Matrix SETU Combined Brochure for PDF

SETU ATA
VoIP Adaptors with FXO, FXS and Multiple SIP Accounts
Internet Telephony offers intrinsic benefits of cost and flexibility. At the
same time legacy telephony infrastructure and habits cannot be
replaced overnight. People desire the best of both worlds - lower cost
of VoIP and convenience of using existing telephony products and
methods.
Matrix SETU ATA Range of Products is designed to meet this
requirement of converting VoIP network to traditional telephony
interfaces and vice-versa. It handles all the complexities of VoIP
technology internally and provides simple telephone interfaces to
make and receive calls.
Let Matrix SETU ATA be your bridge to the new world of IP Telephony!
VoIP ATA With Unique Integration of Technology
Features and Reliability
Matrix SETU ATA is a SIP based Analog Terminal Adaptor (ATA), it
interfaces legacy telephone devices with IP-based networks. It is
specially designed for SOHO users to offer them the advantages of lowtariff Internet Telephony for long distance calls, international calls and
peer-to-peer calls. It can be used with any existing PBX providing users
access to VoIP trunks. It can also be used in a stand-alone mode.
Matrix SETU ATA converts the voice traffic into data packets for
transmission over the Internet. When a telephone number is dialed by a
user, Matrix SETU ATA converts it into an IP call using the SIP protocol
and initiates a call to the dialed number in any part of the globe. Using an
appropriate VoIP service provider, long distance or inter-office call
charges can be reduced significantly or eliminated through peer-to-peer
calling on the IP network.
SETU ATA1S
1-FXS Port, 1-WAN Port,
1-LAN Port and 2-SIP Accounts
Making an outgoing call is as easy as from a normal telephone. Call
progress tones like Dial Tone, Ring Back Tone and Busy Tone are fed to
the caller as per the called number status. The FXS ports can make
outgoing calls on a common or different SIP accounts. In addition,
number based SIP account selection is provided to select the most
economical SIP account for a given outgoing number.
An incoming call from a SIP account can be routed to any one or both
FXS ports. All different CLIP protocols are supported so that the user
can identify the caller before answering the call.
Once a call is established, features like Call Hold, Call Toggle, Call
Transfer, Call Wait and Conference are supported to manage two calls
from the same FXS port. Call forward in different conditions and Do Not
Disturb is also provided.
SETU ATA2S
2-FXS Ports, 1-WAN Port and
2-SIP Accounts
The FXO port allows users to originate a call and terminate it at VoIP
network. The routing logic used between FXO and VoIP helps the user to
dial a destination PSTN number directly over VoIP network. Using FXOVoIP routing logic two or more PBX can be networked to offer seamless
connectivity between the PBXs
Matrix SETU ATA provides two Ethernet ports - one for WAN and the
other for LAN. The user can connect his PC on the LAN port and browse
the internet or check his emails while talking on VoIP calls.
Matrix SETU ATA can also be used with any PBX without changing its
existing infrastructure. PBX users can make voice calls on IP to avail of
the low-tariff of VoIP calls. The users can continue to make and receive
calls without worrying on which network their calls are routed. Matrix
SETU ATA is easy to install and operate. It can be configured using its
built-in web pages served by the internal HTTP server.
SETU ATA211
1-FXO Port, 1-FXS Port, 1-WAN Port,
1-LAN Port and 3-SIP Accounts
PRODUCT APPLICATIONS
RESIDENITAL APPLICATION
Proxy
FXS 2
WAN
IP
FXS 1
ATA2S
BUSINESS APPLICATION
SIP Server/IP PBX
FXS 2
FXS 2
WAN
WAN
FXS 1
FXS 1
ATA2S
ATA2S
VoIP
VoIP
IP
LAN
FXS
WAN
WAN
LAN
FXO
FXO
FXS
PSTN
ATA211
ATA211
PEER-TO-PEER CALLING APPLICATION
xDSL/Router
WAN
FXS2
FXS1
ATA2S
IP
Network
xDSL/Router
WAN
LAN
xDSL/Router
WAN
FXS2
FXS1
FXO FXS
FXS
ATA211
PBX
PBX
ATA211
KEY FEATURES
Auto Configuration
SETU ATA can be configured automatically from a central location. The
configuration details like Registrar Server Address, Authentication User ID, and
User Password are stored in the central server. When user connects SETU ATA to
the network, it automatically downloads its configuration using TFTP. This plug-nplay feature requires the user to enter only the server address provided by the
service provider.
Peer-to-Peer Calling
SETU ATA can make and receive calls from other VoIP users without any Registrar or
Proxy server. Numbers and IP addresses can be assigned to the other VoIP users to
provide direct access across the network. For Peer-to-Peer calling, SETU ATA provides
two options - (i) Peer-to-Peer Number Dialing (ii) IP Address Dialing. Organizations
having multiple locations like branch offices and factories can use this feature to provide
direct dialing between these end-points.
Auto PSNT Fall back
PSTN can be interfaced to the SETU ATA using FXO Port. This port is used to dial
out numbers to the PSTN Network. When Routing the calls from PSTN number to
SIP trunk, it may happen that the Ethernet Link may go down or the SIP Account
used is not registered. So the call will not be routed through SIP and you will get
error tone. To avoid this you can use this feature Auto PSTN Fall Back through
which the call will be automatically get routed through the alternate FXO Port.
PIN Authentication
Incoming calls on FXS or FXO ports of the ATA can be restricted to a specific caller.
The caller has to first prove his authentication before calling to ATA. This feature is
used to avoid the possibility of malicious calls and to avoid misuse of its services.
Automatic Number Translation
SETU ATA supports multiple port types, FXO, FXS and SIP. Whenever a number is
dialed from any of these ports, gateway routes the call to the desired destination
port as per the routing mechanism defined for that port. In certain cases, the dialed
number string is not understood by the network through which the call is to be
routed, so by using Automatic Number Translation the dialed number string is
translated into a number that is understood by the network or ITSP to reach the
desired destination port.
Call Progress Tones and Rings
Matrix SETU ATA supports programmable tones and rings to match those of the
country where it is installed.
CLIP
SETU ATA allows users to program the FXS ports for any of the three CLIP
protocols - DTMF, FSK ITU-T V.23 and FSK Bellcore 202A.
Dial Number Table
Matrix SETU ATA provides a list of programmable numbers or part-numbers with
the preferred SIP account for each entry. When the user dials a number, the SETU
ATA finds the matching number using the “best-fit” logic. It then uses the SIP
account given against this matching number to make that call. This ensures lowest
cost for all the outgoing calls.
Disconnect Signaling
When a call is released from the other side of the internet, the Matrix SETU ATA can
propagate this call release on the FXS in the form of disconnect signal. The device
senses this signal and frees the FXS port.
Fax over IP (FoIP)
The user can send and receive Fax over SIP account, once a Fax machine is
connected to SETU ATA. The SETU ATA supports FoIP using T.38 UDPTL and Pass
Through.
FXO
SETU ATA FXO port should be connected to the PSTN or PBX so that the user can
make PSTN calls from the FXO port.
Incoming Call Routing
Calls arriving from any SIP account can be routed to either one or both FXS ports.
Jeeves (Web based Programming Tool)
Flexible and user friendly windows based software, Jeeves, helps in programming
the features through web browser. This web based programming feature helps
users to configure the SETU ATA from any part of the world once it is connected
with the IP network.
MAC Cloning
When replacing the existing hardware with other, you can simplify the installation
process by copying the MAC Address of your existing PC. In such case, you do not
need to delay the SETUp process by informing your service provider of newly
installed equipment.
Multi Stage Dialing
Multi-Stage dialing is useful for ATAs connected to a SIP Server used for
networking PBX of multiple sites. The user can dial the entire number string, both
the destination number and extension number of the destination PBX together. The
ATA will split the string into two stages, and dial out the destination number first and
on receiving the answering signal it dial extension number. This ensures hassle
free access to PBX extension.
PPPoE
Matrix SETU ATA supports PPPoE client and hence can be used with any xDSL modems
Router
Basic routing capabilities are provided so that LAN port packets can be transferred on
WAN port. This allows the user to browse the internet and check his emails while making
and receiving VoIP calls.
SIP Accounts
Multiple SIP accounts can be programmed and each FXS user can be assigned one of the
SIP accounts for outgoing calls. Dynamic allocation of SIP account is also possible using
Dial Plan.
Speech Volume Setting
SETU ATA allows user to set transmit and receive gain to improve the quality of speech.
Speed Dialing
Frequently used numbers can be programmed in the internal phone book with 99 entries.
The user can dial these numbers by using short codes in place of the complete, long
numbers.
Supplementary Services
SETU ATA supports supplementary service like Call Hold, Call Waiting, Call Toggle, Call
Transfer, Call Forward, Conference, Caller ID, DND and Making another Call. These are
the service provider dependant features.
Surface Mount Technology (SMT)
The Surface Mount Technology is the current semi-conductor packaging technology that
offers reduction in real estate resulting in less heat generation and low power
consumption. This is in turn improves reliability.
System Log
Syslog is one of the built in protocol used extensively for sending debug messages on IP
network. This client/server protocol uses UDP as transport protocol for debugging
process. Logging has several benefits which include troubleshooting, security and
system administration. Debug messages are sent to remote server on IP network for
finding and reducing the number of bugs or defects from a system.
FEATURE LIST
!100Rel/PRACK (RFC 3262)
! Polarity Reversal
! PPPoE
! Programmable Call Progress Tones and Rings
! SIP over TCP
! Speech Volume Setting (Transmit and Receive)
! Speed Dialing
! STUN
! Supplementary Services
! Answer Signaling
! Auto Configuration
! Auto PSTN Fallback
! Automatic Number Translation
! Called Party Number Table
! CLIP (DTMF, FSK-ITU-T V.23, Bellcore 202A)
! Comfort Noise Generation
! DHCP Client/ PPPoE
! Dialed Number Table
! Digest Authentication
! Disconnect Signaling
! Echo Cancellation
! Fax over IP-T.38 and Pass Through
! Flash Timer
! Forward Error Correction (FEC)
! Full Duplex Audio
! Incoming Call Routing
! LED Indications
! MAC Cloning
! Multiple Gateway Support
! Multistage Dialing
! Password Protection
! Peer-to-Peer Calling
! PIN Authentication
Call Forward Unconditionally
Call Forward on Busy
Call Forward on No Reply
Call Hold
Call Toggle
Call Waiting
Caller ID
Call Transfer-Blind
Call Transfer-Attended
Conference 3 Party
Do Not Disturb (DND)
Making Second Call
! Symmetric RTP
! Syslog Client
! Voice Activity Detection
! Web based GUI for Configuration
MATRIX SETU ATA FEATURES LIST
Features List
SETU ATA1S
SETU ATA2S
SETU ATA211
3-Party Conference
Answer Signaling
Auto Configuration
Automatic Number Translation
Auto PSTN Fall Back
Called Party Number Table
CLIP
Comfort Noise Generation
DHCP Client
Dial Plan
Digest Authentication
Disconnect Signaling
Echo Cancellation
Fax Over IP- T.38 and Pass Through
Flash Timer
Forward Error Correction
Full Duplex Audio
LED Indications
Multiple Gateway
Multi-Stage Dialling
Password Protection
Peer-to-Peer Calling
PIN Authentication
Phone Book
Polarity Reversal
PPPoE Client
Call Progress Tone and Rings
Speech Volume Settings
Speech Dialing
Supplementary Services
STUN
Symmetric RTP
Syslog Client
Web based GUI for Configuration
Available
Unavailable
TECHNICAL SPECIFICATIONS
VoIP
VoIP Protocols
Network Protocol
SIP
NAT
Voice CODECS
Line Echo Cancellation
Call Progress Tones
Voice
Fax
Quality of Service
Data Network
Security
FXS Port
Connection
Off Hook Impedance
Loop Limit
Loop Feed
Ringing Voltage
Pulse Dialing
DTMF Dialing and Reception
Caller ID Presentation (CLIP)
Call Maturity
Protection
FXO Port
Connection
Off Hook Line Impedance
Loop Limit
Pulse Dialling
DTMF Dialling and Reception
CLI Reception
Call Maturity
Protection
Power Supply
Input
Connector
: SIP v2, SDP, RTP, RFC 2833
: IPv4, TCP, UDP, DHCP, SNTP,
STUN, HTTP, PPPoE
: Multiple Accounts Out Bound
Proxy Support, Display Name,
User Name, Password, URL,
Proxy URL, Registrar URL,
Registrar Interval
: STUN and NAT Keep Alive
: G.711 A-Law, µ-Law, G.723,
G.729A, G.729B
: G.168 with 8/16/32ms Tail Length
: Dial Tone, Ring Back Tone, Busy
Tone, Error Tone
: Dynamic Jitter Buffer (Adaptive),
Comfort Noise Generation and
Voice Activity Detection
: T.38 and Pass Through
: Layer 3 DIFFServ and TOS
: WAN Port (RJ45), Auto MDIX
10/100 BaseT,
LAN Port (RJ45), Auto MDIX
10/100 BaseT
: Password Protected Administration
: RJ11
: 600 Ù
: 270 Ù (Max) Excluding
Telephone Set
: 39mA (Max)
: 55Vrms @25Hz, 3REN
: 10 PPS and 20PPS @ 1:2, 2:3
and 1:1
: ITUT Q.23 and Q.24
: DTMF, FSK ITU-T V.23 and FSK
Bellcore 202A
: Polarity Reversal
: Solid State (Over Voltage and
Over Current) built-in Secondary
Protection
Power Consumption
SETU ATA1S
SETU ATA2S
SETU ATA211
Mechanical
Dimensions (WxHxD)
SETU ATA1S
: 7.9x10.5x2.7cm
(3.1”x4.1”x1.1”)
: 7.9x10.5x2.7cm
(3.1”x4.1”x1.1”)
: 7.9x10.5x2.7cm
(3.1”x4.1”x1.1”)
SETU ATA2S
SETU ATA211
Unit Weight
SETU ATA1S
SETU ATA2S
SETU ATA211
: 0.45Kgs (1.10lbs) Approx.
: 0.45Kgs (1.10lbs) Approx.
: 0.45Kgs (1.10lbs) Approx.
Shipping Weight
SETU ATA1S
SETU ATA2S
SETU ATA211
: 1.00Kgs (2.20lbs) Approx.
: 1.00Kgs (2.20lbs) Approx.
: 1.00Kgs (2.20lbs) Approx.
Material
SETU ATA1S
SETU ATA2S
SETU ATA211
: ABS Plastic
: ABS Plastic
: ABS Plastic
Installation Mounting
SETU ATA1S
SETU ATA2S
SETU ATA211
: Wall and Table-Top
: Wall and Table-Top
: Wall and Table-Top
Environmental
Operating Temperature
Storage Temperature
Operating Humidity
Storage Humidity
: RJ11
: 600 Ù
: 1500 Ù
: 10 PPS and 20 PPS @ 1:2, 2:3
and 1:1
: ITU-T Q.23 and Q.24
: DTMF, FSK ITU-T V.23 and FSK
Bellcore 202A
: Polarity Reversal
: Solid state (Over Voltage and over
current) built-in secondary
Protection
: 12VDC @1.25A through
External Adaptor
(90-265VAC, 47-63Hz)
: DC Power Jack
: 5W Approx.
: 5W Approx.
: 5W Approx.
: -100C to +500C
(-140F to +1220F)
: -400C to +850C
(-400F to +1850F)
: 5-95% RH (Non-Condensing)
: 0-95% RH (Non-Condensing)
at 400C
SYSTEM CAPACITY AND RESOURCES
Hardware
FXS Ports
FXO Ports
LAN Port
WAN Port
DC Power Jack
SETU ATA1S SETU ATA2S SETU ATA211
1
1
2
1
1
1
1
1
1
1
1
1
COMPLIANCES
EMI/EMC
Conducted Emission
Radiated Emission
Harmonic Current Emission
Voltage Flicker
Electro-static Discharge
Radiated Susceptibility
Electrical Fast Transient
Surge
Conducted Immunity
Power Frequency Magnetic Field
Voltage Interruption & Dips
FCC
Conducted Emission
Radiated Emission
EC Directives
R&TTE 1999/5/EC
LVD 73/23/EEC
EMC 89/336EEC
Safety
IEC 60950-1
: CISPR 22 Class A
: CISPR 22 Class A
: IEC 61000-3-2
: IEC 61000-3-3
: IEC 61000-4-2
: IEC 61000-4-3
: IEC 61000-4-4
: IEC 61000-4-5
: IEC 61000-4-6
: IEC 61000-4-8
: IEC 61000-4-11
: FCC Part 15 Sub Part B Class A
: FCC Part 15 Sub Part B Class A
: 2001 First Edition
VoIP PRODUCTS FROM MATRIX
SETU ATA1S
SETU ATA2S
SETU ATA211
SETU VFX44L
SETU VFX88L
SETU VP236S
SETU VP236SE
SETU VP248SE
SETU VP248PE
SETU VGFX8422
SETU VGFX8440
SETU VGFX8404
SAPEX IPXP250
SAPEX IPXP500
SIP based Analog Telephone Adaptor with 1 FXS Port and 2 Ethernet Ports
SIP based Analog Telephone Adaptor with 2 FXS Ports and 2 Ethernet Ports
SIP based Analog Telephone Adaptor with 1 FXO, 1 FXS Port and 2 Ethernet Ports
SIP based VoIP Gateway with 4 FXS Ports, 1 FXO (PSTN Pass-Through) and 1 Ethernet Port
SIP based VoIP Gateway with 8 FXS Ports, 1 FXO (PSTN Pass-Through) and 1 Ethernet Port
Executive IP-Phone with 2 Lines x 24 Characters LCD Display
Executive IP-Phone with 2 Lines x 24 Characters LCD Display and PoE
Executive IP-Phone with 2 Lines x 24 Characters LCD Display and PoE
Executive IP-Phone with 6 Lines x 24 Characters LCD Display and PoE
Multi-Port VoIP Gateway with 8 VoIP Channels, 4 GSM Channels, 2 FXO and 2 FXS Ports
Multi-Port VoIP Gateway with 8 VoIP Channels, 4 GSM Channels and 4 FXO Ports
Multi-Port VoIP Gateway with 8 VoIP Channels, 4 GSM Channels and 4 FXS Ports
All-Integrated IP PBX for 250 Users
All-Integrated IP PBX for 500 Users
ABOUT MATRIX
394-GIDC, Makarpura,
Vadodara-390010, India.
Phone: +91 265 2630555
Fax: +91 265 2636598
Email: Inquiry@MatrixTeleSol.com
URL: www.MatrixTeleSol.com
SMS ‘MATRIX’ to 99987 55555
V2.R1-Aug ‘09
An ISO 9001 Company, Matrix is a leader in the VoIP, GSM, Key Phone System and PBX market. An innovative, technology
driven and customer focused organization; the company is committed to keep pace with revolutions in the telecom industry.
This has resulted in bringing forth cutting edge products like VoIP Phones, VoIP Gateways, VoIP ATA, GSM FCT, GSM
Gateways, SMB PBX, Enterprise PBX, Hotel PBX, Voice Messaging Products , Communication Security Products and PLCC
Switches. With over 1,500,000 line units installed and growing by over 1500 line units per day, the installed base of Matrix
connects over 15,000,000 calls everyday. Thus, Matrix has gained the trust and admiration of users representing the entire
spectrum of industries. Matrix has won many awards for its innovative products.