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User manual
2008
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)
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Table of Contents
1
3
4
5
6
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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.
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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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.
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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.
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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.
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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Pcs
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.
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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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 9 male connector;
3 connectors 2 (blue and yellow) for connecting the head phones and speakers and the pink
one for connecting a microphone.
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
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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.
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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.
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Incoming class
Optional DB checks:
- translate ANI
- translate DNIS
- portability
- ANI users
Routing table
Outgoing class
Access Out
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trunks (or junctions) channels where through the calls from other networks are received or
sent;
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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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.
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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.
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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 Client Class if the destination is an id from the Client Class
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:
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:
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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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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.
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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.
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// 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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radius_dictionary 0
tx_access_request 1
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db_route 1
db_type 2
Must be always 2
maxsysincallrate 10000
db_card 1
maxsysoutcallrate 10000
consysincallrate 10000
Default is 10000
consysoutcallrate 10000
Default is 10000
set_asci_sip 1
set_asci_h323 1
billing_fields_number 255
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Significance
No debug
Error only
Error + Warning
The parameters used for SIP debug are detailed in the table below:
Debug parameter name
agent_debug
register_server_debug
register_client_debug
database_config_debug
notify_debug
proxy_debug
options_debug
queue_debug 2
console_debug
file_debug
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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
tcp
tls
sip_udp_port
sip_tcp_port
sip_tls_port
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
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Significance
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update_location
proxy
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
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User name
password=301
Password
first_proxy=mysippbx.ro
second_proxy=89.38.11.22
expires=120
nat_refresh=50
Field 2
Public class
Field 3
Field 4
Private class
Field 5
Field 6
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2000
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
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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Significance
radius_billing 1 radius_billing.cfg
radius_billing_alt 1 radius_billing_alt.cfg
radius_dictionary 0
tx_access_request 1
Send
or
not
access_request
for
authentication. The possible values are 0/1
tx_accounting_start 1
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radius_sip_user_auth ip
class_name = the
(direction) of the call
source
class
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
radius_server 192.168.1.11
dictionary 0
auth_port 1812
acct_port 1813
secret 99topexSECRETqutex11
timeout 4
retries 3
NAS_IP 192.168.1.50
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33. ptime
3. ANI in
35. gw_name
4. DNIS in
5. date
21. destination IP
6. time
7. call duration
8. billing units
40. in_billing_profileid
9. SIM used
41. out_billing_profileid
42. in_classid
27. jitter
43. out_classid
28. client id in
44. billing_prefix
13. CIMI
45. price_in
30. protocol_in
31. protocol_out
32. codec
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host=127.0.0.1
user=softswitch
The line above contains the connection string for connection to postgresql database.
the
configuration
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files
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Prepaid
Voice Attendant
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.
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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
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
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.
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
MaxTry(integer maxtry) increments the nrtry variable and if this is >= maxtry starts the
event OnMaxTry()
TxAccessRequest(integer stage, integer tip, integer pin_len, integer timeout, char state)
Send Access Request on RADIUS to Mind
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Digit(digit) run when is received the specifie digit; the value digit=255 implies the event run
at any received digit
DBNotFound() executed when is received not found message from data base
EndCall() executed when the current call is closed with # (special character and allows
ending a current call connected or not)
RadiusAccept()
RadiusReject()
RadiusError()
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[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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Description
help
access in
access out
all queue
billing profiles
classes
debug on
debug off
dialog states
fdwatch connections
fw subscriptions
global rules
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interface ip
license
port subscriptions
prepaid users
quit
register users
resellers
sip pp
sip publish
sip offline
sip online
sip users
sip vm notify
subscriber [port]
subscribers
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ts sock busy
ts sock free
ts sock wait
update config
view allports
view classes
view ports
view routes
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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.
8.2 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:
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Settings for adding new TOPEX multiSwitch equipments in the web interface;
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.
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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.
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Receive Identity when checked, 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)
Cat Call all calls made through the equipment will be closed
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.
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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.
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Max Calls maximum number of calls allowed for incoming and outgoing;
Max Time maximum interval allowed for connection; if this threshold is reached, the connection
is closed;
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;
Type a list that has two options: "Allow" and "Restrict", for establishing the phone numbers
permitted or restricted to that client class;
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 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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After entering the desired options, press "Submit" to add the record to the list. After this, the
list will be displayed as follows:
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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:
-
"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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Billing Group name of the Billing Group to which the prepaid user is associated;
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;
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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Profiles the Billing Profiles associated to the users of the TOPEX multiSwitch equipment
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).
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After entering the data, the list will look like this:
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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.
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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.
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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.
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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Max Calls maximum number of calls allowed for incoming and outgoing;
Max Time maximum interval allowed for connection; if this threshold is reached, the connection
is closed;
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;
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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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"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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Billing Group name of the Billing Group to which the prepaid user is associated;
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;
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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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
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.
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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Max Calls maximum number of calls allowed for incoming and outgoing;
Max Time maximum interval allowed for connection; if this threshold is reached, the connection
is closed;
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;
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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:
-
"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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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:
-
"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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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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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:
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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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.
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10
TECHNICAL DATASHEET
TOPEX multiSwitch
19" unit
Features
Interfaces
Status LEDs
Protocols
Supply voltage
Environment
conditions
Weight
Maximum 15 kg
Dimensions
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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.
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.
Find an adequate location (there must be enough free space around the equipment so that it is
correctly ventilated);
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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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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