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Dcare Technologies India Private Limited

New Siddhapudur, Coimbatore, Tamil Nadu

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Voip Gateways

A VoIP gateway is a gateway device that uses Internet Protocols to transmit and receive voice communications. The general term is ambiguous and can mean many different things. There are many such devices.
A VoIP gateway is a gateway device that uses Internet Protocols to transmit and receive voice communications (VoIP). The general term is ambiguous and can mean many different things. There are many such devices. They are quickly becoming the most common type of voice phone service in many areas.

Sangoma Vega Video MCU
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Video conferencing has become a popular service widely used by organizations. Low cost video endpoints, as well as the popularity of smart phones, have made enterprises think about integrating video capacity to their existing communication networks.

Vega Video MCU is a versatile SIP-based multipoint conferencing unit appliance that allows SMBs and SMEs, with multiple locations, to easily add video conferencing capacity to their existing telephony systems. Providing high quality voice and video conferencing in a single platform increases productivity for enterprises and reduces communication and travel costs.

Ease of Use

Imagine adding video conferencing capability to your existing PBX in 4 easy steps:

  1. Connect VEGA Video MCU to your enterprise network
  2. Configure IP Address information on MCU
  3. Connect to your enterprise IP PBX web interface, query and apply Vega MCU settings
  4. Make your first HD video conferencing call from your video IP phone endpoint
PBX Integration

Our MCU comes with a standardized RESTful API that PBX administrators can query to get necessary information in order to easily provision video conference rooms their existing PBXs.

Dedicated Media Processing Hardware

Our MCU has dedicated DSP resources that allow scalability for high density audio and video conferencing.



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  • Delivery Time: 1 day

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VoIP Gateway MARTIX VOIP ISDN BRI
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Power SupplyExternal Power Adaptor Input 90-265VAC, 47-63Hz Output 12VDC@2A
Number Of PortsDual BRI Ports
Network TypeMATRIXISDNBRI
Compatible BrandMATRIX
ColorBlack
Antenna1
Ethernet PortsEthernet (RJ45)

  • Up to Eight Simultaneous VOIP Calls
  • Two ISDN BRI Ports with Programmable TE/NT Modes
  • Deployable in all SIP based VOIP Networks
  • 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 a List of Denied Numbers
  • Direct Number Dialing on the Desired Network with Automatic Number Translation
  • Call Details Records of 2,000 Calls
  • PIN Authentication to Prevent Unauthorized Usage
  • Caller ID Presentation and Restriction
  • Emergency Number Dialing
  • Web based Configuration and Management


Additional Information:

  • Item Code: MATRIXVOIPISDNBRI
  • Port of Dispatch: Immediate

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Dinstar VoIP Gateway  FXO & FXS Gateways
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Number of Ports/PinsTelephone Port DAG1000-1S 1 FXS port (RJ11) DAG1000-2S 2 FXS ports (RJ11) DAG1000-4S 4 FXS ports
Network Type2G, 3G
Compatible BrandDinstar
Network Speed1 WAN, 10/100Mbps (RJ45) 1 LAN, 10/100Mbps (RJ45)
Model Name/NumberDINSTARDAG1000-1S/2S
AntennaYes
ColorBlack
BrandMatrix
Warranty1 Year

Cost-effective ATA with 1, 2 or 4 FXS ports Fax over IP (T.38 and Pass-Through)Support IPv4 and IPv6 TR069 and SNMP Elastix and Broadsoft Certification Multiple codecs: G.711A/U, G.723.1, G.729A/B, G.726 Fully compatible with leading IMS/NGN and SIP-based IP telephony systems

Additional Information:

  • Item Code: DINSTARDAG10001S2S
  • Port of Dispatch: IMMEDIATE

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Dinstar DAG2000-16S FXS Analog VoIP Gateway
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Compatible BrandDinstar
Power SupplyYes
Ethernet PortsEthernet RJ45
ColorBlack
BrandDinstar
ModelDinstar DAG2000-16S
Warranty1 Year
Supplementary Service Call Waiting Blind Transfer Attend Transfer Call Pickup Call Forward on Busy Call Forwa

•Support SIP/MGCP•Primary/Backup SIP Servers•Flexible Routing and Manipulation•Support Modem/POS•Elastix/Broadsoft Certification•Support IPv4 and IPv6•Data/ Voice/ Management VLAN
•SNMP/TR069/Provision
•Cloud-based Management System and Bandwidth Optimization¿¿¿


Additional Information:

  • Item Code: DinstarDAG20006SFXSAnalogVoIPGateway
  • Port of Dispatch: Immediate

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DINSTARDAG1000-4S4O FXS/FXO Mixed Analog VoIP Gateway
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Network Type2G, 3G
PowerInput 12VDC 2A Power Adapter
Compatible BrandDinstar
Power Supplyyes
DimensionsDimension 242L152W40H mm
Ethernet Interface4
Ethernet PortsEthernet RJ45
Net WeightWeight 1.0KG
ColorBlack
Warranty1 Year
Supplementary ServiceCall Waiting Call Transfer Blind transfer Attendant Transfer Call Forwarding Hotline Call H

• Primary and secondary SIP account• Flexible FXO port groups• Multiple SIP trunks• Caller ID• Flexible routing and manipulation rules• Data/Voice/Management VLAN• T.38 and Pass-through• Friendly Web GUI access,easy to maintain• Power failure /network disconnect bypass

Additional Information:

  • Item Code: DINSTARDAG10004S40FXOFFXS
  • Port of Dispatch: IMMEDIATE

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DINSTARDAG1000-4S/8S FXS Analog VoIP Gateway
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Compatible BrandDinstar
Power SupplyYes
Ethernet Interface4
Ethernet PortsEthernet RJ45
Output Power Output 12VDC 2A
ColorBlack
BrandDinstar
ModelDAG1000 4S 8S
Warranty1 Year
Supplement ServiceCall Waiting Blind Transfer Attend Transfer Call Pickup Call Forward on Busy Call Forward on No

  • Support SIP/MGCP
  • Primary/Backup SIP Servers
  • Flexible Routing and Manipulation
  • Support Modem/POS
  • Elastix/Broadsoft Certification
  • Support IPv4 and IPv6
  • Data/ Voice/ Management VLAN
  • Build-in Firewall and Access Rules
  • SNMP/TR069/Provision
  • Cloud-based Management System and Bandwidth Optimization


Additional Information:

  • Item Code: DAG10004S8SFXSAnalogVoIPGateway
  • Port of Dispatch: Immediate

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DINSTARDAG1000-4O/8O FXO Analog VoIP Gateway
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Compatible BrandDinstar
Power SupplyYes
Ethernet Interface4
ColorBlack
BrandDinstar
ModelDAG1000 4O8O
Warranty1 Year

• Primary and secondary SIP account• Flexible FXO port groups• Multiple SIP trunks
• Caller ID• Flexible routing and manipulation rules
• Data/Voice/Management VLAN• T.38 and Pass-through
• Friendly Web GUI access,easy to maintain

Additional Information:

  • Item Code: DAG10004O8OFXOAnalogVoIPGateway
  • Port of Dispatch: Immediate

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VOIP PRI GATEWAY
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VOIP PRI GATEWAY

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BrandMatrix
ColorBlack
Channel2.1, 4.1
Power SupplyYes
PowerYes

  • 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 a List of Denied Numbers
  • 2,000 Call Detail Records
  • PIN Authentication to Prevent Unauthorized Usage
  • Caller ID Presentation and Restriction
  • SNMP Monitoring
  • TR-069 Auto-configuration
  • VOIP Security over SRTP/TLS Encryption
  • Web based Configuration and Management


Additional Information:

  • Item Code: MatrixPRI
  • Delivery Time: Immediate

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DINSTAR DAG2500486472S FXS Analog VoIP Gateway
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ColorBlack
BrandDinstar
Voice CodecSilence Suppression Comfort Noise Generation(CNG) Voice Activity Detection(VAD) Echo Cancellation
ModelDAG2500-48/64/72S FXS
Supplementary ServiceCall Waiting,Blind Transfer,Attend Transfer,Call Pickup,Call Forward on Busy,Call Forward on No Rep

High density gateway, 48/64/72 FXS ports in 1U sizeOpen-standard SIP supportSupport IPv4 and IPv65KM Maximum Cabling LengthElastix/Broadsoft CertificationMultiple codecs: G.711A/U,G.723.1,G.729A/B, iLBC,AMRFully compatible with leading IMS/NGN, SIP based IP telephony system

Additional Information:

  • Item Code: DINSTARDAG2500486472SFXSAnalogVoIPGateway
  • Port of Dispatch: Immediate

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VoIP Gateway  FXO & FXS Gateways
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Compatible BrandMatrix
Power SupplyYes
Output PowerYes
ColorBlack
BrandMatrix
ProtocolSIPv2, SDP, RTP (RFC 2833)
Warranty1Year

  • Call Detail Records of 2,000 Calls
  • Caller ID Presentation on Analog Station
  • Emergency Number Dialling
  • 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 during IP Network Outage
  • Restrict Unwanted Calls with a List of Denied Numbers
  • SNMP Monitoring
  • TR-069 Auto-configuration
  • VOIP Security Over SRTP/TLS Encryption
  • Web based Configuration and Manageme


Additional Information:

  • Item Code: MatrixVOIPFXOFXSGATEWAYS
  • Port of Dispatch: Immediate

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VoIP GSM Gateway 32 Port SIM Gateway Dinstar
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BrandDinstar
Warranty1 Year
Power SupplyYes
ColorBlack

  • 3G Internet over First SIM (DL/UL - 14.4/5.76 Mbps)
  • Access Codes - Making a New Call and Disconnect Call
  • Allowed and Denied List
  • Automatic Number Translation
  • Backup Configuration
  • Blacklist Callers
  • Block ICMP on WAN
  • Block PING on WAN
  • Call Detail Record (CDR) – 2,000 Numbers
  • Call Minutes - Configurable Up to 9,999 Minutes
  • Call Minutes – Balance Alarm, Scheduled Date
  • Call Progress Tones
  • Call Waiting
  • Caller ID Presentation
  • CDR Filters - SIP Trunks, GSM Ports, Time, Date and PIN
  • Certificate Manager
  • Clone MAC Address
  • Daylight Saving Time
  • Dial Plan
  • Digest Authentication
  • Dynamic DNS
  • Emergency Numbers
  • Fax over IP (FoIP)
  • Firmware upgrade by HTTP/TFTP
  • G.168 Line Echo Cancellation
  • Gain/Volume Adjustment - Microphone/Speaker
  • Quad-band GSM and Penta-band 3G/UMTS Support
  • Hotline
  • Hot Swapping for SIM Cards
  • IMEI Number
  • Message Wait Indication
  • Mobile Network/Band Selection - Auto/Manual
  • Multiple Language Support - English, Italian, Spanish, French, German and Portuguese
  • Multi-stage Dialing
  • NAT/STUN Support
  • Password Protection
  • PCAP Trace
  • Peer-to-peer Calling
  • PIN Authentication
  • QoS (Layer 3 DiffServ and TOS)
  • Selective IP Address for Web Server Access
  • Selective WAN Access - Web Server, FTP, Telnet
  • Signal Strength Indication
  • SIM Balance and Recharge
  • SIM PIN
  • Single Antenna per Four GSM/3G Ports
  • SMS Notification - WAN Link Down, Change of IP Address, Calling Minutes, Data Usage and
  • SIM Balance
  • SNMP Monitoring
  • Soft Restart
  • Static Routing
  • System Debug
  • TLS/SRTP Security
  • TR-069 Auto Configuration
  • USSD
  • Web based Management
  • White List IP Address


Additional Information:

  • Delivery Time: Immediate

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DINSTARDAG2000-8S8O FXS/FXO Mixed Analog VoIP Gateway
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• Primary and secondary SIP account• Flexible FXO port groups• Multiple SIP trunks
• Caller ID• Flexible routing and manipulation rules
• Data/Voice/Management VLAN• T.38 and Pass-through
• Friendly Web GUI access,easy to maintain• Power failure /network disconnect bypass

Additional Information:

  • Item Code: DINSTARDAG20008S8OFXSFXOMixedAnalogVoIPGateway
  • Port of Dispatch: Immediate

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DINSTARDAG3000-112S Access Gateway
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SIP, MGCP, IMS, H.248(Optional)
Primary and secondary SIP account
Dual Power Module
Dual Main Control Board
Modular Design, Easy to Expanding
5km Cabling Length Maximum
Data/ Voice/ Management VLAN
SNMP, TR069
IP Trunk
IPv6
Firmware update automatically
Dinstar Cloud Management


Additional Information:

  • Item Code: DAG3000112SAccessGateway
  • Port of Dispatch: Immediate

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Voip Gateway Sangoma Vega 200g Digital Gateway
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The most resilient VoIP gateway in its class

The Vega 200G VoIP gateway connects digital telephony equipment to IP networks. Each unit purchased is factory configured with a maximum capacity of 60 simultaneous calls.

The available Sangoma Resiliency Enablement Suite (SRES) makes the Vega 200 the most resilient gateway in it is class. In the event of a WAN failure, IP phones behind the Vega gateway can continue to call each other, be routed to a backup switch or connected directly to the PSTN. In addition, the use of the Sangoma Network Appliance Provisioning (SNAP) tool makes the Vega 200 the easiest to provision gateway on the market.

Each E1/T1 interface can be independently configured as network side or terminal side. The Vega 200 gateway can therefore be connected to a PBX or the PSTN. This configuration provides:

  • No disruption to the configuration of existing equipment
  • Flexibility & choice for call routing

Open, non-proprietary interfaces

The Vega 200G gateway supports the following signalling schemes:

  • ETSI, VN4, ISDN
  • NI1, NI2, AT&T 5ESS, DMS100
  • ISO QSIG Basic Call & QSIG feature transparency
  • Channel Associated Signalling (CAS)
  • R2 MFC

All Vega gateways support SIP and T.38 Fax.

The Vega 200G gateway has proven interoperability with a wide range of existing telecommunications and VoIP equipment.

Service Provider Applications:
  • Customer premises gateway for SIP trunking
  • Low-density PSTN gateway
  • Survivability for IP phones
Enterprise Applications:
  • Enterprise VoIP networking
  • PSTN trunking for IP-PBXs
  • Enterprise IP telephony gateway

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Voip Gateway ISDN BRI Card
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Sangoma built its business by designing hardware that simply works, the first time. We have taken the time to ensure our BRI solution delivers.

  • Expand from two to twenty-four ports of BRI with optional telco-grade hardware echo cancellation
  • A single PCI or PCI Express slot hosts the connection for up to 24 ports and ensures common synchronous clocking for all channels
  • Lifetime Warranty (with product registration)

The Sangoma A500 S/T BRI interface card delivers superior audio quality and scalability. Expand from two to twenty-four ports of BRI with optional telco-grade hardware echo cancellation.

A single PCI or PCI Express slot hosts the connection for up to 24 ports and ensures common synchronous clocking for all channels with no signaling issues. The card is 100% software configurable.

Finally, a BRI card that upholds Sangoma’s high standards of quality in engineering and untiring product support.



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  • Delivery Time: 1 day

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Voip Gateway Sangoma Vega 100G
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THE MOST RESILIENT VOIP GATEWAY IN ITS CLASS

The Vega 100 VoIP gateway connects digital telephony equipment to IP networks. Each unit purchased is factory configured with a maximum capacity for 30 simultaneous calls.

The available Sangoma Resiliency Enablement Suite (SRES) makes the Vega 100 the most resilient gateway in it is class. In the event of a WAN failure, IP phones behind the Vega gateway can continue to call each other, be routed to a backup switch or connected directly to the PSTN. In addition, the use of the Sangoma Network Appliance Provisioning (SNAP) tool makes the Vega 100 the easiest to provision gateway on the market.

Each E1/T1 interface can be independently configured as network side or terminal side. The Vega 100 gateway can therefore be connected to a PBX or the PSTN. This configuration provides:

  • No disruption to the configuration of existing equipment
  • Flexibility & choice for call routing

OPEN, NON-PROPRIETARY INTERFACES

The Vega 100G gateway supports the following signalling schemes:

  • ETSI, VN4, ISDN
  • NI1, NI2, AT&T 5ESS, DMS100
  • ISO QSIG Basic Call & QSIG feature transparency
  • Channel Associated Signalling (CAS)
  • R2 MFC

All Vega gateways support SIP and T.38 Fax.

The Vega 100G gateway has proven interoperability with a wide range of existing telecommunications and VoIP equipment.

Service Provider Applications:
  • Customer premises gateway for SIP trunking
  • Low-density PSTN gateway
  • Survivability for IP phones
Enterprise Applications:
  • Enterprise VoIP networking
  • PSTN trunking for IP-PBXs
  • Enterprise IP telephony gateway


Additional Information:

  • Delivery Time: 1 day

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Sangoma A116: 16-Span T1/E1/J1 Card
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The A116 is part of Sangoma’s family of Advanced Flexible Telecommunications hardware product line – using high performance PCI Express interface, providing superior performance in critical systems all over the world. The A116 supports up to 32.8 Mbps of full duplex data throughput or 480 voice calls using 16 T1/E1/J1 spans.

With Sangoma cards, you can take advantage of hardware and software improvements, as soon as they become available. The A116, like all cards in Sangoma’s AFT family, is field-upgradable with our unique unbreakable firmware feature.

Choose the Sangoma A116E and A116DE, equipped with world class DSP hardware to achieve carrier-grade echo cancellation and voice quality enhancement functions for telecommunication systems.

Typical A116 Applications:
  • Call Centers
  • Value-Added Services
  • Large IVR Systems
  • Gateways and Soft Switches


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  • Delivery Time: 1 day

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Sangoma Vega 400G Digital Gateway
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Field-upgradeable

Choose the right license for your business requirements now: 30, 60, 90 or 120 simultaneous voice calls and expand it later to meet your growing business needs with a quick and easy license upgrade.

Configuration’s A Snap

Sangoma’s next generation Vega VoIP gateways are the easiest to provision on the market. It’s a SNAP with the Sangoma Network Appliance Provisioning tool. Keep the configuration of existing equipment; the Vega 400G delivers flexibility and choice when it comes to call routing. Each T1/E1 interface can be set up to connect to a network or a terminal side – e.g. directly to the PSTN or straight to your PBX.

Once Connected, Stay Connected

In the event of a WAN failure, IP phones behind the gateway can continue to call each other, be routed to a backup switch or connect directly to the PSTN. Plus, the Vega 400G comes complete with a pin switching and set of bypass T1/E1 connectors that don’t rely on cross-over cables. The next generation 400G is perfect for redundancy or failover scenarios.

Open, Non-proprietary Interfaces

The Vega 400G supports the following signalling schemes:

  • ETSI, VN4, ISDN
  • NI1, NI2, AT&T 5ESS, DMS100
  • ISO QSIG Basic Call and QSIG Feature Transparency
  • Channel Associated Signalling (CAS)
  • R2 MFC

All Vega gateways support SIP and T.38 Fax. The Vega 400G has proven interoperability with a wide range of existing telecommunications and VoIP equipment.

Enterprise Applications:
  • Enterprise VoIP networking
  • PSTN trunking for IP-PBXs
  • Enterprise IP telephony gateway
Service Provider Applications:
  • Customer premises gateway for SIP trunking
  • Low-density PSTN gateway
  • Survivability for IP phones


Additional Information:

  • Delivery Time: 1 day

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Sangoma STM1 Mux Fiber Multiplexer
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STM1 Mux Fiber Multiplexer Sangoma Fiber Multiplexers (Mux) arrange STM-1 or OC-3 fiber signal into 63 E1 or 84T1 framing, cost-effectively and reliably
  • Cost-effective DS3 bandwidth consolidation
  • Simple configuration via ethernet or RS232
  • Rate Stability
  • Improved Response Time
  • Timely Provisioning
  • Responsive Service and Repair
  • Flexible Customer Solutions
  • Reduce Operational Costs
  • Reliability
  • Security

As demand and bandwidth needs increase in today’s telecommunications and data network, large economies of scale can be achieved by substituting multiple single T1/E1 trunks with a single fiber connection running at 155 Mbps. A STM-1 or OC-3 interface will reduce overall expenses and control costs by combining data and voice applications over one circuit. Depending on the market, the benefits can be seen as soon as you subscribe to more than 20 T1 or E1 circuits simultaneously.

The STM1 Mux is a 155 Mbps fiber multiplexer that interfaces with the STM-1/ OC-3 network via a standards-based SFP transceiver with single mode fiber with LC connectors. The T1/E1 connectors are RJ-48, each carrying 2 T1/E1 per connector, in a DSX pinout.

The STM1 Mux offers a simplified and user-friendly management console accessible via ethernet or RS232. It features a VT100 terminal interface for system management, T1/E1 and fiber interface configuration.



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  • Delivery Time: 1 day

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Sangoma T116: T1/E1/J1 Tapping Card
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The T116 is part of Sangoma’s family of Advanced Flexible Telecommunications hardware product line — it uses the same high-performance PCI Express interface that is providing superior performance in critical systems all over the world.

The T116 supports the passive tapping of up to 240 voice calls using up to 16 T1, E1 or J1 spans.

With Sangoma cards, you can take advantage of hardware and software improvements, as soon as they become available. The T116, like all cards in Sangoma’s AFT family, is field upgradable with unbreakable firmware.

Choose the T116 to collect call control information, telecom protocol information and voice/media.

 

  • Sixteen receive-only spans with optimum PCI-Express interface enables tapping of sixteen one-way or eight two-way conversation
  • Support for Asterisk®, Yate®, and FreeSWITCH® PBX/IVR Projects, as well as other open source and proprietary PBX, Switch, IVR, or VoIP gateway applications
  • All of Sangoma’s AFT products use the same base PCIe interface card. Fully compatible with all commercially available motherboards—proper PCIe-standard interrupt sharing without manual tuning.
  • Dimensions: 2U Form factor: 120 mm x 55 mm for use in restricted chassis.
  • Intelligent hardware: Downloadable FPGA programming with multiple operating modes. Add new features related to voice and/or data when they become available
  • Line decoding: HDB3, AMI, B8ZS
  • Framing: CRC-4, Non CRC4, ESF, SF, D4. Also compatible with Japan’s J1
  • PCI Express: 1 Lane PCI Express bus
  • Maximum operational power: 5.5 W (1.67 A @ 3.3 V)
  • Temperature range: 0 – 50 °C
  • 32-bit bus master DMA data exchanges across PCI interface at 132 Mbytes/sec for minimum host processor intervention
  • Ring buffer DMA handling for minimum host intervention and guaranteed data integrity on high volume systems
  • Supports Robbed Bit Channel Associated Signaling (CAS) and ISDN PRI
  • T1/E1 and fractional T1/E1, multiple channel HDLC per line for mixed data/TDM voice applications
  • Optimized per channel DMA streams and hardware-level HDLC handling unload the host CPU
  • Uses raw bitstream interfaces to support arbitrary non-standard line protocols, such as non-byte aligned monosynch or bisynch
  • WANPIPE® routing stack is completely independent of TDM voice application for total system reliability
  • WANPIPE® supports certified, field-tested, and reliable Frame Relay, PPP, HDLC, and X.25


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  • Delivery Time: 1 day

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Voip Gateway SANGOMA A102: Dual Voice and Data Card
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Part of Sangoma’s award-winning Advanced Flexible Telecommunications product line, the A102 uses the same high-performance PCI or PCI Express interface that is providing superior performance in critical systems all over the world.

The A102 supports up to 60 voice calls or 4.096 Mbps of full-duplex data throughput over two T1, E1, or J1 lines.

Take advantage of hardware and software improvements, as soon as they become available. Like all cards in Sangoma’s AFT family, the A102 is field upgradable with crash-proof firmware.

Choose the Sangoma A102D and A102DE, equipped with world class DSP hardware to achieve carrier-gradeEcho Cancellation and Voice Quality Enhancement functions for your telephone systems.

  • Two T1/E1 spans with a single PCI or PCI-Express interface optimized for high performance voice and data applications
  • Mix T1 channel banks and E1 networks with full channel synchronization. TDM clocking mode lets network timing to be passed from a network-connected DS0 to any or all of the other spans, so both T1 and E1 are supported simultaneously
  • Support for Asterisk®, FreeSWITCH®, and Yate® as well as other Open Source and proprietary PBX, Switch, IVR, and VoIP gateway applications
  • All of Sangoma’s AFT products use the same base PCI interface card. Fully compatible with all commercially available motherboards—proper PCI-standard interrupt sharing without manual tuning.
  • Dimensions: 2U Form factor: 120 mm x 55 mm for use in restricted chassis. Includes high-quality, tested RJ45 cables and short 2U mounting clips for installation in 2U rack-mount servers
  • Intelligent hardware: Downloadable FPGA programming with multiple operating modes. Add new features related to voice and/or data when they become available

 

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Voip Gateway Sangoma D500: 400 2 000 Sessions
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Most IP telephony applications require the use of multiple types of voice codecs, which are used to digitally compress voice signals, to save on bandwidth requirements. While voice signals from the Public Switched Telephone Network (PSTN) always come in the form of the G.711 codec, the VoIP terminal equipment and networks support a variety of different voice codecs including such as G.729, G.726, AMR, G.722, iLBC, etc. VoIP infrastructure most often needs to include the capability to mediate between endpoints supporting different codecs, but this functionality often requires digital signal processing tasks that are costly, resource intensive and can affect the quality of the voice signals if it introduces too much latency and delay.

The D500 card converts numerous simultaneous channels of transcoding from one type of codec (e.g. G.711) to another (e.g. G.729), without affecting latency or using up precious host CPU resources. The card allows running up to 2000 sessions of any-to-any voice codec conversion, with unmatched quality1. All codecs are fully indemnified; no additional licensing is required for their use2.

The D500 works with both Asterisk and FreeSWITCH. With compatible drivers offered by Sangoma, these applications can use the D500 cards as seamless voice transcoding resources. Alternatively, developers and integrators can use the Transcoding API in C for their own application development.

 

Sample Applications:
  • Hosted VoIP
  • PBX with HD Voice
  • IP Network Peering
  • Hosted PBX
  • Call Centers

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Voip Gateway Sangoma PCI Card
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Part of Sangoma’s award-winning Advanced Flexible Telecommunications product line, the A104 uses the same high-performance PCI or PCI Express interface that is providing superior performance in critical systems all over the world.

The A104 supports up to 120 voice calls or 8.192 Mbps of full–duplex data throughput over four T1, E1, or J1 lines.

Take advantage of hardware and software improvements, as soon as they become available — like all cards in Sangoma’s AFT family, the A104 is field upgradable with crash-proof firmware.

Choose the Sangoma A104D and A104DE, equiped with world class DSP hardware to achieve carrier-grade Echo Cancellation and Voice Quality Enhancement functions for your telephone systems.

  • Four T1/E1 spans with a single PCI or PCI-Express interface optimized for high performance voice and data applications
  • Mix T1 channel banks and E1 networks with full channel synchronization. TDM clocking mode lets network timing to be passed from a network-connected DS0 to any or all of the other spans, so both T1 and E1 are supported simultaneously
  • Support for Asterisk®, FreeSWITCH®, and Yate as well as other Open Source and proprietary PBX, Switch, IVR, and VoIP gateway applications
  • All of Sangoma’s AFT products use the same base PCI interface card. Fully compatible with all commercially available motherboards—proper PCI-standard interrupt sharing without manual tuning.
  • Dimensions: 2U Form factor: 120 mm x 55 mm for use in restricted chassis. Includes high-quality, tested RJ45 cables and short 2U mounting clips for installation in 2U rack-mount servers
  • Intelligent hardware: Downloadable FPGA programming with multiple operating modes. Add new features related to voice and/or data when they become available.

 

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Voip Gateway Sangoma A108-PMC: Octal Voice & Data Card
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The A108-PMC is part of Sangoma’s family of Advanced Flexible Telecommunications hardware product line — it uses the same high-performance PMC interface that is providing superior performance in critical systems all over the world.

The A108-PMC supports up to 16.4Mbps of full duplex data throughput or up to 240 voice calls using up to eight T1,E1, or J1 lines.

With Sangoma cards, you can take advantage of hardware and software improvements, as soon as they become available. The A108-PMC, like all cards in Sangoma’s AFT family, is field-upgradable with crash-proof firmware.

  • Eight T1/E1 spans with optimum PMC interface for high performance voice and data applications
  • Support for Asterisk®, Yate® and FreeSWITCH® PBX/IVR Projects, as well as other open source and proprietary PBX, Switch, IVR, or VoIP gateway applications
  • Shares same hardware and driver design as Sangoma’s award-winning AFT products
  • Dimensions: 90 mm x 150 mm
  • Intelligent hardware: Downloadable FPGA programming with multiple operating modes; add new features related to voice and/or data when they become available.

 

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Voip Gateway Sangoma A101: Single Voice & Data Card
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The industry’s first affordable single span T1/E1/J1 card with available Telco-grade hardware echo cancellation — designed for optimum voice support for smaller systems.

  • One T1/E1 span with optimum PCI or PCI-Express interface for high performance voice and data applications
  • Support for Asterisk®, FreeSWITCH®, and Yate® as well as other Open Source and proprietary PBX, Switch,IVR, and VoIP gateway applications
  • Lifetime warranty (with product registration)

The A101 is part of Sangoma’s family of Advanced Flexible Telecommunications hardware product line — it uses the same high-performance PCI or PCI Express interface that is providing superior performance in critical systems all over the world.

The A101 supports up to 2.048 Mbps of full duplex data throughput or up to 30 voice calls over a single T1, E1, or J1 line.

With Sangoma cards, you can take advantage of hardware and software improvements, as soon as they become available. The A101, like all cards in Sangoma’s AFT family, is field-upgradeable with crash-proof firmware.

Choose the Sangoma A101D and A101DE, equiped with world class DSP hardware to achieve carrier-grade echo cancellation and voice quality enhancement functions for your telephone systems.



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Voip Gateway SANGOMA PCI or PCI Express
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The A108 is part of Sangoma’s family of Advanced Flexible Telecommunications hardware product line — it uses the same high-performance PCI or PCI Express interface that is providing superior performance in critical systems all over the world.

The A108 supports up to 16.4 Mbps of full duplex data throughput or up to 240 voice calls using up to eight T1, E1, or J1 lines.

With Sangoma cards, you can take advantage of hardware and software improvements, as soon as they become available. The A108, like all cards in Sangoma’s AFT family, is field upgradable with crash-proof firmware.

Choose the Sangoma A108D and A108DE, equiped with world class DSP hardware to achieve carrier-grade Echo Cancellation and Voice Quality Enhancement functions for your telephone systems.

  • Eight T1/E1 spans with optimum PCI or PCI–Express interface for high performance voice and data applications
  • Mix T1 channel banks and E1 networks with full channel synchronization. TDM clocking mode lets network timing to be passed from a network-connected DS0 to any or all of the other spans so both T1 and E1 are supported simultaneously.
  • Support for Asterisk®, Yate®, CallWeaver® and FreeSWITCH® PBX/IVR Projects, as well as other open source and proprietary PBX, Switch, IVR, or VoIP gateway applications
  • All of Sangoma’s AFT products use the same base PCI interface card. Fully compatible with all commercially available motherboards—proper PCI-standard interrupt sharing without manual tuning.
  • Dimensions: 2U Form factor: 120 mm x 55 mm for use in restricted chassis. Includes high-quality, tested RJ45 port-splitter cables and short 2U mounting clips for installation in 2U rack-mount servers
  • Autosense compatibility with 5 V and 3.3 V PCI busses
  • Intelligent hardware: Downloadable FPGA programming with multiple operating modes. Add new features related to voice and/or data when they become available

 



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SANGOMA B700 Flex BRI Hybrid Voice Card
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Seamlessly integrate fax onto your BRI phone system using only one interface card.
  • Connect up to 4 ports of BRI plus 2 ports of Analog FXO or FXS
  • Common synchronous clocking for all channels for guaranteed fax support
  • Includes telco-grade hardware echo cancellation

With the FlexBRI, now you can connect your analog lines into your BRI system using only one interface card.

Ideal for small or home offices, Sangoma’s industry-leading hybrid FlexBRI solution will seamlessly integrate youranalog fax machine with your BRI phone system, using only one PCI or PCI express slot and will fit in even the smallest 1U servers.

A single PCI or PCI Express slot hosts the connection for up to 4 ports of BRI and 2 ports of analog FXO or FXS and ensures common synchronous clocking for all channels with absolutely no signaling issues. The card is 100% software configurable.

Telco-grade, hardware echo cancellation is included.



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Sangoma T3 Mux M13 Multiplexer
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T3 Mux M13 MultiplexerSangoma’s M13 multiplexer re-arranges DS3 framing signal into 28 T1 framing
  • Cost-effective DS3 bandwidth consolidation
  • M13 and C-bit signaling support
  • External DS3 clock option
  • Simple configuration via ethernet or RS232
  • B8ZS and AMI DS1 signaling support
  • Industry standard CSU and NIU loopback codes

As demand and bandwidth needs increase in today’s telecommunications and data network. Large economies of scale can be achieved by substituting multiple single T1 trunks with a single DS3. A DS3 will reduce overall expenses and control costs by combining data and voice applications over one circuit. The benefits can be reaped as soon as you subscribe to more than 8 T1 circuits simultaneously.

The T3 Mux is a M13 Multiplexer that interfaces with the DS3 network via two BNC coaxial connectors, one for Rx (Input) and one for Tx(Output). The T1 connectors are RJ48, there are two T1 ports per connector, each in a DSXpinout.

The T3 Mux offers a simplified and user-friendly management console accessible via Ethernet or RS232. It features a VT100 terminal interface for system management, DS1 and DS3 Interface configuration.



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Sangoma PRI D100: 30 400 Sessions
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IP telephony applications commonly require the use of multiple voice codecs, used to digitally compress voice signals and save on bandwidth. Voice signals from the Public Switched Telephone Network (PSTN) come in the form of the G.711 codec, but the VoIP terminal equipment and networks can support a variety of different voice codecs, such as G.729, G.726, AMR, G.723.1, G.722, iLBC, etc. The VoIP infrastructure needs the capability to mediate between endpoints supporting different codecs, but this functionality requires digital signal processing tasks that are often costly and resource-intensive, and can affect the quality of the voice signals, if it introduces too much latency and delay.

The D100 card, available in PCI and PCI Express form factors, converts simultaneous channels of transcoding from one type of codec (e.g. G.711) to another (e.g. G.729), without affecting latency or using up precious host CPU resources. The card allows up to 30, 60, 120, 240 or 400 channels of any-to-any voice codec conversion, with unmatched quality¹. All codecs are fully indemnified; no additional licensing is required for their use².

The D100 works with both Asterisk and FreeSWITCH. With compatible drivers, these applications can use the D100 cards as seamless voice transcoding resources.

 

Sample Applications:
  • Call Centers and Remote Agent Pools
  • Hosted iPBX / IP Centrex
  • Distributed PBX
  • IP Network Peering
  • SIP Trunking
  • VoIP Gateways


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Sangoma Vega 5000 Analog Gateway
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The Vega 5000 Analog VoIP Gateway connects up to 50 analog handsets & devices to an IP network. The entire Sangoma Resiliency Enablement Suite (SRES) is available and the rich set of telephony features create an outstanding resilient solution that maintains key communications functions even in the event of a loss of connectivity to the IP-PBX or Hosted Solution. This is all supplemented by two FXO connections for direct connection to the PSTN or to a PBX. The GUI and Web-based SNAP Tool make the Vega 5000 one of the easiest gateways to provision and deploy.

Enterprise Applications:

  • Connect existing analog devices to an IP-PBX, Hosted IP-PBX solution, IP-Based UC platform or an IP-Based Contact Center implementation
  • Customer premises gateway for SIP Trunking
  • PSTN Trunking for IP-PBXs
  • VoIP Enablement of existing Contact Center or UC Installations

Service Provider Applications:

  • Customer premises gateway for SIP Trunking
  • FXS Endpoints for a hosted IP-PBX


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Sangoma D150: 30 400 Sessions
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Most IP telephony applications require the use of multiple types of voice codecs, which are used to digitally compress voice signals, to save on bandwidth requirements. While voice signals from the Public Switched Telephone Network (PSTN) always come in the form of the G.711 codec, the VoIP terminal equipment and networks support a variety of different voice codecs including such as G.729, G.726, AMR, G.722, iLBC, etc. VoIP infrastructure most often needs to include the capability to mediate between endpoints supporting different codecs, yet this functionality often requires digital signal processing tasks that are often costly, resource-intensive and can affect the quality of the voice signals if it introduces too much latency and delay.

The D150 card allows to convert numerous simultaneous channels of transcoding from one type of codec (e.g. G.711) to another (e.g. G.729), without affecting latency or using up precious host CPU resources. The card allows running up to 30, 60, 120, 240 or 400 channels of any-to-any voice codec conversion, with unmatched quality. All codecs are fully indemnified; no additional licensing is required for their use.

 

Sample Applications:
  • Call Centers and Remote Agent Pools
  • Hosted iPBX / IP Centrex
  • Distributed PBX
  • IP Network Peering
  • SIP Trunking
  • VoIP Gateways


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Sangoma B600 Analog Voice Card
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1 port of FXS and 4 ports of FXO. Introductory series optimized for small-budget installations. With B-Series hardware, Sangoma’s premium audio and engineering quality is even more affordable.

For small, stable systems, when the modular design, single-slot expansion and port-customization power of Sangoma’s A200 and A400 series is not required. The Sangoma B600 voice cards deliver customers a substantial cost savings, coupled with the “because it must work” quality and the Wanpipe® high-end system diagnostics, drivers, and utilities for which Sangoma is known worldwide.

Your customers can achieve optimum echo-cancelled voice quality and enhancement by choosing models withTelco-Grade DSP Hardware Echo Cancellation. One DSP echo canceller on the B600D or B600DE voice card provides Telco-grade hardware echo cancellation on all channels with no additional CPU load and minimal additional cost.



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Sangoma Vega 50 Media Gateway
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The Vega 50 VoIP gateway creates a connection between any analog or BRI device and a VoIP network. The Vega 50 can be factory configured to support a variety of configurations. When configured for analog telephony, the Vega 50 can support up to eight FXO ports or eight FXS ports with 2 FXO ports. When configured for BRI/ISDN2, the Vega 50 can support up to 4 BRI interfaces. Each BRI interface can be independently configured as network side or terminal side. Therefore, the Vega 50 media gateway can be connected to both a PBX & the ISDN simultaneously. The GUI and browser-based SNAP tool also make the Vega 50 one of the easiest gateways to provision and deploy.

The Vega 50 is an outstanding choice for a variety of enterprise and service provider application:

Enterprise Applications:
  • Resilient branch office support for hosted or premise-based PBX or IP-PBX
  • FXS endpoints and emergency number calling for hosted or premised-based IP-PBX
  • Support for door entry phones, POS, elevator/lift phones
Service Provider Applications:
  • Customer premises gateway for SIP trunking
  • FXS endpoints and emergency number calling for hosted IP-PBX
Analog
  • 4 FXS + 2 FXO
  • 8 FXS + 2 FXO
  • 4 FXO
  • 8 FXO
BRI / ISDN2
  • 2 BRI interfaces (4 channels)
  • 4 BRI interfaces (8 channels)
  • 8 BRI interfaces (16 Channels, subject to MOQ from customer)

Each BRI interface can be independently configured for connection to the PSTN or to a PBX. The Vega 50 media gatewaycan, therefore, be connected to both a PBX & the ISDN simultaneously. This configuration provides:

  • No disruption to the configuration of existing equipment
  • Flexibility & choice for call routing


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Sangoma A400: 2:24 Port Expandable Analog Voice Card
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The A400 Series of analog voice cards deliver superior audio quality in a 2U card that expands from two to twenty-four ports using a single interface slot.

For optimum echo-cancelled voice quality and enhancement, choose the A400D Model with DSP Daughtercard Hardware Echo Canceller. One DSP echo canceller on the A400D base card will support Telco-grade hardwareecho cancellation on all channels, up to the card’s maximum configuration in 24 ports with no additional CPU load.

To reach this configuration, simply add additional Remora cards to any base card in the A400 Series. A singlePCI or PCI Express slot hosts the connection and ensures common synchronous clocking for all channels.

Like all the cards in the award-winning Sangoma AFT Series Product Line, the A400 and Remora system’s architecture is shared with Sangoma’s A101, A102, A104, A108 and A200 cards, ensuring common 3.3 V or 5 V, high performance, universal PCI or PCI Express compatibility, and crash-proof field upgradable firmware to take advantage of enhancements, as they become available.

Sangoma cards guarantee error-free faxing and click-free audio on analog-digital links.



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Sangoma GSM Card for Voice & SMS
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Choose the W400 card to add GSM voice connectivity and SMS capabilities to IP-PBX applications.

  • Assign PBX extensions and forward calls to mobile devices
  • Send SMS messages over the GSM network
  • Convenient access to SIM cards on card

IP-PBX users will be able to reach mobile phone users on the cellular network, established as mobile-to-mobile calls. Assign PBX extensions and forward calls to mobile devices, which will greatly reduce operating costs.

The W400 also offers the possibility to send SMS messages over the GSM network, enhancing the IP-PBX to act as an SMS gateway. For added flexibility, SMS messages can be sent while a voice conversation is active on the GSM port.

The W400 has a unique design that allows access to SIM cards directly from the front of the card. This added flexibility removes the need to open up computer systems to access SIM cards.



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Sangoma B601 Hybrid Voice Card
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T1/E1/J1 + 4 FXO + 1 FXS

Introducing the Sangoma B-Series — now even cost-conscious customers can opt for Sangoma’s premium audio and engineering quality by choosing the voice card designed specifically for smaller-budget installations.

  • 34 Simultaneous PSTN Calls (Per Interface Slot)
  • Compatible with All Commercially Available Mothercards
  • 5 year Warranty (With Product Registration)
  • Hardware Echo Cancellation
  • Hardware Fax Synching Support

For small, stable systems, when the modular design, single-slot expansion and port-customization power of Sangoma’s A100, A200 and A400 series is not required.

The Sangoma B601 voice cards deliver customers a substantial cost savings by integrating on a single PCI or PCI Express slot both digital and analog interfaces. This, coupled with the “Because it must work” quality and the Wanpipe® high-end system diagnostics, drivers, and utilities for which Sangoma is known worldwide.

Your customers can achieve optimum echo-cancelled voice quality and enhancement with telco-grade DSPHardware Echo Cancellation. One DSP echo canceller on the B601D or B601DE voice card provides telco-grade hardware echo cancellation on all channels with no additional CPU load and minimal additional cost.



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