ATA and GATEWAY. MATRIX GSM FCTs & GATEWAYS

Transcription

1 ATA and GATEWAY MATRIX GSM FCTs & GATEWAYS 1

2 SIMADO GFX11 GSM/3G Fixed Cellular Terminal for Voice Applications OVERVIEW SIMADO GFX11 is a compact fixed cellular terminal used to make and receive calls over GSM/3G network while retaining the existing infrastructure. SIMADO GFX11 eliminates interconnection charges and provides substantial cost savings by turning all calls into mobile-tomobile calls. It easily integrates with any PBX or telephone via GSM/3G network. SIMADO GFX11 is ideal for small offices, remote project sites and retail shops for cost-effective and reliable communication. KEY FEATURES Make and receive mobile calls from conventional analog phones Quad-band GSM and Tri-Band 3G network operation Restrict unwanted calls from dialing with list of denied numbers Hotline extension dialing Country specific programmable call progress tones like Dial Tone, Ring Back Tone and Busy Tone etc. Direct number dialing on the cellular networks with automatic number translation CPC signaling to prevent port blockages and accurate billing Caller ID presentation on analog extension Emergency number dialing, even in absence of SIM card APPLICATIONS GSM/3G Connectivity to Existing PBX Make mobile calls from analog extensions Fixed- to-mobile call cost reduction Maintain existing dialing habits and communication pattern Compatible with all industry leading PBXs Pay-Phone Applications Accurate billing with pulse metering Supports polarity reversal, ensuring accurate billing Authentication to ensure proper usage of the payphone Lock payphone with specific FCT, avoids improper usage of system Elevator Emergency Applications Establishes communication link with the help center during emergency situations like elevator break-down or power cuts Used in various applications like elevators, ATM centers, parkades, hospitals, shopping malls and senior citizen s homes 2

3 SIMADO GFX11 GSM/3G Fixed Cellular Terminal for Voice Applications SPECIFICATIONS Models: SIMADO GFX11 SIMADO GFX11 3G GSM FCT for Voice Applications 3G FCT for Voice Applications Mobile Port: GSM/3G SIM Cards 1 Frequency Bands SIM Interface Compliance FXS Port: Quad-Band GSM850, EGSM900, DCS1800, PCS1900 Tri-Band WCDMA 850/1900/2100 MHz or Tri-Band WCDMA 900/1900/2100 MHz 1.8, 3V SIM Interface ETSI GSM Phase2/2+ Dialing and Reception CLIP Answering and Disconnect Signaling RJ11 Pulse and DTMF Dialing DTMF and FSK CLI Presentation Polarity Reversal for Call Connect/Disconnect Signaling Power Supply: Input Power Consumption External Power Adaptor Input: VAC, 47-63Hz Output: 12VDC@1.25A 5W (Typical) Other: Dimension 13.3 x 19.8 x 4.4 cm (5.26 x 7.80 x 1.76 ) Mounting Options Operating Temperature Wall Mount and Table Top -10 C to +50 C (14 F to +122 F) 3

4 SIMADO GBR Fixed GSM/3G to ISDN BRI Gateway OVERVIEW SIMADO GBR is a fixed GSM/3G-to-ISDN B RI gateway interfacing ISDN devices to the mobile networks. The gateway offers up to 4 GSM/3G channels to connect a digital phone system via single or dual BRI ports. The gateway seamlessly routes calls over the GSM/3G network, bringing significant cost savings on fixed-to-mobile interconnection charges. The gateway serves as an ideal solution for business organizations, remote project sites with limited fixed-line connectivity and temporary office set-ups with limited budget to invest in costly ISDN lines. APPLICATION GSM/3G Trunking for ISDN PBX Easy integration with existing telephony interfaces and devices Make calls to mobile networks using existing telephony instruments Automated call route selection based on least cost algorithms Fixed-line alternate/back-up for remote office sites KEY FEATURES Direct number dialing on the cellular networks with automatic number translation Time/number based call routing Restrict unwanted calls with list of denied numbers Up to 4 GSM/3G SIM support Quad-Band GSM network operation Tri-Band 3G network operation Single Antenna for 4 GSM/3G SIMs Software configurable TE/NT modes ISDN network clock synchronization for error-free communication Caller ID presentation/restriction Emergency number dialing on GSM/3G networks, dial emergency numbers even in the absence of SIM Hotline extension setting Easy programming interface 4

5 SPECIFICATION Models: SIMADO GBR42 SIMADO GBR42 3G SIMADO GBR21 SIMADO GBR21 3G GSM to ISDN BRI Gateway with 4 GSM SIM and 2 BRI Ports 3G to ISDN BRI Gateway with 4 3G SIM and 2 BRI Ports GSM to ISDN BRI Gateway with 2 GSM SIM and 1 BRI Port 3G to ISDN BRI Gateway with 2 3G SIM and 1 BRI Port Mobile Port: GSM/3G SIM Cards 1 Frequency Bands SIM Interface Compliance Antenna ISDN BRI Port: Quad-Band GSM850, EGSM900, DCS1800, PCS1900 Tri-Band WCDMA 850/1900/2100 MHz or Tri-Band WCDMA 900/1900/2100 MHz 1.8, 3V SIM Interface ETSI GSM Phase2/2+ 2.5dBi,50Ω SMA(Male) (Panel Mount) Operational Mode Interface Switch Variant RJ45 Software Configurable NT/TE modes S/T Interface, Point to Point and Point to Multipoint ETSI - EURO ISDN NET3 BRI (BRI NET3) Power Supply: Input Power Consumption External Power Adaptor Input: VAC, 47-63Hz Output: 12VDC@2A 10W (Typical) Other: Dimension Mounting Options Operating Temperature 20.0x28.3x6.0 cm (7.9 x11.1 x2.4 ) Wall Mount and Table Top -10 C to +50 C (14 F to +122 F) 5

6 SIMADO GFXD1111S GSM/3G FCT for Voice and Data Applications OVERVIEW Matrix SIMADO GFXD1111S is a gateway to interface GSM and Plain Old Telephone System (POTS) networks. Incoming calls on GSM port can be routed on FXS or FXO port. Similarly, outgoing calls from FXS port are routed either on GSM or FXO port. While routing calls over GSM/3G network, the unit converts all fixed-to-mobile calls into mobile-to-mobile calls and hence provide significant saving to call costs. PBX users can avail the low tariff of GSM networks by connecting SIMADO GFXD1111S with the PBX without changing their existing infrastructure. SIMADO GFXD1111S routes the calls either on POTS or GSM network depending on the destination numbers dialed by the users. KEY FEATURES Allowed and Denied Lists SIMADO GFXD1111S permits or restricts of certain outgoing numbers for toll-control purpose. It is useful to restrict the dialling of long-distance and international numbers for proper system usage. A number is blocked if its prefix matches any entry in the Denied List. On the other hand, a number is allowed to go through if it matches any entry of the allowed list. Automatic Number Translation SIMADO GFXD1111S modifies the dialled numbers to match dialling formats of the network through which the call is routed. For example, GSM network require dialling of complete numbers including the area codes. If a caller has dialled only the local number, the SIMADO GFXD1111S adds area code as appropriate prefix. Call Progress Tones and Rings SIMADO GFXD1111S provides Call Progress Tones like Dial Tone, Ring Back Tone an Tone etc. Country-specific tones can be selected to match those of the country where it is installed- Emergency Number Dialing SIMADO GFXD1111S allows user dial emergency numbers for services like Police, Fire Brigade or Ambulance without SIM or without network registration. A maximum of 4 emergency numbers can be programmed. These numbers can also be dialed using the FXO. Fall-back Connectivity SIMADO GFXD1111S provides uninterrupted communication; even in non-availability of GSM network. In absence of GSM/3G network, all calls are dialed out from the FXO (PSTN) port, automatically. This feature sets user free from selecting FXO ports manually. In true sense it provides a Lifeline to critical business communication need. Fixed Line Replacement SIMADO GFXD1111S offers voice, data and SMS services over GSM network. It provides fixed line functionality but in a mobile solution, hence offering apt solution for locations lacking fixed line communication infrastructure. With same quality of service and user experience of fixed tie lines, it provides ideal alternative for leased line connectivity. General Packet Radio Services (GPRS Class 10) The GSM engine used in SIMADO GFXD1111S is GPRS Class 10 enabled. The user can browse the internet and check hi s. When it is connected to a proxy, multiple users can share the bandwidth for internet and s. Pulse Tone Generation SIAMDO GFXD111S provides both 12KHz and 16 KHz metering for PCO applications. It generates metering pulse for the PCO monitor connected to its FXS port for accurate billing. Short Message Services (SMS) SIMADO GFXD1111S allow its user to send and receive text messages (SMS) using any PC based SMS client. By connecting PC using serial COMM port, sms can be transmitted and received for various applications and promotional activities. Options like bulk sms and scheduled messaging are also available. Predefined SMS templates can also be used for easy and fast applications. 6

7 APPLICATION GSM/3G and POTS Connectivity to Traditional PBX Fixed-to-Mobile Call Cost Reduction Access GSM and PSTN Network from Analog Extensions Maintain existing Dialing Habits and Communication Patterns Fallback PSTN Connectivity Compatible with all Industry Leading PBX Messaging Application Send and Receive SMS from PC Integration with Various Third-Party Sms Clients Easy and Effective Marketing and Promotional Toll Internet Access over GPRS Internet Accessibility over GSM Network (GPRS Class 10) Send/receive Files, s and Web-browsing Ideal for application like Construction sites, Small Offices, Road- Side Offices and Shopping Centers. KEY FEATURES 12KHz/16KHz Pulse Tone Generation Allowed and Denied Lists Automatic Number Translation Call Divert Call Duration Display Call Hold Call Progress Tones and Rings Call Swap Call Waiting Calling Line Identification and Presentation (CLIP) Calling Line Identity Restriction (CLIR) Configuration Reports Data Service Distinctive Rings Emergency Number Dialling GPRS Class 10 Hotline Least Cost Routing (Time, Number and Combination) Location Information Indication Network Selection Short Message Services (SMS) Signal Strength Indication System Configuration (FXS, RS232C, GSM Port) 7

8 SIMADO GFX44 Fixed GSM/3G FCT to Analog Voice Gateway OVERVIEW SIMADO GFX44 is a fixed configuration GSM/3G-FXS voice gateway. It connects with the existing telephony devices via FXS interface. Up to 4 SIM cards can be placed inside the SIMADO GFX44 gateway allowing up to 4 concurrent calls. The calls to mobile numbers are placed via the GSM/3G SIM cards, reducing higher fixed-to-mobile interconnection charges. The gateway serves as an ideal solution for business organizations, remote project sites with limited fixed-line connectivity and temporary office set-ups. APPLICATION GSM/3G Trunking for Legacy Phone System Make mobile calls from existing telephone instruments Easy integration with existing telephony interfaces and devices Automated call route selection based on least cost algorithms Fixed-line alternate/back-up for remote office sites and call-shops KEY FEATURES Up to 4 GSM/3G SIM support Single Antenna for 4 GSM SIMs Quad-Band GSM network operation Tri-Band 3G network operation Time/number based call routing Direct number dialing on the cellular networks with automatic number translation Restrict unwanted calls with list of denied numbers CPC signaling to prevent port blockages and accurate billing Caller ID presentation on analog phones Hotline extension setting Emergency number dialing on GSM/3G networks, dial emergency numbers even in the absence of SIM Easy programming interface 8

9 SIMADO GFX44 Fixed GSM/3G FCT to Analog Voice Gateway SPECIFICATION Models: SIMADO GFX44 SIMADO GFX44 3G GSM to Analog Voice Gateway with 4 GSM SIM and 4 FXS Ports 3G to Analog Voice Gateway with 4 3G SIM and 4 FXS Ports Mobile Port: Frequency Bands SIM Interface Compliance Antenna Quad-Band GSM850, EGSM900, DCS1800, PCS1900 Tri-Band WCDMA 850/1900/2100 MHz or Tri-Band WCDMA 900/1900/2100 MHz 1.8, 3V SIM Interface ETSI GSM Phase2/2+ 1.8/2.5 *dbi, 50, SMA (Male), Fixed Omni Directional Swivel Antenna1.8/3.0 *dbi, 50, SMA (Male), Omni Directional with cable of 3meters Length FXS Port: Dialling CLI Presentation Call Maturity RJ11 DTMF and Pulse (10/20PPS) DTMF, FSK ITU-T V.23 and FSK Bellcore 202A CLI Polarity Reversal Power Supply: Input Power Consumption External Power Adaptor Input: VAC, 47-63Hz Output: 12VDC@2A 8W (Typical) Other: Dimension Mounting Options Operating Temperature 15.5 x 22.0 x 4.95cm (6.10 x8.66 x1.95 ) Wall Mount and Table Top -10 C to +50 C (14 F to +122 F) 9

10 SETU ATA1S AND SETU ATA2S VoIP-FXS Adaptors OVERVIEW SETU ATA1S and ATA2S are analog telephony adaptors with FXS ports to connect analog telephone instruments, multiple SIP Accounts to register with VoIP carriers (ITSPs) and Ethernet ports to connect IP network. Make and Receive Calls over Internet Access Telephony Features like Call Hold, Call Toggle, Conference and many more Binds with Stand-alone Application and with any existing PBX Provides Web based Interface for ease of Management APPLICATION Make VoIP Calls over Internet Network Cost-effective telephony over internet for teleworkers and SOHOs Integration with existing IT network for VoIP Calling Send/Receive FAX over Internet Allows Point-to-Point Calls over Internet KEY FEATURES Open-Standard SIP Support Allowed and Denied List Auto Provisioning for Mass Deployments Emergency Number Dialing Message Wait Indication for voice message retrieval Call Progress Tones and Rings PIN authentication Service 100 Speed Dial Entries Supplementary Call Management Features Web based Configuration and Management SPECIFICATION Models: SETU ATA1S SETU ATA2S VoIP adaptor with 1 FXS port VoIP adaptor with 2 FXS ports 10

11 SETU ATA1S AND SETU ATA2S VoIP-FXS Adaptors SPECIFICATION Parameters: SETU ATA1S and SETU ATA21S VoIP Protocol SIPv2, SDP, RTP (RFC 2833) Network Protocol IPv4, TCP, UDP, PPPoE, DHCP Server & Client, SNTP, NAT, STUN, HTTP, PPP, Dynamic DNS, Port Forwarding and DMZ Ethernet (RJ45) SIP Accounts 3 Voice Codecs Fax over IP (FoIP) Security G.711 (A/µ Law), G.723 and G.729AB T.38 and Pass-Through SRTP/TLS over SIP FXS Dialing and Reception Answering and Disconnect Signaling RJ11 DTMF and FSK CLI Presentation Polarity Reversal for call connect/disconnect signaling 11

12 SETU ATA211 VoIP-FXO-FXS Adaptors APPLICATION SPECIFICATION Parameters: SETU ATA211 VoIP Protocol SIPv2, SDP, RTP (RFC 2833) Network Protocol IPv4, TCP, UDP, PPPoE, DHCP Server & Client, SNTP, NAT, STUN, HTTP, PPP, Dynamic DNS, Port Forwarding and DMZ Ethernet (RJ45) SIP Accounts 3 Echo Cancellation Voice Codecs Fax over IP (FoIP) Security G.168 with up to 128ms Tail length G.711 (A/µ Law), G.723 and G.729AB T.38 and Pass-Through SRTP/TLS over SIP FXO FXO Port Dialing and Reception CLI Presentation 1 RJ11 Pulse and DTMF DTMF, FSK ITU-T V.23 and FSK Bellcore 202A CLI FXS FXS Port Dialing and Reception Answering and Disconnect Signaling RJ11 DTMF and FSK CLI Presentation Polarity Reversal for call connect/disconnect signaling 12

13 SETU ATA211G VoIP-GSM-FXS Adaptors APPLICATION SPECIFICATION Parameters: SETU ATA211G VoIP Protocol SIPv2, SDP, RTP (RFC 2833) Network Protocol IPv4, TCP, UDP, PPPoE, DHCP Server & Client, SNTP, NAT, STUN, HTTP, PPP, Dynamic DNS, Port Forwarding and DMZ Ethernet (RJ45) SIP Accounts 3 Echo Cancellation Voice Codecs Fax over IP (FoIP) Security G.168 with up to 128ms Tail length G.711 (A/µ Law), G.723 and G.729AB T.38 and Pass-Through SRTP/TLS over SIP FXS FXS Port Dialing and Reception Answering and Disconnect Signaling RJ11 DTMF and FSK CLI Presentation Polarity Reversal for call connect/disconnect signaling Sim Card 1 GSM Frequecy Bands SIM Interface Compliance Quad-Band GSM850, EGSM900, DCS1800, PCS V, 3V SIM Interface ETSI GSM Phase2/2+ 13

14 SETU VGFX Fixed VoIP to GSM/3G-FXO-FXS Voice Gateways OVERVIEW SETU VGFX is a single-box gateway solution for any legacy system or an IP-PBX. It offers seamless connectivity between IP, GSM/3G and POTS (FXO and FXS) networks. Organizations can place calls using preferred networks to lower communication cost and avoid higher interconnection charges. GSM/3G trunking not only saves on mobile call charges but also allows unique applications to be created. Organizations can have virtually free calling between office and field employees using the corporate CUG plans. SETU VGFX easily fits in to any existing data infrastructure and allows low-cost long distance calling through SIP truking. Organization can connect multiple branch offices over the WAN internet for cost-effective internal calling. APPLICATION GSM/3G VoIP Gateway for Traditional PBX Multiple network connectivity on a single platform VoIP access to traditional analog systems Save on recurring fixed-to-mobile call costs Retain dialing habits and patterns Compatible with all TDM PBX GSM/3G POTS Access for IP-PBX Place calls over GSM/3G or POTS from existing IP terminals Seamless integration with existing LAN/WAN infrastructure Deployable in all SIP based VoIP networks Cost-effective calling with advanced routing logics Lower CAPEX and early ROI Peer-to-Peer Calling Communicate with dispersed offices at minimum call costs Replace expensive Tie lines with cost-effective IP network for inter-office communication Make direct IP calls without involving SIP Server Eliminate carrier charges associated with proxy calling Dial by extension numbers between distant locations 14

15 SETU VGFX Fixed VoIP to GSM/3G-FXO-FXS Voice Gateways KEY FEATURES Multiple GSM/3G SIM support Make and receive calls from/to analog terminals over the cellular and IP networks Quad-Band GSM and Tri-Band 3G network operation Open-standard SIP support Multiple SIP proxy registrations Restrict unwanted calls with list of denied numbers Direct number dialing on the cellular networks with automatic number translation Caller ID presentation/restriction Fax over IP (T.38 and PSTN Pass-Through) Emergency number dialing Hotline extension setting SIM balance inquiry and recharge from web-based GUI SNMP Monitoring Message Wait Indication Port Forwarding and DMZ VoIP Security over SRTP/TLS Encryption Web based configuration and management SPECIFICATIONS Models: SETU VGFX8422 SETU VGFX8422 3G SETU VGFX8404 SETU VGFX8404 3G Gateway with 8 VoIP Channels, 4 GSM SIMs, 2 FXO and 2 FXS Ports Gateway with 8 VoIP Channels, 4 3G SIMs, 2 FXO and 2 FXS Ports Gateway with 8 VoIP Channels, 4 GSM SIMs and 4 FXS Ports Gateway with 8 VoIP Channels, 4 3G SIMs and 4 FXS Ports VoIP Port: Protocol SIPv2, SDP, RTP (RFC 2833) SIP Accounts Up to 9 Echo Cancellation Voice Codecs Mobile Port: GSM/3G SIM Cards Up to 4 Frequency Bands SIM Interface Compliance FXO Port: Ethernet (RJ45) G.168 with up to 128ms Tail length G.711 (A/µ Law), G.723 and G.729AB Quad-Band GSM850, EGSM900, DCS1800, PCS1900 Tri-Band WCDMA 850/1900/2100 MHz or Tri-Band WCDMA 900/1900/2100 MHz 1.8, 3V SIM Interface ETSI GSM Phase2/2+ Dialing and Reception CLIP RJ11 DTMF and Pulse Dialing DTMF, FSK ITU-T V.23 and FSK Bellcore 202A CLI FXS Port: Dialing and Reception CLIP Answering and Disconnect Signaling RJ11 Pulse and DTMF Dialing DTMF and FSK CLI Presentation Polarity Reversal for call connect/disconnect signaling 15

16 SETU VFX FAMILY Fixed VoIP to FXO-FXS Gateways OVERVIEW SETU VFX is a range of multi-channel SIP gateway offering seamless connectivity between VoIP and PSTN networks. These gateways offer 4 to 32 FXO/FXS ports to connect TDM based telephony infrastructure to an IP network. For organizations those already migrated to IP, SETU VFX provides POTS trunking option. SETU VFX series offers up to 32 VoIP channels, Fax over IP support and flexible dialing plans. The dedicated signal processing resources and superior protocol set ensures multiple call capabilities with toll-grade voice quality. SETU VFX enables small and medium businesses to create seamless office environment, integrate a traditional phone system into IP network and significantly reduce communication costs.these gateways are ideally suited for business and consumer VoIP services including call centers and multi-location environments. APPLICATION VoIP Gateway for Traditional PBX VoIP access to traditional analog systems Enhance connectivity of existing TDM PBX Maintain existing dialing habits and business communication patterns Multi-Site Connectivity Cost-effective inter-branch office communication Centralized management and monitoring Remote access to call management features of main office IP-PBX system Peer-to-Peer and proxy calling between distant locations Maintain vital business continuity in event of IP network failure KEY FEATURES Call detail records of 2000 calls Caller ID presentation on analog station Emergency number dialing Fax over IP (T.38 and Pass-Through) Hotline extension setting Make and receive calls using analog terminals over the IP network Port Forwarding and DMZ PSTN Pass-Through port to make calls over POTS in case of IP network outage Restrict unwanted calls with list of denied numbers SNMP Monitoring VoIP Security over SRTP/TLS Encryption Web based configuration and management 16

17 SETU VFX FAMILY Fixed VoIP to FXO-FXS Gateways SPECIFICATIONS Models: SETU VFX404 SETU VFX440 SETU VFX808 SETU VFX880 SETU VFXTH0808 SETU VFXTH1616 SETU VFXTH1600 SETU VFXTH3200 SETU VFXTH0016 SETU VFXTH0024 SETU VFXTH0032 VoIP-FXO-FXS gateway with 4 VoIP, 4 FXS and 1 FXO (Pass-Through) Ports VoIP-FXO-FXS gateway with 8 VoIP and 4 FXO Ports VoIP-FXO-FXS gateway with 8 VoIP, 8 FXS and 1 FXO (Pass-Through) Ports VoIP-FXO-FXS gateway with 8 VoIP and 8 FXO Ports VoIP-FXO-FXS gateway with 16 VoIP, 8 FXO and 8 FXS Ports VoIP-FXO-FXS gateway with 32 VoIP, 16 FXO and 16 FXS Ports VoIP-FXO-FXS gateway with 16 VoIP and 16 FXO Ports VoIP-FXO-FXS gateway with 32 VoIP and 32 FXO Ports VoIP-FXO-FXS gateway with 16 VoIP and 16 FXS Ports VoIP-FXO-FXS gateway with 24 VoIP and 24 FXS Ports VoIP-FXO-FXS gateway with 32 VoIP and 32 FXS Ports VoIP Port: Protocol SIPv2, SDP, RTP (RFC 2833) Ethernet (RJ45) SIP Accounts Up to 32 Echo Cancellation G.168 with up to 128ms Tail length Voice Codecs G.711 (A/µ Law), G.723 and G.729AB FXO Port: Dialing and Reception CLIP PSTN Pass-Through RJ11 DTMF and Pulse Dialing DTMF, FSK ITU-T V.23 and FSK Bellcore 202A CLI 1 (SETU VFX404 and SETU VFX808) FXS Port: Dialing and Reception CLIP Answering and Disconnect Signaling RJ11 Pulse and DTMF Dialing DTMF and FSK CLI Presentation Polarity Reversal for call connect/disconnect signaling 17

18 SETU VGB802 Fixed VoIP to ISDN BRI Gateways OVERVIEW SETU VGB is a multi-port fixed VoIP to ISDN BRI gateway offering 8 VoIP channels and 2 ISDN BRI ports. It connects ISDN PBX to IP network for cost-effective and flexible communication. For an IP-PBX, SETU VGB provides ISDN BRI trunking. The gateway enables inbound and outbound calls over VoIP and ISDN BRI networks. Seamless connectivity with existing communication setup, enterprise-grade features and cost-effective communication over IP network help businesses to realize an early return on the investment. Flexible and intelligent call routing options further ensures that communication takes place using the most cost-effective network. The gateway serves as an ideal solution for small businesses and call centers to control operational costs. Service Providers and system integrators addressing business customers can offer a strategic migration to VoIP with protecting their existing investment. Corporates with multi-locational branch offices can avail benefits of VoIP network for making cost-effective interoffice communication. APPLICATION VoIP Gateway for Traditional ISDN PBX VoIP Access to ISDN BRI Systems Maintain existing PBX infrastructure and dialing habits Programmable TE/NT modes for easy integration Compatible with all ISDN PBX ISDN BRI gateway for IP-PBX ISDN BRI trunking for IP-PBX Deployable in all SIP based VoIP networks Reduced call cost for making ISDN BRI calls 18

19 SETU VGB802 Fixed VoIP to ISDN BRI Gateways APPLICATION Peer-to-Peer Calling Place VoIP calls without involving SIP proxy server No need of Static IP, Peer-to-Peer Calls can be established with dynamic Public IP address Dial by extension numbers between distant locations DDI Routing over IP Extends ISDN DDI service over IP from one location to another Receive DDI calls directly on remote branch extensions Route callback over ISDN trunk either from local or remote branch KEY FEATURES Up to 8 Simultaneous VoIP Calls 2 ISDN BRI Ports with programmable TE/NT modes Deployable in all SIP based VoIP network Simultaneous peer-to-peer and proxy calling Multiple SIP proxy registrations ISDN Network clock synchronization for error-free communication Cost-effective calling with intelligent least cost routing Fax over IP (T.38 and Pass-Through) VLAN tagging for advanced networking Restrict unwanted calls with list of denied numbers Direct number dialing on the desired network with automatic number translation Call details records of 2000 calls PIN authentication to prevent unauthorized usage Caller ID presentation and restriction Emergency number dialing Web-based configuration and management 19

20 SETU VGB802 Fixed VoIP to ISDN BRI Gateways SPECIFICATIONS Models: SETU VGB802 Gateway with 8 VoIP Channels and 2 ISDN BRI Ports VoIP Port: Protocol SIPv2, SDP, RTP (RFC 2833) Ethernet (RJ45) SIP Accounts 4 Echo Cancellation G.168 with up to 128ms Tail length Voice Codecs G.711 (A/µ Law), G.723 and G.729AB ISDN BRI Port: Channels Operational Mode Interface Switch Variant Dual BRI Ports RJ45 Software Configurable NT/TE Modes S/T Interface, Point to Point and Point to Multipoint ETSI- Euro ISDN BRI NET3 20

21 SETU VTEP Fixed VoIP to T1/E1 PRI Gateways OVERVIEW SETU VTEP is a compact, dedicated and feature-rich VoIP to T1/E1 PRI gateway. It is a single-span gateway offering 30 simultaneous VoIP to ISDN PRI calls. This in-line device sits between ISDN PBX and T1/E1 PRI line to connect PBX users to the IP network for cost-effective communication. For an IP based system it provides T1/E1 PRI trunking. The gateway efficiently delivers toll-grade voice quality with industry standard voice codecs and advanced QoS techniques. Multiple mounting options and remote management through web-based console adds to the operating ease. The gateway is suitable for SMBs, Large enterprises, VoIP service providers and System integrators for smooth migration to the new-age IP telephony. It helps them to control the communication overheads and realize an earlier return on investment through advanced features and functionalities. With SETU VTEP, multi-branch offices can use their existing broadband connections to setup cost-effective IP communication among them. APPLICATION VoIP Gateway for Traditional ISDN PBX VoIP Access for legacy phone systems with T1/E1 PRI interface Programmable TE/NT modes Retain existing PBX infrastructure and dialing habits Existing terminals can be used to place VoIP calls Network clock synchronization for error-free communication Compatible with most make of ISDN PBX ISDN PRI gateway for IP-PBX ISDN PRI Trunking for IP based System Deployable in all SIP based VoIP networks Connectivity to traditional and wide-spread ISDN PRI network Peer-to-Peer Calling Virtually access trunk connectivity of remote location Connect multiple branch offices over costeffective IP network Reduce higher inter-network call charges Hop-on and Hop-off calls to and from IP and ISDN PRI network 21

22 SETU VTEP Fixed VoIP to T1/E1 PRI Gateways KEY FEATURES Up to 30 simultaneous VoIP to ISDN PRI calls T1/E1 PRI Port with programmable TE/NT modes Deployable in all SIP based VoIP network Register with multiple SIP service providers Simultaneous peer-to-peer and proxy calling ISDN network clock synchronization for error-free communication Fax over IP (T.38 and Pass-Through) VLAN tagging for advanced networking Restrict unwanted calls with list of denied numbers Call details records of 2000 calls PIN authentication to prevent unauthorized usage Caller ID presentation and restriction SNMP Monitoring VoIP Security over SRTP/TLS Encryption Web based configuration and management SPECIFICATIONS Models: SETU VTEP321 Gateway with 32 VoIP Channels and 1 T1/E1 PRI Port VoIP Port: Protocol SIPv2, SDP, RTP (RFC 2833) Ethernet (RJ45) SIP Accounts 32 Echo Cancellation Voice Codecs G.168 with up to 128ms Tail length G.711 (A/µ Law), G.723 and G.729AB ISDN PRI Port: Channels Switch Variant Framing T1-23B+D, E1-30B+D RJ45 T1 RBS - AT&T 5ESS, DMS-100, US Ni2 E1 CAS - ETSI NET5,ITU-T Q.921, ITU-T Q.931 T1 RBS - SF-D4/ESF E1 CAS - CEPT1 (with/without CRC) with CAS MF 22