Topex multiSwitch User manual

TOPEX multiSwitch
User manual
2008

multiSwitch – user manual
© TOPEX 2008 2
About TOPEX
TOPEX is a group of Romanian companies, established in 1990, by 10 enthusiastic engineers
experienced in telecommunications. Its activity is directed to the research, development
and manufacturing of telecom equipment, as well as providing the service for it.
TOPEX has quickly become the most important supplier of IT&C solutions for small and
large companies, as well as for telecommunications operators and providers in Romania.
The company designs equipment for all existing mobile systems (GSM/GPRS/EDGE/UMTS,
CDMA EVDO, HSDPA/TDD), including 3G+ technology.
TOPEX is represented all over Romania by a wide network of local distributors through
which the promotion, administration and product maintenance are running.
Due to our innovation power, authentic flexibility, real respect for our partners and to the
secure solutions that we provide, TOPEX extended its business worldwide. Currently,
TOPEX delivers its products through its distributors to: Italy, Spain, France, Russia, UK,
Turkey, Netherlands, Greece, Republic of Moldavia, Bulgaria, Nigeria etc.
In order to achieve effective and flawless manufacturing for its products, TOPEX has
carefully organized its Research and Development Department along with its production
facility. This allows TOPEX to have maximum control of all the processes involved in the
complex operations related to high-technology electronic manufacturing.
Currently, the Research and Development Department counts 30 specialists and the trend
is ascending.
TOPEX also considers the training and the service as part of the solutions it provides.
Therefore, comprehensive trainings are organized at the Topex factory, complimentary for
the company’s clients. Service is also provided over internet, as all TOPEX solutions are
designed especially to allow this, at the lowest cost.
TOPEX has implemented the quality management system according to the ISO-9001
standard certified by SRAC since 1997, respectively by IQNET since 2002. Since 2001,
TOPEX became a member of I.T.U. (International Telecommunication Union).
The main product lines of our company are:
-GSM/GPRS/UMTS interfaces (analogue, ISDN BRI and PRI, GSM, VoIP gateways) and
SIM servers.
-Advanced interfaces for connecting to different mobile networks (voice/data
UMTS, TDD or HSDPA for fast speed in data transmission, LAN, EVDO router, GPS,
miniatures)
-VoIP gateways, softswitches
-Mixed systems for special communications applications, such as VCSS (air traffic
control, railroad dispatch)
-Telephone switch systems (analogue, digital, private/rural, IP PBX, protocol
converters)

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For further details, please visit our website at: www.topex.ro
WEEE Directive Compliance
WEEE Directive
This symbol applied on the product you purchased or on its packaging means
that the product is not to be recycled in the same manner as domestic
garbage.
According to the EU and Romanian legal provisions, the recycling of EE
equipments must be accomplished separately for the purpose of preserving
the natural resources and to avoid the negative effects upon human health
and environment.
The TOPEX product will be recycled in compliance with current legislation.
The purchased product will not be disposed of as domestic garbage after
ending its lifecycle and will be returned to TOPEX at the following address:
10 Feleacu Str., 014486, or delivered to a DEEE collecting organization that
is agreed by TOPEX.
!Do not dispose of the product yourself as unsorted domestic garbage,
recycle it to protect the environment. Separate the packaging materials and
sort it for recycling.
For additional information, contact us at:
Phone: [phone] or www.topex.ro

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Table of Contents
1BASE CONCEPTS ..................................................................................................................7
1.1 Network switching technology................................................................................................7
1.2 Packet switching ....................................................................................................................8
1.3 IP telephony technology.........................................................................................................8
1.4 IP Voice Coding and compression modes.............................................................................9
2SYSTEM OVERVIEW........................................................................................................... 10
2.1 TOPEX multiSwitch – introduction...................................................................................... 12
2.2 Architecture......................................................................................................................... 13
2.2.1 Client computer........................................................................................................ 13
2.2.2 TOPEX equipment................................................................................................... 14
2.2.3 Enclosure ................................................................................................................. 15
2.2.4 Electric circuit........................................................................................................... 16
2.2.5 PC Server Specifications ......................................................................................... 16
3INTERFACES AND PROTOCOLS....................................................................................... 17
4SYSTEM SERVICES............................................................................................................ 18
4.1 Call Control Flow................................................................................................................. 19
4.2 Client Classes..................................................................................................................... 20
4.2.1 Description............................................................................................................... 20
4.2.2 Main Features .......................................................................................................... 21
4.3 Clients ................................................................................................................................. 21
4.3.1 Description............................................................................................................... 21
4.3.2 Main Features .......................................................................................................... 21
4.3.3 SIP users ................................................................................................................. 22
4.3.4 FXS users ................................................................................................................ 23
4.3.5 ANI Users................................................................................................................. 24
4.3.6 Prepaid users........................................................................................................... 24
4.3.7 Trunks ...................................................................................................................... 25
4.4 Destinations ........................................................................................................................ 25
4.5 Tariffs .................................................................................................................................. 26
4.6 Service instances................................................................................................................ 26
4.7 CDR – Call Detail Records ................................................................................................. 27
5SIP FEATURES....................................................................................................................28
5.1 Security level....................................................................................................................... 29
6TOPEX multiSwitch INSTALLATION.................................................................................... 30
6.1 System access.................................................................................................................... 30
6.2 IP address changing ........................................................................................................... 31
6.3 Time and date changing ..................................................................................................... 31
7TEXT FILES CONFIGURATION .......................................................................................... 34
7.1 “voip.cfg” configuration........................................................................................................ 34
7.2 “group.cfg” configuration..................................................................................................... 35
7.3 “exec.cfg” configuration....................................................................................................... 37
7.3.1 Debug parameters ................................................................................................... 37
7.3.2 Data Base activation................................................................................................ 38
7.3.3 RADIUS activation ................................................................................................... 38
7.3.4 RTP Proxy................................................................................................................ 39
7.3.5 Other parameters..................................................................................................... 39
7.4 “sip_pbx.cfg” configuration.................................................................................................. 41
7.4.1 Debug parameters ................................................................................................... 41
7.4.2 Proxy and register parameters ................................................................................ 42
7.5 “register_users.cfg” configuration ....................................................................................... 45
7.6 “public_private.cfg” configuration........................................................................................ 45
7.7 “trafic.cfg” configuration ...................................................................................................... 46
7.8 RADIUS activation .............................................................................................................. 47
7.9 Data Base Billing................................................................................................................. 49
7.9.1 Pg SQL Billing.......................................................................................................... 50
7.9.2 My SQL Billing ......................................................................................................... 50

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7.10 IVR activation and configuration....................................................................................... 52
7.10.1 Actions and events of IVR service ........................................................................... 52
7.10.2 IVR configuration examples for prepaid şi voice attendant ..................................... 54
7.11 TELNET commands.......................................................................................................... 56
8DESCRIPTION OF THE WEB INTERFACE ........................................................................ 59
8.1 The web administration interface........................................................................................ 59
8.2 Authentication ..................................................................................................................... 59
8.3 The menu bar...................................................................................................................... 60
8.4 The "Server Settings" menu................................................................................................ 61
8.4.1 The "About" option................................................................................................... 61
8.4.2 "SIP Access List" option .......................................................................................... 62
8.4.3 "Settings" option....................................................................................................... 63
8.4.4 The "Equipments" option ......................................................................................... 64
8.5 The "Server Management" menu........................................................................................ 66
8.5.1 The "Client Classes" option ..................................................................................... 66
8.5.1.1 Editing a client class................................................................................................ 67
8.5.1.2 Adding a new client class........................................................................................ 70
8.5.1.3 Adding rules for a client class ................................................................................. 72
8.5.1.4 Setting alarms for a client class .............................................................................. 73
8.5.2 The "Translate ANI" and "Translate DNIS" options ................................................. 74
8.5.3 The "Routes" option................................................................................................. 77
8.5.3.1 Editing a route......................................................................................................... 77
8.5.3.2 Creating a new route............................................................................................... 79
8.5.4 The "Access In" option............................................................................................. 82
8.5.5 The "Access Out" option.......................................................................................... 84
8.5.6 The "Global Rules" option........................................................................................ 85
8.6 The "Services" menu........................................................................................................... 86
8.6.1 The "SIP Aliases" option.......................................................................................... 86
8.6.2 The "Centrex groups" option.................................................................................... 87
8.6.3 The "Hunting Groups" option ................................................................................... 89
8.6.4 The "Pickup Groups" option..................................................................................... 90
8.6.5 The "Forking Groups" option ................................................................................... 91
8.7 The "User Management" menu........................................................................................... 92
8.7.1 The "System" option ................................................................................................ 92
8.7.2 The "SIP" option....................................................................................................... 97
8.7.2.1 Adding a SIP user................................................................................................... 97
8.7.2.2 CLI options for a SIP user..................................................................................... 100
8.7.2.3 Customized rules for SIP users ............................................................................ 101
8.7.2.4 Aliases for SIP users.............................................................................................101
8.7.2.5 The Web Access option........................................................................................ 102
8.7.3 The "ANI" option .................................................................................................... 103
8.7.4 The "FXS" option ................................................................................................... 105
8.7.4.1 Editing the properties of the FXS subscriber ........................................................ 105
8.7.4.2 Adding a new FXS user ........................................................................................ 106
8.7.5 The "Prepaid" option.............................................................................................. 107
8.7.6 The "SIP Locations" option .................................................................................... 109
8.8 The "Billing" menu............................................................................................................. 110
8.8.1 The "Profiles" option .............................................................................................. 110
8.8.2 The "Subscriptions" option..................................................................................... 112
8.8.3 The "Currencies" option......................................................................................... 113
8.8.4 The "Groups" option............................................................................................... 114
8.9 The "Account Properties" menu........................................................................................ 116
8.9.1 The "Call List" option.............................................................................................. 116
8.9.1.1 The "Filters" button................................................................................................ 117
8.9.1.2 Call details............................................................................................................. 119
8.9.1.3 CDR Export........................................................................................................... 121
8.9.2 The "Address Book" option.................................................................................... 122
8.9.3 The "Logout" option ............................................................................................... 123
8.10 The "Testing" menu.........................................................................................................124
8.10.1 The "User Generator" option ................................................................................. 124

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8.10.2 The "Rule Generator" option.................................................................................. 125
8.11 The "Reports" menu........................................................................................................ 126
8.11.1 The "Profitability [Brief]" option .............................................................................. 126
8.11.2 The "Profitability [In]" option................................................................................... 127
8.11.3 The "Profitability [Out]" option................................................................................ 127
8.11.4 The "Reliability [In]" option..................................................................................... 128
8.11.5 The "Reliability [Out]" option .................................................................................. 129
8.12 The "Templates" menu.................................................................................................... 130
8.12.1 The "Class Templates" option ............................................................................... 130
8.12.2 The "SIP User Templates" option .......................................................................... 132
8.12.3 The "FXS User Templates" option......................................................................... 134
8.12.4 The "ANI User Templates" option.......................................................................... 135
8.12.5 The "Prepaid User Templates" option ................................................................... 137
9ACTIVATION AND CONFIGURATION OF THE SERVICES............................................. 138
9.1 SIP users registration........................................................................................................ 138
9.1.1 Creating a client class (directs).............................................................................. 138
9.1.2 Adding SIP users ................................................................................................... 140
9.2 Configuring the Call Waiting option................................................................................... 142
9.3 Configuring the Do Not Disturb option.............................................................................. 144
9.4 Configuring the Call Forward service................................................................................ 145
9.5 Configuring the Call Back service..................................................................................... 146
9.6 Configuring the Call Hunting option.................................................................................. 147
9.7 Configuring the Call Pick-up option................................................................................... 148
9.8 Configuring the Call Forking service................................................................................. 148
9.9 Configuring the Suspend Subscriber service.................................................................... 149
9.10 ACL Configuration........................................................................................................... 150
9.11 Configuring the Call Baring service................................................................................. 151
9.12 Configuring the CLIP / CLIR options............................................................................... 152
9.13 Configuring the Voice Mail.............................................................................................. 153
9.14 Configuring Voice Mail 2 E-mail...................................................................................... 156
9.15 Configuring Missed Calls 2 E-mail.................................................................................. 157
9.16 Flagging the packets as ToS or DSCP........................................................................... 157
9.17 Number Portability........................................................................................................... 158
9.18 Enabling the services directly from the SIP phone......................................................... 159
10 TECHNICAL DATASHEET................................................................................................. 160
11 OPERATING CONDITIONS............................................................................................... 161
11.1 Environment conditions................................................................................................... 161
11.2 Security conditions.......................................................................................................... 161
12 GLOSSARY........................................................................................................................162
This manual is Revision E, October 2008

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1 BASE CONCEPTS
In order to create a quick background of networks and data communications we detail below a
few base concepts related to the network switching technology, packet switching, IP telephony,
compression modes or IP voice coding.
Data communication is a new communication mode combining the computer and
communication, which is the foundation upon which various computer networks can be set up.
The data communication network has been developing for over 30 years. In the course when
human beings enter the information society, data communication is playing a more and more
important role.
1.1 Network switching technology
In a wide area, data communication is to transmit data from a source node to the destination
via the intermediate switching node network. Such a switching node does not care about contents of
data.
Its objective is to provide switching facilities for mobile data between nodes. A termination
device for communication can be called a site. A site can be a computer, terminal, telephone or
another communication device.
A switching device providing communication is called a node. They form a topology after being
connected with each other via transmission links. Each site can be connected with a node. The
collection of all nodes is called a communication network.
In a switching communication network, the data entering the network from a site via inter-node
switching is sent to the destination after being routed. From a simple network, we can specify that:
- Some nodes are only connected to other nodes. The only task of these nodes is to complete
internal data exchange. Other nodes are connected to one or more sites. Except the exchange
function, these nodes also receive data from the connected site and delivers data to the
connected site;
- Generally, the multi-channel multiplexing is adopted for inter-node links. We can also adopt the
frequency division multiplexing (FDM) or time division multiplexing (TDM) mode. In addition, the
network is not all connected, i.e., there is no direct link between each possible node pair.
However, it is always hoped that there are more than one path between each pair of sites to
increase the network reliability.
In the wide area network, two different technologies are adopted: circuit switching and packet
switching. Along the path from the source to destination, there are differences between the modes of
switching information from one line to another for different nodes.
Since the major multiSwitch bearer network is based on the packet switching network, we will
primarily describe the packet switching mode below.

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1.2 Packet switching
In the packet switching network, data are transmitted in short packets which have a maximum
size limit. If a larger packet is to be sent from a source site, this packet will be split into a series of
shorter packets.
Each packet contains a part of user data (or the whole of a shorter packet) and some control
information. The control information should at least contain routing information needed by the network
for sending packets to the destination.
At each node of a path, packets are received, stored for a short period of time and then
transmitted to the next node. Compared with circuit switching, packet switching has the following
advantages:
High line efficiency: a single inter-node link can be dynamically shared by multiple packets. In this
case, packets are queued and sent out from the relevant link as soon as possible. However, in
circuit switching, the time on the inter-node link is allocated in advance with synchronous time
division multi-channel multiplexing. In this case, the time slot allocated to a connection cannot be
occupied by other connections even when it is idle;
The packet switching network can implement data rate conversion. In this case, two sites with
different data transmission rates can exchange packets with each other since each site is
connected to its communication node at the respective rate. However, in circuit switching, the two
sites connected with a circuit should transmit and receive data at the same rate;
In a circuit switching network, when the traffic is high, some calls will be blocked, i.e., the network
will reject new connection requests before the load on the network reduces. However, in a packet
switching network, such packets will still be received but the transmission delay is increased;
Priorities are used. If there are many packet queues to be transmitted for a node, it can transmit
packets with higher priorities in precedence. These packets will have lower delay than those with
lower priorities.
Generally, packet switching does not mean to send the whole packet of a user. Instead, one
packet is divided into several packets that can be saved in the memory.
This increases the switching speed. This mode is applicable to interactive data transmission.
According to services provided by the communication subnet for the termination system,
packet switching can be further divided into datagram and virtual circuit switching.
1.3 IP telephony technology
The major objective of the IP telephony technology is to combine the IP network with the
telephony network. Moreover, IP telephones can be used by not only computer users but also ordinary
telephone users.
The two networks have different characteristics: the IP network is a kind of network
transmitting data information, in which the packet switching technology is applied; while the telephone
network is a kind of network transmitting analog voice signals, in which the circuit switching technology
is applied.
As we know, the characteristic of circuit switching is that a circuit is occupied whenever a call
is connected. It will be occupied all along as long as no party hangs up no matter whether the two
parties are talking to each other.
Generally, a party is listening while the other party is talking. Therefore, in this case, at least
50% of the circuit is not utilized and the circuit utilization ratio is very low.
The packet transmission technology is to divide information data to be transmitted into groups
based on a certain length (i.e., cutting them to “packets”), add an address flag to each “packet”, and
then transmit them in the store-forward mode.

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In this case, each session packet does not monopolize a circuit. Instead, it is sent only when
the circuit is idle. In this way, multiple sessions can share one channel asynchronously. Thus, the
circuit utilization ratio is considerably increased.
Furthermore, the digital compression technology is adopted in packet transmission. Therefore,
the circuit utilization ratio is many times higher than that of circuit switching. In addition, the charging
mode of packet transmission is irrelevant to the distance. This tremendously reduces the IP costs.
At present, with the application of multiple QoS-ensuring technologies such as the queue,
priority, RSVP, VPN and MPLS, the IP network technology is developing towards a higher rate and
better QoS.
Moreover, with the IP telephone technology, the communication cost can be saved
considerably. This determines its tremendous market potential.
With the driving of the market, more and more research institutes, international
standardization organizations, manufacturers etc. are devoting themselves to the development of IP-
related technologies, thus enabling it to reach the degree of commercialization.
1.4 IP Voice Coding and compression modes
The transmission of real time voices via the IP network is different from that of ordinary data.
In the former case, the relevant application devices must meet the real time of voices. The voice
packet transmission requires the network to provide sufficient bandwidth in time.
Therefore, for most of the current IP networks that do not provide so high rates, the voice
compression technology is the key for implementing IP voice communication. Now, we will present a
brief description of the frequently used voice coding and compression modes at present:
PCM - Pulse code modulation is the earliest digital voice technology, which does not include any
compression algorithm. It transmits voice signals with the 64kbps bandwidth, i.e., taking 8,000
samples per second and acquiring an 8-digit voice signal per sample. PCM is the standard coding
mode adopted in G.711.
CELP - Code excited linear prediction (CELP) is the most advanced voice transmission technology
at present. The CELP algorithm is to compare analogue signal samples with curves in the
predefined code book; send codes in the code book closest to these analogue signal samples to
the receiving end; and regenerate original signals after comparison again with the code book at
the receiving end. The sampling interval of original signals is very short. Therefore, the
regenerated signals are very close to the original signals after being filtered. CELP is the basis of
numerous advanced patented voice compression modes. Voices can be compressed to 5.3 kbps,
8 kbps or 9 kbps.
CS-ACELP - Conjugate structure algebra code excited linear prediction (CS-ACELP) or G.729 is the 8
kbps voice compression and coding standards of International Telecommunications Union (ITU).
CS-ACELP is a new algorithm, which is able to encode 8kbps voice signal bit streams (while the
rate of ordinary PCM signals is 64 kbps). The bandwidth efficiency is eight times as that of PCM
and four times as that of 32 kbps ADPCM. At present, CS-ACELP is the most welcome voice
encoding/decoding plan.
When actually selecting a voice compression algorithm, it is necessary to take various factors
into consideration. For example: the pursue of higher bit rates guarantees sound voice quality but
requires to occupy more system resources. While lower bit rates will influence voice quality and
increase delay.
Therefore, to keep better voice quality in the precondition of lower bit rates is the principle for
compression algorithm selection.

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2 SYSTEM OVERVIEW
At present, two totally independent networks exist: the PSTN network and the data network,
which provide the voice service and basic data service respectively.
Network separation and isolation of operation & maintenance have been keeping the general
network operation & maintenance costs on a high level, and furthermore, a network cannot provide
complicated convergence services, although the network convergence has been an inevitable trend.
Hierarchical models are adopted for the MultiSwitch - based Next Generation Network (NGN).
The entire network can be divided into five levels: Application Level, Call Management Level,
Infrastructure Level, Interface Equipment Level and Customer Premises Level, as shown in Figure 1.
Figure 1 – Next Generation Network based on the MultiSwitch Technology
Below will be described all the component levels of the Next Generation Network architecture
starting from the base to the top (as you can see in Figure 1):
Customer Premises Level – it refers to the terminals which are accessible to the common users. This
level include among other equipments: SIP Phones, GSM terminals, modems, Soft Phones,
computers, analogue phones, etc.
Interface Equipment Level – this layer refers to various access gateways and new types of access
terminal devices related to the current network. It implements interworking between the devices
located at the “Customer Premises Level” and the “Infrastructure Level”. This level contains
several gateways manufactured by TOPEX such as Media Gateway, GSM / UMTS / CDMA
Gateway and SS7 Signalling Gateway.
Infrastructure Level – The Infrastructure Layer refers to a packet switching network composed of
backbone transmission equipment such as IP router or broadband ATM switch, which is the
bearer basis of a MultiSwitch network.
Call Management Level – refers to MultiSwitch control units, which completes integrated control
processing functions such as call processing control, access protocol adaptation, interconnection
and interworking and provides an application support platform for the entire network.

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Application Level - The Application Layer provides a network with various applications and services,
client-oriented integrated intelligent services and service customization. The applications included
here are:
- Voice messaging;
- Voice portal;
- Prepaid application;
- Billing application;
- Video services.
Standard interfaces are used for communication between layers. Under the control of the core
equipment (i.e., the MultiSwitch equipment) and based on division of labour and cooperation of work,
the related equipment implements various service functions of the system.
In MultiSwitch architecture, the MultiSwitch control equipment is the core, which is
independent of the bottom-layer bearer protocols and implements functions such as call control, media
gateway access control, resource allocation, protocol processing, routing, authentication and
accounting.
The MultiSwitch control equipment can provide all basic call services, supplementary services
and point-to-point multimedia services a PSTN can provide.
Furthermore, with the cooperation of the Service Layer equipment (SCP) and Application
Server, the equipment also can provide users with traditional intelligent services, value-added IP
services, diverse third-party value added services and new intelligent services.

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2.1 TOPEX multiSwitch – introduction
Since a traditional voice network is a closed network with monopolized resources, it has
become a common understanding in the telecom industry that the packet network (typically, the
Internet), with the advantages such as open architecture, low costs and large scale, will replace the
PSTN.
Thus it becomes the basic frame of the next generation of convergence networks and that the
construction of the next generation of networks will be based on current packet networks. It is
necessary for carriers to consider resource utilization and investment protection during construction of
future networks. On one hand, carriers should trace the latest technologies and on the other hand,
they should try to utilize existing technologies and resources.
Thus, carriers can provide users with large numbers of services economically and rapidly to
make the highest profits, without the need of large-scale network alteration. The solution of smooth
transition from existing networks to the next generation networks is the key to the problem. The
MultiSwitch solution based on multiSwitch technology is just a mainstream solution to smooth network
evolution.
A MultiSwitch is a very important device in a telecommunication network. It connects the calls
from a telecommunication operator to another one only through software. The old routing call through
hardware devices is now replaced by the MultiSwitches.
The MultiSwitch is a new switching solution running on a hardware platform – a computer - to
improve and even to replace the switching function of the C.O. (Central Office).
Figure 2 – Software system of TOPEX MultiSwitch control equipment

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Designed in the distributed mode, the software system of TOPEX MultiSwitch has the
hierarchical and modular features. The schematic diagram is shown in Figure 2.
The Device (Protocol) Adaptation Layer is responsible for accessing various external standard
protocols such as H.248, MGCP, H.323 and SS7, converting them to unified internal messages and
sending them to the Call Server for proper processing.
For future protocols, we can implement the upgrading of the system smoothly just by adding
the corresponding software adaptation module to this layer.
As the control core of the system, the Call Server provides unified call control. The Resource
Manager is responsible for allocating various call-related media resources, for example, controlling the
media server to play service tones. The BICC / SIP (Bearer Independent Call Control) module
supports interworking between peer entities (MultiSwitch control equipment).
The Service Manager is responsible for providing interaction between the MultiSwitch control
equipment and the upper-level SCP and Application Server. The Data Manager provides an unified
access interface to the internal database.
The MultiSwitches achieve the same operations like the usual switch equipments and they are
completely transparent to the final user.
A call has three stages until it is ended:
•The client and client classes identification;
•The identification of services and routes to destination;
•The call accomplishment.
The TOPEX MultiSwitch application manages the calls using two concepts: the “Client
Classes” and “Clients”.
2.2 Architecture
The TOPEX MultiSwitch application is delivered by TOPEX Company in two variants:
•installed directly on the client’s computer;
•included into equipment which can be mounted in a rack.
In both cases, the manufacturer offers to the client a software licence key. This licence key is
different for each TOPEX application. It supplies a better protection of the equipment and unauthorized
use of this one.
2.2.1 Client computer
In the first situation, the software application is installed by TOPEX on a computer supplied by
the client. Also, TOPEX install on this computer the Gentoo Linux 2006 operating system required for
the proper operation of the MultiSwitch application.
The computer where the TOPEX MultiSwitch application will be installed must accomplish the
minimum requirements shown below:
•Pentium 4 processor with 3 GHz frequency;
•1 GB RAM memory;
•100 GB hard disk free space;
•DVD RW.
The system is delivered by TOPEX with a default IP address 192.168.1.20.This address is on
a label applied on the client computer enclosure.

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2.2.2 TOPEX equipment
The second delivery variant of the multiSwitch application is the installation on a TOPEX
equipment.
The equipment has a compact design, it is reliable and has a 1U height and 19” width to be
mounted in a correspondent rack.
In fact, it is a computer enclosed in a metallic case.
On the front panel, the equipment has a cover with notches for the ventilation. The cover can
be locked with a key delivered by TOPEX to prevent the unauthorized access to the front panel of the
equipment.
Figure 3 – TOPEX MultiSwitch – general view
Also, there is a label applied on the TOPEX multiSwitch equipment enclosure. This label
contains the default IP address of the equipment which is 192.168.1.20.
The equipment is supplied through the power supply cable included in the TOPEX MultiSwitch
equipment from the 230 V A.C. / 50 Hz network.
Package content:
When you unpack the TOPEX multiSwitch equipment box, you must find the next items:
Item Pcs
TOPEX MultiSwitch equipment in a 1U height case 1
Mounting kit for a 19” rack 1
Protection key for the front panel 2
Internal supply cable 1
Power supply cable 1
User’s Manual 1
Warranty Certificate 1

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2.2.3 Enclosure
The TOPEX MultiSwitch unit is enclosed in a 19” case with 2U height. The front panel includes
a cover with notches that is opening with a key delivered by TOPEX. The access to the front panel of
the equipment can be locked to prevent unauthorized access.
When the cover is unlocked you will have access to the front panel of the equipment.
Figure 4 – TOPEX MultiSwitch - Front panel
On the front panel the equipment are located:
•the reset button – colored in red (labeled RESET);
•the power button – colored in green (labeled POWER);
•2 x cooling fans for assuring the proper ventilation of the equipment;
•DVD – ROM unit for reading CDs or DVDs;
•2 x LEDs of the electric circuit.
Figure 5 – TOPEX MultiSwitch – Back panel
On the back panel of the TOPEX MultiSwitch equipment we have:
•a central panel with the connectors of the electrical circuit;
•the special connector for the power supply cable on the left upper part of the back panel;
•2 x sets of notches for the equipment ventilation (located on the left and right extremities of the
back panel).

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2.2.4 Electric circuit
The card contains a powerful processor controlling the whole PCI bus of the system.
The electric circuit includes multiple printed circuit cards and a hard disk for the operating system
(Gentoo Linux 2006), MultiSwitch application and storage of data.
It features on the front panel:
•green LED (labeled with the symbol) – shows the equipment power supply when it lights up;
•orange LED (labeled with the symbol) – shows the hard disk activity when it lights up.
On the back panel of the electric circuit we have:
•2 x PS/2 connectors for mouse and keyboard connection;
•4 x USB-A connectors (2 connectors in the left side, near the mouse / keyboard connectors, and
2 connectors in the right side, under the Ethernet connector);
•DB 27 female connector for the printer connection;
•DB 9 male connector;
•DB 15 female connector for the PC monitor connection;
•Ethernet connector for the local network;
•3 connectors – 2 (blue and yellow) for connecting the head phones and speakers and the pink
one for connecting a microphone.
2.2.5 PC Server Specifications
TOPEX multiSwitch application can be installed on several PC servers which must achieve a
minimum configuration described earlier. In case of the PC Servers delivered by TOPEX along with
the multiSwitch application installed the usual server is a HP Proliant DL140G3 type.
The HP PC Server specifications are detailed below:
Model HP Proliant DL140 G3 5140
Processor Dual-Core Intel® Xeon® processor 5140 (2.33 GHz, 65 W, 1333 MHz FSB)
Cache Memory 4MB (1 x 4MB) Level 2 Cache or 8 MB (2 x 4MB) Level 2 Cache
RAM Memory 1 GB FBD PC2-5300 DDR2 Fully Buffered DIMMs (667 MHz) – 2 x 512 kit
Network
Controller Two embedded NC320i PCIe Gigabit Server Adapters
Storage
Controller HP 8 Internal Port SAS HBA with RAID
Internal storage 36 GB 15 K SAS Hot plug hard drive 3.5”
Optical Drive HP DL320 RoHS DVD RW
Form Factor Rack 1U
Installation Kit HP 5140 DL140 G3

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3 INTERFACES AND PROTOCOLS
This chapter introduces the connection of the product with the entire network, the provided
internal/external interfaces and used signaling / protocols. It enables users to have a full
understanding of the connection modes of the equipment and other products.
The TOPEX MultiSwitch control equipment is the control core in the multiSwitch system, which
also serves as the external interface of the entire system. It supports multiple signaling protocols,
which can provide interactions with other networks such as PSTN, H.323 and SIP.
In addition, the multiSwitch control equipment also provides the Ethernet interface for
connection with the data network.
The TOPEX MultiSwitch control equipment is a multi-protocol entity, which interacts and
coordinates with other equipments in the multiSwitch network via various standard protocols
(interfaces) to perform functions needed in the system together.
The TOPEX MultiSwitch control equipment supports the following protocols:
Call processing protocol – SIP, H.323, ISUP, TUP over IP;
Transmission control protocol - TCP, UDP, SCTP and TCAP / M3UA;
Media control protocol – SIP and H.323;
Service application protocol - RADIUS;
Maintenance management protocol – SNMP.
Figure 6 – Typical application of protocols

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4 SYSTEM SERVICES
The TOPEX MultiSwitch control equipment has powerful services capability. The MultiSwitch
application inquires the MultiSwitch database which communicates with the administration and
configuration web based interface.
Usually the database is installed on a different hardware platform than the MultiSwitch
application, in order to prevent supplementary charge of the MultiSwitch machine.
Although in certain cases, when the data base is not to large, the MultiSwitch application and
the data base are installed on the same hardware structure. The TOPEX MultiSwitch control
equipment can use either a Postgre SQL or a My SQL database.
Figure 7 – Database interconnection
The basic voice service refers to the service of which the major purpose is to ensure normal
conversation between subscribers. It is a service collection instead of a specific service. It includes the
basic call service, basic PSTN voice service and such voice services as the supplementary service
and traditional intelligent service.
Here, the basic call service refers to the local, domestic and international automatic incoming
call and outgoing call services of various terminals. The terminal type includes the ordinary phone
accessed via IAD, Soft Phone and various IP Phones based on H.248, MGCP, SIP and H.323. The
basic call service is provided by the multiSwitch control equipment independently.

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4.1 Call Control Flow
The route followed by a call entered in the multiSwitch equipment is different from call to call.
The call control flow depends on the calls type, if it is IP to IP, IP to TDM, TDM to IP, TDM to TDM or
IP/TDM to a special service. These cases are detailed below:
IP to IP case
Access In => Incoming Class => Optional DB checks => Routing table => Outgoing class =>
Access Out
TDM to TDM case
Incoming Class => Optional DB checks => Routing table => Outgoing class
IP to TDM case
Access In => Incoming Class => Optional DB checks => Routing table => Outgoing class
TDM to IP
Incoming Class -> Optional DB checks -> Routing table -> Outgoing class -> Access Out
IP/TDM to special service
Access In (IP case) -> Incoming Class -> Optional DB checks -> Routing table -> Service |
Hunting | Port
From the multiSwitch input trunk until multiSwitch output trunk, a call must pass through
several tables like Access In, Incoming Class, Routing Table, Outgoing Class, Data Base, Access Out.
Access In Is used only for IP call (SIP/H323).
For TDM calls port is directly assigned to incoming class.
It assign an incoming class for each call according to:
- IP/netmask
- TCP/UDP port
- protocol (SIP/H323)
- Prefix
- Number of digits from DNIS
Incoming class
Each call has an incoming class
Settings from incoming class will be applied to the calls assigned to
it.
Optional DB checks:
- translate ANI
- translate DNIS
- portability
- ANI users
Here you can activate some optional database checks see bellow.
Routing table
It route calls to one of the following:
- outgoing class
- special service
- hunting service
- directly to a specific TDM port
Outgoing class
Settings from outgoing class will be applied to the calls routed to it.
Access Out
Is used only for IP call (SIP/H323).
For TDM calls port is directly assigned to outgoing class.
Specify the IP:Port, Protocol (SIP/H323) and
Transport(UDP/TCP/TLS) where the calls are sent.
Relation between Outgoing Class and Access Out is one to one

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4.2 Client Classes
4.2.1 Description
The client classes are divided in two categories:
•trunks (or junctions) – channels where through the calls from other networks are received or
sent;
•subscribers – the common users CLASS 5 – residential users or prepaid users.
There are two types of MultiSwitches:
•Class 4 MultiSwitch – used for the control of calls from different trunks (IP, PSTN) to VoIP
network ;
•Class 5 MultiSwitch – contains local users (subscribers) and interconnection with other operators
– trunks.
The MultiSwitch developed by TOPEX can act as Class 4 MultiSwitch and also as Class 5
MultiSwitch.