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TOPEX multiSwitch

User manual

2008

multiSwitch user manual

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 companys 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)

TOPEX 2008

multiSwitch user manual


For further details, please visit our website at: www.topex.ro
WEEE Directive Compliance
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.
WEEE Directive

! 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: +4021 408.39.00 or www.topex.ro

TOPEX 2008

multiSwitch user manual

Table of Contents
1

3
4

5
6

BASE 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
SYSTEM 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
INTERFACES AND PROTOCOLS....................................................................................... 17
SYSTEM 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
SIP FEATURES .................................................................................................................... 28
5.1 Security level....................................................................................................................... 29
TOPEX multiSwitch INSTALLATION.................................................................................... 30
6.1 System access .................................................................................................................... 30
6.2 IP address changing ........................................................................................................... 31
6.3 Time and date changing ..................................................................................................... 31
TEXT 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 i voice attendant ..................................... 54
7.11 TELNET commands.......................................................................................................... 56
8
DESCRIPTION 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
9
ACTIVATION 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

TOPEX 2008

multiSwitch user manual

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 IPrelated 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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multiSwitch user manual

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 clients 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

Mounting kit for a 19 rack

Protection key for the front panel

Internal supply cable

Power supply cable

Users Manual

Warranty Certificate

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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

orange LED (labeled with the

symbol) shows the equipment power supply when it lights up;


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
Cache Memory
RAM Memory
Network
Controller
Storage
Controller
Internal storage
Optical Drive
Form Factor
Installation Kit

Dual-Core Intel Xeon processor 5140 (2.33 GHz, 65 W, 1333 MHz FSB)
4MB (1 x 4MB) Level 2 Cache or 8 MB (2 x 4MB) Level 2 Cache
1 GB FBD PC2-5300 DDR2 Fully Buffered DIMMs (667 MHz) 2 x 512 kit
Two embedded NC320i PCIe Gigabit Server Adapters

TOPEX 2008

HP 8 Internal Port SAS HBA with RAID


36 GB 15 K SAS Hot plug hard drive 3.5
HP DL320 RoHS DVD RW
Rack 1U
HP 5140 DL140 G3

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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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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

Incoming class

Optional DB checks:
- translate ANI
- translate DNIS
- portability
- ANI users
Routing table

Outgoing class
Access Out

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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
Each call has an incoming class
Settings from incoming class will be applied to the calls assigned to
it.
Here you can activate some optional database checks see bellow.

It route calls to one of the following:


- outgoing class
- special service
- hunting service
- directly to a specific TDM port
Settings from outgoing class will be applied to the calls routed to it.

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.

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4.2.2 Main Features


The main features of a client class are:
to allow or to restrict calls to certain phone numbers depending on the DNIS Dialed Number
Identification Service;
to allow or to restrict calls from certain phone numbers depending on the ANI Automatic
Number Identification;
to modify ANI or DNIS for a client class the source and destination numbers;
to route the call depending on a client class;
to charge the call depending on the client class;
to limit the calls numbers from or to a certain client class for IN calls and OUT calls. For instance, a
client class can make a certain number of calls and also it can receive a limited number of calls.
For that he will establish the number of received calls from every client class thus the total numbers
of received calls to not be overflow.
Notice: This feature is specially used to the IP traffic in the Class 4 multiSwitches!
to select the channels searching mode on TDM. There are three possibilities:
increasing from channel 1 to channel N;
decreasing from channel N to channel 1;
circular for instance 1 2 N 1 2 ...
where the N parameter depend on the number of the E1 streams used. This feature is specially used
in the SS7 trunks.
0 Warning!
To avoid congestions, it is necessary that the two ends of the connection to have different
searching modes on TDM.
For example, if one user has the increasing mode and the other has the decreasing mode, the
chance of congestion is almost zero.

to establish the list of codecs on the input and output. For example if an operator send traffic with
a voice codec that is not recognized by the destination the call could not be deployed.

4.3 Clients
4.3.1 Description
The clients represent the second concept of the TOPEX MultiSwitch and also an integrated
part of the Client Classes. There are several types of clients:
) SIP SIP users;
) FXS analogical subscribers;
) ANI or postpaid clients identifiable through ANI.
The MultiSwitch has a data base with the ANI codes of his subscribers. When a subscriber
wants to make a call, the MultiSwitch identifies his ANI code in the data base and allows him to
achieve the call.
) Prepaid clients identifiable through PIN code.
It is a similar procedure to the ANI identification. The difference is that the MultiSwitch has a
data base with PIN codes instead the ANI codes. Usually the clients who use this option dial a certain
number to insert the PIN code. If the PIN code is retrieved in the MultiSwitch data base, they can dial
the destination number. If the user dial directly the destination number and he will not be identified on
any options above he will be automatically rejected.
Notice: This option is used when the user can not be identified through any other options above.
) Trunks junctions links with other operators.

4.3.2 Main Features


The clients have several features described below:

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Language - the messages language that the user received from the MultiSwitch. These options are:
English and Romanian.
Client Class - a call from a certain client can be routed and charged different from another depending
on his client class.
Restrictions - a client can have restriction to dial certain numbers or it can be restrict for dialing on
any number - he can only be dialed.
A client can have restrictions for the received calls. He can choose certain users to be not able to call
him.
Client state we have four options for the status of a client that can be:
Active;
Cant call;
Cant be called;
Cant call and cant be called.
Showing ANI - when a client makes a call, there are three possibilities for his ANI number:
ANI is displayed on destination
ANI is hidden on destination
ANI is not present
Translate we have two possibilities: translate ANI and translate destination. The numbers from a
specified client class can be translated into numbers recognized by the destination.

4.3.3 SIP users


It represents a simple category of clients. It can be the usual SIP phone or a computer. We
have several services for this type of users:
User name / Password SIP user name and registration password
Restrictions the SIP user can choose the subscribers who can or can not call him.
Call forward this option allows the forwarding of a call when the user can not answer the call and he
chooses a specified destination to answer the respective call. There are several possibilities when this
option is used:
destination is busy
destination not answers
destination is not available the user is not registered
combination of the all options
Call hold the called part can use this option to switch off the actual call for answering to a more
important call that intervenes. After the important call is ended, the previous call is retaken.
Call wait it is used to switch between several calls at the same time. Instead the Call hold option
where a call must be ended before take over another one, in the Call wait option the calls can be
switched random and it is not necessary to end a call to take over another which is waiting.
Call wake up the client can set up a time when the MultiSwitch will call him. He can choose a
message that the TOPEX MultiSwitch will send to him.
Redial this option allows the user to call the last dialed number just pressing the redial key.
Do not disturb the client can choose a time period when he cant be disturbed. Between the
specified hours or days, any calls addressed to this client will be rejected.
Call back this option allows a user to drop a call and to call back the respective subscriber who
called him.
Call to this option allows the calling of a second number and the connection with the call back
number called.
Call hunting priority - it is used to establish the priority to route a call from a busy destination to
another one. The algorithm must be a non linear one to avoid overcharging of a certain user. In the
case of a linear algorithm, when the destination number is busy, the call will be routed always to the
same user which will be overwhelmed.

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Call hunting when a call is directed to certain user and this one can not answer, the call will be
forwarded to another user. If also the second user is busy, the call is routed to a third user. The order
of routing the calls is established depending on the Call hunting priority parameter.
0 Warning: To accomplish the hunting operation, the users must be in the same hunting group!
Call pick up when a phone rings and the user is not there or he can not answer, another user can
take over the respective call. The user which wants to answer the calls must dial a code from his
terminal to take over the call. Then the first phone will stop to ring and the call is answered by the
second user.
0 Warning: To accomplish the pick up operation, the two users must be in the same pick up group!
Voice mail the service that sends voice messages to a specified mail address. These messages
appear when:
destination is busy
destination not answers
destination is not available the user is not registered
combination of all these options
Video allows the achievement of the video calls
Messages allows sending messages such as: On Line, Busy, Be Right Back, Away, On the
phone, Out to lunch or Appear offline.

4.3.4 FXS users


The FXS users are the usual analogical subscribers. The parameters of these users are
detailed below:
ANI the ANI code of the subscriber;
Client id the identification code for a client from the clients table;
Has allow in option which establish the clients who can call a certain client;
Has deny in option which establish the clients who can not call this client;
Call forward - this option allows the forwarding of a call when the user can not answer and he
chooses a specified destination to answer the respective call. The options for using this service are the
same with those for the SIP users;
Voice mail the service that sends voice messages to a specified mail address. The cases for
messages appearance are the same with those for the SIP users;
Call hunting priority - it is used to establish the priority to route a call from a busy destination to
another one. The algorithm must be a non linear one to avoid overcharging of a certain user. In the
case of a linear algorithm, when the destination number is busy, the call will be routed always to the
same user which will be overwhelmed;
Call hunting when a call is directed to certain user and this one can not answer, the call will be
forwarded to another user. If also the second user is busy, the call is routed to a third user. The order
of routing the calls is establish depending on the Call hunting priority parameter;
0 Warning: To accomplish the hunting operation, the users must be in the same hunting group!
Call pick up when a phone rings and the user is not there or he can not answer, another user can
take over the respective call. The user which wants to answer the calls must dial a code from his
terminal to take over the call. Then the first phone will stop to ring and the call is answered by the
second user;
0 Warning: It is necessary that the two users to be in the same pick up group to realize the pick up
operation!
Call state the status of the call which can be:
idle;
dialing;
alerting;
ringing;

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connected as calling;
connected as called.

Notice: There are other services for the FXS users but we do not detail them here because they are
similar to the SIP users services. These services are:
Call wait;
Call wake up;
Redial;
Call hold;
Do not disturb;
Call back;
Call to.

4.3.5 ANI Users


Also, they are named postpaid clients. They are identified through the ANI number the
number of the calling part.
The specific features of this category of users are detailed below:
Client id the client identification code.
Call back - this option allows to a user to drop a call and to call back the respective subscriber who
called him.
Call to - this option allows the calling of a second number and the connection with the call back
number called.
SMS2Mail this option allows sending SMS messages from a phone terminal to a computer

4.3.6 Prepaid users


The prepaid users are identified through the PIN code. Usually they must dial a specified
number where they will insert the PIN code. The MultiSwitch checks the inserted PIN code in its data
base, and if it is retrieved, the client will have the permission to make a call to the destination he
wants.
The parameters of the prepaid users are:
- Client id the client identification code;
- Activation date the date when the credit was activated;
- Initial credit the initial value of the prepaid client credit before any recharging;
- Current credit the current value of the prepaid client credit;
- Expire date the date when the current credit expires in the format yyyy-mm-dd hh-mm-ss;
- Distributor code id the identification code of the distributor of the recharging card.

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4.3.7 Trunks
They are channels where through the calls from other networks are received or sent.
There are two types of routing for trunks: when traffic is sent through IP or through TDM. The options
are:
Routes in / Routes out the IN and OUT routes for TDM traffic. The parameters are:
port the switch port (slot)
client id the id from the clients table
IP Routes IN / IP Routes OUT the IN and OUT routes for IP traffic. The parameters are:
IP the IP Address or the IP mask
port number of the port on which the call is made
proto the protocol used for sending the call. It can be SIP or H 323
clientid the id of a client
0 Warning! For IP traffic, the IP parameter for the IP Routes OUT can be only the IP Address and
for IP Routes IN it can be the IP Address or the IP / Mask.

4.4 Destinations
After the Client Class and Client identification are made, the next step is the identification of
the destination route. This feature of the TOPEX MultiSwitch shows the way routing is made.
The routing operation is based on several parameters described below:
Client Classes Id the identification code from Client Classes
Prefix the destination number of the call
Action the operation made by the switch depending on the destination number. There are several
actions that can be accomplished:

the call can be routed to a subscriber

the call can be routed to a Client Class if the destination is an id from the Client Class

the call can be routed to a service if the destination is a service number

the call can be routed to a hunting group

the call can not be routed

the call can be routed on LCR if the destination is lcr from the LCR table

Tax type the way how the call is charged. There are several methods for charging calls developed
by the TOPEX MultiSwitch:

charging through the taxid pulses

charging on time, where taxid is the id on the table tax on time

charging depending on the geographical areas (labeled from 0 to 9).

Fork the number of forked calls to reach the destination.


Mod the way the call is routed depending on call parameters. The options are:

normal routing

it is searched the destination with he best ASR Answer Seizure Ratio the number of
successful answered calls divided by the number of all the tries to call

it is searched the destination with the best ACD Average Call Duration

Max connection time after this time is out of date the call is dropped.
Max digits the maximum number of digits for a destination. It is set to 20. All the digits sent beyond
this number will be ignored.
Digits time out this option interfere when the allocated time to dial the digits has expired.

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4.5 Tariffs
Usually there are only three tariffs depending on time: for week days, for week-end and for
holidays. The week days are labeled from 1 to 7 and the holidays are labeled 8. The two parameters
are: day and tariff.
We have three types of charging:

Tax id pulses charging this method establishes a fix charge for the client. For a tax id pulse, a
time interval is set. The conversation will have a certain number of those time intervals and the
correspondent number of tax id pulses. At the end of the conversation the cost of the call is
calculated through the multiplication of the number of tax id pulses and the cost for a tax id pulse.

Tax on time charging this option sets the time charging, where the tax id is an id from the tax on
time table. There are two time intervals: t1 and t2. Until a specified moment of the call t1 the
charge is made with a tax id and after this interval the charge will be made with another tax id
corresponding to the t2 interval.
For instance, the client makes a call. He is charged when the other end of the connection answers.
At the answer the charge is at 1 minute. After this period he will be charged at 30 seconds.

Tax zone special charging depending on the geographical area. There are 10 geographical
areas labeled from 0 to 9. The tax zone parameters are:

tariffs id is the tariff from the Tariff table


start time represents the time when the tariff for a certain area begins
end time represents the time when the tariff for a certain area ends
t...z0 ... t...z9 is the time in milliseconds for a tax pulse in area number 0 ... 9

4.6 Service instances


Represents the services achieved by the TOPEX MultiSwitch. The parameters of this table
are:
Name the name of the respective service.
Service Type the type of the service. There are several types of services such as: prepaid,
postpaid, postpaid SSD, composite, account management, subscriber.
PIN length the dimension of the PIN code the number of digits.
Digits time out this option interfere when the allocated time to dial the digits has expired.
Ask language the client receives a message that tells him to select the desired language such as:
For English press key 1, For Romanian press key 2.
Play tone before dest instead receiving a message, the destination receives a tone.
Nr error the number of errors for a call. When this number is reached, the call is dropped.
Drop call if no ANI if the ANI number is not recognized, the PIN code is asked to the user. If it is not
recognized the call is dropped.
Minute beep file it is a beep message that appears at every talk time minute.
Play credit shows the available credit of the client.
Play talk time shows the talk time of the client calls
Ask new dest on call end this option establishes if you want to make another call after the current
one was ended.
Drop pay phone play file if it is defined, the calls from the pay phones are dropped.
Welcome file the welcome message received by the client from the TOPEX MultiSwitch.
Transfer on error number it implies the Nr error parameter. Usually when this error number is
reached the call is dropped. Through this option the call will not be dropped anymore when the error
number is reached. The call will be transferred to another destination chose by the client.

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Drop call on not credit for dest When the client has not enough credit to reach the destination, he
is asked if he want to call another destination. If the credit is to low for any destination, the call is
dropped.

4.7 CDR Call Detail Records


This option shows the detail records of the calls made and received by a client. It has several
parameters described below:
Number the number where the client called or from where he was called.
Date / Time the date and time of the calls accomplishment.
Duration the time elapsed for every conversation made or received by a client in the format hh-mmss.
Client_id the identification number of the client
Client_type the type of the client. It can be SIP user, FXS user, ANI (Postpaid) user, Prepaid user or
a trunk.
PIN the PIN code of the client.
Calling_category the category of the called client.
Calling_number_type the structure of the called client number.
Forward_number the number where the call was forwarded if the initial destination was not
available for answer.
Forward_reason the reason why the call was forwarded to another destination. The call is
forwarded if the client is busy, he dont answer the phone or he is not available (is not registered).

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SIP FEATURES

TOPEX multiSwitch carrier grade softswitch is based on a power full SIP stack which contains
several modules like:
SIP Registrar allows the registration of SIP users (devices) in TOPEX multiSwitch
SIP User Agent - endpoint SIP interface for IP to IP or IP to TDM calls
SIP Proxy Server allows the calls between SIP users
The events supported b

TOPEX multiSwitch equipment covers the complexity of the inferior level of the network in
order to generate integrated complex services.
The system uses a large dimension database as a data platform and offers several back-up
modes for the tax files, in order to guarantee data tax guarantee and the complete accomplishment of
the actual requirements in telecommunications. Also the system features a high reliability.
Through CLASS 5 services supported by TOPEX multiSwitch are:
Call Waiting
Call Hold/Retrieve
Do Not Disturb
Call Forwarding (No Answer, Busy Number, Unconditional)
Call Back
Call Hunting
Call Forking
Multiple Subscriber Aliases
Anonymous Call Rejection
Access Control Lists for Inbound and Outbound
Call Baring Incoming/Outgoing
Suspend Subscriber
Calling in / out intra-group and out-group
CLIP / CLIR
Web Agenda
Portal for customer to change its features
Centrex Emergency call
Centrex Call forward
Centrex Call Pick-up
Centrex Call Hunting
Voice Mail
Voice Mail 2 e-mail
Meet-me conference
SIP Video call
The system offers the complete set of telephony services to the clients connected from the
private IP networks, without no limitation of the IP address diagram.

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TOPEX multiSwitch can support up to 3000 simultaneous calls with H.323 signaling
and 10.000 simultaneous calls with SIP signaling.

5.1 Security level


Regarding the network security, the system supports setting different levels of
administration, thereby preventing the achievement of unsafe operations on the equipment.
For the users who attempt to illegally access the system, the maintenance terminal provides a
connection to real-time information on any unauthorized attempts to access the system.
For connections that may threaten the safety of the equipment, such as repeated failed logins,
IP addresses unsafe or very high traffic will be available prompts alarms.
In addition, the TOPEX multiSwitch offers some tools to test security, which will automatically
reject applications from unauthorized connection and report them to the system.
For user security, the system TOPEX multiSwitch system offers an authentication code
equipment for each device that has valid permission to access.
The module contains an analysis of data traffic, which may help to highlight the abnormal
behavior of the network or any attempts of fraud.

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TOPEX multiSwitch INSTALLATION

TOPEX multiSwitch installation must achieve several stages:


- Physical installation connection of Ethernet cables
- Power supply
- System connection and configuration
First of all the multiSwitch server must be connected to the data network through an Ethernet
cable. Then power supply the equipment through the on/off button located on the back side of the
equipment.

6.1 System access


The connection to the system can be made directly (local connection) or remote through an
Ethernet connection from a PC.
In the case of local connection the access to the multiSwitch is made through the command:
ssh tpxadm@<IP address of TOPEX MultiSwitch>
The remote connection is made through SSH connection from a Linux PC or through an
application which performs SSH under a different operating system (for instance Putty).
When you use Putty make the settings according to the figure :

Figure 8 Ethernet connection through Putty


In the Putty main window check the SSH option, insert the IP address of TOPEX MultiSwitch,
insert the number of port 2212 and press the Open button in order to access the TOPEX
MultiSwitch structure.
The authentication data required for the login are:
-

SSH port 2212;

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-

User name tpxadm;

Password u53rp455.

The direct connection as root is not allowed. You must enter to the prompt the su -
command after you are logged in, and insert the 5y5t3mp455 password. The firewall is open by
default.
For the serial connection use a serial cable which is inserted in the DB9 connector from the
back panel of the TOPEX multiSwitch enclosure and in the serial connector from a PC.
Use the HyperTerminal from Windows with parameters 115.200, 8, N, 1 in order to establish a
serial communication with the TOPEX multiSwitch.

6.2 IP address changing


After the root connection with the su- command and the password 5y5t3mp455, the client
must change the IP address of the equipment.
The IP address modification is made by accessing the file located at /etc/conf.d/net which
can be edited with nano or mcedit Linux applications.
The file must look like the example below:
config_eth0=( "192.168.1.20 netmask 255.255.0.0" )
routes_eth0=( "default via 192.168.1.2" )
The DNS server is set in the /etc/resol.conf file.
After the files content modification you must restart the network interface through the
command:
/etc/init.d/net.eth0 restart

6.3 Time and date changing


After the root connection on the equipment the main application is stopped through the
command:
/etc/init.d/softswitch stop
Here you have two options for the time and date settings:
a Set the time and date and is written in BIOS:
date MMDDHHMMYY
/sbin/hwclock --systohc
b Time zone is set and it is synchronize with the remote server pool.ntp.org
ln -sf /usr/share/zoneinfo/"timezone" /etc/localtime
/etc/init.d/ntp-client restart
The multiSwitch application is then restarted:
/etc/init.d/softswitch restart

The configuration of multiSwitch equipment requires two steps:

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1 Text files configuration
2 Web based interface configuration
The internal multiSwitch structure is based on text files configuration which contains several
parameters of the equipment that must be changed according to each client requirements. The text
files includes several features and services which must be enabled from the configuration files.
The web based management interface is used to activate the final configuration at the end
user level.
In the next chapters we will describe the text files and the web based interface with its features
and parameters.

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TEXT FILES CONFIGURATION

TOPEX multiSwitch equipment central application is based on text files which contains the
main parameters of the system. These files are .cfg type files and can be accessed from the Linux
file structure of the equipment.

7.1 voip.cfg configuration


In the voip.cfg configuration file are set several parameters and protocols referred to VoIP
communication. The location of this file is /mnt/app/cfg/voip.cfg. In this file can be set the VoIP
protocols, DTMF digits, VAD, codecs.
The main lines of configuration file are:
dtmfRTP 101 100 1
dtmfINFO 1/0
The line includes the DTMF digits according to the RFC 2833 standard and the DTMF digits
sent through INFO signaling.
vad 0
Activation / deactivation of Voice Activity Detection the process of separating conversational
speech. The primary function of a VAD is to provide an indication of the presence of speech in order to
facilitate speech processing as well as possibly providing delimiters for the beginning and end of a
speech segment.
audio_codecs 18 20 8 20 4 20 0 20
The line above describes the voice codecs used by the equipment and the packets time rate.
Each codec has assigned a standard number. For instance:
18 corresponds to G.729
8 corresponds to G.711 A law
4 corresponds to G.723.1
0 corresponds to G.711 law
Between the codecs name is included the time rate the time interval between two voice
packets.
The following lines starting with voipgw, publicvoipgw, msp and forkmsp are not used on
multiSwitch.
The last two lines from the voip.cfg configuration files are related to H.323 VoIP protocol.
Initially multiSwitch has implemented only SIP protocol. For the H.323 support these two lines must be
activated.
h323 192.168.1.1 9010
forkh323 /mnt/app/bin/h323_apc -p9010
The first line includes the activation command for H.323 VoIP protocol including the IP
address of the equipment and the communication port. The second line runs the core application of
H.323 module.

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7.2 group.cfg configuration


Group.cfg configuration file establishes the cards definition for the system. In the
multiSwitch case we do not have a physical card and it is used a virtual VoIP card.
The first line establishes if the multiSwitch equipment is master or slave. Value 1 is for
master and 0 for slave.
m1
In the case of standalone multiSwitch equipment, it is always set as master. When it is
involved a larger architecture with media or signaling gateways and softswitches, the multiSwitch is set
as master and the gateways are set as slave.
Slave means another main system application that is configured to accept connections from
master. It can run on the same machine or a remote machine.
So if is the same application then you need to have the same directory structure as from app
directory. If is running on the same machine (ex slave RTPproxy) you need to duplicate the app
directory; usualy it is done by making a copy of 'app' to 'apprtpproxy' for example and after that making
changes on cfg directory cards and ports are installed only on master.
The next line includes several parameters related to card group, group number, type, etc.
g 0 2 195.114.116.235 9000 1 0
The significance of each field is described below:
field 1: g = means group line. For each group must be added in card.cfg at least one or more
cards. For each card you need to install also the ports each group can have up to 16 cards
installed. Cards number will be in range [(gr_nr*16) ... ((gr_nr*16)+16)]. For instance if gr_nr =
3, cards will be in range [3*16 ... 3*16+16] = [48 ... 64]
Note: if gr_nr > 12 you cannot have voip lines attached to group line; this mean that you need to add
them in slave configuration
field 2: group number; on versions over 4.1.90 you can see the number of groups with which
the main application was built by running centrala v and look for NRGROUP value
field 3: group type: 0=FXS|FXO|GSM|CDMA...; 1=E1; 2=VoIP|RTPproxy
field 4: connection IP to slave. If IP is different than 0.0.0.0 or 0 means that the master will try
to connect to this IP and port from next field; so in this case you don't need a second line
starting with voip; this line will appear in slave group.cfg. Keep in mind that always the
master is trying to connect to slaves defined here in this file
field 5: connection Port to slave
field 6: is significant only if it is set to 1 which means that is a Eones group not used for
multiSwitch
field 7: reserved for future development
For multiSwitch must be defined the rtp_ip parameter instead of a physical VoIP card.
rtp_ip 0 192.168.0.0/16 192.168.1.193
rtp_ip 0 0.0.0.0/0 89.38.173.23
The format of rtp_ip field is: rtp_ip <group_nr> <ip_class(format: ip/netmask_len)> <rtp ip
used for this class>.

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When the group type is 2 the following lines will appear:
voip 0 2 127.0.0.1 9081 fork /mnt/app/bin/rtpproxy -p 9081
voip 0 2 195.114.116.235 9671 fork /mnt/app/bin/mspd -p 9671 --trace-cmd -v --mem
16 --gw-mac 00:19:AA:D2:3C:25 -m 00:52:C2:40:3E:43 195.114.116.239 --log
/mnt/app/out/%d-%m-%y_mspd.log The fields significance is described below:
field 1: voip always
field 2: group number to which this line refer
field 3: group type, always 2
field 4: connection IP to rtpproxy/mspd
field 5: connection port to rtpproxy/mspd
field 6: fork always
field 7: path to rtpproxy/mspd
Next fields are option passed to rtpproxy/mspd application

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7.3 exec.cfg configuration


The exec.cfg configuration file includes several parameters related to debug process of
protocols, data base type, RADIUS interconnection, RTP Proxy, DTMF tones, softswitch name or
ASCII protocols.

7.3.1 Debug parameters


In this section are detailed the parameters referred to the debug of protocols and services. For
debug configuration the following parameters are available:
set_cfg_debug 0-5
set_telnet_debug 0-5
set_cctl_debug 0-5
set_app_debug 0-5
set_alr_debug 0-5
set_oam_debug 0-5
set_db_debug 0-5
set_dbp_debug 0-5
set_vm_debug 0-5
set_dlg_debug 0-5
set_srv_debug 0-5
set_gsm_debug 0-5
set_fdw_debug 0-5
set_connect_debug 0-5
set_acc_debug
set_serial_debug 0-5
set_q921_debug 0-5
set_q931_debug 0-5
set_db_config_debug 0-5
set_h323_debug 0-5
set_debug_pbx_h323 0-5
set_r2s_debug 0-5

// configuration
// telnet
// call control
// application
// alarms
// OAM
// db forked clients
// db pool
// voice mail
// dialogic
// servicii pbx
// gsm not used on multiSwitch
// fdwatch
// Connect Matrix debug not used
// SS7 ACC debug not used
// Serial debug
// ISDN Q921 debug not used
// ISDN Q931 debug not used
// Database Config debug
// H323 debug from H323 log
// H323 communication debug from centrala log
// R2S debug not used

The notation 0-5 means that the respective parameter can take any integer value in this
range. Each number from 0 to 5 has a different signification:

0 = no debug
1 = ERROR debug
2 = ERROR + WARN debug
3 = info debug
4 = full debug
5 = verbose debug

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7.3.2 Data Base activation


The following lines are referring to the process poll activation / deactivation
establishes the database type. The supported format for data base by TOPEX multiSwitch
are PostGre SQL, My SQL and Microsoft SQL. The value 0 means deactivation and 1
activation.
pgsql_billing 0/1
mysql_billing 0/1
mssql_billing 0/1
pgsql_prepaid_pool 0/1
pgsql_ani_pool 0/1
pgsql_simserver_pool 0/1
rtptx_pool 0/1
rtprx_pool 0/1
dns_pool 0/1

7.3.3 RADIUS activation


The next lines includes several parameters related to RADIUS AAA (Authentication,
Authorized, Accounting).
TOPEX multiSwitch can be interconnected with an external billing system. At this
moment multiSwitch supports 3 billing dictionaries: TOPEX, Quintum and Mind.
radius_billing 1

// 1 for activation and 0 for deactivation

radius_dictionary 0

// 0=Topex; 1=Quintum; default 0

tx_access_request 1

// sends or not access request for authentication

tx_accounting_start 1 // sends or not accounting_start for billing. The value 1


indicates activation and 0 deactivation. You can use billing also without activate
accounting_start parameter. You can activate only accounting_stop.
radius_auth cli
// establishes the information sent in the access_request
packet in the username field for authentication on trunk input calls.
The possible values for radius authentication on trunks are:
cli = Caller ID, ANI
ip = IP source of the call
class_name = source class (direction) of the call
radius_sip_user_auth ip
// establishes the information sent in the
access_request package in the username field for authentication on the calls
initiated by the SIP users
The possible values for radius authentication on SIP users are:
cli = Caller ID, ANI
ip = IP source of the call
The command below command enables / disables the CDR (Call Detail Records) writing in
Mind format assuring the interconnection with an external billing system developed by Mind.
mind_cdr 1/0

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7.3.4 RTP Proxy


The lines below establishes the number of RTP Proxy ports opened for the communication.
The RTP Proxy application assures the pass of RTP packets and signaling through the multiSwitch
main application.
rtp_proxy_range 10
rtp_proxy_port 15000
The command above indicates the number of the first RTP Proxy port opened for
communication. If the number of used ports is 10 and the first port is 15000, the used ports for RTP
Proxy communication will be 15000, 15001, ..., 15009.
rtp_ip 192.168.0.0/16 192.168.1.107
rtp_ip 0.0.0.0/0 89.38.123.34
The rtp_ip parameter is used for calls through RTP Proxy from multiSwitch main application.

7.3.5 Other parameters


This section allows sending DTMF tones, T.38 and pass-through fax, activate console,
voicemail or telnet activation.
dtmf_off 0/1
When it is set the value 0, DTMF is sent at tone ON, and if it is set the value 1 send DTMF
at tone OFF.
t38 0/1
This parameter is used to activate or deactivate T.38 fax relay. The activation of T.38 is made
through the value 1.
ITU-T Recommendation T.38 describes the technical features necessary to transfer facsimile
documents in real-time between two standard Group 3 facsimile terminals over the Internet or other
networks using IP protocols. The Recommendation allows the use of either TCP or UDP depending on
the service environment.
TOPEX multiSwitch supports both T.38 and G.711 inband fax. In the case that T.38 fax relay
is activated and the equipment receives a G.711 inband fax packet, the multiSwitch equipment will
automatically switch to bypass mode (G.711 inband).
For disabling the T.38 fax is used the value 0 in the command above.
block_cons 0/1
For this command you must always use value 1
taxstarttime 0/1
The parameter taxstarttime is used to establish the beginning of calls tax in billing. The value
1 put connect time of calls in billing and value 0 put end time of calls in billing.
In this file you can also enable or disable the voicemail service which allows the users to listen
missed calls. Voicemail messages are stored on the equipment hard disk and are available for each
user who activates this feature.
voice_mail 0/1

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VoIP DTMF tones are activated through the line below:
voip_dtmf 0/1
The configuration file supports changing the telnet port for the communication with the
equipment:
telnet 23
Also you can activate or not the remote telnet access. Value 0 rejects remote telnet access
and value 1 accepts telnet remote access.
remote_telnet 0/1
The line:
name topex
suggests the gateway name. If it is set this name precede the name of cdr, log and alr files from the
/mnt/app/out/ directory.
The line
sip_trying_timeout 1000
indicates the timeout for waiting a provisional answer from destination. The time is measured in msec
and the default value is 1000 msec 1 second.
parameter
will
play
the
content
of
the
The
play_accessin
/mnt/app/raw/acl_reject_<codec>.<language> file in case that the call is rejected from accessin
mismatch.
play_accessin 0/1
This parameters allows playing a voice message like: Sorry you dont have the right to
access this address for the users who are not in the Access In list.
Also the exec.cfg file allows to define several (by default 10) emergency numbers like 112
or 911 which will be reserved for these special destinations.
emergency_dnis 112,911,961
To set the dimension of the log file is used the command below. The default value is 1GB
(1000000 KB). After the log exceeds this value, in the data base will not be written anything. You need
to delete or move the file in order to allow another log file creation.

db_route 1

Loading configuration from database. Reads cards


configuration from database. By default the value is 0.
Read routes configuration from database. By default the value
is 0.

db_type 2

Must be always 2

maxsysincallrate 10000

Maximum simultaneous incoming calls supported by the


system. By default is 10000
Maximum simultaneous outgoing calls supported by the
system. By default is 10000

db_card 1

maxsysoutcallrate 10000
consysincallrate 10000

Default is 10000

consysoutcallrate 10000

Default is 10000

set_asci_sip 1

ASCII protocol, now default it is enabled. Enable ASCII


protocol on SIP interface

set_asci_h323 1

Enable ASCII protocol on H.323 interface

billing_fields_number 255

TOPEX 2008

Specifies the number of fields written in billing. The default


value is 43. In this case you will have 43 fields written in billing.
The value 255 writes all the fields available. You can see the
range of configurable fields on telnet with command "billing
fields number".

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7.4 sip_pbx.cfg configuration


The configuration file sip_pbx.cfg contains several parameters dedicated for the main
application debug, SIP users registration, QoS parameters or transport protocols.

7.4.1 Debug parameters


The debug parameters of the multiSwitch main application can take values between 0 and 5.
For values higher than 5 are reserved for developers only.
debug 6
The values 0 to 5 have the following advantages:
Value

Significance

No debug

Error only

Error + Warning

Error + Warning + Info

Error + Warning + Info + Debug

Error + Warning + Info + Debug + Verbose

The parameters used for SIP debug are detailed in the table below:
Debug parameter name

Significance of debug parameters

agent_debug

Debug parameter used for User Agent calls

register_server_debug

Debug parameter used for SIP users registration to


multiSwitch

register_client_debug

Debug parameter used for the softswitch registration to


another proxy

database_config_debug

Debug parameter for loading configuration from database

notify_debug

Debug for SUBSCRIBE / NOTIFY / PUBLISH

proxy_debug

Debug parameters for proxy calls

options_debug

Debug for OPTIONS requests

queue_debug 2

Debug parameter for internal queues

console_debug

Debug for console. 0 is disabled and 1 is enabled. The default


value is 1.

file_debug

Debug for files. 0 is disabled and 1 is enabled. The default


value is 1.

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The dimension of the log file is also established in this file through the parameter
max_log_size. The dimension is in kilobytes and the default value is 1GB (1000000 KB). Make sure
to put a dimension under 2GB, which represents by default the maximum file size on many Linux
systems.
max_log_size 1000000

7.4.2 Proxy and register parameters


The sip_pbx.cfg configuration file contains also parameters related to the SIP operational
modules of the equipment.
The transport parameters are detailed in the table below:
Transport
parameter name

Significance of transport parameters

tcp

Enable or disable TCP protocol (Transmission Control Protocol) for


transport. 0 = disabled, 1 = enabled. The default value is 0

tls

Enable or disable TLS protocol (Transport Layer Security). 0 =


disabled, 1 = enabled. The default value is 0. It is a cryptographic
protocol which provides secure communication through the
multiSwitch

sip_udp_port

Established the port number of the UDP SIP communication. The


default port number is 5060

sip_tcp_port

Established the port number of the TCP SIP communication. The


default port number is 5060

sip_tls_port

Established the port number of the TLS SIP communication. The


default port number is 5061

The registration activation of SIP users is made through the register parameter. The SIP
users can be registered from several locations like data base, files, memory, etc. The parameter can
take values between 0 and 4. Each significance is detailed below:
Value

TOPEX 2008

Significance

Not used default value

Postgresql database, fork process for each request;


deprecated

Load user settings to memory from text file

Postgresql database process poll; deprecated

Load user settings to memory from database

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The register and proxy parameters are detailed in the table below:
Proxy parameter name

Significance of register and proxy parameters

update_location

Update online users memory status to database table sip


location
Value 0 = disable, 1 = enable and the default value is 1

proxy

Enable the proxy parameter of multiSwitch. The values


available are 0 = disable, 1 = enable. This parameter must
be always enabled!

record_route 1
register_auth 0
invite_auth 0
default_register_expire 600
default_subscribe_expire 600
default_publish_expire 600
forced_register_expire 600
forced_subscribe_expire 600
forced_publish_expire 600
tos

heartbeat 9001

redundancy client/server
192.168.1.193:8001
user_redirect 0
register_client_random 0

database_row_chunk 50

user_agent
max_call_time 3600
session_expires 1800
min_session_expires 90
[credentials]
192.168.1.100 306 306

TOPEX 2008

Must be always enabled. If enabled all dialog messages


will go through the SIP proxy
0=disable, 1=enable, default 1
Allows the register of SIP users. By default is disabled.
0=disable, 1=enable, default 0
Authenticate incoming invite request on UA
0=disable, 1=enable, default 0
Timer in seconds. It is used if the incoming
REGISTER/SUBSCRIBE/PUBLISH don't have an expire
header/parameter in request
Timer in seconds. It forces the expire refresh interval to
value configured here. It has higher priority than
default_xxx_expire
Type Of Service, integer value, default 0. The ToS
parameter represents a field in an IP packet header that
specifies the service level required for the packet
Remote heartbeat UDP port where the KEEP_ALIVE
messages are sent
0=disable sending of KEEP_ALIVE messages
default 0
The same port must be configured also in heartbeat
application at app_port or second_app_port
Configure the ip:port of server redundant equipment. One
equipment must be client and the other one must be server.
A TCP socket is used; one equipment is server and the
other one is client
Enable a SIP user to redirect (forward) a call from SIP
phone instead using call forward setting from SSW
0=disable, 1=enable, default 0
Enable the random registration of clients
0=disable, 1=enable, default 0
Activates the number records read from the data base at
once. For instance the records can be sipusers,
sipuseralias, etc
It must be an integer number between 10 and 500. The
default value is 50
String value, default version of centrala application (run
centrala -v to see version)
Max call time for proxy calls. If this value is reached the
proxy will cut the current call. The time is set in seconds
0=disable; default 0
The time in seconds after which a session expires. It must
be in the interval 90 and 7200. The default value is 1800
The minimum time in seconds after which the session
expires. It must be in the interval between 90 and 7200. The
default value is 90
Credentials used for outgoing INVITE authorization on User
Agent
- realm(ip/dns_name) user_name password

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For multiSwitch equipments with more than one network interface (for instance one for local
class 192.168.0.0/16 and one for public class) is used the [interface_ip] parameter.
Here can be configured different IP/DNS_name for each source/destination IP class. It is also
useful for equipments behind NAT; in this case you put signaling IP the IP of NAT and bind IP the IP of
network interface used in that NAT.
Note: Remember that always last rule must have class 0.0.0.0/0, application is searching from up to
down and must always must one class!
If you don't have a dns name for a specific ip address you can omit dns name from config. You
can use "interface ip" telnet command to check the values read by main application from the file into
the memory and see also the fd created for each interface.
In case of redundancy only the master will create sip udp sockets when virtual ip is up. If
centrala is slave the sip udp sockets will be closed.
The line bellow is for calls from private class 192.168.0.0/16 received on local network card:
192.168.0.0/16 192.168.1.1 192.168.1.1 local.turu.ro
The next line is for calls from public class 80.27.127.0/24 received on local network card (NAT
case)
80.27.127.0/24 80.27.127.10 192.168.1.1 public.turu.ro
The line bellow is for calls from class 0.0.0.0/0 received on public network card
0.0.0.0/0 89.249.83.193 89.249.83.193 public.turu.ro
An important parameter is the country code used for example if you want to add a country
code for call initiated. The syntax format is:
add_country_code 0[1-9][0-9]{4,} 004
By the SIP users to destinations on UA (other that SIP users). In this way you can keep in
routing table only prefixes with country code (E164 format). This change is also called E164 format
number conversion.
The ani_override_display parameter is used if enabled ANI info received from CCTL (call
control flow) will replace presentation info. It is enabled by default because most of the carriers don't
send this info, except British Telecom
# enable=1/disable=0; default 1
ani_override_display 1

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7.5 register_users.cfg configuration


The register_users configuration file located in mnt/app/cfg/register_users.cfg allows the
registration to one or more SIP registrar servers It can be used also as a testing tool for SIP registrar.
Allows the register to a alternative SIP server in order to test the first one functionality.
For instance to register a SIP phone with the number 301 the following parameters must be
inserted.
[301]

User name

password=301

Password

first_proxy=mysippbx.ro

ip/dns_name of first server to which the client try to register

second_proxy=89.38.11.22

ip/dns_name of second server to which the client try to


register. In case it does not succeed to the first one

expires=120

Registration time (in seconds) offered in REGISTER message


to the server. Registration time used is the one received from
server; depending of the server configuration registration time
can be the one sent in REGISTER or any other value

nat_refresh=50

Time interval in seconds at which the client send keep alive


messages to the server in order to keep NAT connection
open recommended value is between 20 and 60 seconds

7.6 public_private.cfg configuration


In this file can be made the calls filtering made by SIP users according to the public and
private IP address. Also it allows defining a maximum number of concurrent calls for each pair of
classes public and private.
If this file is missing from the configuration in text mode (exist also the mode with configuration
from the data base) causes the calls reject on SIP with the message 403 Forbidden.
Bellow it is an example of a public_private.cfg file content:
class1 89.38.188.0 24 192.168.0.0 16 100
class2 172.27.0.0 16 192.168.1.0 24 100
The fields significance is detailed in the table below:
Field 1

Singular identification name. Can be any name but it must be unique

Field 2

Public class

Field 3

Netmask length for the public class

Field 4

Private class

Field 5

Netmask length for the private class

Field 6

Maximum number of concurrent calls

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7.7 trafic.cfg configuration


Trafic.cfg file contains parameters regarding logs activation on HDD, reroute release cause,
translate release cause or test calls. The file contains several lines which must be used with the
default values like:
asr 30
The logs activation on HDD is made through the line below which includes 4 fields with the
next significance:
debug 1 0 2000 1
1

activation / deactivation log file

logging file start port

2000

logging file end port

Log file write on HDD

When a call is rejected is must be rerouted on an alternative route. For this it is used the next
parameter:
rerouteoncause 34 1/0
The first parameter signifies the Q.850 cause received from destination. The second
parameter enable/disable reroute call on cause specified.
If a call is rejected with a certain release cause and you want to send to the destination
another cause you must use the translate_cause parameter:
translatecause 133 34
The first parameter represents the Q.850 cause received from destination. The second
parameter represents the Q.850 cause sent to the source of the call.
In order to enable / disable a test call generator complete the lines below as follows:
run 0
pause 10
con 30

// connection time

777 98 98 0

//must be created the SENDCALL direction

traffic_run 0 4 test
traffic_idle 15 1
traffic_sel 4 5
traffic_con 1 5
traffic_call 777 100
traffic_call 777 100
traffic_call 777 100
traffic_call 777 100

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7.8 RADIUS activation


RADIUS (Remote Authentication Dial In User Service) is a networking protocol that provides
centralized access, authorization and accounting management (AAA) for people or computers to
connect and use a network service.
It is a protocol for carrying authentication, authorization, and configuration information
between a Network Access Server which desires to authenticate its links and a shared Authentication
Server.
The key features of RADIUS protocols are:
Client/Server Model - A Network Access Server (NAS) operates as a client of RADIUS. The
client is responsible for passing user information to designated RADIUS servers, and then
acting on the response which is returned.
RADIUS servers are responsible for receiving user connection requests, authenticating the
user, and then returning all configuration information necessary for the client to deliver service to the
user. A RADIUS server can act as a proxy client to other RADIUS servers or other kinds of
authentication servers.
Network Security - Transactions between the client and RADIUS server are authenticated
through the use of a shared secret, which is never sent over the network. In addition, any
user passwords are sent encrypted between the client and RADIUS server, to eliminate the
possibility that someone snooping on an unsecured network could determine a user's
password.
Flexible Authentication Mechanisms the RADIUS server can support a variety of methods
to authenticate a user. When it is provided with the user name and original password given by
the user, it can support PPP PAP or CHAP, UNIX login, and other authentication mechanisms.
Extensible Protocol - All transactions are comprised of variable length Attribute -LengthValue 3-tuples. New attribute values can be added without disturbing existing implementations
of the protocol.
TOPEX multiSwitch uses RADIUS protocol for the interconnection with an external billing
system. For instance, at this moment, multiSwitch RADIUS interface supports three types of RADIUS
dictionary TOPEX, Quintum and Mind.
To activate the RADIUS interconnection with an external system you must configure first the
/mnt/app/cfg/exec.cfg configuration file. The lines which must be modified are detailed in the table
below and also were presented in chapter 8.3.3. Radius activation.
Radius parameter

Significance

radius_billing 1 radius_billing.cfg

Enable / disable RADIUS AAA main pool.


The last field is the name of the configuration
file used by this pool.

radius_billing_alt 1 radius_billing_alt.cfg

Enable / disable alternative pool for AAA


RADIUS.
The last field is the name of the configuration
file used by this pool.

radius_dictionary 0

0 = TOPEX; 1 = Quintum; default 0

tx_access_request 1

Send
or
not
access_request
for
authentication. The possible values are 0/1

tx_accounting_start 1

Send or not accounting_start for billing.


The possible values are 0/1. The billing can
be made also without sending accounting

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start just only with accounting stop
radius_auth cli

Establishes the information sent in the


access_request packet in the username
field for authentication on input calls on
trunk.
The possible values are:

radius_sip_user_auth ip

cli = caller id, ANI

IP = source IP of the call

class_name = the
(direction) of the call

source

class

Establishes the information sent in the


access_request packet in the username
field for authentication initiated by SIP users.
The possible values are:
-

cli = caller id, ANI

IP = source IP of the call

After the exec.cfg file is configured, completing the lines presented above, you must
configure the /mnt/app/cfg/radius_billing.cfg file. The lines included in the radius_billing.cfg
configuration file which must be modified are detailed in the table below:
Parameter name

Significance

debug 2

0= no debug, 1=minimum debug, 2=full


debug
Radius server IP address

radius_server 192.168.1.11
dictionary 0

Must be the same dictionary as exec.cfg

auth_port 1812

UDP port for authentication of radius packets

acct_port 1813

UDP port for accounting radius packets

secret 99topexSECRETqutex11

Shared secret between NAS and RADIUS


server. In radius server the same value of
secret must be configured for this client
Value in seconds for waiting response from
RADIUS server
The maximum number of repeated requests
before to give up
The network interface IP address used for
sending radius packet

timeout 4
retries 3
NAS_IP 192.168.1.50

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7.9 Data Base Billing


TOPEX multiSwitch equipment support three types of data base PostGre SQL, MySQL and
Microsoft SQL. Each variant of data base must be configured according to the corresponding
parameters.
The number of fields from billing is variable and can be configured from the exec.cfg
configuration file. By default in billing are written the first 43 fields. The specified value must be
displayed in telnet at the command billing fields number. If the number of fields is 255 then all the
billing fields are exported.
The order of billing fields is presented below:
1. type

17. source signaling IP

33. ptime

2. source transcoding port

18. source signaling PORT

34. client id out

3. ANI in

19. source RTP IP

35. gw_name

4. DNIS in

20. source RTP port

36. ANI out

5. date

21. destination IP

37. DNIS out

6. time

22. destination port

38. original ANI

7. call duration

23. destination RTP IP

39. connected DNIS

8. billing units

24. destination RTP port

40. in_billing_profileid

9. SIM used

25. call session id

41. out_billing_profileid

10. destination transcoding


port

26. packet loss

42. in_classid

11. release cause

27. jitter

43. out_classid

12. release Q850 code

28. client id in

44. billing_prefix

13. CIMI

29. class name out

45. price_in

14. call selection time

30. protocol_in

15. GSM cell

31. protocol_out

16. class name in

32. codec

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7.9.1 Pg SQL Billing


In order to activate Billing in PostGre SQL you need to configure the files
/mnt/app/bin/pgsql_sip_pool and /mnt/app/cfg/pgsql_sip_pool.cfg.
The data base tables are created automatically by the multiSwitch application for the current
month and for the next two months. The billing tables are in the format billing_yyyy_mm.
Warning: When the multiSwitch main application was updated with a new version where were added
new fields in billing you must change manually the billing tables (for the current month and for the
months for which it has been created) and to add the new fields!
Example for adding a new field in billing:
ALTER TABLE billing_2008_06 add column in_billing_profileid INT NOT NULL DEFAULT 0;
The multiSwitch application includes a PostGre SQL billing pool for redundancy.
First of all you must configure the file /mnt/app/cfg/exec.cfg as follows:
pgsql_billing 1/0 [configuration file name]
The command enable/disale pgsql billing pool. The field configuration file name is optional
and if it is set it specifies the configuration file which will be read on start. The default file is
pgsql_sip_pool.cfg.
pgsql_billing_alt 1/0 [configuration file name]
The command enable / disable the pgsql_billing pool for redundancy. The field configuration
file name is optional and if it is set it specifies the configuration file which will be read on start. The
default file is pgsql_sip_pool.cfg.
The second step is to configure the file /mnt/app/cfg/pgsql_sip_pool.cfg. The changes
made on this file are loaded automatically by pgsql_status application.
debug 0
The line above indicates that no debug is made on Pg SQL billing. 0 = no debug, 1 = min
debug, 2 = full debug.
conn_string
dbname=softswitch
password=99softswitch11

host=127.0.0.1

user=softswitch

The line above contains the connection string for connection to postgresql database.

7.9.2 My SQL Billing


In order to activate billing in My SQL you will need
/mnt/app/bin/mysql_client and /mnt/app/cfg/mysql_client.cfg.

the

configuration

In the exec.cfg file the is inserted the line:


mysql_billing 1
Then is created the data base:
mysql -p
create database billing
use database billing

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The My SQL data base structure is presented below:
CREATE TABLE billing(
id INT NOT NULL AUTO_INCREMENT, PRIMARY KEY (id),
tip CHAR(3),
port_src SMALLINT,
identity VARCHAR(32),
number VARCHAR(32),
date date,
time time,
duration INT,
units SMALLINT,
SIM CHAR(2),
port_dest SMALLINT,
`release` VARCHAR(15),
rel_Q850 SMALLINT,
CIMI VARCHAR(30),
selection SMALLINT,
GSM_cell VARCHAR(10),
direction VARCHAR(20),
IP_s VARCHAR(20),
PORT_s INT,
IP_RTP_s VARCHAR(20),
PORT_RTP_s INT,
IP_d VARCHAR(20),
PORT_d VARCHAR(20),
IP_RTP_d VARCHAR(20),
PORT_RTP_d INT,
session_id VARCHAR(20),
jitter INT,
packet_loss INT,
client_id INT,
direction_out VARCHAR(20),
proto_in VARCHAR(20),
proto_out VARCHAR(20),
codec VARCHAR(10),
ptime INT,
out_clientid INT,
gw_name VARCHAR(30),
id_out VARCHAR(32),
nr_out VARCHAR(32),
orig_ani VARCHAR(32),
con_dnis VARCHAR(32),
in_billing_profileid INT,
out_billing_profileid INT,
price_in FLOAT(9,2),
currency_in VARCHAR(10);
call_type VARCHAR(32);
);
The alteration of data base after an update is made like in the example below:
ALTER TABLE billing_2007_07 ADD COLUMN orig_ani VARCHAR(32);
ALTER TABLE billing_2007_07 ADD COLUMN con_dnis VARCHAR(32);
Note: The parameter billing_2007_07 is the billing table from month 07 year 2007.

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7.10 IVR activation and configuration


IVR (Interactive Voice Response) service is a technology that allows a machine to detect
voice and touch tones using a normal phone call. The IVR system can respond with pre-recorded or
dynamically generated audio to further direct callers on how to proceed.
IVR systems can be used to control almost any function where the interface can be broken
down into a series of simple menu choices. Once constructed IVR systems generally scale well to
handle large call volumes.
TOPEX multiSwitch application allows achieving IVR services configurable from the text files
such as:
-

Prepaid

Voice Attendant

The service code for TOPEX IVR is 33.


In the configuration file of the IVR service are defined several states. Each state can have
actions and events..
Each state is defined between square brackets [state name]. Each state can contain 2 key
words: "do" and "event"
do contains only actions; these actions are made when is made the transition in the
respective state
event contains one or several actions associated to a event
The actions are separated through the character ';'. After the last action is not necessary the
character ';'. The character '>' delimits an event of its actions. The commented lines start with ';'
Certain actions may trigger events, such as the following actions are no longer run; the
implementation moves on the new event created.
For the IVR service is written an API in the "ivr" file stored in /mnt/app/cfg/ivr. This file is put
into the field "play_file" of routing.
Files that will be run on the IVR service will be stored on the HDD in the directory
/mnt/app/raw/IVR/ in the format <file name>_ <codec>.<language>.
The codec must from 2 numbers, for instance: 00=PCMU(G711u), 08=PCMA(G711a),
18=G729, and the language must correspond with the language set as parameter in the SetLang()
action; it must have 2 characters, ex: ro, en.
IVR text files are loaded into memory when reading routes, they can be viewed with the telnet
command ivr<file name>. This represents a further check that you have configured correctly.

7.10.1 Actions and events of IVR service


The configuration file is based on ations and events detailed below:
Actions:
PlayFile (string file_name, integer repeat, integer background_foreground) plays a file,
where:
- file_name the name of the played file
- repeat specifies if runs from the begin after file end. The possible values are 0/1
- background_foreground species the play mode of the file;

This way can be play a silence file in order to sent RTP to destination (to not hear noise on RTP
packets lack). The possible values are 0=background and 1=foreground.

PlayCredit() plays the credit of a prepaid user

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PlayTime() plays the maximum time which can be used on a specified destination

StopPlay (integer background_foreground) stop the current file in the specified mode.
The options are 0=background; 1=foreground; 2=both

Dial (integer dnis_len) send the call in routing and looks for a destination; start of the
OnDial() event

dnis_len specifies the minimum number of digits after which is run the action Dial

SetTimer(integer timer_nr, integer timer_value) - set a timer specified in timer_nr;


timer_value is in miliseconds

KillTimer(integer timer_nr) stops the specified timer

GoTo(string stare) goes in the specified state

AddToDNIS() is used in conjuction with the event Digit(dig); add digits to DNIS

AddToPIN() is used in conjuction with the Digit(dig) event; adds digits to PIN number

CheckPrepaidPIN(integer pin_len, integer check_update_credit) verifies the PIN code in


data base and loads th settings of the prepaid user in case the PIN code is found

check_update_credit if is 1 then is verified the credit update at the end of the call, in other
case is made only PIN authentication

It is useful for clients which wants to define users which must authenticated on ANI and / or
PIN without verify / decrement the credit.

CheckPrepaidANI(integer check_update_credit) verifies ANI in the data base and loads


the settings of the prepaid user when ANI is find

check_update_credit if it is 1 then the credit is verified and it is made the update to the credit at the
end of the call, in the other case is made only an ANI authentication;
It is usefull for clients which want to define users which are authenticated on ANI and / or PIN
without verify / decrement the credit.

Release(integer code) release the call with the specified code; the code must be specified
according with Q850

SetLang(string[2] language) specifies the language of the voice messages; 2 characters


used for instance ro, en

MaxTry(integer maxtry) increments the nrtry variable and if this is >= maxtry starts the
event OnMaxTry()

ChangeANI(integer ignore, string insert) ignore or insert in ANI

ChangeDNIS(ignore, insert) ignore or insert in DNIS

ChangePIN(ignore, insert) ignore or insert in PIN

TxAccessRequest(integer stage, integer tip, integer pin_len, integer timeout, char state)
Send Access Request on RADIUS to Mind

stage - on Mind the authentication is made in two steps; 1=authentication; 2=resolve


tip authentication type: 1=PIN; 2=ANI
pin_len PIN code length; this action is executed only when the PIN code is higher or equal with the
pin_len parameter
timeout athis action set automatically the timer 0 with the specified value in the timeout parameter
(in milliseconds)
state the state for answer reception

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Events:

Timeout(timer_nr) run when the timer timer_nr expires;

Digit(digit) run when is received the specifie digit; the value digit=255 implies the event run
at any received digit

EndPlay() run when at the end of the ruled file

PrepaidCreditFinished() runs if the prepaid user has no credit

PrepaidCreditOK() runs if the prepaid user credit is > 0

PrepaidNotFound() runs if after verification CheckPrepaidPIN(), CheckPrepaidANI() no


result was return

OnDial() executed by the Dial() action

DBResult() executed when is received an answer from the database

DBNotFound() executed when is received not found message from data base

DBError() executed on error from data base

OnMaxTry() executed by the MaxTry() action

EndCall() executed when the current call is closed with # (special character and allows
ending a current call connected or not)

RadiusAccept()

RadiusReject()

RadiusError()

7.10.2 IVR configuration examples for prepaid i voice attendant


Voice Attendant
[first]
do=PlayFile(music,1)
event=Digit(255)>StopPlay();AddToDNIS();Dial(3);SetTimer(0,5000)
; this line is commented
event=Timeout(0)>Release(41)
event=OnDial>KillTimer(0)
Prepaid
[idle]
do=PlayFile(select_language,0)
event=EndPlay()>SetTimer(0,5000)
event=Digit(1)>StopPlay();SetLang(ro);PlayFile(enter_pin, 0);GoTo(wait_pin)
event=Digit(2)>StopPlay();SetLang(en);PlayFile(enter_pin, 0); GoTo(wait_pin)
event=Timeout(0)>PlayFile(temporary_failure,0);GoTo(release)
; this line is commented
[wait_pin]
event=EndPlay()>SetTimer(0,5000)
event=Digit(255)>AddToPIN();CheckPIN(4);SetTimer(0,5000)
event=Timeout(0)>PlayFile(temporary_failure,0);GoTo(release)
event=OnCheckPIN()>GoTo(wait_db);SetTimer(0,10000)
[wait_db]
event=Timeout(255)>PlayFile(pin_error,0);GoTo(release)
event=DBResult()>ParsePrepaid()
event=DBNotFound()>PlayFile(pin_error,0);GoTo(wait_pin);ChangePIN(32,);MaxTry(3)
event=DBError()>PlayFile(pin_error,0);GoTo(release)
event=CreditOK()>PlayCredit();GoTo(dialdigits)
event=CreditFinished()>PlayFile(pin_no_credit,0);GoTo(release)

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event=OnMaxTry()>PlayFile(pin_error,0);GoTo(release)
[dialdigits]
event=EndPlay()>SetTimer(0,5000)
event=Digit(#)>Dial(255)
event=Digit(255)>AddToDNIS()
event=Timeout(0)>Dial(255)
event=OnDial()>KillTimer(0)
event=EndCall()>PlayFile(enter_new_number,0)
[release]
event=EndPlay()>Release(31)

Prepaid with RADIUS Server from Mind

[idle]
do=PlayFile(select_language,0,1)
event=EndPlay()>PlayFile(silence,1,0);SetTimer(0,5000)
event=Digit(1)>SetLang(ro);PlayFile(enter_pin, 0,1);GoTo(wait_pin)
event=Digit(2)>SetLang(en);PlayFile(enter_pin, 0,1); GoTo(wait_pin)
event=Timeout(0)>PlayFile(temporary_failure,0,1);GoTo(release)
[wait_pin]
event=EndPlay()>PlayFile(silence,1,0);SetTimer(0,5000)
event=Digit(255)>SetTimer(0,5000);StopPlay(1);AddToPIN();TxAccessRequest(1,1,4,4000,wait_auth,
0)
event=Timeout(0)>PlayFile(temporary_failure,0,1);GoTo(release)
event=OnMaxTry()>PlayFile(pin_error,0,1);GoTo(release)
[wait_auth]
event=Timeout(0)>PlayFile(temporary_failure,0,1);GoTo(release)
event=RadiusAccept()>KillTimer(0);PlayCredit();GoTo(dialdigits)
event=RadiusReject()>KillTimer(0);MaxTry(2);PlayFile(pin_error,0,1);GoTo(wait_pin);ChangePIN(32,)
event=RadiusError()>KillTimer(0);PlayFile(pin_error,0,1);GoTo(release)
event=OnMaxTry()>PlayFile(pin_error_final,0,1);GoTo(release)
[wait_rezolve]
event=Timeout(0)>PlayFile(temporary_failure,0,1);GoTo(release)
event=RadiusAccept()>KillTimer(0);Dial(255)
event=RadiusReject()>KillTimer(0);PlayFile(temporary_failure,0,1);GoTo(release)
event=RadiusError()>KillTimer(0);PlayFile(temporary_failure,0,1);GoTo(release)
event=OnDial()>KillTimer(0)
[dialdigits]
event=EndPlay()>PlayFile(silence,1,0);SetTimer(0,5000)
event=Digit(#)>TxAccessRequest(2,1,0,4000,wait_rezolve,0)
event=Digit(255)>StopPlay(1);AddToDNIS();SetTimer(0,5000)
event=Timeout(0)>TxAccessRequest(2,1,0,4000,wait_rezolve,0)
event=OnDial()>KillTimer(0)
event=EndCall()>PlayFile(enter_new_number,0,1)
[release]
event=EndPlay()>Release(31)

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7.11 TELNET commands


Telnet (Telecommunication network) is a client-server network protocol based on connectionoriented reliable transport. Usually, this protocol is used for establishing a connection through the TCP
23 port, where an equivalent application is listening.
The Telnet protocol is used to remotely connect to an equipment, to diagnose the problems of
such equipment, query the remote application regarding certain of its parameters.
The Telnet commands used for multiSwitch equipment are detailed in the table below:
Command

Description

help

Displays all the commands and their description

access in

Allows the view of the loaded "Accessin" configuration

access out

Allows the view of the loaded "Accessout" configuration

all queue

Allows you to view all the calls in the call queue

billing fields number

Indicates the number of billing fields in the file "exec.cfg"

billing profile [profile id]

Indicates the ID for the specified profile

billing profiles

Displays all the billing profiles

count all online users

Displays the number of online users

count offline users

Displays the number of offline users

count sip users

Displays the number of online SIP users

classes

Displays the classes defined on the equipment

debug on

Turns on the debug function

debug off

Turns off the debug function

dialog states

Displays the call status for each conversation

fdwatch connections

Indicates all the fdwatch connections available (telnet,


httpd etc)

forking group [group


number]

Indicates the SIP users from the selected forking group

fw subscriptions

Indicates all the events redirected by the Proxy SIP

global rules

Displays the list of global administration rules

global translate prefix

Displays the list of the translated prefixes

hunting group [group


number]

Indicates the SIP users from the hunting group

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interface ip

The IP interface used

ivr [filename] [resellerid]

Indicates the IVR file; the reseller ID is optional

kill call [port]

Ends the call

license

Indicates licensing information

pickup group [group


number]

Indicates the SIP users from the pick-up group

port subscriptions

Indicates all the events on the port

prepaid users

Displays all the prepaid users

queue [queue number]

Displays the number of the call queue

quit

Quits the application

register users

Displays the list of registered SIP users

reseller rules [reseller id]

Displays the list of rules for a specified reseller ID

reseller translate prefix


[reseller id]

Displays the list of translated prefixes for a specified


reseller ID

resellers

Displays the resellers defined in the equipment

ring state remote

Displays the remote queue connected to the call state

search online [pattern]

Searches for a pattern in the SIP online users list

search user [pattern]

Searches for a pattern in the SIP users list

show sip online [username]

Indicates all the details for the SIP online users

show sip user [username]

Indicates all the details for the SIP users

sip pp

Indicates the access list for the public/private IP class

sip publish

Indicates all the events published for the selected online


user

sip offline

Indicates the list of offline SIP users

sip online

Indicates the list of online SIP users

sip online forked

Indicates the list of forked online SIP users

sip users

Displays the list of SIP users

sip vm notify

Displays the voice mail list for SIP users

subscriber [port]

Displays a list of settings for FXS subscribers on that port

subscribers

Displays a list of FXS subscribers

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tls sock [fd]

Indicates the details of the specified TCP/TLS socket

ts sock busy

Indicates the TCP/TLS sockets in the busy calls queue

ts sock free

Indicates the TCP/TLS sockets in the calls queue

ts sock wait

Indicates the TCP/TLS sockets in the pending calls queue

update config

Updates the configuration of the memory database

view allports

Displays all the available ports

view class [classid]

Displays details for a specified class ID

view classes

Displays the list of customer classes, defined by


equipment

view ports

Displays the installed ports that have the status !=FREE

view port [port number]

Displays the specified port

view route [routeid]

Displays the details of the specified route

view routes

Displays the list of available routes

web callback [A username]


[B username] [A class id]

Connecting the calls in the order: call A, call B

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DESCRIPTION OF THE WEB INTERFACE

TOPEX multiSwitch has the Linux Gentoo distribution installed as operating system and a
series of applications built around the "central" application, ensuring the softswitch function.
The equipment has an administration, configuration and monitoring web-based interface, that
allows configuring the calls that go through the equipment, as well as adding, configuring and deleting
new users in the associated database.
It also allows enforcing strict rules for allowing/disallowing certain users and calls and allows
the administration and configuration of the billing system.
You can easily access the dedicated interface from this PC or from a remote PC through the
IP network.

8.1 The web administration interface


The web interface allows the operator to offer the customers phone services, view tax
information, statistics, services, voicemail viewing and listening etc.
The web interface access is performed through a web browser and is password-protected.
All that is needed on the PC is an installed browser web. TOPEX multiSwitch equipment is
delivered by TOPEX with the default IP 192.168.1.20.

8.2 Authentication
To access the configuration page for TOPEX multiSwitch equipment from a remote computer,
you must enter the default IP address as URL: http://192.168.1.20.
After entering this address in the web browser, the "Authentication" window will appear on the
screen and you will be prompted to enter your user name and password:

Figure 1 Authentication in the administration page for TOPEX multiSwitch


The default user name is admin and the default password is 99admin11. Fill in these fields
and press the "LogIn" button at the bottom of the authentication window.
If you authentication data you entered is wrong (either the user name or the password), the
following message will be displayed above the window: "Authentication": "Error Incorrect User /
Password", as you can see in Figure 2.

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Figure 2 Wrong authentication

8.3 The menu bar


On the top of the administration page is the bar that contains the menus for the equipment
interface. These menus are used to navigate through the web interface of TOPEX multiSwitch.

Figure 3 Menu Bar


The web interface of the equipment contains the following menus:
Server Settings Here you can add, edit or delete equipments (Media Gateways) connected to
multiSwitch and you can view details of the Softswitch web interface;
Server Management contains customer classes (directions), the routing table, rules that can be
applied, resellers etc.
Services creating aliases for SIP users, Centrex, Hunting, Forking or Pick-up groups
User Management this menu contains the customer list, grouped in several submenus. You can
edit or delete properties for users registered for Softswitch: FXS, ANI, Prepaid or SIP;
Billing the billing files details, including the call list, billing or generation of detailed billing form;
Testing generate users and rules for testing
Reports in this section are displayed the statistics about reseller margin and profits;
Templates creation, for each class or user, of a template that can be applied to more users.

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8.4 The "Server Settings" menu


The "Server Settings" menu includes several sections about the web interface version, the
user access lists, general settings for the equipment(s) interconnected to TOPEX multiSwitch.
The Server Settings menu includes the options:
-

About details about the Softswitch web interface;

SIP Access List contains the locations of the SIP users;

Settings for adding new TOPEX multiSwitch equipments in the web interface;

Equipments includes equipments that can be interconnected to the TOPEX multiSwitch


equipment;

Reload config enforces the application of changes made on the equipment. Usually, TOPEX
multiSwitch checks whether there are changes in the database, at a fixed interval (15 seconds),
and by using this command, the changes are applied instantly.

Figure 4 "Server Settings" menu options

8.4.1 The "About" option


The "About" option under "Server Settings" opens a window where the web interface version
of the TOPEX multiSwitch is displayed.

Figure 5 The "Server Settings" menu "About" option


The window displays the equipment name, the software version (2.1.2. in this case ) and the
message "Copyright 2008 Topex S.A.". If you click the company name, you will be redirected to the
website at http://www.topex.ro.

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8.4.2 "SIP Access List" option


Under the "SIP Access List" menu, you can setup the IP address domains from which calls
can be received, as well as the maximum number of calls.
The default value is 0.0.0.0/0, which allows access to the equipment for all calls, from all IP
addresses, without limitation.

Figure 6 The "Server Settings" menu "SIP Access List" option


For adding a new access list, press the "New Record" button and fill the fields in the window
that will appear on screen.

Figure 7 Adding a new access list


The fields that must be filled are:
Class Name the name of the new access list
Public the public IP address
Private the private IP address
Max Calls the maximum number of calls allowed from previous addresses
After filling in the fields, press "Submit", and the record will be added to the list.
To change settings for each record, use the "Edit" button, from the "Action" region, and after
you finished making your changes, press "Submit" to validate them.
You can delete records by pressing "Delete" from the "Action" region.

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8.4.3 "Settings" option


This menu allows setting up local and global parameters in the Softswitch and adding new
equipments that will be managed through the web interface.

Figure 8 The "Server Settings" menu "Settings" option


To enter global parameters, press the "New Parameter" button, in the middle region of the
screen, below the parameter list (by default without records); a new window will be displayed:

Figure 9 Adding new parameters


In the field Name enter the name of the new parameter, and in the field Value, enter its value.
To add the parameters, press Add.
After entering the parameters, this is how the list will look:

Figure 10 Displaying the added parameters


To change the value of a parameter that was already entered, press "Change"; after entering
the new value, press "Apply" to commit the change.
If you want to delete a parameter from the list, use the "Delete" option. A dialogue window will
be displayed, asking if you confirm this action; press "OK".
In case you want to customize the settings for a TOPEX multiSwitch, press the "New
Softswitch" button, near the New Parameter option, and you will be able to enter a new Softswitch as
well as local parameters for it. The following window will appear:

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Figure 11 Adding a new equipment in Softswitch Parameters
Enter the name and press "Submit". The parameters will be displayed as follows:
"Local Parameters" (no records by default), with the displayed message: "No parameters
were defined";
"Global Parameters", where the common parameters for all Softswitches are displayed.
The figure below shows an example:

Figure 12 Displaying the global parameters for a Softswitch


In the list of global parameters, check the Make Local option near each parameter. This will
allow the change of each global parameter in a local one (specific for a customized Softswitch). After
checking the Make Local option, the parameter will be added in the Local Parameters list, being now
a local as well as a global parameter.
For example, the reconfig_centrexAlias parameter, after checking its Make Local option, will
also be displayed in the list of local parameters, with the same name:

Figure 13 Translating a global parameter into a local one


To enter more local parameters, press New Parameter, enter the name and value for the
parameter, then press Add.
Using the "Change" and "Delete" options, you can change or delete the records entered.
Next to the New Parameter option, you can see the New Softswitch option, which allows you
to add a new Softswitch to the list.

8.4.4 The "Equipments" option


In this section of the "Server Settings" menu, you can add and manage equipments that are
directly interconnected with TOPEX multiSwitch in a complete voice solution.
The "Equipments" window contains a list of the equipments connected to TOPEX
multiSwitch. In the example below, there is no equipment connected to the Softswitch.

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Figure 14 The "Equipments" option


To add a new equipment, press the "New Entry" button, at the top of the page. In the window
that appears, fill in the following fields:

Figure 15 The "Equipments" option


Name - name of the equipment to be added in the TOPEX multiSwitch configuration
Type type of the equipment. The available options are:
- TOPEX multiAccess VoIP E1 GSM Gateway with 30 voice channels
- TOPEX Eones Media and Signaling Gateway that can contain a maximum of 300 voice
channels. It can be used only for signaling conversion E1 ISDN E1 SS7 or E1 Gateway to VoIP
(ISDN/SS7 to SIP&H.323).
- TOPEX mGu TOPEX miniGateway Media Gateway for the 2E1-VoIP conversion; can be
equipped with various analogue interfaces: FXS, FXO, E&M, BL
- Unknown / Foreign some other equipment, manufactured by TOPEX or by some other
manufacturer, and which can be interconnected to multiSwitch. In order to be interconnected, the
equipment must support the VoIP SIP or H.323 signaling protocols.

Figure 16 List of the equipment that can be interconnected


Protocol the protocol used to interconnect the Softswitch and the rest of the equipments. Use the
option "Topex".
Command IP The IP address of the equipment being added to the structure
Port the port number through which TOPEX multiSwitch will communicate with the rest of the
equipments.

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8.5 The "Server Management" menu


The "Server Management" menu includes sections for administration of the database, which
includes administration of the customer classes, translations, routes, access and SIP locations. Under
this menu, you can define the customer classes or directions, you can declare call routes or define
customized and general restriction rules.

Figure 17 The "Server Management" menu


The options of the "Server Management" menu are:
Client Classes the customer classes from the database
Translate ANI number translation (adding a prefix) depending on ANI
Translate DNIS number translation (adding a prefix) depending on DNIS
Routes defining the call routes
Access In access list for incoming calls
Access In access list for outgoing calls
Global Rules general call rules
Operators defines the mobile operators depending on the number - under development
Devices under development

8.5.1 The "Client Classes" option


This menu contains all the client classes from the database of the TOPEX multiSwitch
equipment
Note: The client classes can be changed according to your options. You can choose your own
suggestive names for the client classes. For example, you can create a client class for all SIP users,
named "SIP Users" or "Utilizatori SIP".
In the figure below, you can see the names of client classes, as well as the corresponding
parameters.

Figure 18 Description of the client classes


When first accessing the Client Classes menu, you will see the window presented above.
Along the names of the client classes, you will see the following parameters:
-

Id the identification number of the client class;

Name name of the client class;

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-

Rules In/Out - incoming/outgoing restriction rules applied to the specified client class;

Transcoding the transcoding feature for each client class - used for clients with different
codecs.

Max Calls In / Out maximum number of incoming and outgoing calls for that class;

Call Rate In / Out the rate for the incoming and outgoing calls;

ANI Ignore / Insert / Max the ANI prefixes to be ignored, added and the maximum incoming
number of digits for ANI;

DNIS Ignore / Insert / Max the DNIS prefixes to be ignored, added and the maximum incoming
number of digits for DNIS;

Action the option available for that class: Edit ("Edit" button) or Delete ("Delete" button) the
class.

8.5.1.1 Editing a client class


If you want to edit an existing client class, press "Edit" in the "Action" region, and a window
similar to the one below will appear on the screen. You can change the following client class
parameters:

Figure 19 Editing a client class


Name name of the client class;
Type type of the client class: Available options are "Class" or "Port";
Trunk If this option is checked, the client class will be displayed as a route for "Destination Class"
(the Reseller can select that client class as route to "Destination Class");
Billing Profile IN and Out the billing profile used for that client class. Billing profiles are created in
the Billing Profiles region;
Search Mode allows you to select the search mode for channels on TDM. Available options are
"Up" ascending search mode (from channel 1 to channel N), "Down" descending search mode
(from channel N to channel 1) or "Circular" circular search mode (for example 1 2 N 1 2 ...);

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RTP Proxy In / Out represents an application that runs on the TOPEX multiSwitch equipment and
transfers UPD packets from one port to another. RTP Proxy is used for transferring the RTP voice
packets behind the NAT. For each call, two UDP ports are opened. The application directs the UPD
packets received on one port to another port. Here, you can select one of the following options: "Not
Used", "Used except users in the same NAT" or "Always Used";
Transcoding In / Out you can check this option to use the transcoding for that client class;
Rules In / Out the incoming and outgoing rules; select "In" and/or "Out" if you intend to enforce
incoming and/or outgoing rules Depending on the option you choose here, you must fill in the following
fields:
Max Calls maximum number of calls allowed for incoming and outgoing
Call Rate number of calls per second, incoming and outgoing
Max Taxunits maximum number of units (pulses) charged
Taxunits number of units charged
Max Time maximum interval allowed for connection; if this threshold is reached, the
connection is closed;
Time number of seconds for the call;
TOS Type of Service packet prioritization service;
Translate the option to add a prefix in front of a number; ANI and/or DNIS numbers can be selected.
The following six fields must be filled in depending on the selections made under this option:
Ignore from ANI the prefix to be deleted from the ANI number Ignores the specified
number of digits from the caller's ID (maximum 20 digits);
Insert into ANI the prefix to be added to the ANI number Adds the specified digits
(maximum 16 digits) to the Caller ID;
Ignore from DNIS the prefix to be deleted from the DNIS number. Specifies the number of
digits to be ignored from the numbers received on that direction (client class);
the first x digits from the received number will be ignored (maximum 20 digits);
Insert into DNIS the prefix to be added to the number received on that direction
(maximum 16 digits can be entered);
Max ANI /DNIS the maximum number of incoming digits for ANI/DNIS;
Sign 1-6 options for setting the signaling parameters. The "Sign 1", "Sign 2" and "Sign 3" fields can
be extended by pressing the "Edit" button. The options for these fields are explained below.

Figure 20 Options for the "Sign1" field


-

Receive Identity when checked, the Caller ID is transmitted to the selected direction.

Send Identity - The caller ID is transmitted to the selected direction.

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-

Load balancing algorithm establishes the algorithm for changing SIM cards in GSM interfaces.
When using "0", the LCR algorithm is used, and when using "1", the load balancing algorithm for
GSM cards is used. This option is used when interconnecting with a multiAccess TOPEX
equipment.

Coupling of the ring-back tone - the system allows for a second call to be placed, until the
receiver of the first call will pick up the call.

Audio Level level of the audio signal. Values can be between 0 (maximum level) and 7
(minimum level)

Verify CLIR check settings for Calling Line Identification Restriction

GOODASR used for equipments with GSM modules

Cat Call all calls made through the equipment will be closed

Test Net used for equipments with GSM modules

Check Call Back the call that will come through this direction will be checked by the call back
function. The call back table will be analyzed with the Caller ID.

Module CDMA used for equipments with CDMA modules

Figure 21 Options for the "Sign 2" field


Transit Q.850 transfers the Q.850 code to E1 ISDN
Load Balancing algorithm on SIM Index - used for equipments with GSM modules
Calculate Tax Pulses calculates the tax pulses for incoming calls
Get Tax get tax information
Check Credit checks the credit
Load Credit reloads the credit
13 PABX Wait Dial Tone wait dial tone for analogue junctions
Delay CDMA - used for equipments with CDMA modules
Ring Back Detect detects a ring back
Delay Reset SS7 - used for signaling gateway equipments
Search Up / Search Down establishes the rules by which the channels are selected on ISDN
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Check Credit First checks the credit
Priority used for routes with the same prefix
H323 No Tunnel H245 deactivates the method of detection for negotiation processes via the H245
protocol
H323 No Fast Start deactivates the method of detection for negotiation processes via the H323
protocol
Play CLI Error error message for an incorrect caller ID
Answer CLI Error activate error message for incorrect caller ID

Figure 22 Options for the "Sign 3" field


Tx Channel ISDN User used for equipments with E1 ISDN trunks
Check DNIS Checks the DNIS
Check ANI Checks the ANI
Translate DNIS translates the DNIS (changes the prefix for the destination)
Limitation Time the limitation time
Cut on Error DB free the call in case of a database error
Limit On Cost limit the costs

8.5.1.2 Adding a new client class


You can create a new route by pressing the "New Class" button. In the Client Classes / New
Class window you can change the following settings:
Billing Group associate the client class with a previously defined billing group
Name name of the client class to be created
Type select the Class option to create a client class. The other available option is "Port".
Trunk If this option is checked, the client class will be displayed as a route for "Destination Class"
(the Reseller can select that client class as route to "Destination Class");
Billing Profile the billing profile used for that client class. Billing profiles are created in the Billing
Profiles section;
Search Mode used for routing when same prefix routes are used. The available options are:
-

Up The route will be selected from the last to the first. Depending on its position in the routing
table, the last route in the group will have the highest priority.

Down The route will be selected starting from the first route to the last. Depending on its position
in the routing table, the first route in the group will have the highest priority.

Circular - The route will be selected circularly.

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RTP Proxy In / Out represents an application that runs on the TOPEX multiSwitch equipment and
transfers UPD packets from one port to another. RTP Proxy is used for transferring the RTP voice
packets behind the NAT. For each call, two UDP ports are opened. The application directs the UPD
packets received on one port to another port. Here, you can select one of the following options: "Not
Used", "Used except users in the same NAT" or "Always Used";
Transcoding In / Out the transcoding feature for each client class - used for clients with different
codecs;

Figure 23 Creating a client class


Rules the incoming and outgoing rules; select "In" and/or "Out" if you intend to enforce incoming
and/or outgoing rules. Depending on the option you choose here, you must fill in the following fields:
-

Max Calls maximum number of calls allowed for incoming and outgoing;

Call Rate number of calls per second, incoming and outgoing;

Max Taxunits maximum number of units (pulses) charged;

Max Time maximum interval allowed for connection; if this threshold is reached, the connection
is closed;

Max Time maximum duration of the call;

TOS Type of Service packet prioritization service.

Translate he option to add a prefix in front of a number; ANI and/or DNIS numbers can be selected:

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-

Ignore from ANI the prefix to be deleted from the ANI number. Ignores the specified number of
digits from the caller's ID (maximum 20 digits);

Insert into ANI the prefix to be added to the ANI number. Adds the specified digits (maximum
16 digits) to the Caller ID;

Ignore from DNIS the prefix to be deleted from the DNIS number. Specifies the number of digits
to be ignored from the numbers received on that direction (client class); the first x digits from the
received number will be ignored (maximum 20 digits);

Insert into DNIS the prefix to be added to the number received on that direction (maximum 16
digits can be entered);

Max ANI /DNIS the maximum number of incoming digits for ANI/DNIS;

Sign 1-6 signaling parameters for configuring the equipment.


After configuring the parameters you desire, press "Submit" to create the client class.

8.5.1.3 Adding rules for a client class


For a newly created client class you can also establish a series of applicable rules.
On the left of the window, you can see the following options:

Main main settings described above;


The "Rules" option establishes rules for ANI and DNIS: what ANI and/or DNIS numbers
are allowed or restricted.

The window also contains five enterable fields, as described below:


-

Name name of the client class to which the rules apply

Type a list that has two options: "Allow" and "Restrict", for establishing the phone numbers
permitted or restricted to that client class;

DNIS phone number of the destination;

ANI phone number of the caller party;

Actions by default, if the list is empty, this will only contain the "Add" button, for adding
restriction rules for the client class. When a rule has been added to the list, one other button will
appear in the "Actions" region the "Delete" button, for deleting that record.

For example, if you select the "Restrict" option for the source number "0722222222", and the
destination number "0212121212", then no call will be permitted from that source to that destination.
If you select "Restrict" for a DNIS number, all the calls placed to that number will be rejected.
If only an ANI number was restricted, all the calls from that number will be restricted, regardless of the
destination called.

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Figure 24 Editing a client class - Rules region


If you press "Submit" without filling in the ANI or DNIS fields, the following error message will
appear on screen:

To delete a restriction rule, press the


"Delete" button from the "Action" region. A
window will appear on screen, with the message
"Are you sure you want to delete this rule";
press "OK".

8.5.1.4 Setting alarms for a client class


The "Alerts" option refers to the alarm type used. The following are available:
ASR (Answer Seizure Ratio) represents the rate between the number of calls successfully answered
and the total number or call retries;
ACD (Average Call Duration) the average duration of all the calls placed by a subscriber or by a
client class;
NER (Network Efficiency Ratio) is one of the simplest and most useful analyses for traffic
management. The rate between the number of correct disconnection causes and the number of total
disconnection causes is calculated;
ABNSR the Answer Bid Ratio parameter the rate between the number of successful calls and the
number of total calls.

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Figure 25 Editing a client class - Alerts region


Leave the default value 0 for all these parameters if you do not wish to activate alerts (this
option is inactive).
To activate it, enter values between 0 and 100 (which represent percentages) for ASR,
ACD and ABNSR.
For ACD (measured in seconds), enter a value between 1 and 240-300 (seconds).
This option is used in the Statistics menu from the web interface: if the values entered
here are higher than the ones displayed in Statistics, then the Class field and the values are
displayed in red!!!
After entering the values in these fields, press "Update" to add them to the list.

8.5.2 The "Translate ANI" and "Translate DNIS" options


These two options from the web interface of the TOPEX multiSwitch equipment are used to
translate the call prefixes depending on the ANI (source number) and the DNIS (destination number).
"Translate DNIS" is used in the Number Portability application.
The two menus, "Translate ANI" and "Translate Prefix", are identical, so we will only describe
one of them below "Translate ANI".
By default, as in the example below, there is no prefix translation rule defined.

Figure 26 The Translate "ANI" option


For this, use the "New Rule / Import from File" option, on the left of your screen

Figure 27 Translate ANI New Rule


You have two options for adding prefixes. The first is to load the records from a file in the
equipment's database: press the Browse, button, which allows you to search the desired file.
The second option is to enter the options manually:
Description a short description of the client (for example, the name of the switch system);

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Prefix the phone number to which the prefix will be added. It can also be in the format "075%",
which means that all numbers starting with "075" will be prefixed with the prefix from the "Add Prefix"
field; do not add "%" if you enter the complete phone number);
After filling in all the data, press "Submit" to commit them. You can view these records in
the "Main" option.

Figure 28 Displaying the records in the Translate ANI/DNIS table


The fields in this table are:
ID the identification number for that group;
Description a short description for the group (for example, the network name);
Prefix the phone number to which the prefix will be added;
Add Prefix the prefix to be added to the number entered in the "Prefix" field;
Action a field where you can change the data ("Edit") or delete them ("Delete").
Using the "Group Update" option, you can add the same prefix for all the records that have
the same name in the "Descriptions" field:

Figure 29 Changing the prefixes of a group with the "Group Update" option
The fields in this window are:
Group the group to which the prefix will be added;
Add Prefix the prefix to be added to the number entered in the "Prefix" field;
After configuring the options, you must press "Submit" for the prefix to be added to the list.
To filter the prefixes, use the "Filter" option, under "Menu". To select only certain numbers,
enter the number in the "Filter Prefix" field, and press Enter. All the records with that phone number
will be added (from the "Prefix" field).
For example, to select numbers that start with "075", enter "075" in the "Filter Prefix" field
and press "Enter":

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Figure 30 Filtering prefixes with the "Filter" option

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8.5.3 The "Routes" option


The "Routes" option allows you to define the routes required by the calls to reach their
destination. The menu contains a list of routes, in table form.
The routing table routes the calls towards an outgoing class, service or port, depending on the
prefix and on the incoming class.
The routing algorithm searches the number digit by digit, from left to right, trying to find the
corresponding destination route. The prefix list is browsed from top to bottom and the most specific
route is searched. If a route was matched, the call is routed on the corresponding outgoing class.

Figure 31 The "Routes" option

8.5.3.1 Editing a route


To edit a route's parameters, press "Edit" in the "Action" region. The editing window will
appear on screen, as shown below:

Figure 32 Editing a route

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The editable fields are:
Name name of the route;
Source Class name of the source client class; the incoming call class, check one of the created
classes or enter the keyword "Any " for any source class
Prefix the prefix associated to that route
Action defines the action type; the action can be "DIR", "DIRIP", "LCR", "SERV", "HUNT" or
"PORT"; any call with the prefix from the "Prefix" field will be redirected to the class with the ID from
the "Dest" field, if the selected option is "DIR";
Destination Class final destination for the prefix (the class with that ID represents the destination);
Service Type service type associated to that route. It is active only if the "Service" option is
selected in the "Action" region.
ANI Ignore / Insert the digits to be ignored from the caller ID sent through "Class Dest" (maximum
20 digits); the "Insert" field is used to add to the caller ID sent through "Class Dest" (maximum 16
digits);
DNIS Ignore / Insert the number of digits to ignore (skip) from the number sent through "Dest".
There is a list of values, with values from 0 to 20; the "Insert" field allows adding digits to the number
sent through "Class Dest"
Sign 1 - 6 fields for setting the signaling parameters. The parameters can be configured only when
creating that route.
Max connection time - the maximum duration allowed for the call;
Search Mode the search mode, depending on several parameters. The options available are ASR,
ACD, Priority, Down, Up, Circular, Percent, Fork Answer, Fork Ringing. If there are several
destinations with the same prefix and incoming class, one of the routing algorithms is used.
Play File option used for the voice mail service - name of the file to be run
Billing Profile IN / Out - the billing profile used for that client class. Billing profiles are created in the
Billing Profiles section;
Start Time the start time of the day, for that route's validity;
End Time the end time of the day, for that route's validity;
Days days of the week when that route is valid.

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8.5.3.2 Creating a new route


To create a new route, press the "New Route" button. The "Routing / New Route" window will
appear, where you can fill in the fields with the desired values. Next to the previously described
options, there are several more options you need to configure in order to finish entering a new route.

Figure 33 Creating a new route


Next to the Sign 1, Sign 2, Sign 4 fields, you have the Edit option. If you press this option,
new fields will appear:

Figure 34 New Route Sign 1


Number of Seconds number of seconds to wait for the number to be entered - last digit of the
parameter;
Number of Digits number of digits - values can be up to 20 (inclusive).

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Simulate Connection option to simulate the connection;
Restrict ID option to restrict the identity;
Check Operator option to check the operator;
Simulate Tax option to simulate the tax;
Retry Attempt option to re-establish the connection;
Transit transit option
Alloc BSS option to allocate BSS. Used when the answer tone must be identified to declare the
answer to a call. This option is useful when the application must differentiate between a call without
answer call and a call that is answered after such a tone. For example, if you want to avoid being
taxed by a voicemail !

Figure 35 New Route Sign 2


The "Sign 2" and "Sign 4" options are used for equipments that have SS7 routes defined.
Check Called Party NAI option to check the destination party for the call, and for this option, the
last two digits of the code next to Sign 2 are reserved.
If it is enabled, you can choose one of the following options:
Subscriber the number called is one of the subscribers;
Unknown the number called is unknown;
National the number called is a national number;
International the number called is an international number;
UK Specific the number called is UK specific.
The nature of the address for the number called is changed by enabling the first two options
"Check Called Party NAI" and "Override Called Party NAI".
The "Called Party NAI" parameter - Subscriber, Unknown, National, International and UK
Specific can be changed to "Override Called Party NAI", which contains the same list as the first
one.
Moreover, the "Calling Party NAI" parameter can be overridden by selecting a value from
"Override Calling Party NAI" and enabling the "Override Calling Party NAI" option.
The "Type of Media Required" region the route will only be available for the specified media
type. Possible values are "speech", "64k_unrestr" and "3K1Hz_audio". The "Translation Occurred"
option is used to indicate to SS7 that a number translation has occurred.

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Figure 36 New Route Sign 4


To use a higher priority for the incoming category, you must check the "Check IN Category"
and "Override IN Category" options. In the "IN Category" list select the incoming category to be
replaced by "Override IN Category". Possible values are: "unknown", "op_french", "op_english",
"op_german", "op_russian", "op_spain", "op_rsrv1", "op_rsrv2", "op_rsrv3", "notused", "ord_subscr",
"prio_subscr", "data_call", "test_call", "payphone", "uk_oper_call", "uk_admin_diverted".

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8.5.4 The "Access In" option


The "Access In" option allows you to associate calls from different IP addresses to incoming
client classes. It controls the IP access, more specifically the incoming traffic to TOPEX multiSwitch.
Before entering in the routing table, the call is checked by Access IN. The call must enter the
Softswitch with the parameters from Access IN, and if the call does not respect these parameters, it
will be rejected. Is used only for IP calls (SIP/H323).
To configure this option, use the "New Rule" button, which will open the "Access IN" window:

Figure 37 Entering a new rule in the "Access In" option


Each call has an incoming class associated, depending on the parameters:
IP / Netmask the IP or IP class from which calls are sent;
Port the incoming port (TCP/UDP) to which the calls are sent; if the field value is 0, then the port will
be configured depending on the SIP protocol: 5060 or H323:1720;
Protocol the protocol to which the IP belongs; this can be SIP, H323, R2S, CAS, ISDN, SS7,
depending on the equipment used
Class select the associated IP/Netmask class
Port (range) not used for multiSwitch
Prefix the prefix added to a translated number, required for calls incoming to the Softswitch with
different prefixes but from the same IP source. In case two such prefixes exist, the user can assign
two different directions (client classes) for the same IP source;
No. Digits this parameter controls the number of digits for each call that enters the Softswitch from
the specified IP;
End Cause the error message that will appear on screen in case the number of digits is not declared
in the previous field; the default value is 34;
Insert into ANI this field is used to add the specified digits to the Caller ID; the maximum number of
digits allowed is 16;
Ignore from ANI the number of digits to be ignored from the Caller ID; it can have a value between
0 and 20;

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Insert into DNIS the DNIS prefix to be added
Press the "Submit" button to add the record to the list, which will look like this:

Figure 38 Access list for incoming calls


If you want to change the entered data, use the "Edit" option, from the "Action" region. A
window will appear, where you can make the desired changes. You must then press "Submit" to
confirm your changes.
Note 1: In the IP field, you can either use a single IP (for example, 192.168.144.57), or use a IP range
(for example, "192.168.1.0 / 24").
If you use a single IP, without specifying the range, only calls sent from that IP will be
accepted. It is the same situation when you use the range / 32.
For example, when you use 192.168.1.0 / 24, the allowed IPs are 192.168.1.x;
For example, if you use 192.168.0.0 / 16, the allowed IPs are 192.168.x.x;
For example, if you use 192.0.0.0 / 8, the allowed IPs are 192.x.x.x;
But if you use 0.0.0.0 / 0, then all the x.x.x.x IPs are allowed (which means that calls are
accepted from all IPs!).

To delete a record, use the


"Delete" option, from the same
region, "Action". The following
window will appear on screen, and
you must select OK:

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8.5.5 The "Access Out" option


The "Access Out" option stores the destination IPs for each outgoing client class and controls
the outgoing traffic in TOPEX multiSwitch.
Access Out is used only for IP calls (SIP/H323). In Access Out you enter information about the
call destination.
For configuration, press the "New Rule" button, to enter records. A new window will appear,
with the configuration options:

Figure 39 The "Access Out" option


The user defines the following options:
Class the source name of the client class;
Host the destination IP;
Port port used for signaling (for example 1720 for H323 or 5060 for SIP);
Protocol the signaling protocol used (can be SIP or H323);
Transport The transport protocol used
Media Parameters the media parameters used
After configuring them, the records will be displayed as follows, and you will be able to edit
or delete them by using the "Edit" and "Delete" options in the Action region:

Figure 40 Displaying the records in the "Access Out" menu

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8.5.6 The "Global Rules" option


The "Global Rules" option allows you to define general rules for call administration. The most
popular application of this option is the Call Baring service.
The Call Barring service allows you to restrict the incoming or outgoing calls made to or by a
user.

Figure 41 The "Global Rules" option


Name name of the rule;
Type the rule type: to allow ("Allow") or restrict ("Restrict") the calls;
DNIS the DNIS number (received);
ANI the ANI number (caller ID);
After filling in the fields above, press the Submit button. You can fill in either the ANI or the
DNIS field.
Otherwise, the following error message will appear:

Figure 42 Error message when filling in ANI and DNIS

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8.6 The "Services" menu


The "Services" menu includes options for services associated to SIP users: SIP aliases,
Centrex, Hunting, Forking or Pick-up groups.
The options for the "Services" menu are:
-

SIP Aliases includes the aliases associated to a SIP client

Centrex Groups defining the Centrex groups

Hunting Groups - defining the Hunting groups

Pickup Groups - defining the Pickup groups

Forking Groups - defining the Forking groups

Figure 43 The "Services" menu

8.6.1 The "SIP Aliases" option


The "SIP Aliases" option includes all the aliases associated to a SIP user registered in the
database of the TOPEX multiSwitch equipment. When you enter the "SIP Aliases" menu, the following
window will appear on screen:

Figure 44 The "SIP User Alias" window


The window contains the following parameters:
ID the ID number of the SIP user;
Class the client class to which the SIP user belongs;
Username name of the SIP user;
Alias alias of the SIP user;
Name name for the alias;

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Press the "View" button to view the "Aliases" window under the "SIP Users" menu, where
aliases are defined.

Figure 45 Displaying the aliases in the "User Management SIP" menu


Note that there are two types of aliases: Global and Centrex. The first alias can be used at
global level, and the other can be used at Centrex group level.

8.6.2 The "Centrex groups" option


Here you can define the user groups named Centrex groups ("Centrex Groups"). Centrex
(Central Office Exchange Service) is a service offered by local telephony providers, that includes the
latest services for the users, so that they do not need to purchase these services separately.
Centrex represents a set of specialized solutions (generally for voice), where the equipment
that supports the call control and offers various services is owned by the service provider and located
at the provider level. As Centrex frees the client of costs and responsibilities related to owning the
equipment, it can be considered an external solution.
The client is saved the costs of technological updates (for example, the continuous update of
the switch system structure), and the phone company can offer a new set of services. In many cases,
Centrex has replaced the switch system. Central Office has effectively become a huge switch system
for all the local subscribers.
In most cases, Centrex (sold under different names depending on region) offers the clients a
level of service control that it equal or higher than PBX. In some cases, the phone company places the
Centrex equipment at a level that is accessible to the client.
Usually, the Centrex service includes Direct Inward Dialing (DID), sharing the same system for
several locations of the company, allocating a self-administration line, as well as advanced monitoring.
Call control and service logistics are clearly referring the required functions for processing a
phone call and for providing specific features. Below are some call control examples and some service
functions:
Detecting a pick-up and providing a dial tone;
Interpreting the dialed numbers to determine the call destination;
Determining that the called party is available, busy, or has the call redirect option
activated, and the applying the corresponding action (for example, ring tone, busy tone,
waiting tone, delivering the call to the voicemail or redirecting the call to another party);
Detecting the answer of the called party, ending the call by one of the parties and
registering the corresponding tax information.

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Figure 46 Adding a Centrex group


To add a Centrex group to the existing list, enter the name of the group in the "Group Name"
field, then press the "Add" button. The new group will be added to the list, in ascending order, in the
next free position.
To delete a group from the list, press "Delete" in the "Actions" region. The group will
automatically be deleted.
The "Actions" regions also contains a "View" button, which displays a window with details
about that group. In the window title bar you can see the name of the Centrex group. For example, in
the figure below, "Centrex Groups / Topex".

Figure 47 Viewing Centrex groups


The window contains details about the users included in that Centrex group. The fields
described here are:
User ID Identification number for the registered user;
Description a relevant description of the subscriber;
Username user name of that client (for example, the phone number used inside the Centrex group);
Alias aliases used to call a certain user included in the Centrex group. A user can have more aliases
through which they can be called.
The properties of each user in a Centrex group can be edited using the "Edit" button to the
right of the field. If you press that button, you will access the "Aliases" window in the "SIP Users"
menu.

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8.6.3 The "Hunting Groups" option


The "Hunting Groups" option allows you to define hunting groups for SIP users.
When a call is directed to a user and for some reason they cannot answer, the call will be
redirected to another user. If the second user is also busy, the call will be routed to a third user. This is
the Call Hunting service. The call routing algorithm is established depending on the "Call hunting
priority" parameter.
0 Warning: To accomplish the hunting operation, the users must be in the same hunting group!
Call hunting priority - it is used to establish the routing priority of a call from a busy destination to
another one. The algorithm must be a non linear one to avoid overcharging a certain user. In the case
of a linear algorithm, when the destination number is busy, the call will always be routed to the same
user, which will be overwhelmed;
To activate the Call Hunting option for a SIP user, first you have to create a hunting group in
the Services Hunting Groups menu.

Figure 48 Creating a hunting group


Access the configuration page for the SIP user for which you want to activate the hunting
service - User Management menu - SIP.

Figure 49 Call Hunting for SIP users


In the "Call Hunting Group" field, select the previously created hunting group to which the SIP
user will belong.

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8.6.4 The "Pickup Groups" option


The "Pickup Groups" option allows you to define pickup groups for SIP users.
To activate the Call Pickup option for a SIP user, first you have to create a pick-up group in
the Services Pickup Groups menu.

Figure 50 Creating a pickup group


Then, access the configuration page for the SIP user for which you want to activate the pickup
service - User Management menu - SIP.

Figure 51 Call Pickup for SIP users


In the "Call Pick Up Group" field, select the previously created pickup group to which the SIP
user will belong.

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8.6.5 The "Forking Groups" option


The "Forking Groups" option allows you to define forking groups for SIP users.
The Call Forking service can be used in two different ways. In the first case, you can
associate several SIP users to the same forking group. When a SIP user is called, then all the users in
that group will be called as well.
The second case refers to several SIP phones with the same user name and password, but
not the same IP address. For example, if you have several SIP phones in different locations (home,
office), with the same user name and password. If you include the SIP phones in a forking group, then
when one of the phones is called, all the SIP phones in that group will ring.
To activate the Call Forking option for a SIP user, first you have to create a forking group in
the Services Forking Groups menu.

Figure 52 Creating a forking group


Then, access the configuration page for the SIP user for which you want to activate the forking
service - User Management menu - SIP.

Figure 53 Call Forking for SIP users


In the "Forking Group" field, select the previously created forking group to which the SIP user
will belong.

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8.7 The "User Management" menu


This menu contains options for administration of the user interfaces. Here, all the client types
from the TOPEX Soft Switch database are grouped. The menu contains several submenus with details
about each client included in the TOPEX Soft Switch equipment structure. The submenus that
describe the client types are:
System system users;
SIP SIP users with IP phones;
FXS FXS subscribers that use common phone terminals;
ANI subscribers detected by their identity (phone number);
Prepaid users that have their calls restricted by the current credit, clients that use prepaid
cards;
SIP Locations list of the SIP users locations in the database

Figure 54 The "User Management" menu

8.7.1 The "System" option


The "System" option is dedicated for defining the system administrators for the TOPEX
multiSwitch equipment. Contains a list with these users and the functions they perform in the system.

Figure 55 System Users


To add a new system user, use the "New User" option. The following window will appear on
screen: "System User / New User":

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Figure 56 Adding a system administrator


Web User the Web user;
Web Password the web connection password;
Confirm confirmation of the password;
Role the system user role; this can be:

Administrator - user with unrestricted rights

Reseller - user with restricted rights;

Name name of the system user;


E-mail the user's e-mail address
Address the system user's address;
Additional Info additional information.

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After entering the desired options, press "Submit" to add the record to the list. After this, the
list will be displayed as follows:

Figure 57 Displaying the system users


In the top menu, some options appear which are significant for the system users:
ID the user's identification number;
Name name of the user;
Role role of the user;
Web Access web access;
User web user;
Actions you can change ("Edit") or delete ("Delete") the users in the list; the records with
administrator role can only be edited, not deleted;
Use the "Edit" option if you want to change the details for system users. In the window that
opens, you can see the following fields:
Web User name of the web user;
Web Password password for accessing the web page;
Confirm password confirmation field;
Role role for the system user - this is a configured option and cannot be changed by editing it;
If the role of the system user is Reseller, there are some default restrictions when entering
data in the web interface. On the other hand, the administrator has all the rights.
For Reseller, two submenus will be displayed in the editing window, to the left of the screen:
Main and Limits.

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Figure 58 Editing the properties for system users


The options in this window are:
Web Access the functions that system users can have are:
Enabled definitive, activated access;
Locked not allowed (password protected);
Can Edit Users has the right to edit users;
Can use global Classes has the right to use the global classes;
Can Create Trunks has the right to define trunks;
Can View Classes has the right to view classes;
Can Edit Classes has the right to edit classes;
Can View Routing has the right to view the routing;
Can Edit Routing has the right to change the routing;
Can View Users has the right to view the users;
Can Edit Users has the right to edit user options;
Can View Calls has the right to view calls;
Can View Billing has the right to view billing;

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Can Edit Billing has the right to edit billing.
Name name of the system user;
E-mail the system user's e-mail address
Address the system user's address;
Additional Info additional information about the system user
The "Limits" submenu contains options for restrictions applied to system users.
Note! The restrictions are only displayed for Reseller, not for Administrator, as the latter is granted all
the system rights.
The options displayed in this window are:
Name name of the reseller (default);
Max. User Classes maximum number of user classes allowed - default is "100";
Max. Client Classes maximum number of allowed clients default is "10";
Max. Access In maximum number of records allowed for incoming calls;
Max. Access Out maximum number of records allowed for outgoing calls;
Max. Routes maximum number of records allowed in the "Routes" menu;
Max. Customers maximum number of records in buyers;

Figure 59 The System menu - "Limits" submenu


Max. Accounts maximum number of accounts that can be entered;
Max. Centrex maximum number of Centrex groups that can be entered in the web interface;

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8.7.2 The "SIP" option


The "SIP" option contains the list of SIP users in the database of the TOPEX multiSwitch
equipment. It allows adding new users, editing existing accounts and deleting users.
By default, the window with defined SIP users will appear on screen:

Figure 60 SIP Users

8.7.2.1 Adding a SIP user


To add a new user, press the "New User" button or "New User From Template" in case you
want to create a predefined user.
In the SIP Users window, you will configure the parameters for the SIP user to be added in the
database of the TOPEX multiSwitch equipment.

Figure 61 Adding a SIP user


Billing group name of the billing group to which the SIP user will belong
Name name of the SIP user;
Class the previously created client class to which the SIP user will belong;

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Username and Password the user name and access password for the SIP client;
Centrex Group allows you to select the Centrex group to which the user will belong. Select one of
the Centrex groups previously defined in the Services Centrex Groups section.
Call Pickup Group allows you to select the Pick-up group to which the user can belong. Select one
of the Pickup groups previously defined in the Services Pickup Groups section.
Call Forking Group allows you to select the forking group to which the user can belong. Select one
of the forking groups previously defined in the Services Forking Groups section.
Call Hunting Group allows you to select the hunting group to which the user can belong. Select one
of the hunting groups previously defined in the Services Hunting Groups section.
Call Hunting Priority - select the priority for the call hunting option. You can select a value from 0 to
10. The value "0" has the highest priority;
RTP_Proxy you can select one of the following options: "Not Used", "Used except users in the same
NAT" or "Always Used";
Transcoding - the transcoding feature for each client - used for clients with different codecs;
Public / Private Access List for public and private IPs. The "Public" option is for public IPs and the
"Private" option is for private IPs in the SIP message header.
Rules Incoming / Outgoing incoming/outgoing rules for calls made/received by the SIP user.
Rejects Call with no ANI equivalent of the Anonymous Reject service rejects anonymous calls without ANI.
Publish Presence option that allows viewing a subscriber's status by other subscribers registered in
the equipment - presence service.
Do Not Disturb If this is checked, all the calls to this user are rejected
Options for call forwarding Call Forward

Call forward enables call forwarding when the user is in one of the statuses that determines
call routing;

Selective forwarding allows selective forwarding for the calls received by this SIP user

Call forward state this feature contains several statuses that determine the call forwarding:
-

"Offline" the user is not in the network

"Busy" the user is busy

"No Answer" the user does not answer

"Always" all the calls received by the user are forwarded. If you want to forward the
calls in all the situations above, select all the available options.

Number [Offline] the phone number to which the call is forwarded in case the user is in the
"Call Forward State Offline" status;

Number [No Answer] the phone number to which the call is forwarded in case the user is in
the "Call Forward State No Answer" status;

Number [Busy] the phone number to which the call is forwarded in case the user is in the
"Call Forward State Busy" status;

Number [Always and Selective] the phone number to which the call is forwarded in case
the user is in the "Call Forward State Always" forwarding status or in case the "Selective
Forwarding" option is selected.

Call Wait enables the call waiting service


Call Wait State the call wait status - checks if that option is also enabled in the phone
Voice Mail enabling the voicemail service
Voice Mail Number the number of the voicemail to which the calls will be forwarded;

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Voice Mail State this option contains several statuses that determine activation of the voicemail:
"Offline" the user is not in the network, "Busy" user is busy, :No Answer" user does not answer,
"Always" all the calls received are forwarded to the voicemail;
VoiceMail 2 Email state administrator setting for the voicemail 2 email service
VoiceMail 2 Email enabling the voicemail 2 email service
Missed Calls to Email enabling the missedcalls 2 email service, which sends missed calls
notifications to a specified e-mail address
Missed Calls to Email State administrator setting for the missed calls 2 email service
Missed Calls to Email Email the e-mail address where the missed calls notifications will be sent
Missed Calls to SMS enables the missedcalls 2 SMS service, which sends missed calls SMS
notifications to a specified phone number
Missed Calls to SMS State administrator setting for the missed calls 2 SMS service
Missed Calls to SMS Number phone number to which the missed calls notifications will be sent
Billing Profile the billing profile to which the SIP user will belong The profile is predefined in the
Billing Profiles menu.
After entering the desired parameters, press the Submit button to register the SIP user in the
database of the TOPEX multiSwitch equipment.
To change the details for already existing SIP users, press the "Edit" button, which will open
the "SIP User" window. Here, you have the option "Client id", which allows you to access the page
with details about the SIP users.
To delete a SIP user from the equipment database, press the corresponding "Delete" button.

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8.7.2.2 CLI options for a SIP user
At the top left, you can notice all the submenus for the Users / SIP option. The previously
open window is the Main window.
This option allows you to enable/disable the CLIP and CLIR options. In the SIP Users window,
from the left menu, choose the CLI option.

Figure 62 Configuring CLIP / CLIR


For a SIP user with a client ID and Use Name, you can configure the following options,
regardless of whether they belong or not to a Centrex group:
CLI Proxy displays the Caller ID for calls placed among SIP users. The field contains a list with all
the aliases associated to that SIP user;
Proxy Privacy enables the CLI Proxy option
CLI User Agent displays the Caller ID for calls placed by SIP users to external locations other than
SIP, which can be PSTN, IP junctions etc. The alias selected here will be displayed at the destination
instead of the source phone number;
User Agent Privacy enables the CLI User Agent option
CLI Centrex displays the Caller ID for calls among the SIP users in the same Centrex Group. For all
the calls received from those SIP users, the destination will see the alias selected in this field.
Centrex Privacy enables the CLI Centrex option
Display Name name displayed with the User Alias
Display Name Privacy enables the option to display the Caller ID at the destination

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8.7.2.3 Customized rules for SIP users
The "Rules" button allows you to edit the call restrictions for the SIP user.

Figure 63 Rules for SIP users


At the top of the window, you can see the "Client ID" fields (if you select the grayed out text,
you will enter the User Management / System menu).
The "Type" field has three options: "Allow", "Restrict" and "Forward" for SIP users calls.
You can allow, restrict or forward phone numbers for SIP users.
For example, if you select the "Restrict" option, enter "Restriction" in the "Name" field, then
enter the number "0212000000" in the "DNIS" field and the phone number "0720000000" in the "ANI"
field, then press the "Add" button to add the record to the list. All calls placed from the number
"0720000000" to the number "0212000000" will be rejected.

8.7.2.4 Aliases for SIP users


If you press the Alias option, you can access the window with the following options:

Figure 64 Aliases for SIP users


Here you can add aliases that can be used for that client. The page contains only one field,
"Alias", and the "Add" button. For the aliases you add to the list, you can check either the "Alias"
option (global alias, valid for all groups), or the "Centrex" option (aliases that are only valid for the
Centrex Group selected in the Main menu). After entering the alias in the specified field, press the
"Add" button to add it to the list. You can delete an alias using the "Delete" option.

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8.7.2.5 The Web Access option
The "Web Access" option opens a window displaying the users' web access:

Figure 65 Web access for SIP users


The fields for this option are the following:
Reseller administrative role of the user;
Billing Group billing group to which the user belongs - connects to the "Groups" submenu from the
Billing menu;
Web User the web user;
Web Password the web access password;
Confirm field to confirm the previously entered password;
Role user level role;
Web Access field with two options: Enabled (access enabled) and Locked (access restricted);
Name name of the system user;
E-mail the user's e-mail address
Address the user's address;
Additional Info additional information about the user;
After filling in the fields, press the "Submit" button.

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8.7.3 The "ANI" option


When you access the "ANI Users" menu in the administration web page, the following window
will appear on screen:

Figure 66 The "ANI Users" menu


The ANI users, also known as postpaid users, are identifiable by means of the call number.
The "ANI Users" window has the following options:
Reseller administration group to which the user belongs
ID identification number for that ANI user - postpaid client;
Name name of the ANI user;
Class client class to which that client belongs;
Changed Class alternative client class for the ANI user;
ANI identity - phone number of the caller;
Prefix determines if the ANI user will have a valid prefix associated;
CallBack State the status of the ANI user. The options for this feature are "Inactive" or "On Call";
Action options used for ANI users the options are "Edit" or "Delete".
If you press the "Edit" button, the following window will appear:

Figure 67 Editing the properties of an ANI user


In this window, you can change the following properties for the selected ANI user:
Class select the client class for the ANI user (for example ANIuser);
Changed Class select an alternative class for the ANI user;
ANI phone number of the postpaid client;
Prefix select this option if you want to use the prefix associated with the ANI user;
CallBack State the status of the ANI user. This option contains 4 statuses: "On Call", "On SMS",
"SMS to EMail" and "Allow In";

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CallBack phone number used for the call back option. This option is disabled by default, but it is
enabled when you select one of the options "On Call" or "On SMS" in the "CallBack State" field;
CallTo phone number of the third party. This option is also enabled by checking one of the options
"On Call" or "On SMS" in the "CallBack State" field;
SMS to EMail The e-mail address where SMS messages are sent. This option is enabled only if you
check the "SMS to EMail" checkbox next to the "CallBack State" field.
By default, the three fields at the bottom of the window ("CallBack", "CallTo" and "SMS to
EMail") are grayed out. When you select one or both of the "On Call", "On SMS" options, the
"CallBack" and "CallTo" fields will be enabled. The "SMS to E-Mail" is enabled by checking the "SMS
to EMail" checkbox. If the "Allow In" option is selected, the "On Call" and "On SMS" fields are disabled.
After you make your desired changes, you must press the "Submit" button (at the bottom of
the editing window) to save your changes.
The "Billing Group" button allows you to switch to Billing / Groups, with details about the
group to which the selected ANI user belongs, displayed in the Members submenu.
Also in this window, you can see the "Client ID" button, which will open another window with
details about the ANI user, such as Client ID, Class, User Type (in our case, ANI user), Web User,
Web Password and the confirmation field, e-mail address, a shirt description of the ANI user, web
access (with three options: User, Billing and Disable), name of the ANI user, address and additional
information about that user.
To add new ANI users, use the "New Entry" option, at the bottom of the main window. The
window that will open is similar to the editing window:

Figure 68 Adding ANI users


In this windows there are the same fields as in the previous one, and some additional fields:
Name name of the ANI user;
Billing Group billing group to which the user belongs; must be selected from the list;

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8.7.4 The "FXS" option


The "FXS" menu contains details about the FXS users - common analogue subscribers. When
you access this option, the "Subscribers" window will appear on screen:

Figure 69 The "Subscribers" menu


The parameters for FXS users are:
Reseller administrative group to which the user belongs;
ID identification number of the FXS user;
Name user name;
Class client class to which that user belongs;
ANI subscriber identity (Automat Number Identification);
Port physical port for connecting an analogue terminal;
Pickup Group number of the pickup group to which the subscriber is associated;
Hunting Group number of the hunting group to which the subscriber belongs;
Hunting Priority priority of the hunting group;
Action you have two options: editing the subscriber's properties ("Edit" button) or deleting the
subscriber ("Delete" button).

8.7.4.1 Editing the properties of the FXS subscriber


To edit the parameters for an analogue subscriber, press the "Edit" button to change the
parameters. The following window will appear on screen:

Figure 70 Editing the parameters of the FXS subscriber


In the editing page for FXS users, you can change the following parameters:
Class client class to which that user belongs;
Port the physical port number for the FXS phone terminal;
ANI identity of the FXS subscriber;

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Call forward enables call forwarding when the user is in one of the statuses that determines call
routing;
Call forward state this feature contains several statuses that determine call forwarding: "Offline"
the user is not in the network, "Busy" the user is busy, "No Answer" the user does not answer
and "Always" all the calls received by the FXS user are forwarded. If you want to forward the calls in
all the situations above, select all the available options.
Call forward number the phone number to which the call is forwarded. This option is used if the
subscriber is in one of the forwarding statuses or if at least one option is checked;
Voice Mail check this option to send voicemail to an unspecified e-mail address;
Voice Mail State voicemail is sent only if the subscribers are in one of the following statuses:
"Offline", "Busy", "No Answer" or "Always";
Voice Mail Number phone number for the voicemail;
Voice Mail to Email State check the corresponding checkbox to enable this option;
Voice Mail to Email The e-mail address where voicemail messages are sent.;
Call Pickup Group this option associates the FXS subscriber to a pickup group;
Call Hunting Group this option associates the FXS user to a hunting group;
Call Hunting Priority select the priority for the call hunting option. You can select a value from 0 to
10. The value "0" has the highest priority;
After you finish configuring the settings in this window, press "Submit" to save the new
parameters.

8.7.4.2 Adding a new FXS user


At the bottom of the "Subscribers" window is the "New User" button that allows you to add
another FXS user in the database.

Figure 71 Adding a new FXS user


The page contains approximately the same parameters as the editing page for FXS users,
except the "Billing Groups" field.

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8.7.5 The "Prepaid" option


This section contains all the details about users that identifiable by their PIN code. The users
must dial a certain code to enter the PIN code. The Softswitch checks if the PIN code entered is
correct and if it corresponds to the one in the database.
If TOPEX multiSwitch accepts the PIN code, the user can place a call to the destination. The
"Prepaid Users" window is similar to the one below, where you can see three prepaid user records:

Figure 72 The "Prepaid" menu


In Figure 72 you can see the identification number for the prepaid client, the description
(name) of the client, the client class to which the user belongs, the language used, the identity (ANI),
the current credit, Play time, Play Credit and the "Action" region with the "Edit" and "Delete" buttons.
The currency is determined depending on the client's necessities: EURO, USD etc.
To change the properties of prepaid clients, press the "Edit" button and the following window
will appear on screen:

Figure 73 Changing the properties of prepaid clients


In this page, you can change the client class for the prepaid user, the language used, the PIN
code, the value of the current credit and the ANI number. You can also use the "Play time" and "Play
Credit" options. Press the "Submit" button to save these settings.
Also in this window, you will see the "Client ID" field, which contains a button that will open
the user details window.
To add another prepaid user in the database, press the "New User" button at the bottom of
the main window of the "Prepaid" menu (Figure 72). The following window will appear on screen:

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Figure 74 Adding a new prepaid client in the database


To add a new prepaid client in the database, fill in the following fields:
-

Billing Group name of the Billing Group to which the prepaid user is associated;

Name name of the prepaid user;

Type type of the prepaid account. The available options are "Private" and "Shared";

Class list of the names for all the classes defined in the database. You can choose the
corresponding class for the new prepaid client;

Language selects the language used for voicemail. The two available options are English and
Romanian;

PIN the PIN code (Personal Identification Number) corresponding to the new prepaid client;

Re / Charge with the value to be recharged and the currency;

ANI Caller ID (Automatic Number Identification);

Play time check to activate the "Play time" feature for prepaid clients. In case the client receives
a dial tone and dials the destination number, they will receive a vocal prompt that will inform them
about the time available for that call;

Play credit check to activate the "Play credit" feature for prepaid clients. Before dialing the
destination number, the prepaid client will receive a vocal prompt that will inform them about the
current available credit for calls.

After filling in all the fields in this window, press the "Submit" button at the bottom of the
window to save the parameters of the new prepaid client.

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8.7.6 The "SIP Locations" option


This menu contains a list with the locations of all the SIP users in the TOPEX multiSwitch
database that were active at any moment. In this window, you can only see the locations of the SIP
users online at the time when the equipment's web interface was accessed.
If you press the "SIP Locations" button, a window will appear on screen, which will be similar
to the one below. In the example below, there is no online user.

Figure 75 The "SIP Locations" menu


The list of SIP locations contains several parameters, such as::
ID the identification number of the SIP location;
Client ID - the identification number of the registered user
Username name of the SIP location;
IP IP address of the Softswitch;
Contact has the following format<Username>@<IP address of the SIP location>:<Port number>;
Expire2 Softswitch registration time for a client. If this time (in seconds) expires, the SIP location will
be offline.
For example, if the "Expire2" parameter is 35 seconds (see figure above) and the client does not
register to Softswitch in this interval, the SIP location of that client will be offline.
RTP Proxy the application that transfers UDP packets from one specified port to another;
Transcoding this field contains two options: "No" or "Yes";
Call State indicates the status of the SIP location;
PID process identification number;
Details details (corresponding parameters) related to the SIP locations..

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8.8 The "Billing" menu


The "Billing" section in the administration web interface contains information about the
parameters of the calls placed via the TOPEX multiSwitch equipment.

Figure 76 Options for the "Billing" menu


The "Billing" menu contains the following options:
-

Profiles the Billing Profiles associated to the users of the TOPEX multiSwitch equipment

Subscriptions the subscription types for users

Currencies unique calculation of the invoices, then calculation in the currency selected by the
user;

Groups users are placed in groups and invoices are generated based on these groups (not for
each individual user).

8.8.1 The "Profiles" option


This section presents the method for calculating the invoice by an algorithm The cost of a
call is divided in two: the cost for the caller and the cost for the service provider.
To generate the billing, you must create billing profiles. A billing profile can be associated to
a SIP user, prepaid user or to a class. The application will search billing for the configured profile, in
order to enter the price. The billing profiles configured for users (SIP, prepaid etc.) have a higher
priority than the ones configured by classes.
If the price does not correspond to the billing profile of the users, the application will search
the billing profile of the class (only if it is configured)..
The fields in this menu are the following:
Reseller administrative group to which the reseller belongs;
ID the identification number of the profile;
Name name of the billing profile;
Currency currency in which the invoice is calculated;
Actions the options in this field are used for editing the invoice specific features ("Edit") or for
editing the data specific to this option ("Properties");
Press the "New Profile" button to enter a new profile; the window that opens is:

Figure 77 The "Billing Profiles" option


Enter the name of the billing profile and the currency in which the invoice will be calculated,
then press the "Submit" button. The new group will be added to the list, in the next free position. To
edit, press the "Properties" button, in the "Actions" region.

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After entering the profile, in order to be able to add the call details, press the "Edit" button; a
new window will open, that will contain the "New Period" option. Press this option:

Figure 78 Adding details about the calls placed


The fields in this window are:
Start start date of the invoice - the format is yyyy-mm-dd;
End end date of the invoice - the format is yyyy-mm-dd;
T0 the interval that determines the first time period that is not taxed, in seconds (for example, the
first 5 seconds in a call);
T1 the interval that determines the first time period that is not taxed and is indivisible (for example,
the first minute - 60 seconds - in a call);
Price for T1 determines the price for the T1 period in EUR;
T2 determines the number of seconds after T1, at which taxing will occur (meaning that after T1
seconds, taxing will occur every T2 seconds);
Price for T2 the price determined for each T2 interval in EUR;
Min. Cost the minimum cost for the call;
After entering these details, which are required to calculate the invoice, press the "Submit"
button to save the settings.
Note: If the calculated price is lower than the value in the field Min. Cost, then Min. Cost will be used
to calculate the invoice!

Figure 79 Displaying the billing details


You can edit these details using the "Edit" option in the Actions region; after completing your
changes, press "Submit" to save them.

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8.8.2 The "Subscriptions" option


This section allows you to add details related to user subscriptions, which means including
all the facilities from the user subscriptions.
Enter a new profile using the "New Profile" option. The window that appears on screen will
be the following:

Figure 80 Entering a new profile in the "Subscriptions" submenu


The fields in this window are:
Reseller the administrative group to which the profile belongs;
Profile Name name of the subscription type;
Currency currency in which the invoice is calculated;
Invoice Generation the period for generating the bill; this period can be one of the following: "Every
Week" weekly, "Every Two Weeks" once every two weeks, "Every Month" monthly;
After entering these data, press the "Submit" button, to add the profile to the list, in the next
available position.
In the main window, the following additional fields will appear next to the ones previously
filled in:
ID the identification number of the entered profile;
Actions the editing options("Edit") and the property details options ("Properties").
If you choose the "Properties" option, you can edit the profiles entered, but only the Invoice
Generation field can be changed (period for invoice generation); after making the desired changes,
press "Submit" to save.
The "Edit" option allows you to enter the period when the facilities are valid. Press the "Edit"
option, and in the window that appears, press the "New Period" option. The window that appears on
screen is the following:

Figure 81 Entering the period of validity


Enter the start and end date for the subscription options (in the format yyyy-mm-dd), and
press the "Submit" button. A new window will appear, with several fields, and at the bottom of the
window you can see the "New Entry" option. Choosing this option will bring the following window on
screen:

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Figure 82 The window for entering the subscription options


Fill in the following fields:
Name name of the profile;
Value the value of the invoice, in EUR;
Percent the minimum percentage of payment;
Volume Min the maximum volume;
Class choose the client class for which the invoice is created;
Actions includes the "Save" button, for saving the previous options.

8.8.3 The "Currencies" option


The "Currencies" region allows you to use a unique tax system at the Softswitch level. The
invoices are calculated in this system and then converted to the desired currency.
When opening this menu, the "Currency Manager" window appears. At the bottom, you can
see the "New Currency" option, used to enter tax variants.

Figure 83 Entering a new tax unit


The fields are:
Reseller administrative group;
Currency tax unit;
Rate conversion rate for the TAX unit; for example, if 1 tax unit in the Softswitch corresponds to 1
EUR, then enter the digit 1 in the first field of this option, and EUR in the other field.
Actions when entering matches between the equipment tax unit and the currency in which the bill is
calculated, the previously entered records are also displayed. The "Actions" region contains the
options: "Cancel" (to cancel a record that is being entered) and "Submit" (press this button to add the
record to the list).

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After entering the data, the list will look like this:

Figure 84 Tax units Currency Manager


In the main window of this menu you can see the same fields as above. Next to the Actions
field is the option to edit the records. After making the desired changes, press "Submit" to validate the
changes.

8.8.4 The "Groups" option


This section of the Billings menu allows you to create client groups, so that you can then
generate invoices for a whole group, and not for each individual user.
When you access the Billing "Groups" menu, you can see the option to enter a new
group: "New Group", under the table.

Figure 85 Entering a new billing group


The fields in this menu are the following:
Reseller administrative group;
Name name of the billing group;
Subscription the subscription type of the user;
Billing Address the billing address;
Additional Information additional information;
After filling in these fields, press the "Submit" button to add the group to the list, in the next
free position.

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The list will look like this:

Figure 86 Displaying the billing groups in the Billing menu

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8.9 The "Account Properties" menu


The "Account Properties" menu contains details related to the calls placed by users registered
in the database of the TOPEX multiSwitch equipment, allowing you to generate text files with CDRs for
billing; it contains the list of users in the system and the option of quitting the interface.
The options of the "Account Properties" menu are:
-

Call List list of calls placed, generation of CDR files

Address Book list of users registered in the system

Logout option to quit the web interface

Figure 87 The "Account Properties" menu

8.9.1 The "Call List" option


In this section are presented all the calls placed via the TOPEX multiSwitch equipment. Each
call in the list has several parameters that can be displayed in the main window, according to the client
requirements.
By default, the main page for the "Call List" option is similar to the one below:

Figure 88 The "Call list" option


At the top of the main page of the "Call List" menu is the "Filters" region, containing several
fields that will be detailed below.
The region contains three fields where you have to enter the time period for which to display
the calls placed via the TOPEX multiSwitch equipment.
These fields have the following meaning:
-

The first list indicates the year and month, in the format yyyy-mm;

The second list, labeled "From", indicates the starting period - the day when billing starts;

The third list, labeled "To", indicates the day when the billing period ends.

For example, if you select


, then in the call details
section you will see all the calls placed on March 13, 2007, together with the selected parameters.

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8.9.1.1 The "Filters" button


When accessing the "Filters" field, a menu similar to the one below will appear on screen. The
"Filters" menu contains the call parameters previously selected. Initially, all fields are grayed out.
Press the "Edit" option, and in the window that appears, press the "New Period" option.

Figure 89 The "Billing" menu "Filters" button


To activate the fields in this menu, you must check the checkbox corresponding to the
selected parameter.
The parameters initially displayed in this window are:
Type type of the call source can be a local subscriber ("TLI" option) or a junction ("TJI" option).
Next to the "Type" option, you can see a checkbox and a grayed out field. Check the
checkbox and the field next to it will be activated. Here, you can enter one of the two options available
for the "Type" parameter: TLI or TJI.
For example, if you enter the "TLI" option and you press the "Show / Filter" button, only calls
placed to local subscribers will be displayed.
Caller phone number of the caller party.
You can only choose to view the calls placed by a single client. Enter the desired number in
this field and press the "Show / Filter" button. You will see only the calls placed by the selected client.
Destination phone number of the destination party.
In this case, you can opt to view the calls received by a client. Enter the desired phone
number in this field and press the "Show / Filter" button. You will see all the calls received by a user.
Time start time of a call the format is hh:mm:ss. The current time is the one configured in the
TOPEX multiSwitch equipment.
You can select a certain period to view all the calls placed in that period. For example, if in the
two fields next to the "Time" parameter you enter the values "01:00 PM:00 01:50 PM:00", you can
see all the calls placed in this period.
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Duration duration of a call in seconds.
You can opt to view all the calls that have a minimum duration, which you specify in this field.
You only have to enter the minimum duration in the first field of the "Duration" parameter. For
example, if you enter "100", only the calls that have a minimum duration of 100 seconds will be
displayed.
If you want to view all the calls that have their duration included in a certain range, all you
have to do is fill in both fields and press the "Show / Filter" button.
For example, if the values entered are "30 - 60", then all the calls will be displayed that have
their duration within this range.
If you enter an invalid value in the "Duration" field, such as 30.5, you will get an error similar to
the one presented in Figure 90.

Figure 90 Duration error


Units number of the tax units used for calls;
Release call ending mode. There are several possibilities in case a call is ended. The possible
options are: ACONG, AINEX, AOK, ARELS, ATOUT, BBUSY, BCONG, BOK and BRELS.
The message indicates the mode in which the call was ended. The first character in each
message indicates the party that ended the call: A = caller or B = the destination party. The following
characters represent keywords that detail the call ending mode:
CONG indicates an equipment congestion not enough resources are available;
INEX the equipment does not recognize the caller. There are no defined routes;
OK indicates an answer from the destination;
RELS connection was freed in other way than reverse dial tone or busy tone;
TOUT indicates that the time counter has expired;
BUSY indicates the call ended after a busy tone situation.
In order to view only the calls that ended due to a certain cause, select the desired message
and press the "Show / Filter" button.
For example, if you select "BBUSY", all the calls will be displayed for which the destination is
busy and cannot answer.
A more complex filter can be seen in the next
image.
In this case, if you press the "Show / Filter"
button, only local calls will be displayed (TLI option)
that were placed by the user with the phone number
"311" to the destination "312", during the period
between 01:00 PM and 01:50 PM, with a minimum
duration of 3 seconds and a maximum duration of 100
seconds and in which the destination ended the call.
Q850 disconnection cause according to the ITU-T Q.850 standard. A number will be displayed (from
1 to 128), corresponding to a special disconnection cause. For example:
- 3 "No route to destination" This cause indicates the fact that the destination cannot be
reached, as the network through which the call is routed cannot serve that destination.

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-

21 "Call rejected" The equipment that sent this cause does not accept the call, although it
could accept it, as it is neither busy, nor incompatible. This cause can also be generated by the
network, indicating that the call was not deleted due to an additional restriction service The
diagnostics field can contain additional information related to additional services and rejection
causes.
111 "Protocol error unspecified" this cause is used to report a protocol error that occurred;
only in case no other protocol error is applicable.

CIMI 15 digit code referring to the SIM card used to place a call;
Selection the time range (in seconds) elapsed from the moment the call was received in Softswitch
and the destination party answered;
Packet Loss number of packets lost during the call. This parameter must have the value "0";
Jitter is an abrupt and undesirable variation of one of more signal features, such as the interval
between successive pulses, the amplitude of two successive cycles or the frequency/phase of
successive cycles. The jitter is a very important parameter in the design of communication lines. The
jitter value must be "0" to ensure adequate equipment quality;
Session ID identification number for each call session - this parameter consists of 8 characters. This
parameter is unique and only matches one session;
SIM number of the SIM card in the GSM Gateway;
GSM Cell GSM cell in the mobile network where the SIM card used for the call is registered. This
option is available for a call routed via a GSM port;
Source Direction direction for the source port - this is the incoming direction;
Source Port physical position of the source port - the number of the port used by the call;
Sig. Source IP IP address of the equipment that places the call (for example, the IP address of a
SIP phone);
Sig. Source Port port number of the source terminal;
RTP Source IP IP address for the source RTP;
RTP Source Port the number of the source RTP port;
Dest. direction name of the outgoing direction;
Dest. Port number of the destination port;
Sig. Dest IP IP address of the destination;
Sig. Dest Port physical position of the destination port;
RTP Dest IP destination RTP IP address;
RTP Dest Port number of the destination RTP port;
In Client ID incoming client identification number;
Out Client ID outgoing client identification number;
After selecting the desired parameters from the list above (checking the corresponding
checkboxes), all you must do is press the "Show / Filter" button, on the right of the "Filters" window.

8.9.1.2 Call details


In the "Call List" menu, under the "Filters" field, you will notice the main region of the window,
where all calls and their parameters are included. These parameters will be displayed for the calls that
go through the Softswitch in the time period indicated by the three buttons next to the "Show / Filter"
button.

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Figure 91 Displayed calls


On the title bar of this window, several options allow you to access a statistics page. The
current page is indicated by the highlighted number.
Next to the page number, you can see two buttons, with the following meaning: ">" allows you
to go to the next page and ">>" jumps to the last page in the call list.
In this section, only the parameters selected in the "Fields" menu will be displayed. In the
example above, the parameters indicate the call type ("type" field), the source phone number ("Caller"
field), the destination phone number ("Destination" field), the time the call started ("Time" field), the
duration of the call ("Duration" field) and the call release cause ("Release" field).
At the bottom of the page, you can see a statistics of the total number of calls and total
duration (in seconds) for the selected time period, as shown below.

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8.9.1.3 CDR Export


You can export CDR files (Call Detail Records) using the "Export" button at the bottom of the
"Call List" window.
The following dialog window will appear on screen, prompting you to save the text file:

Figure 92 Save text file


The file name contains the period for which CDRs are exported and has the ".csv" extension,
to allow easy handling in a database management application.
The file format is: "CDR-yyyymmdd-yyyymmdd.csv".
The generated file will contain all the parameters selected in "Call List", in text format, as
shown below:

Figure 93 Format of the CDR file

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8.9.2 The "Address Book" option


The "Address Book" option contains a list with users registered in the TOPEX multiSwitch
equipment, as well as their contact information (first name, last name, phone number, fax, e-mail,
company).
This represents the web address book of the equipment, allowing you to add and delete
records from the database.
To add a new equipment user, press the "New Entry" button, under the list of registered
users.

Figure 94 Adding a new user


In the "Address Book" window, fill in the following fields:
Private check this option to keep the user's data private
First Name user's first name
Last Name user's last name
Company client's company
E-Mail user's e-mail address
Phone (1) user's phone number
Phone (2) user's phone number
Mobile - user's mobile phone number
Fax user's fax number

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After filling in the fields described above, press "Submit". The following window will appear:

Figure 95 Address Book User Details


To edit the properties of the newly added user, press the "Edit" button at the bottom of the
window.
To delete the user from the equipment list, press the corresponding "Delete" button:

Figure 96 User list

8.9.3 The "Logout" option


Press the "Logout" button to quit the web interface for administration and configuration of the
TOPEX multiSwitch equipment.

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8.10 The "Testing" menu


The "Testing" menu is used, as the name suggests, for testing the TOPEX multiSwitch
equipment.
The menu allows you to generate users and rules and to simulate real situations that might
occur in the equipment.
The options of the "Testing" menu are:
User Generator allows you to generate users registered in the Softswitch
Rule Generator allows you to generate rules applicable to the users and calls in the equipment.

Figure 97 The "Testing" menu

8.10.1 The "User Generator" option


The "User Generator" option allows you to generate users in a certain client class, to test the
TOPEX multiSwitch equipment capabilities.

Figure 98 The "User Generator" option


The window above includes the following options:
Reseller reseller name
Billing Group billing group to which the users belong
Class client class that contains the users
Number of digits number of digits
Username user name together with the prefix and the suffix
Number of accounts number of accounts to be generated
After filling in the fields described above, press the "Submit Query" button at the bottom of
the window.

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8.10.2 The "Rule Generator" option


The "Rule Generator" option allows you to add configuration rules for each user's calls and for
each previously created client class the "User Generator" option.

Figure 99 The "Rule Generator" option


At the top of the "Rule Generator" window, you can see a warning message stating that the
routing rules generated in this section are for test purposes only and will not be used in other
purposes, such as call generation.
Below this message, there is a window with the reseller name and the number of rules to be
generated for the reseller - the "Number of rules" field.
After entering the number of rules to be generated, press the "Submit Query" button.

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8.11 The "Reports" menu


The "Reports" menu contains a series of extremely useful client reports, used to generate a
statistics for the number of incoming minutes, their cost, the margin resulted after routing and other
management modes for these features.
The "Reports" menu contains the following options:
Profitability [Brief] profitability report - separate balances for each client;
Profitability [In] profitability report for incoming calls - In balance
Profitability [Out] profitability report for outgoing calls - Out balance
Reliability [In] reliability report for incoming calls
Reliability [Out] - reliability report for outgoing calls

8.11.1 The "Profitability [Brief]" option


The "Profitability [Brief]" option renders a profitability report for a certain client. This is the
difference between the acquisition price and the selling price, in other words, the reseller's profit.
This is a detailed report for each client, indicating the receipts, sales as well as sources and
destinations.

Figure 100 Profitability report [Brief]


At the top of the window, there is a "Filters" section that includes the period for which the
report is generated and the currency used.
The report is displayed at the bottom of the window; it is a In/Out report for each Billing group,
with all the required parameters: calls, minutes, price, cost, percentage or value.

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8.11.2 The "Profitability [In]" option


The "Profitability [In]" option is used for the profitability report for a certain client, for incoming
calls.

Figure 101 The "Profitability [In]" option


The fields for this option are:
Billing Group select the billing group for which the report will be generated
Interval the time interval starting and ending dates - for which ROI reports will be generated
Detailed if checked, a detailed report will be generated

8.11.3 The "Profitability [Out]" option


The "Profitability [Out]" option is used for the profitability report for a certain client, for outgoing
calls.

Figure 102 The "Profitability [Out]" option


The fields for this option are:
Billing Group select the billing group for which the report will be generated
Interval the time interval starting and ending dates - for which ROI reports will be generated
Detailed if checked, a detailed report will be generated

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8.11.4 The "Reliability [In]" option


The "Reliability [In]" option presents a reliability report for incoming calls. This represents
statistics for each single client, depending on the generated CDRs (Call Detailed Records).
The reliability reports are calculated offline.

Figure 103 - The "Reliability [In]" option


The fields for this option are:
Billing Group select the billing group for which the report will be generated
Interval the time interval starting and ending dates - for which profitability reports will be generated
Detailed if checked, a detailed report will be generated

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8.11.5 The "Reliability [Out]" option


The "Reliability [Out]" option presents a reliability report for incoming calls. This represents
statistics for each single client, depending on the generated CDRs (Call Detailed Records).
The reliability reports are calculated offline.

Figure 104 - The "Reliability [In]" option


The fields for this option are:
Billing Group select the billing group for which the report will be generated
Interval the time interval starting and ending dates - for which ROI reports will be generated
Detailed if checked, a detailed report will be generated

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8.12 The "Templates" menu


The "Templates" menu contains a series of templates (patterns) that can be applied to clients
and client classes to facilitate the adding and registration in the database of the TOPEX multiSwitch
equipment.
The "Templates" menu contains the following options:
Class Templates template for client classes
SIP User Templates template for SIP users
FXS User Templates template for FXS users
ANI User Templates template for ANI users
Prepaid User Templates template for Prepaid users

Figure 105 The "Templates" menu

8.12.1 The "Class Templates" option


The "Class Templates" option contains templates for the client classes to be added in the
database of the TOPEX multiSwitch equipment.
You can create a template for a client class, and when adding a client class from the "Server
Management" menu, you can use the "New Class from Template" option, which will simplify your work.
This option is recommended when massive data volumes must be entered in the system.
The fields that have to be filled in are similar to the ones in the "Client Classes" menu.
Billing Group associate the client class with a previously defined billing group
Name name of the client class to be created
Type select the Class option to create a client class. The other available option is "Port".
Trunk If this option is checked, the client class will be displayed as a route for "Destination Class"
(the Reseller can select that client class as route to "Destination Class");
Billing Profile the billing profile used for that client class. Billing profiles are created in the Billing
Profiles section;
Search Mode used for routing when same prefix routes are used. The available options are:
-

Up The route will be selected starting from the last to the first. Depending on its position in the
routing table, the last route in the group will have the highest priority.

Down The route will be selected starting from the first route to the last. Depending on its position
in the routing table, the first route in the group will have the highest priority.

Circular - The route will be selected circularly.

RTP Proxy In / Out represents an application that runs on the TOPEX multiSwitch equipment and
transfers UPD packets from one port to another. RTP Proxy is used for transferring the RTP voice
packets behind the NAT. For each call, two UDP ports are opened. The application directs the UPD
packets received on one port to another port. Here, you can select one of the following options: "Not
Used", "Used except users in the same NAT" or "Always Used";

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Transcoding In / Out the transcoding feature for each client class - used for clients with different
codecs;

Figure 106 Creating a "Class Template"


Rules the incoming and outgoing rules; select "In" and/or "Out" if you intend to enforce incoming
and/or outgoing rules. Depending on the option you choose here, you must fill in the following fields:
-

Max Calls maximum number of calls allowed for incoming and outgoing;

Call Rate number of calls per second, incoming and outgoing;

Max Taxunits maximum number of units (pulses) charged;

Max Time maximum interval allowed for connection; if this threshold is reached, the connection
is closed;

Max Time maximum duration of the call;

TOS Type of Service packet prioritization service.

Translate the option to add a prefix in front of a number; ANI and/or DNIS numbers can be selected:
-

Ignore from ANI the prefix to be deleted from the ANI number. Ignores the specified number of
digits from the caller's ID (maximum 20 digits);

Insert into ANI the prefix to be added to the ANI number. Adds the specified digits (maximum
16 digits) to the Caller ID;

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-

Ignore from DNIS the prefix to be deleted from the DNIS number. Specifies the number of digits
to be ignored from the numbers received on that direction (client class); the first x digits from the
received number will be ignored (maximum 20 digits);

Insert into DNIS the prefix to be added to the number received on that direction (maximum 16
digits can be entered);

Max ANI /DNIS the maximum number of incoming digits for ANI/DNIS;

Sign 1-6 signaling parameters for configuring the equipment.


After configuring the parameters you desire, press "Submit" to create the client class.

8.12.2 The "SIP User Templates" option


The "SIP User Templates" option contains templates for SIP users to be added in the
database of the TOPEX multiSwitch equipment.
You can create a template for a SIP user, and when adding a user from the "User
Management - SIP" menu, you can use the "New User from Template" option, which will simplify your
work.
This option is recommended when massive data volumes must be entered in the system.
The fields that must be filled in are similar to the ones in the "SIP" menu.
Billing group name of the billing group to which the SIP user will belong
Name name of the SIP user;
Class the previously created client class to which the SIP user will belong;
Username and Password the user name and access password for the SIP client;
Centrex Group allows you to select the Centrex group to which the user will belong. Select one of
the Centrex groups previously defined in the Services Centrex Groups section.
Call Pickup Group allows you to select the Pick-up group to which the user can belong. Select one
of the Pickup groups previously defined in the Services Pickup Groups section.

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Figure 107 Adding "SIP User Template"


Call Forking Group allows you to select the forking group to which the user can belong. Select one
of the forking groups previously defined in the Services Forking Groups section.
Call Hunting Group allows you to select the hunting group to which the user can belong. Select one
of the hunting groups previously defined in the Services Hunting Groups section.
Call Hunting Priority - select the priority for the call hunting option. You can select a value from 0 to
10. The value "0" has the highest priority;
RTP_Proxy you can select one of the following options: "Not Used", "Used except users in the same
NAT" or "Always Used";
Transcoding - the transcoding feature for each client - used for clients with different codecs;
Public / Private Access List for public and private IPs. The "Public" option is for public IPs and the
"Private" option is for private IPs in the SIP message header.
Rules Incoming / Outgoing incoming/outgoing rules for calls placed/received by the SIP user.
Rejects Call with no ANI equivalent of the Anonymous Reject service rejects anonymous calls without ANI.
Publish Presence option that allows viewing a subscriber's status by other subscribers registered in
the equipment - presence service.
Do Not Disturb If this is checked, all the calls to this user are rejected
Options for call forwarding Call Forward

Call forward enables call forwarding when the user is in one of the statuses that determines
call forwarding;

Selective forwarding allows selective forwarding for the calls received by this SIP user

Call forward state this feature contains several statuses that determine the call forwarding:

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-

"Offline" the user is not in the network

"Busy" the user is busy

"No Answer" the user does not answer

"Always" all the calls received by the user are forwarded. If you want to forward the
calls in all the situations above, select all the available options.

Number [Offline] the phone number to which the call is forwarded in case the user is in the
"Call Forward State Offline" status;

Number [No Answer] the phone number to which the call is forwarded in case the user is in
the "Call Forward State No Answer" status;

Number [Busy] the phone number to which the call is forwarded in case the user is in the
"Call Forward State Busy" status;

Number [Always and Selective] the phone number to which the call is forwarded in case
the user is in the "Call Forward State Always" forwarding status or in case the "Selective
Forwarding" option is selected.

Call Wait enables the call waiting service


Call Wait State the call wait status - checks if that option is also enabled in the phone
Voice Mail enables the voicemail service
Voice Mail Number the number of the voicemail to which the calls will be forwarded;
Voice Mail State this option contains several statuses that determine activation of the voicemail:
"Offline" the user is not in the network, "Busy" user is busy, "No Answer" user does not answer,
"Always" all the calls received are forwarded to the voicemail;
VoiceMail 2 Email state administrator setting for the voicemail 2 email service
VoiceMail 2 Email enabling the voicemail 2 email service
Missed Calls to Email enabling the missedcalls 2 email service, which sends missed calls
notifications to a specified e-mail address
Missed Calls to Email State administrator setting for the missed calls 2 email service
Missed Calls to Email Email the e-mail address where the missed calls notifications will be sent
Missed Calls to SMS enables the missedcalls 2 SMS service, which sends missed calls SMS
notifications to a specified phone number
Missed Calls to SMS State administrator setting for the missed calls 2 SMS service
Missed Calls to SMS Number phone number to which the missed calls notifications will be sent
Billing Profile the billing profile to which the SIP user will belong The profile is predefined in the
Billing Profiles menu.

8.12.3 The "FXS User Templates" option


The "FXS User Templates" option contains templates for analogue users to be added in the
database of the TOPEX multiSwitch equipment.
You can create a template for a FXS user, and when adding a client from the "User
Management - FXS" menu, you can use the "New User from Template" option, which will simplify your
work.
This option is recommended when massive data volumes must be entered in the system.
The fields that must be filled in are similar to the ones in the "FXS" menu.

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Figure 108 Adding "FXS User Template"


Class client class to which that user belongs;
Port the physical port number for the FXS phone terminal;
ANI identity of the FXS subscriber;
Call forward enables call forwarding when the user is in one of the statuses that determines call
routing;
Call forward state this feature contains several statuses that determine call forwarding: "Offline"
the user is not in the network, "Busy" the user is busy, "No Answer" the user does not answer
and "Always" all the calls received by the FXS user are forwarded. If you want to forward the calls in
all the situations above, select all the available options.
Call forward number the phone number to which the call is forwarded. This option is used if the
subscriber is in one of the forwarding statuses or if at least one option is checked;
Voice Mail check this option to send voicemail to an unspecified e-mail address;
Voice Mail State voicemail is sent only if the subscribers are in one of the following statuses:
"Offline", "Busy", "No Answer" or "Always";
Voice Mail Number phone number for the voicemail;
Voice Mail to Email State check the corresponding checkbox to enable this option;
Voice Mail to Email The e-mail address where voicemail messages are sent.;
Call Pickup Group this option associates the FXS subscriber to a pickup group;
Call Hunting Group this option associates the FXS user to a hunting group;
Call Hunting Priority select the priority for the call hunting option. You can select a value from 0 to
10. The value "0" has the highest priority;

8.12.4 The "ANI User Templates" option


The "ANI User Templates" option contains templates for ANI users to be added in the
database of the TOPEX multiSwitch equipment.
You can create a template for a ANI user, and when adding a client from the "User
Management - ANI" menu, you can use the "New User from Template" option, which will simplify your
work.
This option is recommended when massive data volumes must be entered in the system.
The fields that must be filled in are similar to the ones in the "ANI" menu.

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Figure 109 Adding a "ANI User Template"


In this window, you can change the following properties for the selected ANI user:
Class select the client class for the ANI user (for example ANIuser);
Changed Class select an alternative class for the ANI user;
ANI phone number of the postpaid client;
Prefix select this option if you want to use the prefix associated with the ANI user;
CallBack State the status of the ANI user. This option contains 4 statuses: "On Call", "On SMS",
"SMS to EMail" and "Allow In";
CallBack phone number used for the call back option. This option is disabled by default, but it is
enabled when you select one of the options "On Call" or "On SMS" in the "CallBack State" field;
CallTo phone number of the third party. This option is also enabled by checking one of the options
"On Call" or "On SMS" in the "CallBack State" field;
SMS to EMail The e-mail address where SMS messages are sent. This option is enabled only if you
check the "SMS to EMail" checkbox next to the "CallBack State" field.

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8.12.5 The "Prepaid User Templates" option


The "Prepaid User Templates" option contains templates for prepaid users to be added in the
database of the TOPEX multiSwitch equipment.
You can create a template for a prepaid user, and when adding a client from the "User
Management - Prepaid" menu, you can use the "New User from Template" option, which will simplify
your work.
This option is recommended when massive data volumes must be entered in the system.
The fields that must be filled in are similar to the ones in the "Prepaid" menu.

Figure 110 Adding a "Prepaid User Template"


To add a new prepaid client in the database, fill in the following fields:
-

Billing Group name of the Billing Group to which the prepaid user is associated;

Name name of the prepaid user;

Type type of the prepaid account. The available options are "Private" and "Shared";

Class list of the names for all the classes defined in the database. You can choose the
corresponding class for the new prepaid client;

Language selects the language used for voicemail. The two available options are English and
Romanian;

PIN the PIN code (Personal Identification Number) corresponding to the new prepaid client;

Re / Charge with the value to be recharged and the currency;

ANI Caller ID (Automatic Number Identification);

Play time check to activate the "Play time" feature for prepaid clients. In case the client receives
a dial tone and dials the destination number, they will receive a vocal prompt that will inform them
about the time available for that call;

Play credit check to activate the "Play credit" feature for prepaid clients. Before dialing the
destination number, the prepaid client will receive a vocal prompt that will inform them about the
current
available
credit
for
calls.

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ACTIVATION AND CONFIGURATION OF THE SERVICES


9.1 SIP users registration

Adding and registering a SIP user in the TOPEX multiSwitch equipment configuration implies
a series of steps that will be detailed below. The same procedure as configuring and activating the
Class 5 services is used via the web interface for administration and configuration of the equipment.
The process of registering SIP users has two steps:
1 creating a client class (directs) to which the added user will belong
2 adding and registering the SIP user

9.1.1 Creating a client class (directs)


Access the Server Management Client Classes menu.

Figure 111 Client Classes


You can create a new route by pressing the "New Class" button. In the Client Classes / New
Class window you can change the following settings:
Billing Group associate the client class with a previously defined billing group
Name name of the client class to be created
Type select the Class option to create a client class. The other available option is "Port".
Trunk If this option is checked, the client class will be displayed as a route for "Destination Class"
(the Reseller can select that client class as route to "Destination Class");
Billing Profile the billing profile used for that client class. Billing profiles are created in the Billing
Profiles section;
Search Mode used for routing when same prefix routes are used. The available options are:
-

Up The route will be selected from the last to the first. Depending on its position in the routing
table, the last route in the group will have the highest priority.

Down The route will be selected starting from the first route to the last. Depending on its position
in the routing table, the first route in the group will have the highest priority.

Circular - The route will be selected circularly.

RTP Proxy In / Out represents an application that runs on the TOPEX multiSwitch equipment and
transfers UPD packets from one port to another. RTP Proxy is used for transferring the RTP voice
packets behind the NAT. For each call, two UDP ports are opened. The application directs the UPD
packets received on one port to another port. Here, you can select one of the following options: "Not
Used", "Used except users in the same NAT" or "Always Used";
Transcoding In / Out the transcoding feature for each client class - used for clients with different
codecs;

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Figure 112 Creating a client class


Rules the incoming and outgoing rules; select "In" and/or "Out" if you intend to enforce incoming
and/or outgoing rules. Depending on the option you choose here, you must fill in the following fields:
-

Max Calls maximum number of calls allowed for incoming and outgoing;

Call Rate number of calls per second, incoming and outgoing;

Max Taxunits maximum number of units (pulses) charged;

Max Time maximum interval allowed for connection; if this threshold is reached, the connection
is closed;

Max Time maximum duration of the call;

TOS Type of Service packet prioritization service.

Translate he option to add a prefix in front of a number; ANI and/or DNIS numbers can be selected:
-

Ignore from ANI the prefix to be deleted from the ANI number. Ignores the specified number of
digits from the caller's ID (maximum 20 digits);

Insert into ANI the prefix to be added to the ANI number. Adds the specified digits (maximum
16 digits) to the Caller ID;

Ignore from DNIS the prefix to be deleted from the DNIS number. Specifies the number of digits
to be ignored from the numbers received on that direction (client class); the first x digits from the
received number will be ignored (maximum 20 digits);

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Insert into DNIS the prefix to be added to the number received on that direction (maximum 16
digits can be entered);

Max ANI /DNIS the maximum number of incoming digits for ANI/DNIS;

Sign 1-6 signaling parameters for configuring the equipment.


After configuring the parameters you desire, press "Submit" to create the client class.

9.1.2 Adding SIP users


Access the User Management SIP menu, and in the SIP Users window, press the "New
User" button.

Figure 113 SIP Users list


In the SIP Users window, you will configure the parameters for the SIP user to be added in the
database of the TOPEX multiSwitch equipment.
Billing group name of the billing group to which the SIP user will belong
Name name of the SIP user;
Class the previously created client class to which the SIP user will belong;
Username and Password the user name and access password for the SIP client;
Centrex Group allows you to select the Centrex group to which the user will belong. Select one of
the Centrex groups previously defined in the Services Centrex Groups section.
Call Pickup Group allows you to select the Pick-up group to which the user can belong. Select one
of the Pickup groups previously defined in the Services Pickup Groups section.

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Figure 114 Adding a SIP user


Call Forking Group allows you to select the forking group to which the user can belong. Select one
of the forking groups previously defined in the Services Forking Groups section.
Call Hunting Group allows you to select the hunting group to which the user can belong. Select one
of the hunting groups previously defined in the Services Hunting Groups section.
Call Hunting Priority - select the priority for the call hunting option. You can select a value from 0 to
10. The value "0" has the highest priority;
RTP_Proxy you can select one of the following options: "Not Used", "Used except users in the same
NAT" or "Always Used";
Transcoding - the transcoding feature for each client - used for clients with different codecs;
Public / Private Access List for public and private IPs. The "Public" option is for public IPs and the
"Private" option is for private IPs in the SIP message header.
Rules Incoming / Outgoing incoming/outgoing rules for calls made/received by the SIP user.
Rejects Call with no ANI equivalent of the Anonymous Reject service rejects anonymous calls without ANI.
Publish Presence option that allows viewing a subscriber's status by other subscribers registered in
the equipment - presence service.
Do Not Disturb If this is checked, all the calls to this user are rejected
Options for call forwarding Call Forward

Call forward enables call forwarding when the user is in one of the statuses that determines
call routing;

Selective forwarding allows selective forwarding for the calls received by this SIP user

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Call forward state this feature contains several statuses that determine the call forwarding:
-

"Offline" the user is not in the network

"Busy" the user is busy

"No Answer" the user does not answer

"Always" all the calls received by the user are forwarded. If you want to forward the
calls in all the situations above, select all the available options.

Number [Offline] the phone number to which the call is forwarded in case the user is in the
"Call Forward State Offline" status;

Number [No Answer] the phone number to which the call is forwarded in case the user is in
the "Call Forward State No Answer" status;

Number [Busy] the phone number to which the call is forwarded in case the user is in the
"Call Forward State Busy" status;

Number [Always and Selective] the phone number to which the call is forwarded in case
the user is in the "Call Forward State Always" forwarding status or in case the "Selective
Forwarding" option is selected.

Call Wait enables the call waiting service


Call Wait State the call wait status - checks if that option is also enabled in the phone
Voice Mail enabling the voicemail service
Voice Mail Number the number of the voicemail to which the calls will be forwarded;
Voice Mail State this option contains several statuses that determine activation of the voicemail:
"Offline" the user is not in the network, "Busy" user is busy, "No Answer" user does not answer,
"Always" all the calls received are forwarded to the voicemail;
VoiceMail 2 Email state administrator setting for the voicemail 2 email service
VoiceMail 2 Email enabling the voicemail 2 email service
Missed Calls to Email enabling the missedcalls 2 email service, which sends missed calls
notifications to a specified e-mail address
Missed Calls to Email State administrator setting for the missed calls 2 email service
Missed Calls to Email Email the e-mail address where the missed calls notifications will be sent
Missed Calls to SMS enables the missedcalls 2 SMS service, which sends missed calls SMS
notifications to a specified phone number
Missed Calls to SMS State administrator setting for the missed calls 2 SMS service
Missed Calls to SMS Number phone number to which the missed calls notifications will be sent
Billing Profile the billing profile to which the SIP user will belong The profile is predefined in the
Billing Profiles menu.
After entering the desired parameters, press the Submit button to register the SIP user in the
database of the TOPEX multiSwitch equipment.

9.2 Configuring the Call Waiting option


The Call Waiting option is configured in the web interface of the TOPEX multiSwitch
equipment. Enter the User Management SIP menu and open the SIP Users window.

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Figure 115 Enabling Call Waiting


Check the Call Wait or Call WaitState checkboxes, depending on the rights granted to the
client. "Call Wait" is the administrator setting, and "Call Wait State" is the user setting. The SIP user
will have the Call Waiting service enabled.
The Call Waiting function also has to be configured from the user's phone terminal. Which is
why user's phones must support this feature.

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9.3 Configuring the Do Not Disturb option


From the equipment web interface, enter the User Management SIP menu and open the
SIP Users window.

Figure 116 Enabling Do Not Disturb


To enable this service, check the "Do Not Disturb" option, in the "SIP Users" window. This
option enables the rejecting of all calls forwarded to that SIP user.

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9.4 Configuring the Call Forward service


The Call Forward service allows forwarding of the calls when the user is in one of the call
forward statuses (Call forward state).
From the equipment web interface, enter the User Management SIP menu and open the
SIP Users window.

Figure 117 Configuring the Call Forward service


To configure the Call Forward service, you must configure the following options:

Call forward enables call forwarding when the user is in one of the statuses that determines
call routing;

Call forward state this feature contains several statuses that determine the call forwarding:
-

"Offline" the user is not in the network

"Busy" the user is busy

"No Answer" the user does not answer

"Always" all the calls received by the user are forwarded. If you want to forward the
calls in all the situations above, select all the available options.

Number [Offline] the phone number to which the call is forwarded in case the user is in the
"Call Forward State Offline" status;

Number [No Answer] the phone number to which the call is forwarded in case the user is in
the "Call Forward State No Answer" status;

Number [Busy] the phone number to which the call is forwarded in case the user is in the
"Call Forward State Busy" status;

Number [Always and Selective] the phone number to which the call is forwarded in case
the user is in the "Call Forward State Always" forwarding status or in case the "Selective
Forwarding" option is selected.

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9.5 Configuring the Call Back service


To enable the Call Back service for a certain route, use the web interface of the TOPEX
multiSwitch equipment.
Enter the Server Management Client Classes menu. Edit the client class for which you
wish to enable the Call Back service. Check the "Check Call Back" option in the Sign 1 field.

Figure 118 "Check Call Back" option


Now the Call Back option is enabled for the selected client class.
This service is configured from the ANI table - Server Management ANI menu.

Figure 119 Configuring the Call Back service in the ANI table
CallBack State the status of the ANI user. This option contains 4 statuses: "On Call", "On SMS",
"SMS to EMail" and "Allow In";
CallBack phone number used for the call back option. This option is disabled by default, but it is
enabled when you select one of the options "On Call" or "On SMS" in the "CallBack State" field;
CallTo phone number of the third party. This option is also enabled by checking one of the options
"On Call" or "On SMS" in the "CallBack State" field;
SMS to EMail The e-mail address where SMS messages are sent. This option is enabled only if you
check the "SMS to EMail" checkbox next to the "CallBack State" field.
By default, the three fields at the bottom of the window ("CallBack", "CallTo" and "SMS to
EMail") are disabled. When you select one or both of the "On Call", "On SMS" options, the "CallBack"
and "CallTo" fields will be enabled. The "SMS to E-Mail" field is enabled by checking the "SMS to
EMail" checkbox. If the "Allow In" option is selected, the "On Call" and "On SMS" fields are disabled.

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9.6 Configuring the Call Hunting option


When a call is directed to a user and for some reason they cannot answer, the call will be
redirected to another user. If the second user is also busy, the call is routed to a third user. This is the
Call Hunting service. The order of routing the calls is established depending on the "Call hunting
priority" parameter;
0 Warning: To accomplish the hunting operation, the users must be in the same hunting group!
Call hunting priority - it is used to establish the priority to route a call from a busy destination to
another one. The algorithm must be a non linear one to avoid overcharging of a certain user. In the
case of a linear algorithm, when the destination number is busy, the call will be routed always to the
same user which will be overwhelmed;
To activate the Call Hunting option for a SIP user, first you have to create a hunting group in
the Services Hunting Groups menu.

Figure 120 Creating a hunting group


Access the configuration page for the SIP user for which you want to activate the hunting
service - User Management menu - SIP.

Figure 121 Call Hunting for SIP users


In the "Call Hunting Group" field, select the previously created hunting group to which the SIP
user will belong.

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9.7 Configuring the Call Pick-up option


To activate the Call Pickup option for a SIP user, first you have to create a pick-up group in
the Services Pickup Groups menu.

Figure 122 Creating a pickup group


Then, access the configuration page for the SIP user for which you want to activate the pickup
service - User Management menu - SIP.

Figure 123 Call Pickup for SIP users


In the "Call Pick Up Group" field, select the previously created pickup group to which the SIP
user will belong.

9.8 Configuring the Call Forking service


The Call Forking service can be used in two different ways. In the first case, you can
associate several SIP users to the same forking group. When a SIP user is called, then all the users in
that group will be called as well.
The second case refers to several SIP phones with the same user name and password, but
not the same IP address. For example, if you have several SIP phones in different locations (home,
office), with the same user name and password. If you include the SIP phones in a forking group, then
when one of the phones is called, all the SIP phones in that group will ring.
To activate the Call Forking option for a SIP user, first you have to create a forking group in
the Services Forking Groups menu.

Figure 124 Creating a forking group


Then, access the configuration page for the SIP user for which you want to activate the forking
service - User Management menu - SIP.

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Figure 125 Call Forking for SIP users


In the "Forking Group" field, select the previously created forking group to which the SIP user
will belong.

9.9 Configuring the Suspend Subscriber service


The "Suspend Subscriber" service for a SIP user is enabled from the editing menu for that
user's parameter.

Figure 126 Enabling the Suspend Subscriber service


Enter the User Management - SIP menu and open the SIP Users window, then press the
"Edit" button next to the user for which you wish to enable the service.
The "Suspend Subscriber" service is enabled by selecting the "Disabled" option in the
"Account State" menu.

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9.10 ACL Configuration


Access Control List is a type of Black List/White - it specifies the IP addresses that have the
right to place calls, the number of minutes allowed, the number of concurrent calls, CPS (calls per
second) or the maximum allowed cost for each IP in the list.
The ACL option is configured from the Server Management Access In menu. In the
"Access In" window, configure the following parameters:

Figure 127 "Access In" window


IP / Netmask IP address and source netmask for ACL
Port the port by which it is allowed to communicate with the address(es)
Protocol the protocol used. Available options are: SIP, H.323, R2S, CAS, ISDN, SS7
Class client class associated to the IP
Prefix the prefix to be added to a translated number, required for the calls that enter the Softswitch
with different prefixes, but from the same IP source. In case two such prefixes exist, the user can
assign two different directions (client classes) for the same IP source;
No. Digits this parameter controls the number of digits for each call that enters the Softswitch from
the specified IP;
End Cause the release cause in case the number of digits is not declared in the previous field; the
default value is 34;
Ignore from ANI the number of digits to be ignored from the Caller ID; it can have a value between
0 and 20;
Insert into ANI this field is used to add the specified digits to the Caller ID; the maximum number of
digits allowed is 16;
Ignore from DNIS the prefix to be deleted from the DNIS number. Specifies the number of digits to
be ignored from the numbers received on that direction (client class); the first x digits from the received
number will be ignored (maximum 20 digits);
Insert into DNIS the prefix to be added to the number received on that direction (maximum 16 digits
can be entered).

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9.11 Configuring the Call Baring service


The Call Barring service allows you to restrict the incoming or outgoing calls made to or by a
user.
To configure the "Call Baring" service, enter the "Server Management" menu and access
Global Rules.

Figure 128 Configuring the Call Barring service


Name name of the rule;
Type the rule type: to allow ("Allow") or restrict ("Restrict") the calls;
DNIS the DNIS number (received);
ANI the ANI number (caller ID);
After filling in the fields above, press the Submit button. You can enter either the ANI or the
DNIS field.

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9.12 Configuring the CLIP / CLIR options


The CLIP and CLIR options are enabled from the configuration page for SIP users features User Management SIP menu. In the SIP Users window, from the left menu, choose the CLI option.

Figure 129 Configuring CLIP / CLIR


For a SIP user with a client ID and User Name, you can configure the following options,
regardless of whether they belong or not to a Centrex group:
CLI Proxy displays the Caller ID for calls placed among SIP users. The field contains a list with all
the aliases associated to that SIP user;
Proxy Privacy enables the CLI Proxy option
CLI User Agent displays the Caller ID for calls placed by SIP users to external locations other than
SIP, which can be PSTN, IP junctions etc. The alias selected here will be displayed at the destination
instead of the source phone number;
User Agent Privacy enables the CLI User Agent option
CLI Centrex displays the Caller ID for calls among the SIP users in the same Centrex Group. For all
the calls received from those SIP users, the destination will see the alias selected in this field.
Centrex Privacy enables the CLI Centrex option
Display Name name displayed with the User Alias
Display Name Privacy enables the option to display the Caller ID at the destination

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9.13 Configuring the Voice Mail


The voicemail files are recorded in Romanian and English for the G711u (symbol 0) and G729
(symbol 18) codecs.

Enabling the Voice Mail

To enable the Voice Mail for a SIP user, you have to create a route to the
VM_NOANSWER_BUSY service.
In the web interfaceHome / Server Management / Routes create a new route by pressing the
"New Route" button.
In the "Routing / New Route" window, configure the following parameters:

Figure 130 Configuring the Voice Mail

Name enter the route name - for example "Voicemail". The name of the route must not contain
spaces.
Prefix enter a prefix for example "777"
Action select "Service"
Service Type select VM_NOANSWER_BUSY.

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Enter the User Management SIP menu and open the SIP Users window. Enable the
voicemail service by checking the Voicemail and VoicemailState checkboxes.

Figure 131 Enabling the VoiceMail from the web interface


The prefix of the route must come before the user name in the "Voice Mail Number" field,
from the SIP User settings. For example, if the prefix is 777 (as in our case), for user 301 you will enter
in the Voice Mail Number field the value 77301.
If you activate voicemail with noanswer, in the voicemail.cfg file (in the Softswitch) you have
to configure the value for ring_timeout (see the example below). This parameter indicates the
duration in milliseconds for which the caller can ring the destination before the voicemail is activated.
In the example below, a duration of 15 seconds was used.
Use the Putty utility to create a SSH connection to the address that matches the equipment,
then enter the user name and password and after the connection is established, access the
configuration file, "voicemail.cfg".

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Example voicemail.cfg the configuration file for the voice-mail service

Lines that start with a "#" are comments and will be ignored.
# in milliseconds
ring_timeout 15000
# maximum number of messages that can be received by a subscriber
max_msg 15
# maximum duration of a received message (in seconds)
max_time 30
# range of UDP ports used to run the voice messages
begin_rtp_port 15000
end_rtp_port 16000
# if the value is 1, voicemail can be used without transcoding; this is useful if no transcoding machine
exists
# in this case, all SIP phones must be configured to use the same codec
same_codec 1

Example prepaid.cfg

define_language
1 ro
2 en
The first language in the list is the default one.

Reading the messages

For the messages to be read, a route must be created to the READ_VOICE_MAIL.service.


Messages are numbered with two digits; this means that to select the first message, the user has to
press 01 instead of 1, and so on until 9. If the user does not enter 2 digits, they will be disconnected on
timeout.

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9.14 Configuring Voice Mail 2 E-mail


The VoiceMail 2 Email service allows voice mail (audio files) to be sent to a e-mail address
specified by the user.
In the SIP User settings from the web interface, you have to check/fill in the following
options/fields (the first two fields enable the voicemail service, the next two enable sending messages
to an e-mail):

Figure 132 Enabling VoiceMail 2 Email


VoiceMail enables or disables the voicemail option. This setting is accessible only to the system
administrator
VoiceMailState enables or disables the voicemail option in various situations (Offline, Busy,
NoAnswer, Always), this setting is accessible for the user and the system administrator
VoiceMail to Email State enables or disables the voicemail2mail service, this setting is accessible
for the user and the system administrator
VoiceMail 2 eMail the e-mail address where the voicemail files will be sent, this setting is accessible
for the user and the system administrator
In order for ".wav" voicemail files to be sent, the configuration file in the multiSwitch must be
edited: /mnt/app/bin/smtpmail.cfg.
Use the Putty utility to create a SSH connection to the equipment IP (default value
192.168.1.20). Use port 2212, user name tpxadm and password u53rp455. Direct root access is not
allowed. This will be performed after connecting, using the su- command and the password
5y5t3mp455.
In the file smtpmail.cfg configure the IP address of the SMTP server that the smtpmail client
will use to send e-mails and the port through which the SMTP server listens. The two lines to be
configured are:
smtp_server_ip 86.17.6.23
smtp_server_port 25
After enabling the de voicemail2email service, voicemail messages will be sent from the SIP
subscriber to the selected e-mail. The message is automatically converted to wav format by the central
application, so that any common player can play it (for ex. winamp, windows media player etc.).
After the message is sent to the e-mail address, it is deleted from the equipment HDD. In the
sent e-mail, the date and time when the message was received, as well as the identity of the message
sender, are displayed.

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9.15 Configuring Missed Calls 2 E-mail


The Missed Calls 2 Email service allows messages to be sent to an e-mail address specified
by the user.
In the SIP User settings from the web interface, you have to check/fill in the following
options/fields (the first two fields enable the voicemail service, the next two enable sending messages
to an e-mail):

Figure 133 Enabling Missed Calls to Email


Missed Calls to eMail enables or disables the MissedCalls2Email service, this setting is accessible
only for the system administrator
Missed Calls to eMail enables or disables the MissedCalls2Email service, this setting is accessible
for the user and the system administrator
Missed Calls to eMail eMail = the e-mail address that will receive the e-mails with the information
about the missed call, this setting is accessible for the user and the system administrator
In order for ".wav" voicemail files to be sent, the configuration file in the multiSwitch must be
edited: /mnt/app/bin/smtpmail.cfg.
In the file smtpmail.cfg configure the IP address of the SMTP server that the smtpmail client
will use to send e-mails and the port through which the SMTP server listens. The two lines to be
configured are:
smtp_server_ip 86.17.6.23
smtp_server_port 25
After enabling the MissedCalls2Email service, e-mails will be sent with information about
missed calls (1 e-mail per missed call).
In the sent e-mail, the date and time when the message was received, as well as the identity
of the message sender, are displayed.

9.16 Flagging the packets as ToS or DSCP


In order to flag the packets RTP with a ToS or DSCP different from the signaling packets,
access the /mnt/app/bin/rtpproxy options file.
Use the Putty utility to create a SSH connection to the equipment IP (default value
192.168.1.20). Use port 2212, user name tpxadm and password u53rp455. Direct root access is not
allowed. This will be performed after connecting, using the su- command and the password
5y5t3mp455.
Use the following command:
rtpproxy t <value tos>

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9.17 Number Portability


For the Number Portability service, use the Translate DNIS application. For each incoming
call, the equipment will search for the DNIS in the database and will fetch a new prefix. Depending on
this new prefix, the call is routed as per the routing table.
The ported numbers are entered in this table and the new prefix associated to each operator
uniquely identifies the operator (for ex. 072 Vodafone, 074 Orange).
The prefix is translated in the Sign3 (bit val 0x00000008) field from the class (direction)
settings Server Management Client Classes menu.

Figure 134 Translate DNIS


In /mnt/app/cfg/exec.cfg add/modify the following line: "pgsql_ani_pool 1".
In /mnt/app/cfg/pgsql_sip_pool.cfg set up the database to which the connection is
performed (see chapter 8.18 Enabling PostGreSQL Billing).
The numbers and their associated prefixes are automatically entered in the translateprefix
and translateprefix_regex tables from the web interface the Server Management Translate ANI
and Translate DNIS menus.
If regular expressions are used, the web interface automatically enters them in the
translateprefix_regex table (not indexed), otherwise, the numbers are entered in the translateprefix
table (indexed).
The equipment will automatically search both tables at each call.

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9.18 Enabling the services directly from the SIP phone


The service activation codes are used to activate certain SIP services directly from the IP
phone, without using the web interface.
The equipment administrator must first create routes for each service. The (SIP) user can
activate these service by using the following combinations:
Call Forward the route created to SERV_SIP_CAL_FORWARD_ON_OFF 22
route_prefix + 1 digit(1=ON/0FF) + 2 digits(mask) + forward_number(only when ON)
Mask values:
- OFFLINE
1
- BUSY
2
- NO_ANSWER
4
- ALWAYS
8
Call Waiting the route created to SERV_SIP_CALL_WAITING_ON_OFF 39
route_prefix + 1 digit(1=ON/0=OFF)
Do Not Disturb the route created to SERV_SIP_DND_ON_OFF 40
route_prefix + 1 digit(1=ON/0=OFF)
CLIP_CLIR the route created to SERV_SIP_CLIP_CLIR 38
route_prefix + 1 digit(1=CLIP/0=CLIR)
Voice Mail the route created to SERV_SIP_VOICE_MAIL_ON_OFF 23
route_prefix + 1 digit(1=ON/0FF) + 2 digits(mask) + voicemail_number(only when ON)
Mask values:
- OFFLINE
1
- BUSY
2
- NO_ANSWER
4
- ALWAYS
8

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10

TECHNICAL DATASHEET
TOPEX multiSwitch
19" unit

Metal case with the height of 1U

Features

Operating system: Gentoo Linux


Interconnects the calls between different operators
Switching solution that runs on a hardware platform - a PC
Enhances and even replaces the C.O. switching functions (Central
Office)
Integrated Number Portability

Interfaces

2 PS/2 connectors for mouse and keyboard


1 DB 27 mother connector, for printer
1 DB 15 mother connector, for display
1 DB 9 father connector
1 RJ 45 Ethernet connector, for the local network
4 USB A connectors (grouped 2 by 2, on the left and right sides
of the back panel of the electrical circuit)
3 circular connectors for headphones, speaker and microphone (on
the right side of the panel that holds the electrical circuit
connectors)
Yellow LED (noted with the symbol
power supply

Status LEDs

Protocols

Supply voltage

) indicates the equipment

Orange LED (noted with the symbol


) indicates the hard disk
activity
2 mounted LEDs for the Ethernet connector, on the back panel of
the equipment:
The orange LED indicates the connection to the Ethernet
interface
The green LED indicates the Ethernet interface activity
SIPv.1, SIPv.2, H323v.2, H323v.4
Support for T.38 fax and passthrough
NAT Traversal
RTP / RTCP (ToS, DSCP flagging)
SRTP and SSL
TCP, UDP, TLS
RADIUS
230 VC.A. / 50 Hz

Environment
conditions

Temperature range for:


Operating: 50.... 500 C
Storage: 00.... 600 C
Humidity from 10 to 85 % without vapors

Weight

Maximum 15 kg

Dimensions

426.2 x 43.2 x 609.6 mm

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11

OPERATING CONDITIONS

To ensure the correct operation of the TOPEX multiSwitch equipment, you must abide to the
environment and security restrictions presented below.

11.1 Environment conditions


The TOPEX multiSwitch equipment must benefit of the best possible environment conditions
in order to operate correctly. For this purpose:

Do not use the equipment in flammable or explosive environments or in locations where toxic
gas accumulations can occur;

Do not use it in an environment with a high EMI (electromagnetic interferences) degree, near
copiers, PC displays, TV or other domestic appliances or high power equipments such as
electrical engines and radiators.

Temperature range: 5 - 50 0C;

Relative humidity: from 10 to 85 % without vapors

11.2 Security conditions


For the equipment to operate safely, you must:

Find an adequate location (there must be enough free space around the equipment so that it is
correctly ventilated);

Have a reliable internet connection;

Avoid dust and prolonged exposure to solar radiations;

Carefully handle the equipment, to avoid mechanic impact;

Do not power the equipment to a supply power higher than 230 V A.C.;

Ensure that the power plug used for the equipment is in good functioning status and is not
deteriorated.

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12

GLOSSARY

ACD Abbreviation for Average Call Duration the average call duration for a user's calls. It is
expressed in minutes.
ANI Abbreviation for Automatic Number Identification a feature of intelligent service in the
phone network, allowing subscribers to display the phone numbers of the callers. This service is
distributed most often by sending the DMTF digital tone together with the call.
ANI is often used by emergency center dispatches to save phone numbers from which emergency
calls are made, to be able to identify these callers afterwards. For example, in a call center, the ANI
displays to the operator the caller's number, in real time. One of the possibilities is redirecting the calls
to different persons in different geographic areas.
ASR Abbreviation for Answer Seizure Ratio this is a rate between the number of successfully
placed calls and the total number of call attempts. Since the busy signals or other types of call rejects
are considered as unsuccessful attempts, the calculated value for the ASR rate depends on the user
behavior.
ATM Abbreviation for Asynchronous Transfer Mode (ATM) this represents a network and
datalink protocol that codifies the data traffic in fixed cells that have small sizes (53 bytes; 48 data
bytes and 4 header bytes). This is a variant of the variable dimension packets (frames) in the packet
switching network. ATM is a oriented technology, where a connection is established between two
points before the data transfer itself is started.
CO Abbreviation for Central Office this is a component of the phone network. This is a building
where the home or office phone lines are interconnected with a much bigger switching system. In
metropolitan areas, CO are actually rather LOs (Local Offices), as they serve a local area. The "CO"
term is known since the first days of telephony, when the phone company actually had a central office
in each area.
The phone cable between the CO and the home or office is called "local loop" and in most cases is the
last phone network part that still uses analogue voice signals in copper wires. The maximum distance
between a CO and a home or an office is 5 kilometers. A CO in a metropolitan area can provide
services to a number of over 100.000 local loops.
CLIP Abbreviation for Calling Line Identification Presentation this represents an additional
service used to display the caller's number. When a call is initiated, the mobile switching center of the
subscriber transmits the caller's identity to the destination MSC. Then, the destination MSC verifies if
the phone that was called has the CLIP service enabled. If the service is enabled, the caller's identity
will be displayed at entry. CLIP functions together with CLIR (Calling Line Identification Restriction) to
offer an advanced identification version in GSM networks.
DNIS Abbreviation for Dialed Number Identification Service a phone service that displays to the
called party the number that the caller party has dialed. DNIS operates by sending the digits in DTMF
format to the destination, where a special application can read, display of provide these digits for call
center programming.
For example, a company can use a different free toll number for each of its line of products. If a call
center handles calls for all the product lines, the switch that receives the call can examine the DNIS
and then send the adequate welcome message. Another example of free toll numbers is multi-lingual
identification. A number can be setup for clients that speak Spanish, another one for clients that speak
German etc.

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E1 The European format for digital transmissions, equivalent to the T1 format is USA. E1 has a data
rate of 2.048 Mbps, containing 32 channels, each of them having 64 Kbps, and includes two channels
reserved for signaling and control. The US standard, T1, has 24 channels, each having 2.048 Kbps,
and the data rate is 1.544 Mbps. E1 and T1 lines can be interconnected for international usage.
FXS Abbreviation for Foreign Exchange Station a phone interface that sends dial tone and
generates call voltage. A standard phone is entered in such an interface to receive telephony services.
An example of a FXS interface is the switch system. FXS can be any device that from the point of view
of a telephone represents a switching system. This system must be capable of offering power voltage
to the connected phone, call signal and dial tone, as well as capable of knowing when the phone is on
or off the hook, when it sends or receives voice messages.
H.323 an ITU-T recommendation that defines the protocols which sustain the audio-video
communication sessions via any network with packet switching. Currently, this standard is
implemented in several real-time internet applications, such as NetMeeting or Ekiga (the latter uses
the OpenH323 variant).
H.323 is a component of the H.32x protocol series, which provides communications over ISDN, PSTN
or SS7. H.323 is frequently used for Voice over IP (VoIP, internet telephony or IP telephony) and IPbased videoconferencing.
H.323 was initially created to provide a transport mechanism for multimedia applications in local
networks, but it rapidly evolved to the much bigger requirements of VoIP networks. One of the critical
features of the H.323 standard is the availability of a standards set which define not only a basic call
model, but also additional services. H.323 was the first VoIP standard that adopted the RTP IETF
standard for audio and video transport over IP networks.
IP Abbreviation for Internet Protocol. This is the TCP/IP protocol that determines the way data is
sent from one computer to another over internet. The messages are divided into data packets, routed
from the source network to the destination network and reassembled in the correct order to obtain the
original message. As each message is divided in a number of packets, each packet can be sent
through a different route over internet. This means it is possible for the data packets to be received in
a different order than the sending order. The IP protocol only delivers them. It is the TCP protocol
(Transmission Control Protocol) that orders the packets in the correct order.
The IP protocol is a connection-less protocol. IP corresponds to the level 3 network in the OSI
communication model (Open Systems Interconnection).
IP Address Abbreviation for Internet Protocol Address. The numerical address of a network
device or resource, in the format specified by the IP protocol (Internet Protocol). In the current format,
IPv4, the IP address is a sequence of 32 bits (4 bytes) divided in 4 groups of decimal numbers, each
group separated by a full stop. Each number can have a value between 0 and 255. An example would
be "127.0.0.1" or "213.154.120.170". The four numbers in the IP address are used for different
purposes, to identify a particular network and subsystem (host) of that network.
ISDN Abbreviation for Integrated Services Digital Network. This represents the worldwide digital
network that offers high speed connection among terminal devices (phone, fax, PC) for a wide range
of telecommunication services, using the existing telephony infrastructure.
ISUP Abbreviation for ISDN User Part this represents a component of the Signaling System #7,
used to establish phone calls in the public phone network with circuit switching. When a phone call is
established between one subscriber and another, several phone switching systems will be implied,
even from different countries. In order for the call to be correctly established, the switches signal, in
the network, the call information such as the caller/destination phone number, via ISUP messages.
INAP Abbreviation for Intelligent Network Application Part signaling method used in the
intelligent network architecture. This represents a component of the SS7 protocol suite, usually above
the TCAP (Transaction Capabilities Application Part) level. INAP is a signaling protocol between a
SSP (Service Switching Point), the media resources of the network (intelligent devices) and a
centralized database, SCP (Service Control Point).
LCR Abbreviation for Least Cost Routing This is the call delivery method that uses the lowest
cost network. Most of the times, there are several networks that can deliver a call, at different costs.
The LCR Softswitch contains a database with networks and the corresponding destinations and costs,
and will connect the call using the lowest cost network. This process determines a search for the
optimal traffic routes and a reduction in the call cost.

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MGCP Abbreviation for Media Gateway Control Protocol also known as H.248 and Megaco, this
represents a standard protocol for handling signaling and session management, required for a
multimedia conference. The protocol defines a communication method between the media gateway,
that converts the data from a format specific to the circuit switching network and the gateway media
controller. MGCP can be used to configure, maintain and terminate calls between several
terminations. Megaco and H.248 refers to a more advanced version of the MGCP protocol.
MPLS Abbreviation for MultiProtocol Label Switching - this represents a data transport
mechanism that simulates some of the properties of the circuit switching network in a packet switching
network. MPLS operates at a OSI model level which is generally located between the 2nd level
(datalink) and the 3rd level (network) and is often used as level 2.5.
MPLS was designed to offer a data transport service for the clients in circuit switching networks and
for those in packet switching networks. It can be used to transport several traffic types, such as IP,
ATM, SONET packets and Ethernet frames.
NAT Abbreviation for Network Address Translation also known as network masquerading,
native address translation or IP-masquerading) implies rewriting of the source address and/or
destination while the IP packets pass over a router or firewall. Most systems use NAT to activate
several subsystems in a private network, so that they can access the internet using a single public IP
address. Depending on the specifications, the routers are not required to have this behavior, but many
network administrators consider NAT to be a convenient technique, which makes it widely used.
However, NAT can bring complications in the communication between systems.
NMS Abbreviation for Network Management Systems this represents a hardware and software
combination used for network monitoring and administration. The individual elements of the network
(NE) are administered via administration systems.
PIN Abbreviation for Personal Identification Number - A secret code known only by the user and
the system. The code is used to identify the user in the system. The user must enter this code to gain
access into the system. When the system receives the PIN code, it checks in the database whether
the PIN code exists and whether it matches the user and depending on this, grants or rejects the user
the right to authenticate in the network.
PLMN Abbreviation for Public Land Mobile Network this is a network designed and managed by
the ROA (Recognized Operating Agency) for the purpose of providing radio and terrestrial
telecommunication services to the users. The access to PLMN services is made through an interface
that implies radio communications between mobile phones or other wireless equipment and radio
transmitters or basic radio stations. PLMN network are interconnected with themselves and with PSTN
for phone communications or with service providers for internet data and access.
PSTN Abbreviation for Public Switched Telephone Network (s). This represents the classical
phone network based on copper wire that can deliver voice and data. Analogue telephony (mostly) or
digital telephony are used to provide voice services to users that access this network over an
analogue network.
RADIUS Abbreviation for Remote Authentication Dial In User Service represents an AAA
protocol (Authentication, Authorization and Accounting) for applications such as network
authentication or IP mobility. The protocol is designed to function in local and roaming situations.
Moreover, RADIUS is widely used by VoIP providers. It is used to connect a SIP terminal (such as a
broadband phone) to a SIP Registrar using the authentication, and then to the RADIUS server using
RADIUS. Sometimes, RADIUS can be used to collect CDR files (Call Detail Records) that are used
afterwards for example for charging the subscribers.
RSVP Abbreviation for Resource ReSerVation Protocol this is a network level protocol designed
to reserve resources in a network. RSVP does not transport application data, as it is rather a control
protocol over internet, similar to ICMP, IGMP or routing protocols.
RSVP can be used either by subsystems, or by routers, to request or deliver specific QoS levels
(Quality of Service) for data streams or data volume. RSVP defines the mode in which applications
choose their reservations and abandon their resources after they are not useful anymore. RSVP itself
is not a routing protocol, but it was created to work with current and future routing protocols.
RTP Abbreviation for Real Time Transport Protocol defines a standard format for the packets for
audio and video transmission over internet. It was developed by the Audio-Video Transport Working
Group in IETF and first published in 1996 as RFC 1989, obsoleted by the RFC 3550 standard in 2003.

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The RTP protocol does not communicate over a standard TCP or UDP port. The only specification
used is the one about UPD communications being made over an even port, and the higher level
uneven port is used for RTCP communications (RTP Control Protocol).
SCP Abbreviation for Secure Copy SCP is a safe method for transferring files between a local and
a remote system, using the SSH protocol (Secure Shell).
SCTP Abbreviation for Stream Control Transmission Protocol represents a transport level
protocol defined by the IETF Signaling Transport (SIGTRAN) group. As a transport protocol, SCTP
operates like TCP or UDP. Indeed, it provides some services that are similar with TCP - ensuring
reliability or sequential transport for messages, with bottleneck control.
SIP Abbreviation for Session Initiation Protocol a protocol proposed as a standard for initiating,
modifying and ending a session that involves multimedia elements such as video, voice, instant
messages, online games and virtual reality. This, along with H.323, is one of the signaling protocols
used for VoIP.
SIP clients frequently use the TCP or UDP port 5060 to connect to IP servers or to other SIP
terminals. First, the SIP protocol is used to initiate and terminate voice or video calls. However, SIP
can be used in any application where an initiation session is required. All voice or video
communication is made separately via transport protocols such as RTP.
SMS Abbreviation for Short Message Service involves sending short text messages to/from
mobile phones. The messages must contain only text, without pictures or graphic, and no more than
160 alphanumerical characters. Mobile phone network operators use a separate data channel to send
SMS messages. You can send SMS messages to another mobile subscriber, the operator can send
you various settings via SMS or trading companies can send customized messages for each client.
SNMP - Abbreviation for Simple Network Management Protocol part of the protocol suite for
internet. SNMP is used in network administration systems, to monitor the peripherals in conditions that
can warrant an administrative attention. It consists of a set of network administration standards,
including application level protocols, a database schema and a data objects set.
SS7 Abbreviation for Signaling System 7 a global telecommunications standard established by
ITU-T. The standard defines the procedures and the protocol through which the elements from the
PSTN (public switched telephone network) exchange information through a digital signaling network in
order to establish wired or wireless calls, routing and control.
The definition that ITU assigned to the SS7 protocol allowed for the development of national standards
such as ANSI (American National Standards Institute) or Bell Communications Research (Telcordia
Technologies), used in North America and ETSI (European Telecommunications Standard Institute),
used in Europe. SS7 represents an architecture for signals to sustain the call establishing, taxing,
routing and data exchange functions of the PSTN public network. It identifies functions performed by a
signaling network and a protocol to activate its performances.
TDM Abbreviation for Time Division Multiplexing this is a type of digital multiplexing where two
or more temporal channels are derived from a given frequency specter by intercalating the pulses that
represent bits on different channels.
In some TDM systems, successive pulses represent bytes on successive channels. In other systems,
different channels are switched using the channels from a group of successive temporal pulses (also
known as "time slots"), voice channels in the E1 / T1 systems. The important difference between the
time division multiplexing and packet switching is the fact that time slots are allocated in advance to
different channels.
TUP Abbreviation for Telephone User Part. Performs the transport of the user's phone message on
the signaling datalink, via signaling units. The signaling information for each message is the signal
data field of the corresponding unit and contains an integer number of bytes. Initially, it contains the
label and one or more signals and/or indications.
VoIP Abbreviation for Voice over IP. Technology that describes the hardware and software category
that enables human voice calls over internet. Voice signals are converted to data packets which are
then transmitted over public phone networks, avoiding the PSTN network costs. VoIP application can
be used with a microphone and common PC speakers, but they can also be used with IP phones or
VoIP speakers, which provide an experience that is no different from classical telephony. Lately, the

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quality and reliability of the VoIP technology evolved so much that some users completely waived the
standard telephony contracts, in favor of the VoIP telephony.
VPN Abbreviation for Virtual Private Network represents a private communication network
commonly used by a company or by several companies or organizations that desire to communicate in
a confidential manner over a public network. VPN traffic can be transported over a public network
infrastructure (such as internet) in the higher level of standard protocols, or over a private network with
a SLA level (Service Level Agreement) defined between the VPN client and the VPN service provider.

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INDEX FOR THE IMAGES


Figure 8 Authentication in the administration page for TOPEX multiSwitch ...................................... 59
Figure 9 Wrong authentication ........................................................................................................... 60
Figure 10 Menu Bar............................................................................................................................ 60
Figure 11 "Server Settings" menu options ......................................................................................... 61
Figure 12 The "Server Settings" menu "About" option .................................................................... 61
Figure 13 The "Server Settings" menu "SIP Access List" option .................................................... 62
Figure 14 Adding a new access list.................................................................................................... 62
Figure 15 The "Server Settings" menu "Settings" option ................................................................ 63
Figure 16 Adding new parameters ..................................................................................................... 63
Figure 17 Displaying the added parameters ..................................................................................... 63
Figure 18 Adding a new equipment in Softswitch Parameters........................................................... 64
Figure 19 Displaying the global parameters for a Softswitch ............................................................. 64
Figure 20 Translating a global parameter into a local one ................................................................. 64
Figure 21 The "Equipments" option.................................................................................................... 65
Figure 22 The "Equipments" option.................................................................................................... 65
Figure 23 List of the equipment that can be interconnected .............................................................. 65
Figure 24 The "Server Management" menu....................................................................................... 66
Figure 25 Description of the client classes......................................................................................... 66
Figure 26 Editing a client class........................................................................................................... 67
Figure 27 Options for the "Sign1" field ............................................................................................... 68
Figure 28 Options for the "Sign 2" field .............................................................................................. 69
Figure 29 Options for the "Sign 3" field .............................................................................................. 70
Figure 30 Creating a client class ........................................................................................................ 71
Figure 31 Editing a client class - Rules region ................................................................................... 73
Figure 32 Editing a client class - Alerts region ................................................................................... 74
Figure 33 The Translate "ANI" option................................................................................................. 74
Figure 34 Translate ANI New Rule.................................................................................................. 74
Figure 35 Displaying the records in the Translate ANI/DNIS table .................................................... 75
Figure 36 Changing the prefixes of a group with the "Group Update" option .................................... 75
Figure 37 Filtering prefixes with the "Filter" option ............................................................................. 76
Figure 38 The "Routes" option ........................................................................................................... 77
Figure 39 Editing a route .................................................................................................................... 77
Figure 40 Creating a new route.......................................................................................................... 79
Figure 41 New Route Sign 1 ........................................................................................................... 79
Figure 42 New Route Sign 2 ........................................................................................................... 80
Figure 43 New Route Sign 4 ........................................................................................................... 81
Figure 44 Entering a new rule in the "Access In" option .................................................................... 82
Figure 45 Access list for incoming calls ............................................................................................. 83
Figure 46 The "Access Out" option .................................................................................................... 84
Figure 47 Displaying the records in the "Access Out" menu.............................................................. 84
Figure 48 The "Global Rules" option .................................................................................................. 85
Figure 49 Error message when filling in ANI and DNIS ..................................................................... 85
Figure 50 The "Services" menu.......................................................................................................... 86
Figure 51 The "SIP User Alias" window ............................................................................................. 86
Figure 52 Displaying the aliases in the "User Management SIP" menu ......................................... 87
Figure 53 Adding a Centrex group ..................................................................................................... 88
Figure 54 Viewing Centrex groups ..................................................................................................... 88
Figure 55 Creating a hunting group.................................................................................................... 89
Figure 56 Call Hunting for SIP users.................................................................................................. 89
Figure 57 Creating a pickup group ..................................................................................................... 90
Figure 58 Call Pickup for SIP users ................................................................................................... 90
Figure 59 Creating a forking group..................................................................................................... 91
Figure 60 Call Forking for SIP users .................................................................................................. 91
Figure 61 The "User Management" menu.......................................................................................... 92
Figure 62 System Users ..................................................................................................................... 92
Figure 63 Adding a system administrator........................................................................................... 93

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Figure 64 Displaying the system users .............................................................................................. 94
Figure 65 Editing the properties for system users.............................................................................. 95
Figure 66 The System menu - "Limits" submenu ............................................................................... 96
Figure 67 SIP Users ........................................................................................................................... 97
Figure 68 Adding a SIP user .............................................................................................................. 97
Figure 69 Configuring CLIP / CLIR................................................................................................... 100
Figure 70 Rules for SIP users .......................................................................................................... 101
Figure 71 Aliases for SIP users........................................................................................................ 101
Figure 72 Web access for SIP users................................................................................................ 102
Figure 73 The "ANI Users" menu ..................................................................................................... 103
Figure 74 Editing the properties of an ANI user ............................................................................... 103
Figure 75 Adding ANI users ............................................................................................................. 104
Figure 76 The "Subscribers" menu .................................................................................................. 105
Figure 77 Editing the parameters of the FXS subscriber ................................................................. 105
Figure 78 Adding a new FXS user ................................................................................................... 106
Figure 79 The "Prepaid" menu ......................................................................................................... 107
Figure 80 Changing the properties of prepaid clients ...................................................................... 107
Figure 81 Adding a new prepaid client in the database ................................................................... 108
Figure 82 The "SIP Locations" menu ............................................................................................... 109
Figure 83 Options for the "Billing" menu .......................................................................................... 110
Figure 84 The "Billing Profiles" option ............................................................................................... 110
Figure 85 Adding details about the calls placed............................................................................... 111
Figure 86 Displaying the billing details ............................................................................................. 111
Figure 87 Entering a new profile in the "Subscriptions" submenu .................................................... 112
Figure 88 Entering the period of validity........................................................................................... 112
Figure 89 The window for entering the subscription options............................................................ 113
Figure 90 Entering a new tax unit..................................................................................................... 113
Figure 91 Tax units Currency Manager......................................................................................... 114
Figure 92 Entering a new billing group............................................................................................. 114
Figure 93 Displaying the billing groups in the Billing menu.............................................................. 115
Figure 94 The "Account Properties" menu ....................................................................................... 116
Figure 95 The "Call list" option ......................................................................................................... 116
Figure 96 The "Billing" menu "Filters" button ................................................................................ 117
Figure 97 Duration error ................................................................................................................... 118
Figure 98 Displayed calls ................................................................................................................. 120
Figure 99 Save text file..................................................................................................................... 121
Figure 100 Format of the CDR file ................................................................................................... 121
Figure 101 Adding a new user ......................................................................................................... 122
Figure 102 Address Book User Details ......................................................................................... 123
Figure 103 User list .......................................................................................................................... 123
Figure 104 The "Testing" menu........................................................................................................ 124
Figure 105 The "User Generator" option.......................................................................................... 124
Figure 106 The "Rule Generator" option .......................................................................................... 125
Figure 107 Profitability report [Brief]................................................................................................. 126
Figure 108 The "Profitability [In]" option ........................................................................................... 127
Figure 109 The "Profitability [Out]" option ........................................................................................ 127
Figure 110 - The "Reliability [In]" option .............................................................................................. 128
Figure 111 - The "Reliability [In]" option .............................................................................................. 129
Figure 112 The "Templates" menu................................................................................................... 130
Figure 113 Creating a "Class Template" .......................................................................................... 131
Figure 114 Adding "SIP User Template" .......................................................................................... 133
Figure 115 Adding "FXS User Template"......................................................................................... 135
Figure 116 Adding a "ANI User Template"....................................................................................... 136
Figure 117 Adding a "Prepaid User Template" ................................................................................ 137
Figure 117 Client Classes ................................................................................................................ 138
Figure 118 Creating a client class .................................................................................................... 139
Figure 119 SIP Users list.................................................................................................................. 140
Figure 120 Adding a SIP user .......................................................................................................... 141
Figure 121 Enabling Call Waiting ..................................................................................................... 143
Figure 122 Enabling Do Not Disturb................................................................................................. 144

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Figure 123 Configuring the Call Forward service ............................................................................. 145
Figure 124 "Check Call Back" option ............................................................................................... 146
Figure 125 Configuring the Call Back service in the ANI table ........................................................ 146
Figure 127 Creating a hunting group................................................................................................ 147
Figure 127 Call Hunting for SIP users.............................................................................................. 147
Figure 128 Creating a pickup group ................................................................................................. 148
Figure 129 Call Pickup for SIP users ............................................................................................... 148
Figure 130 Creating a forking group................................................................................................. 148
Figure 131 Call Forking for SIP users .............................................................................................. 149
Figure 132 Enabling the Suspend Subscriber service ...................................................................... 149
Figure 133 "Access In" window ........................................................................................................ 150
Figure 134 Configuring the Call Barring service .............................................................................. 151
Figure 135 Configuring CLIP / CLIR................................................................................................. 152
Figure 136 Configuring the Voice Mail ............................................................................................. 153
Figure 137 Enabling the VoiceMail from the web interface.............................................................. 154
Figure 138 Enabling VoiceMail 2 Email............................................................................................ 156
Figure 139 Enabling Missed Calls to Email...................................................................................... 157
Figure 140 Translate DNIS............................................................................................................... 158

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The manufacturer reserves the right to modify the product and the user guide in
order to enhance them, without prior notice. The manufacturer warrants that the product
operates in good conditions, provided it was correctly installed and the storage and usage
guidelines were followed. The warranty exclusively implies the repair or replacement of
the defective unit.
The warranty does not include any direct or indirect loss of profit. The
manufacturer is not responsible for any direct, indirect, special, accidental or
consequential damages that result from using the TOPEX multiSwitch equipment.
No part of this guide can be copied, owned in a distribution system or transmitted
in any form, or by any method, be it electronic, mechanic, recording or any other, without
the written permission from TOPEX S.A.
This certifies that the TOPEX multiSwitch is designed according to legal provisions
regarding the responsibility for the quality of delivered products, responsibility fulfilled by
the quality parameters specified in the "User Manual" and that it is adequate for the
purpose it was created for. It also warrants that the equipment will operate according to
the company's documentation.
We welcome any comments, suggestions and proposals related to our product and
we are looking forward to receiving your messages:

TOPEX S.A.
10 Feleacu street, district 1,
014186 Bucharest, Romania
Phone: +40-21 408.39.00
Fax: +40-21 408.39.09
E-mail: topex@topex.ro
Web:http://www.topex.ro

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