cancel them before they are sent, and therefore alarms with no pre-alarm, are given higher priority than alarms in pre-alarm state. The Emergency Button could be an example of an alarm which would be configured without pre-alarm. Thus, when the Emergency Button is pressed you want to be sure the alarm is sent. However, if another alarm was already in pre-alarm state, it could potentially be cancelled, and if the Emergency Button alarm was ignored in this case, no alarm would be sent. This is the reason alarms with no pre-alarm, are given higher priority than alarms in pre-alarm state. For detailed description on how to alarm please see Alarms HOW-TO guide. Link is found in Appendix. SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 72 | P a g e 5.16 Statistics The statistic feature is divided into five administrative web pages, which can be accessed from any base. 1. System 2. Calls 3. Repeater 4. DECT data 5. Call quality All five views have an embedded export function, which exports all data to comma separated file. By pressing the Clear button, all data in the full system is cleared. 55.16.1 System data The system data web is accessed 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 BASE STATION NAME OPERATION/DURATION D-H:M:S BUSY BUSY DURATION D-H:M:S SIP FAILED HANDSET REMOVED SEARCHING FREE RUNNING DECT SOURCE CHANGED DESCRIPTION Base IP address and base station name from management settings Operation is operation time for the base since last reboot. Duration is the operation time for the base since last reset of statistics, or firmware upgrade. Busy Count is the number of times the base has been busy. Busy duration is the total time a base has been busy for speech (8 or more calls active). Failed SIP registrations count the number of times a SIP registration has failed Handset removed count is the number of times a handset has been marked as removed Base searching is the number of times a base has been searching for its sync source Base free running is the number of times a base has been free running Number of times a base has changed sync source 5.16.2 Free Running explained First, state Free running is NOT an error state, but is a simple trigger state, indicating that some changes have to be made to ensure continuous DECT synchronization. SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 73 | P a g e The state Free running tells the application that the base has not received any synchronization data from its synchronization source base station in the last 10 seconds. The reason for this can be several:

1. The two bases are using the same DECT slots and can therefore not see each other. 2. Many simultaneous voice or data calls. 3. Suddenly change of environment (Closing a fire door) 4. Distortion of DECT frequency (around 1.8MHz) Either by other DECT systems or other equipment. When the Free running state is trigged, several recovery mechanisms are activated:

1. Move DECT slot to avoid using same DECT slot as its synchronization source base state. 2. Use information from all other base station, how they are seeing this base station in the DECT air. This is marked by changing to state Assisted lock The state Assisted lock can be stabile for a long time and normally change to state Locked again. The state Free Running can also change back to state Locked again. If the base is in state Free running and the synchronization source base station is not seen and no data is available for the assisted lock mechanism, the base station will change to a new state after 2 minutes:

1. 2. If the base station does NOT have any active calls, the base will change to state Searching. If the base station has an active call, this base will change to state Sync lost. After the call is released, the state will change to state Searching. 5.16.3 Call data The call data web is accessed 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 BASE STATION NAME OPERATION TIME/DURATION COUNT DROPPED DESCRIPTION Base IP address and base station name from management settings Total operation time for the base since last reboot or reset Duration is the time from data was cleared or system has been firmware upgraded. Counts number of calls on a base. Dropped calls are the number of active calls that were dropped. E.g. if a user has an active call and walks out of range, the call will be counted as a dropped call. An entry is stored in the syslog when a call is dropped. SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 74 | P a g e NO RESPONSE DURATION ACTIVE MAX ACTIVE CODECS HANDOVER ATTEMPT SUCCESS HANDOVER ATTEMPT FAILED AUDIO NOT DETECTED No response calls are the number of calls that have no response, e.g. if an 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. Call duration is total time that calls are active on the base. Active call shows how many active calls that are active on the base (Not active DECT calls, but active calls). On one base there can be up to 30 active calls. Maximum active calls are the maximum number of calls that has been active at the same time. Logging and count of used codec types on each call. Counts the number of successful handovers. Counts the number of failed handovers. Counts the number of times where audio connection was not established. 55.16.4 Repeater data Screenshot The table is organized with headline row, data pr. base rows and with last row containing the sum of all base parameters. SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 75 | P a g e DESCRIPTION Base IP address and base station name from management settings 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 Count is the number of times the repeater has been busy. Busy duration is the total time a repeater has been busy for speech (5 or more calls active). Maximum active calls are the maximum number of calls that have been active at the same time. Repeater searching is the number of times a repeater has been searching for its sync source In case the sync source is not present anymore the repeater will go into lock on another base or repeater and show recovery mode Number of times a repeater has changed sync source Number of wideband calls on repeaters Number of narrowband calls on repeaters PARAMETERS IDX/NAME OPERATION D-H:M:S BUSY BUSY DURATION D-H:M:S MAX ACTIVE SEARCHING RECOVERY DECT SOURCE CHANGED WIDE BAND NARROW BAND 55.16.5 DECT data The DECT data web is accessed by http://ip/DectStatistics.html and data is organized in a table as shown in below example. Screenshot PARAMETERS FREQUENCY SLOTX DESCRIPTION Number of the DECT slot frequency Number of connections that have been active on each frequency SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 76 | P a g e 55.16.6 Call quality The Call quality web is accessed by http://ip/CallQuality.html and the data is organized in a table as shown in below example. Screenshot PARAMETERS BASE STATION NAME TYPE CALL COUNT LOCAL/REMOTE SIDE JITTER[MS]

ROUND TRIP LATENCY [MS]

PACKET LOSS [%]

R-VALUE DESCRIPTION Base IP address and base station name from management settings Call:

Relay conn:

Count the number of calls Local:

Remote:

Measures how the RTP packets are received, the lower the Jitter is the better Measures the time it takes for RTP packets to reach it destination. Percentages of packets lost. A way to measure call quality, from 0-120 USER SATISFACTION LEVEL MOS R-Factor MAXIMUM USING G.711 4.4 93 VERY SATISFIED 4.3-5.0 90-100 SATISFIED 4.0-4.3 80-90 SOME USERS SATISFIED 3.6-4.0 70-80 MANY USERS DISSATISFIED 3.1-3.6 60-70 NEARLY ALL USERS DISSATISFIED 2.6-3.1 50-60 MOS-VALUE NOT RECOMMENDED 1.0-2.6 Less than 50 MOS measures subjective call quality for a call. MOS scores range from 1 for unacceptable, to 5 for excellent. VOIP calls often are in the 3.5 to 4.2 range See table above. SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 77 | P a g e 5.17 Generic Statistics The statistic feature is divided into five sections, which can be accessed from any base. 1. DECT Statistics 2. DECT Synchronization statistics 3. RTP Statistics 4. IP Stack Statistics 5. System Statistics By pressing the Expand all fields you can see statistics hour by hour and by pressing the Reset all statistics button all data in the full system is cleared. DEFAULT VALUES Vary Vary Vary DESCRIPTION Headline of the different statistics Vary for point to point Data from the last 24 hours PARAMETER PARAMETER VALUE 24 HR DATA Screenshot:

PARAMETERS TOTAL NUMBER OF DLC INSTANCE MAX CONCURRENT DLC INSTANCES CURRENT NUMBER OF DLC INSTANCES TOTAL NUMBER OF TIMES IN MAX DLC INSTANCES IN USE DESCRIPTION The lifetime total count of instantiated DLC instances. The lifetime highest concurrent count of instantiated DLC instances. The current count of instantiate DLC instances. The number of times we reach the currently highest count of DLC instances. SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 78 | P a g e TOTAL TIME SPEND IN MAX DLC INSTANCES IN USE AVERAGE FREQUENCY X USAGE THIS HOUR

(MAX 100 PER SLOT) AVERAGE EVEN SLOT USAGE THIS HOUR (MAX 100 PER SLOT) AVERAGE ODD SLOT USAGE THIS HOUR (MAX 100 PER SLOT) PERCENTAGE TIME OF X SLOTS USED THIS HOUR TOTAL CODEC USAGE (G.711A, G.711U, G.726, G.729) TOTAL CHO SUCCESS Total number of forced PP moves The time we have spent in the highest concurrent number of instantiated DLC instances. The average use of frequency number X. The value is 100 if the frequency is fully used by a slot in the measured time frame. The average use of even numbered slots. The average use of odd numbered slots. The percentual time that X number of DECT slots are used during the given hour

(compared to other slot counts). This shows what codec, that have been used. The number of times we instantiate RTP stream using either codec. The number of times connection handover is successfully made. The lifetime total count that this base forces PP moves. SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 79 | P a g e 55.17.1 DECT Synchronization Statistics DECT Synchronization statistics is related to this base station only. Screenshot:

PARAMETERS CURRENT SYNCHRONISATION STATE CURRENT SYNCHRONISATION CHAIN TIMESTAMP FOR LAST CHANGED SYNCHRONISATION CHAIN HOURLY NUMBER OF SYNCHRONISATION CHAIN CHANGES TOTAL NUMBER OF SYNCHRONISATION CHAIN CHANGES TIME IN SYNCHRONISATION STATE: MASTER TIME IN SYNCHRONISATION STATE: LOCKED TIME IN SYNCHRONISATION STATE: FREE RUNNING TIME IN SYNCHRONISATION STATE: LOCKED ASSISTED TIME IN SYNCHRONISATION STATE: SYNC LOST TIME IN SYNCHRONISATION STATE: SEARCHING TIME IN SYNCHRONISATION STATE: UNKNOWN DESCRIPTION The current DECT sync state (e.g. Master, Searching, Free Running, etc). The current DECT sync source Fp Id of this base. Timestamp of the last time this base changed DECT sync source. The number of times this base changed DECT sync source in the current hour. The lifetime total count of times this base changed DECT sync source. Time this hour where this base station has had the state Master Time this hour where this base station has had the state Locked Time this hour where this base station has had the state Alien Free Running Time this hour where this base station has been in lock assisted Time this hour where this base station has not been in Sync Time this hour where this base has been searching for its sync source Time this hour where this base station has not been in unknown state SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 80 | P a g e 55.17.2 RTP Statistics RTP statistics are related to this base station only. Screenshot:

DESCRIPTION The lifetime total count of instantiated RTP streams. The lifetime highest concurrent count of instantiated RTP streams. The time we have spent in the highest concurrent count of instantiated RTP streams. The current count of instantiated RTP streams. PARAMETERS TOTAL RTP CONNECTIONS

(INCLUDING CONNECTION TYPE INFORMATION, E.G. EXTERNAL, RELAY, RECORDING) MAX CONCURRENT RTP CONNECTIONS

(INCLUDING CONNECTION TYPE INFORMATION, E.G. EXTERNAL, RELAY, RECORDING) TOTAL TIME SPEND IN MAX RTP CONNECTIONS IN USE CURRENT RTP CONNECTIONS

(INCLUDING CONNECTION TYPE INFORMATION, E.G. EXTERNAL, RELAY, RECORDING) SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 81 | P a g e 55.17.3 IP - Stack statistics IP - Stack statistics is related to this base station only. Screenshot:

PARAMETERS TOTAL CONNECTIONS OPEN MAX CONCURRENT CONNECTIONS OPEN CURRENT CONNECTIONS OPEN TOTAL NUMBER OF TX MESSAGES TOTAL NUMBER OF RX MESSAGES TOTAL NUMBER OF TX ERRORS DESCRIPTION The lifetime total count of used sockets. The lifetime highest concurrent count of used sockets. The current count of used sockets. The lifetime total count of transmitted IP packets. The lifetime total count of received IP packets. The lifetime total count of errors occurred during IP packet transmission. 5.17.4 System Statistics System Statistics is related to this base station only. Screenshot:

PARAMETERS UP TIME CURRENT CPU LOAD CURRENT HEAP USAGE MAX HEAP USAGE

(%) MAIL QUEUE ROS_SYSLOG MAIL QUEUE ROS_X DESCRIPTION The time the base has been running consecutively. The current load percentage of CPU. This is refreshed once every 5 seconds. The current use of heap in Bytes. The peak usage of heap in percentage. Size of internal mail queue for syslog. Size of internal mail queue. SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 82 | P a g e 5.18 Diagnostics This page provides information about the Ethernet connection to each base station and Extension. 55.18.1 Base Stations Screenshot PARAMETERS BASE STATION NAME ACTIVE DECT EXT

(MM/CISS/CCOUT/CCIN) ACTIVE DECT REP

(MM/CISS/CCOUT/CCIN) ACTIVE RTP

(LCL/RX BC) ACTIVE RELAY RTP

(LCL/REMOTE) LATENCY [MS]

(AVG.MIN/AVERAGE/AVG.MAX) DESCRIPTION Base IP address and base station name from management settings Number of active DECT MAC connections to extensions in the different base stations. Types of connection is (mm/Ciss/CcOut/CcIn) Number of active DECT MAC connections to repeaters in the different base stations. Types of connection is (mm/Ciss/CcOut/CcIn) Number of active RTP Streams used. Types of stream (Local RTP stream/Broadcast Receive RTP stream) Number of active RTP Relay Streams used. Types of stream (Local RTP Relay stream/Remote RTP Relay stream) Ping latency between base station performed by base index 0. Average Minimum delay/Average/Average Maximum delay) 5.18.2 Extensions Screenshot PARAMETERS IDX NO OF HS RESTARTS LAST HS RESTART

(DD/MM/YYYY HH:MM:SS) DESCRIPTION Extension Index number Number of times that the Handset has restarted Date and time of the last time the Handset has restarted SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 83 | P a g e 55.18.3 Logging The Diagnostics/Logging page allows you to collect system diagnostics information into a zip file. Screenshot PARAMETERS RSX INTERNAL TRACING PCAP INTERNAL TRACING TRACING PACKETS TO/FROM THIS BASE

(EXCEPT AUDIO) TRACE AUDIO PACKETS TO/FROM THIS BASE TRACE RECEIVED BROADCAST PACKETS TRACE RECEIVED PACKET WITH DESTINATION MAC BETWEEN TRACE RECEIVED ETHERTYPE TRACE RECEIVED IPV4 PROTOCOL TRACE RECEIVED TCP/UDP PORT INFO DESCRIPTION Enable/Disable. Only RTX engineers can debug the traces If selected, all Ethernet packets sent to/from the base stations MAC address are traced. Broadcast packets sent from the base are also being traced. If selected, RTP streams to/from the BS are traced. Audio packets are filtered by the port number used for RTP packets which is set on the web page If selected, all broadcast packets received by the BS are traced. If selected, each byte of the received destination MAC is checked if it is in the trace range If selected, the user can select 3 received Ethertypes to trace If selected, the user can select 3 received IPv4 protocols to trace If selected, the user can select 3 received TCP/UDP ports to trace. SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 84 | P a g e DOWNLOAD TRACES FROM Choose from which base stations to download the traces all of them or just the current one The following information is added to the zip file. 1: RSX trace (Good practice is to enable RSX internal tracing) 2: Syslog(s) 3: SIPLOG 4: Statistics 5: Home/Status page (HTML format) 6: Config file(s) 7: Error reason (entered by the user) 8: Requested BS(s) information about what base stations is in the trace) SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 85 | P a g e 5.19 Configuration This page provides non-editable information showing the native format of entire SME 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:

Using the SME VoIP Configuration interface to make changes. Each page of the web interface is a template for which the user can customize settings in the configuration file.

Retrieving the relevant configuration file from the TFTP and modify and enter new changes. This should be done with an expert network administrator.

Navigate to the settings page of the VoIP SME 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. An example of contents of settings is as follows:

~RELEASE=BEATUS_FP_V0400_B0001

~System Mode=51/51

%GMT_TIME_ZONE%:0x06

%COUNTRY_VARIANT_ID%:0x12

%COUNTRY_REGION_ID%:0x00

%TIMEZONE_BY_COUNTRY_REGION%:0x01

%DST_BY_COUNTRY_REGION%:0x01

%DST_ENABLE%:0x02

%DST_FIXED_DAY_ENABLE%:0x00

%DST_START_MONTH%:0x03

%DST_START_DATE%:0x00 For detailed description on how to use provisioning please see Provisioning HOW-TO guide. Link is found in Appendix. SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 86 | P a g e 5.20 Sys log This page shows live feed of system level messages of the current base station. The messages the administrator sees here depend on what is configured at the Management settings. The Debug logs can show only Boot Log or Everything that is all system logs including boot logs. The Debug log is saved in the file format <Time_Stamp>b.log in a relevant location in the TFTP server as specified in the upload script. A sample of debug logs is as follows:

0101000013 [N](01):DHCP Enabled 0101000013 [N](01):IP Address: 192.168.10.101 0101000013 [N](01):Gateway Address: 192.168.10.254 0101000013 [N](01):Subnet Mask: 255.255.255.0 0101000013 [N](01):TFTP boot server not set by DHCP. Using Static. 0101000013 [N](01):DHCP Discover completed 0101000013 [N](01):Time Server: 192.168.10.11 0101000013 [N](01):Boot server: 10.10.104.63 path: Config/ Type: TFTP 0101000013 [N](01):RemCfg: Download request of Config/00087b077cd9.cfg from 10.10.104.63 using TFTP 0101000014 [N](01):accept called from task 7 0101000014 [N](01):TrelAccept success [4]. Listening on port 10010 0101000019 [N](01):RemCfg: Download request of Config/00087b077cd9.cfg from 10.10.104.63 using TFTP 0101000019 [W](01):Load of Config/00087b077cd9.cfg from 10.10.104.63 failed To dump the log simply copy and paste the full contents. 5.21 SIP Logs This page shows SIP server related messages that are logged during the operation of the SME 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:Ext003@192.168.10.10:5080>;tag=3o5l314 To: <sip:Ext003@192.168.10.10:5080>

Call-ID: p9st.zzrfff66.ah8 CSeq: 6562 REGISTER Contact: <sip:Ext003@192.168.10.101:5063>

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 paste the full contents. SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 87 | P a g e Appendix How-To setup a Dual-Cell System This chapter we describe how to setup a dual-cell system, add and synchronize one or two base stations to the network. NOTE: It is possible to have RTX8660 and RTX8663 in the same chain. Adding Base stations Here are the recommended steps to add Base stations to network:

STEP 1:

Connect the Base station to a private network via standard Ethernet cable. STEP 2:

Use one of the two methods to determine the base station IP address (see chapter 3.5 for more details). 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 on the handset. If not, use the second method by typing the ipdect address I the browser, followed by the MAC address of the Base station. STEP 3:

Open a browser on the computer and type in the IP address of the base. Press Enter to access the base Login to the base station. The default input for Username/Password is: admin/admin. Once you have been authenticated, the browser will display the front end of the SME Configuration Interface. The front end will show relevant information of the base station. Screenshot SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 88 | P a g e CCountry and Time Server Setup STEP 4:

Navigate to the Country page and configure its country and time settings. Use the PC time feature or enter the relevant NTP server address and press the Save and Reboot button. Make sure there is contact to the Time server otherwise the Dual-cell feature will not work. You can verify whether the Time server is reachable by rebooting the base station and verifying that the correct Time Server IP address is still in place. Screenshot SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 89 | P a g e SSIP Server (or PBX Server) Setup STEP 5:

Create the relevant SIP server (or PBX Server) information in the system. Each service provider/customer should refer to a SIP server vendor on how to setup SIP servers. a. Click the link Server at the left-hand column of home page. This is the place where you can add your SIP server for base station use. b. Next, from the Server page, click on the Add Server URL and enter the relevant SIP server information (an example is shown below). c. Choose Disabled on NAT adaption parameter if NAT function of the SIP aware router is not enabled. Enter the relevant parameters based on the description in the table below. Click Save. Screenshot SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 90 | P a g e AAdd an extension and handset STEP 6:

Add an extension before you move to the Dual Cell page. Go to Extensions Add Extension. Fill in the extension data and check the checkbox for adding a new handset. If you wish to replace an already existing handset check the box with the relevant IPEI number of the handset. Press Save. Screenshot You will now see the extension on the extension page. Usually you do not need to fully register the extension, but since the handset is checked in the above guide, the next step is to add the extension. Click on the Handset link and check the box of the handset that is not registered and select Register handset. Thereafter, go to the main menu of the handset and go to Connectivity Register type the password of 0000 and after a minute, the handset should be registered. For more details, please go to Handset guide.. Screenshot STEP 7:

SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 91 | P a g e Click on Dual-cell URL link in the SME VoIP Configuration to view the current Dual cell settings status of the current base station. Brand new base stations have Dual-cell system feature disabled by default Screenshot STEP 8:

Next, the system administrator needs to create and Enable Dual Settings profile for the current base station. On the Dual-cell settings Page, choose Enable option from the drop-down menu of the Dual-cell system parameter. Enable the Dual-cell debug option if the system administrator wants some Dual-cell related logs to be catalogued by the system. Screenshot STEP 9:

On the same Dual-cell Settings page > Enter the relevant values for System chain ID respectively. The System chain ID is a geographically unique DECT cell identity allocated to bridge several base stations together in a chain. An example is 55555. SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 92 | P a g e NOTE: Do NOT use a chain ID similar to an extension. Screenshot Click on Save button to keep modified changes of dual-cell settings into the base station. Screenshot NOTE: That after you save the entries, the System information changes status to Unchained Allowed to Join as Primary NOTE: The Dual-cell data synchronization ONLY works when the relevant Time Server is set in the system before Server/Subscriber profile is added or created. Refer to STEP 4. IMPORTANT: Base stations must be rebooted after the time server has been set. STEP 10:

Logon to the base station that you want to connect to the Dual-cell system. STEP 11:

Navigate to the Dual-cell page and Enable Dual-cell system and enter the Chain ID that you used on the first base Station. STEP 12:

Press Save and Reboot IMPORTANT: It takes up to 5 minutes (synchronization time) to add a new base station to a Dua-cell System. Screenshot SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 93 | P a g e Appendix - Adding Extensions This section describes how to register the wireless handset to a Dual-cell system. NOTE: Minimum one server must be registered to the base (system), otherwise a handset cannot be registered to the system. Please see chapter 0. STEP 1:

SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 94 | P a g e Login to a base station. STEP 2:

Select Extensions URL and click Add extension link STEP 3:

Fill out the form and click Save. In the example below, we add the extension 529 and this SIP account got the same number as Authentication User Name, Password and Display Name. Screenshot STEP 4:

On the right-hand side there is a Select Handset table. In order to activate the registration of a handset to the current extension, check the box on the Add Handset parameter and click Save Screenshot STEP 5:

On the Extensions and Handset menu click the Handset link for registering a handset. Afterwards, check the box on the extension you wish to add the handset and click Register handset(s). The registration will be open for 5 min. SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 95 | P a g e STEP 6:

To start the registration procedure on the handset, go to the Connectivity menu and select Register. Select an Empty parameter and type in the PIN code of 0000. After a while the handset is registered, and the idle display is shown. STEP 6:

To check if the handset has been registered, go to the Extensions and Handset menu and verify that the unique IPEI of the handset is displayed in the IPEI column. NOTE: 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) is not displayed immediately on the web page. Screenshot SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 96 | P a g e STEP 7:

Verify the SIP registration by SIP State in left of the IPEI. NOTE: The web page must be manually updated by pressing F5 to see that the handset is SIP registered; otherwise the handset SIP state is not displayed immediately on the web page. Repeat STEP 2-7 for each handset you want to register. Appendix - Firmware Upgrade Procedure This step-by-step chapter describes how to upgrade or downgrade base station(s) and/or handset(s) / repeater (s) to the relevant firmware provided by RTX. Network Dimensioning In principle, several hardware and software components should be available or be satisfied before base station/handset update can be possible. The minimum hardware and software components that are required to be able update via TFTP include the following (but not limited to):

Handsets

Base stations

TFTP Server (Several Windows and Linux applications are available)

DHCP Server (Several Windows and Linux applications are available)

Workstation (e.g. Normal terminal or PC)

Any standard browser (e.g. Firefox)

Public/Private Network SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 97 | P a g e TFTP Configuration This section illustrates TFTP Server configuration using SolarWinds vendor TFTP Server. Create the following relevant folders as shown in the snap shots and choose defaults settings for the remaining options and save. SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 98 | P a g e NOTE: If TFTP server timeout settings are too short firmware upgrade might not complete. Recommended time out setting is more than 3 seconds. Create Firmware Directories The admin from the service providers side must create the relevant firmware directory in the server where both old and new firmware(s) can be placed in it. (See the STEP above) BBase:

On the TFTP server root, create directorys as in screenshot. Copy Base station firmware to the named directory. IMPORTANT: The 8663 directory name cannot be changed. SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 99 | P a g e HHandsets/Repeaters:

On the TFTP server root, create directory 8430 or 8630 or 8830 or 8930 or 4024 depending on type. Copy handset/repeater firmware to the named directory of each model. IMPORTANT: The 8430, 8630,8830 and 8930 directory names cannot be changed. Handset Firmware Update Settings Scroll down and Click on Firmware Update URL link in the SME VoIP Configuration Interface to view the Firmware Update Settings page. Screenshot Type IP address and firmware path followed by save. For Http download the firmware update server settings must be entered as follows:

Screenshot SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 100 | P a g e Handset(s) and Repeater Firmware Upgrade On the Firmware Update Settings page enter the relevant handset/repeater firmware for each type and Branch name (e.g. 440 for v440 for Required Version) and (e.g.01 for Branch 01 for Required Branch) to upgrade or downgrade > press Save button to initialize the process of updating all handsets. Screenshot NOTE: To disable handset/repeater firmware process type version 0 in the required version field, followed by the save button. It is recommended to use version 0 after all units are upgraded. NOTE: For handset TFTP/HTTP download only one handset type can be downloaded at the same time. In case two handset models are defined for fwu at the same time fwu will fail. 55.21.1 Monitor handset firmware upgrade Handset firmware upgrade status is monitored on the handset extensions page, FWU Process Colum. If the status says Off it means that the Required Version and Branch is set to 0 as it should be unless youre in process of updating/downgrading the firmware. The firmware Upgrade/Downgrade process has 6 states:

Initializing In progress (% from 0-100)

Verifying (% 0-100)

Waiting for charger (The handset must be placed in charger and NOT removed until it reboots)

Off Complete Screenshot SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 101 | P a g e Handset firmware update time from start to complete takes 20- 40 minutes. MMonitor Repeater firmware upgrade Repeater firmware upgrade status is monitored on the Repeater page, right column. Repeater firmware upgrade time from start to complete takes minimum 20 minutes. Verification of Firmware Upgrade The firmware upgrade is confirmed by the FWU Progress status in the FWU Colum 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 selecting Settings > Status. This menu will list information regarding Base station and Handset firmware versions. 5 Base Station(s) Firmware Upgrade On the Firmware Update Page Base stations are updated in the same way as handsets and other extensions. After entering Required Version and Required Branch choose Save/Start Update button > select OK button from the dialog window to start the update/downgrade procedure. The relevant base station(s) will automatically reboot and retrieve the firmware specified from the server and update itself accordingly. SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 102 | P a g e The base firmware update behavior is: Base will fetch the fwu file for approximately 3 minutes, then reboot and start flashing the LED - indicated by LED fast flashing for approximately 3 minutes and reboots in new version. NOTE: All on-going voice calls are dropped from the base station(s) immediately after the firmware update procedure starts. BBase firmware confirmation Base station firmware version status in a dual-cell environment can be seen in the dual-cell base station group overview page, column 4 (Version). Screenshot Verification of Firmware Upgrade If the firmware Upgrade/Downgrade does not start, you can check the syslog to see if the path is right. Syslog information when Management Syslog level is set to Debug

[ FWU Downloading File tftp://10.1.24.103/FwuPath/8663/8663_v0440_b0001.fwu]

[ Base FWU started]

[ Base FWU ended with exit code 2101 (NE_FILE_TRANSFER_EOF): End of file]

This is the path when the base station expects to find the firmware: tftp://10.1.24.103/FwuPath/8663/8663_v0440_b0001.fwu Check if the firmware file is in the correct directory. SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 103 | P a g e Appendix Multiline Feature This section describes how to register the wireless handset to a system with active multiline feature. One handset will be able to support up to 4 lines (4 different SIP accounts) ... A handset only supports 2 call appearances. The limitation of maximum 1000 terminals in the system is maintained, and the maximum number of SIP registrations, one base station can handle, is maintained. With 4 lines pr. terminal maximum number of terminals registered in a system are 250. With 1-line pr. terminal maximum number of terminals registered in a system are 1000. Still the limitation of 30 SIP accounts registered pr. base is maintained. With 4 lines (SIP accounts) pr. terminal maximum number of terminals registered pr. base is 7. The 4 SIP accounts pr. terminal follow the location of the terminal similar. With multiline feature enabled 200 contacts in contact list is possible. How to setup Multiline. Step 1:

Register handset as described in chapter 7 (Appendix Adding Extensions). Step 2:

Add a multiline to a handset by creating a new extension but instead for New Handset select the existing handset that you want to add the multiline to. (in this case Handset Idx 1) Step 3:

The extension will now show in the extension list with a new Idx , but the same IPEI, whereas on the handset list, the Idx will be the same and the extension column will show the available extensions for the current handset. NB: the handset must be rebooted for the changes to take effect. SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 104 | P a g e The Extension will now have two numbers 522 and 529. When making call the user can chose which line to call from. Simply enter the number to call and press line. Select the desired line and press the green Off-hook button to place the call from this line. SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 105 | P a g e Appendix - Functionality Overview So far, we have setup our SME VoIP system. Next, in this chapter we list what features and functionalities are available in the system. The SME VOIP 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 SME VOIP network functionalities is:

Outgoing/incoming voice call management: The SME VOIP system can provide multiple priority user classes. Further, up to

3 repeaters can be linked to a Base-station. Internal handover: User locations are reported to SIP Server to provide differentiated services and tariff management. Within a DECT traffic area, established calls can seamlessly be handover between Base-stations using connection handover procedures. Security: The RTX SME VOIP system also supports robust security functionalities for Base-stations. Most security2 functionalities are intrinsically woven into the SME VOIP network structure so that network connections can be encrypted, and terminal authentication can be performed. Gateway Interface CONNECTOR INTERFACES POWER LAN INTERFACE INTERNET PROTOCOL:

KEYS LED INDICATOR RF FREQUENCY BANDS OUTPUT POWER SENSITIVITY ANTENNA SOFTWARE UPGRADE DOWNLOADABLE Version 1:

Connector: Ethernet PoE (Ethernet adaptor for normal power) IEEE 802.3af: Power class 2 (3.84 6.49W) DC plug: 5VDC 2A Version 2:

Connector: Ethernet PoE (Ethernet adaptor for normal power) IEEE 802.3af: Power class 2 (3.84 6.49W) Standard : 10BASE-T(IEEE 802.3 100Mbps) Connector: RJ45 8/8 IPv4 IPv6

1 x Reset key One Status LED (red, green, orange) 1880 1895 MHz (Taiwan) 1880 1900 MHz (EMEA, AUS) 1910 1920 MHz (Brazil) 1910 1930 MHz (LATAM, Chile) 1920 1930 MHz (USA, Canada) These are software settings and need to be set when packed in factory. 250 mW (EMEA, Taiwan, Brazil, LATAM) 160 mW (Chile, Australia) 140 mW (Canada, USA)

-92 dBm Two antennas for diversity Remote firmware update HTTP/HTTPS/TFTP 2 With active security with authentication 4 channels are supported SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 106 | P a g e Detail Feature List CODECs G.711 A-LAW & U-LAW G.722 G.726 G.729 SIP RFC2327 RFC2396 RFC2833 RFC2976 RFC3261 RFC3262 RFC3263 RFC2543 RFC3264 RFC3265 RCF3326 RFC3311 RFC3325 RFC3420 RFC3326 RFC3489 RFC3515 RFC3550 RFC3581 RFC3665 RFC3842 RFC3891 RFC3892 RFC3960 RFC4475 SIPS IN-BAND DTMF SRTP WEB SERVER OTHER FEATURES BOOT TIME QUALITY OF SERVICE IP QUALITY AUTOMATIC DST TONE SCHEME Yes/Yes Yes Yes, 32 Kbps A/AB (including VAD) maximum 4 simultaneously call. Note: Only with additional module, this is an extra option that requires a board connector mounted in Gateway. Per default not mounted. SDP: Session Description Protocol Uniform Resource Identifiers (URI): Generic Syntax In-Band DTMF/Out of band DTMF support The SIP INFO method SIP 2.0 Reliability of Provisional Responses in the Session Initiation Protocol (PRACK) Locating SIP Servers (DNS SRV, redundant server support) Session Initiation Protocol (HOLD Option) Offer/Answer Model with SDP Specific Event Notification The Reason Header Field for the Session Initiation Protocol The Session Initiation Protocol UPDATE Method P-Asserted Identity Internet Media Type message/sipfrag The Reason Header Field for the Session Initiation Protocol (SIP) STUN REFER: Call Transfer RTP: A Transport Protocol for Real-Time Application Rport Basic Call Flow Examples Message Waiting Indication Replace header support The Session Initiation Protocol (SIP) Referred-By Mechanism Early Media and Ringing Tone Generation in the Session Initiation Protocol (SIP) Session Initiation Protocol (SIP) Torture Test Messages Secure SIP RFC2833 With authentication 16 calls incl. relays are supported. Embedded web server HTTP Max 60 seconds Type of Service (ToS) including DiffServ Tagging, and QoS per IEEE 802.1p/q Warning Network outage, VoIP service outage Adaptive Jitter Buffer support Yes Tone Scheme is based on CADENCE configuration of the following tones:

Dial Tone Outside Dial Tone Prompt Tone Busy Tone Reorder Tone Off Hook Warning Tone Ring Back Tone SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 107 | P a g e Yes (but PKI deployment/enrollment is not supported) Call Waiting Tone Confirm Tone MWI Dial Tone CFwd Dial Tone Holding Tone Conference Tone Secure Call Indication Tone Page Tone Mute Tone Unmute Tone System Beep Call Pickup Tone Yes (same as RTX MultiLine feature) Supported through provisioning Yes The 2nd base station in a dual cell setup must support automatic installation and not require any web-interface installation. Yes Hardware ready, software not included VLAN (802.1p/q) Yes Yes For secure connections (SCA-256) For configuration download. For configuration download. Yes (AES 128 and 256 key) For secure configuration download Yes, for secure configuration download (Digest authentication using MD5, SHA128, SHA256) Yes/Yes/Yes For internet clock synchronization 66, 160, 159, 150, 60, 43, 125 Yes No SIP TOS and RTP TOS should be supported Device should be able to function normally behind the firewall Device should follow DNS SRV priorities, timeout and A-Rec Device must support TR069 In case of SIP registration server failover, device should be able to register to alternate server Connectionless handover Yes Yes Support for Root CA Certificate Download A new field will be added in the provisioning tab to upload/add new root CA. Upload protocols: http, https, tftp SUPPORT FOR ROOT CA CERTIFICATE DOWNLOAD SHARE NUMBERING

*CODES VOIP CALLER PIN AND ASSOCIATED DIAL PLAN ZERO TOUCH INSTALLATION ETHERNET FEATURES IPV4 IPV6 VLAN DHCP SUPPORT STATIC IP TLS 1.2 TFTP HTTP ENCRYPTED TFTP CONFIG FILES HTTP CLIENT HTTPS TCP/IP/UDP SNTP DHCP OPTION DNS SERVER LANSYNC IEEE1588 TOS SUPPORT NAT TRAVERSAL DNS SRV TR069 REGISTRATION FAILOVER AND FALLBACK DECT DECT CAP CAT-IQ V1.0 DECT ULE ROOT CA POWER SUPPLY FEATURES LED INDICATOR No SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 108 | P a g e COLORS SEPARATE ADAPTOR SWITCH MODE ADAPTOR POWER SPEC MULTI-PLUG ADAPTOR GENERAL TELEPHONY HANDSET SUPPORT VOIP ACCOUNTS SIMULTANEOUS CALLS CALL FEATURES PHONE BOOK CONTACT LIST ENTRIES CALL DEFLECTION DO NOT DISTURB CALL FORWARD UNCONDITIONAL CALL FORWARD NO ANSWER CALL FORWARD BUSY EMERGENCY CALL SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential N/A Yes Yes, EUP II approved 100-240 VAC 50-60Hz 5V2A No 10 simultaneous NB calls supported/cell. Total 10 simultaneous call supported in a single cell configuration. Total 20 simultaneous call supported in a dual cell configuration 20 VoIP accounts 4 Wideband calls (g.722). 10 narrowband calls (PCMA, PCMU, G.726) pr base station. Codec Negotiation Codec Switching Auto Echo Cancellation (AEC) Missed call notification Voice message waiting notification Date and Time synchronization Parallel calls Common parallel call procedures Call transfer unannounced Call transfer announced Call back on Busy Conference Call Waiting (including Call Waiting Caller ID) Calling line identity restriction Outgoing call Call Toggle Incoming call Line identification Multiple Lines Multiple calls Call identification Calling Name Identification Presentation (CNIP) Calling Line Identification Presentation (CLIP) Caller ID Blocking (hiding the caller ID if call is private) Selective/Anonymous Call Rejection Call Hold List of registered handsets Hot line and Warm Line Calling Distinctive Ringing - Calling and Called Number Advanced Inbound and Outbound Call Routing Independent Configurable Dial Plans (1 per port) Music on Hold Common Phonebook: Broadsoft Directory LDAP, XML or csv file load Up to 3000 (depends on size of the entries) Yes Yes Yes Yes Yes Yes 109 | P a g e FCC Warning This equipment has been tested and found to comply with the limits for a Class B digital device, pursuant to part 15 of the FCC Rules. These limits are designed to provide reasonable protection against harmful interference in a residential installation. This equipment generates, uses and can radiate radio frequency energy and, if not installed and used in accordance with the instructions, may cause harmful interference to radio communications. However, there is no guarantee that interference will not occur in a particular installation. If this equipment does cause harmful interference to radio or television reception, which can be determined by turning the equipment off and on, the user is encouraged to try to correct the interference by one or more of the following measures:

Reorient or relocate the receiving antenna. Increase the separation between the equipment and receiver. Connect the equipment into an outlet on a circuit different from that to which the receiver is connected. Consult the dealer or an experienced radio/TV technician for help. Changes or modifications to this equipment not expressly approved by the party responsible for compliance could void the users authority to operate the equipment. This device complies with part 15 of the FCC Rules. Operation is subject to the following two conditions: (1) This device may not cause harmful interference, and (2) this device must accept any interference received, including interference that may cause undesired operation.

SAR tests are conducted using standard operating positions accepted by the FCC with device transmitting at its highest certifed power level in all tested frequency bands, although the SAR is determined at the highest certified power level, the actual SAR level of the device while operating can be well below the maximum value. Before a new model device is an available for sale to the public, it must be tested and certified to the FCC that it does not exceed the exposure limit established by the FCC, tests for each device are performed in positions and locations as required by the FCC. For body worn operation, this model device has been tested and meets the FCC RF exposure guidelines when used with an accessory designated for this product or when used with an accessory that contains no metal.

This equipment complies with FCC radiation exposure limits set forth for an uncontrolled environment. This equipment should be installed and operated with minimum distance 20cm between the radiator& your body. This transmitter must not be co-located or operating in conjunction with any other antenna or transmitter located or operating in conjunction with any other antenna or transmitter. ISEDC Warning This device contains licence-exempt transmitter(s)/receiver(s) that comply with Innovation, Science and Economic Development Canadas licence-exempt RSS(s). Operation is subject to the following two conditions:

1. This device may not cause interference. 2. This device must accept any interference, including interference that may cause undesired operation of the device. SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 110 | P a g e Cet appareil est compatible avec la licence de lInnovation, la Science et le dveloppement conomique du Canada lexemption des normes RSS. Le fonctionnement est sujet aux deux (2) conditions suivantes :

(1) Cet appareil peut ne pas causer de linterfrence, et

(2) Cet appareil doit accepter linterfrence, incluant de linterfrence qui peut causer un mauvais fonctionnement de cet appareil.

SAR tests are conducted using standard operating positions accepted by the ISEDC with device transmitting at its highest certifed power level in all tested frequency bands, although the SAR is determined at the highest certified power level, the actual SAR level of the device while operating can be well below the maximum value. Before a new model device is an available for sale to the public, it must be tested and certified to the ISEDC that it does not exceed the exposure limit established by the ISEDC, tests for each device are performed in positions and locations as required by the ISEDC. For body worn operation, this model device has been tested and meets the ISEDC RF exposure guidelines when used with an accessory designated for this product or when used with an accessory that contains no metal.

This equipment complies with ISEDC radiation exposure limits set forth for an uncontrolled environment. This equipment should be installed and operated with minimum distance 20cm between the radiator& your body. This transmitter must not be co-located or operating in conjunction with any other antenna or transmitter located or operating in conjunction with any other antenna or transmitter. Pour le combin Les tests SAR sont faits en utilisant les normes de positions dopration acceptes par lISEDC avec les appareils mettant les plus hauts niveaux de puissance certifis sur toutes les bandes de frquences, mme si le SAR est dtermin dtre du plus haut niveau de puissance certifi, le niveau SAR actuel de lappareil peut tre sous la valeur maximale de fonctionnement. Avant quun nouveau modle dappareil ne soit disponible pour la vente au public, celui-ci doit tre soumis des tests de certification par lISEDC lesquels nexcdent aucunement la limite dexposition issue par lISEDC, lesquels sont des tests effectus sur chaque appareil dans des positions et endroits requis par lISEDC. Pour lusure de construction de ce modle dappareil, celui-ci a t test et rencontre les lignes directrices mises par lISEDC RF pour lexposition, lorsquil est utilis avec un accessoire conu pour ce produit ou utilis avec un accessoire qui ne contient aucun mtal. Pour la base Cet quipement est conforme avec les limites dexposition la radiation de lISEDC mises dans un environnement contrl. Cet quipement devrait tre install et fonctionnel avec un minimum de distance entre le radiateur et votre corps dau moins 20 cm. Ce transmetteur ne doit pas tre co-situ prs dune autre antenne ou en conjonction avec un autre transmetteur. This Class B digital apparatus complies with Canadian ICES-003. Cet appareille numrique de classe B est conforme aux normes canadiennes ICES-003. SME VOIP SYSTEM GUIDE 4.7 Proprietary and Confidential 111 | P a g e

EXHIBIT C - EUT INTERNAL PHOTOGRAPHS EUT Cover off View 1 EUT Cover off View 2 EUT Main Board Top View EUT Main Board Top Shielding off View EUT Main Board Bottom View EUT Main Chip View

EXHIBIT B - EUT EXTERNAL PHOTOGRAPHS EUT All View EUT Front View EUT Rear View EUT Top View EUT Bottom View 1 EUT Bottom View 2 EUT Left View EUT Right View EUT Chassis Front View EUT With Chassis Front View EUT Chassis Top View EUT RJ45 Cable View EUT DC Cable View EUT Adapter 1 View EUT Adapter 1 Label View EUT Adapter 2 View EUT Adapter 2 Label View

EXHIBIT A - FCC ID LABEL AND LOCATION FCC ID Label FCC ID Label Location Information The label shown shall be permanently affixed at a conspicuous location on the device and be readily visible to the user at the time purchase (Labeling requirements per 2.925)

Model: 8328 SIP-DECT SINGLE BASE STATION Label Label Location Information Model: RTX9431 Label Label Location Information

RTX Hong Kong Ltd. ADD: 8/F Corporation Square,8 Lam Lok Street, Kowloon Bay, Kowloon, Hong Kong Tel: +852 24873718 Fax: +852 24806121 Email: epe@rtx.dk Attestation Letter Date: 2023-06-05 FEDERAL COMMUNICATIONS COMMISSIONS Authorization and Evaluation Division 7435 Oakland Mills Road Columbia, MD 21046 Ref: Attestation Statements CFR 47 2.911(d)(5)(i) Filing FCC ID: T7HX9431 RTX Hong Kong Ltd. certifies that the equipment for which authorization is sought is not covered equipment prohibited from receiving an equipment authorization pursuant to section 2.903 of the FCC rules. Sincerely Yours, Signature Erik Pedersen Engineering Director CFR 47 2.911(d)(5)(i) Attestation Letter

RTX Hong Kong Ltd. ADD: 8/F Corporation Square,8 Lam Lok Street, Kowloon Bay, Kowloon, Hong Kong Tel: +852 24873718 Fax: +852 24806121 Email: epe@rtx.dk Attestation Letter Date: 2023-06-05 FEDERAL COMMUNICATIONS COMMISSIONS Authorization and Evaluation Division 7435 Oakland Mills Road Columbia, MD 21046 Ref: Attestation Statements CFR 47 2.911(d)(5)(ii) Filing FCC ID: T7HX9431 RTX Hong Kong Ltd. certifies that as of the date of the filing of the application, the applicant is not identified on the Covered List as an entity producing covered equipment. Sincerely Yours, Signature Erik Pedersen Engineering Director CFR 47 2.911 (d)(5)(ii) Attestation Letter

RTX Hong Kong Ltd. ADD: 8/F Corporation Square,8 Lam Lok Street, Kowloon Bay, Kowloon, Hong Kong Tel: +852 24873718 Fax: +852 24806121 Email: epe@rtx.dk Date: 2023-06-05 FEDERAL COMMUNICATIONS COMMISSIONS Authorization and Evaluation Division 7435 Oakland Mills Road Columbia, MD 21046 Subject: FCC Class II Permission change for FCC ID: T7HX9431 Original Grant Date: 11/15/2019 Dear Sir/Madam, This is to request a Class II permission change to our product name: VOIP Phone,

(FCC ID: T7HX9431), the device is identical to the previously certified except for the changes as below for details

(1) Add model 8328 SIP-DECT SINGLE BASE STATION.

(2) Add trade mark.

(3) Add an alternative PCB with the following:

a. Changed PoE chip from Si3402 to MP8007. b. Changed voltage regulator from LDO to DCDC. c. Changed Ethernet bias to 2.7V d. Add LED driving circuit. e. Add one output coupling capacitor in RF output circuit. f. Add one backup decoupling capacitor for RF selector. g. Changed Ethernet surge protection component. This authorization is valid until further written notice from the applicant. Please contact me if you have any questions or need future information regarding this application. Sincerely Yours, Signature Erik Pedersen Engineering Director

Declaration of Similarity to Type 12thJune 2023 Model No.:

Multiple Model(s) No.:

RTX9431 8328 SIP-DECT SINGLE BASE STATION To whom it may concern, We, RTX Hong Kong Ltd. hereby confirm that the Trademark of RTX and ALE listed in the following table are identical as follows:

Electrical designs PCB layout Produced at the same location The only differences between these models are the followings for marketing purpose:

Color Cosmetic details Trade name and Logo Model Number Software but does not affect RF parameter Construction design/Physical design/Enclosure Related circuit of POE Items No. Trademark 1 2 RTX Alcatel-Lucent Enterprise Model Number RTX9431 8328 SIP-DECT SINGLE BASE STATION DC input port PoE input port Yes No Yes Yes All type approval reports of RTX models are also valid for RTX9431 and 8328 SIP-DECT SINGLE BASE STATION. All above models FCC ID: T7HX9431 Declared by, RTX A/S Erik Pedersen (Director of Engineering) RTX Hong Kong Ltd. 8/F., Corporation Square, 8 Lam Lok Street, Kowloon Bay, Kowloon, Hong Kong Tel Fax Web E-mail

: +852 2487 3718

: +852 2480 6121

: http://www.rtx.hk

: info@rtx.dk

RTX Hong Kong Ltd. ADD: 8/F Corporation Square,8 Lam Lok Street, Kowloon Bay, Kowloon, Hong Kong Tel: +852 24873718 Fax: +852 24806121 Email: epe@rtx.dk FCC Authorization Date: 2023-06-05 FEDERAL COMMUNICATIONS COMMISSIONS Authorization and Evaluation Division 7435 Oakland Mills Road Columbia, MD 21046 Subject: Agent Authorization To whom it may concern:

We, RTX Hong Kong Ltd., the undersigned, Hereby authorizes China Certification ICT Co., Ltd (Dongguan) to act on its behalf in all matters relating to application for Equipment authorization, including the signing of all documents relating to these matters. All acts carried out by China Certification ICT Co., Ltd (Dongguan) on our behalf shall have the same effect as our own action. We, RTX Hong Kong Ltd., the undersigned, hereby certify that we are not subject to a denial of federal benefits, that includes FCC benefits, pursuant to Section 5301 of the Anti-Drug Abuse Act of 1988, 21 U.S.C. 862. This authorization is valid until further written notice from the applicant. Sincerely Yours, Signature Erik Pedersen Engineering Director QA-FR-170-B

RTX Hong Kong Ltd. ADD: 8/F Corporation Square,8 Lam Lok Street, Kowloon Bay, Kowloon, Hong Kong Tel: +852 24873718 Fax: +852 24806121 Email: epe@rtx.dk FCC Confidential Authorization Date: 2023-06-05 FEDERAL COMMUNICATIONS COMMISSIONS Authorization and Evaluation Division 7435 Oakland Mills Road Columbia, MD 21046 Subject: Confidentiality Request regarding application for certification of FCC ID: T7HX9431 In accordance with Sections 0.457 and 0.459 of the Commissions Rules, RTX Hong Kong Ltd. hereby requests long-term confidential treatment of information accompanying this application as outlined below:

Block Diagram Schematics Operation Description The above materials contain proprietary and confidential information not customarily released to the public. The public disclosure of these materials provides unjustified benefits to its competitors in the market. Sincerely, Signature Erik Pedersen Engineering Director

USA REPRESENTATIVE LETTER OF ATTESTATION USA Representative Company Name: RTX America Inc. Contact Name: Yash Singh Company Address: 10620 Treena St, Suite 230, San Diego, CA 92131, USA Telephone No: (858) 935 6152 Email: ysh@rtx.dk FRN: 0033504135 To: FCC ATTENTION: TCB and/or Certification and Engineering Bureau This letter is to confirm that we have accepted the responsibility to act as USA Representative future FCC on behalf of certification/registrations obtained during the period of the agreement which ends at the specified date below. the Applicant noted below for all existing and Applicant:

Company Name: RTX Hong Kong Ltd. FCC Grantee code: T7H Contact Name: Erik Pedersen Company Address: 8/F Corporation Square, 8 Lam Lok Street, Kowloon Bay, Kowloon, Hongkong Telephone No: +45 96322334 Email: epe@rtx.dk This Agreement is valid until Jan. 31 2028, ____________________________________________________________________________________________________________ ___________________________________________________ ____________________________________________________________________ igngngnggngngngngngngngngngngngngngngngngngngngngngngngngngngngnngngnggngggnggnnngngnggnggngngnngng ededdededededededededededededdedededededededededededededededededededeedededeedededdeddedeeedeeeedeeee b b bbbbbbbbb b bbbbb bbb b bbbbb b bb bb bbb b bbbbbbbbbbbbbbbby:yyyyyyyyyyyyyyyyyyyyyyyyyyyyyyyyyyyyyy Yash Singh / Senior Busin Signed by: Yash Singh / Senior Business Director date 28th Feb, 2023 RTX America Inc. 10620 Treena St, Suite 230, San Diego, CA 92131, USA Telephone No: (858) 935 6152 Email: ysh@rtx.dk Web:

http://www.rtx.dk

RTX A/S Stroemmen 6 9400 Noerresundby, Denmark Tel:

Web:

VAT (CVR) No.: DK17002147

+45 96322300 http://www.rtx.dk

RTX Hong Kong Ltd. Address: 8/F Corporation Square, 8 Lam Lok Street, Kowloon Bay, Kowloon, Hong Kong Tel.: +852 2487 3718 Fax: +852 2480 6121 E-mail: TED@rtx.hk FCC Confidential Authorization Date: 2019/09/10 FEDERAL COMMUNICATIONS COMMISSIONS Authorization and Evaluation Division 7435 Oakland Mills Road Columbia, MD 21046 Subject: Confidentiality Request regarding application for certification of FCC ID: T7HX9431 In accordance with Sections 0.457 and 0.459 of the Commissions Rules, RTX Hong Kong Ltd. hereby requests long-term confidential treatment of information accompanying this application as outlined below:

Block Diagram Schematics Operation Description The above materials contain proprietary and confidential information not customarily released to the public. The public disclosure of these materials provides unjustified benefits to its competitors in the market. Sincerely Yours, Signature Ted Chong Engineering Manager

RTX Hong Kong Ltd. Address: 8/F Corporation Square, 8 Lam Lok Street, Kowloon Bay, Kowloon, Hong Kong Tel.: +852 2487 3718 Fax: +852 2480 6121 E-mail: TED@rtx.hk FCC Authorization Date: 2019/09/10 FEDERAL COMMUNICATIONS COMMISSIONS Authorization and Evaluation Division 7435 Oakland Mills Road Columbia, MD 21046 Subject: Agent Authorization To whom it may concern:

We, RTX Hong Kong Ltd., the undersigned, Hereby authorizes Bay Area Compliance Laboratories Corporation to act on its behalf in all matters relating to application for Equipment authorization, including the signing of all documents relating to these matters. All acts carried out by Bay Area Compliance Laboratory Corporation on our behalf shall have the same effect as our own action. We, the undersigned, hereby certify that we are not subject to a denial of federal benefits, that includes FCC benefits, pursuant to Section 5301 of the Anti-Drug Abuse Act of 1988, 21 U.S.C. 862. This authorization is valid until further written notice from the applicant. Sincerely Yours, Signature Ted Chong Engineering Manager QA-FR-170-B