Aria Telecom Solutions Private Limited

MTG200 PRI VoIP gateway provides the ability to terminate (or originate) T1/E1 circuits directly into IP telephony system or call center, Asterisk-based distributions such as Elastix, FreePBX, etc., or Freeswitch. It provides the ability to handle 120 concurrent calls maximum with G711 codec and 64 concurrent calls with G729.

It supports ISDN PRI, SS7, and R2 signaling so that provides good connectivity to different PBX system.

Features:

• VoIP trunk gateway
• Support echo cancellation & voice decoding function
• Friendly GUI
• Simple configuration
• Strong handling capacity
• Voice codec option: G.711A/U,G.723.1 (option), G.729A/B(option), iLBC(option)

• Comfort Noise Generation (CNG)
• Voice Activity Detection (VAD)
• Adaptive (Dynamic) Jitter Buffer (DJB)
• HTTP/Telnet configuration
• Firmware upgrade by TFTP/Web
• DTMF mode: RFC 2833, SIP INFO and INBAND
• T.38/ Pass-Through FAX over IP

• 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.
• Eight T1/E1 ports with optimum PCI or PCI-Express interface for high-performance voice and data applications
• Support for Asterisk®, Yate®, CallWeaver® and FreeSwitch® PBX/IVR Projects
• Compatible with all commercially available motherboards
• Carrier-grade onboard echo cancellation
• A108E and A108DE PCI Express: 1 Lane PCI Express bus
• Supports Robbed Bit Channel Associated Signaling (CAS) and ISDN PRI

A101 Single Port Sangoma PCI Card

• Single E1/T1 port with PCI interface
• Support for Asterisk®, FreeSWITCH®, and Yate® as well as other Open Source and proprietary PBX, Switch, IVR, and VoIP gateway applications.
• Fully compatible with all commercially available motherboards
• Support Robbed Bit Channel Associated Signaling (CAS) and ISDN PRI
• Carrier-Grade hardware Echo Cancellation options

Supported OS:

• Windows® 2003, Windows® XP, Windows® Server 2008, Windows® Vista, Windows® 7
• Linux (all versions, releases, and distributions from 1.0 up)

Aria Telecom Solutions Pvt. Ltd. provides Digium interface cards to connect, IP PBX system, GSM gateway, IVR etc to the public telephone network. Digium cards are the high-performance cost-effective digital telephone network interface.

Here, some key features of Single span digital cards:

• Up to 24 (T1/J1) or 30 (E1) simultaneous calls
• PCI Express
• One RJ48 Interface Port
• Support T1/E1/PRI environments that require high-performance
• Protocol support: ISDN PRI, CAS, Robbed-Bit
• Selectable T1, E1 or J1 Mode / Half-Length, Half-Height, Digital Card

One Port PRI Gateway

Key features:

• VoIP trunk gateway
• Simple configuration and friendly GUI
• Support echo cancellation, DJB, VNG, VAD and Qos
• Strong handling capacity
• Voice codec option: G.711A/U,G.723.1 (option), G.729A/B(option), iLBC(option)

• Power: input:100-240V,50-60Hz output:12VDC,1.0A
• Operation Humidity: 10%-90% no condensation
• Weight: 0.75kg
• Temperature(operation): 0 ???~ 45 ???
• Storage: -20 ???~80 ???
• Dimension(w/D/H): 225*150*38mm

• One T1/E1 port with optimum Asterisk PCI interface.
• 30 simultaneous voice calls (or 2,048 Mbps full duplex data)
• Support for Asterisk®, FreeSwitch™, and Yate as well as other Open Source and proprietary PBX, Switch, IVR, and VoIP gateway applications
• Support Linux and Windows OS
• Carrier-Grade hardware echo cancellation options
• Line decoding: HDB3, AMI, B8ZS
• Support ISDN PRI
• Supports Robbed Bit Channel Associated Signalling (CAS)
• Autosense compatibility with 5 V and 3.3 V Asterisk PCI busses

• Two T1/E1 spans with a single PCI or PCI-Express interface optimized for high-performance data and voice applications
• Support 60 voice calls or 4.096 Mbps of full-duplex data throughput over two T1, E1, or J1 lines
• Support for Asterisk®, FreeSWITCH®, and Yate® as well as other Open Source and proprietary PBX, Switch, IVR, and VoIP gateway applications
• Available Sangoma A102D and A102DE, equipped with world-class DSP hardware to achieve carrier-grade Echo Cancellation and Voice Quality Enhancement functions for your telephone systems
• Line decoding: HDB3, AMI, B8ZS.
• Fully compatible with all commercially available motherboards - proper PCI-standard interrupt sharing without manual tuning.

• Four T1/E1 ports with a single PCI or PCI-Express interface
• Mix T1 channel banks and E1 networks with full channel synchronization
• Support for Asterisk®, FreeSwitch™, and Yate as well as other Open Source and proprietary PBX, Switch, IVR, and VoIP gateway applications
• Support most commercially available servers and motherboards
• PBX Support: FreePBX,Asterisk, FreeSWITCH, PBXact etc.
• Linux and Windows Supported
• Dimensions: 2U Form factor: 120 mm x 55 mm
• Line decoding: HDB3, AMI, B8ZS
• A104E and A104DE PCI Express: 1 Lane PCI Express bus

Aria Telecom Solutions Pvt. Ltd. provides world-class quality and affordable Asterisk PRI card in all over India. We provide a series of cost-effective and high-performance passive voice logger cards. Aria CF cards allow tapping on T1/E1/J1. The CF card automatically configures for all supported ISDN variants.

Features:

• 24 (T1) or 30 (E1) channels
• Easy to install
• External TAP Adaptor not required
• 2U rack/server mountable
• OS: Linux Kernel:2.4 to Latest
• Line coding: HDB3, AMI, B8ZS

• Allow up to 60(E1) or 48(T1) Channels
• Easy to install
• Career grade optional hardware echo cancellation
• Tapping capable
• 2U rack/server mountable
• Line coding: HDB3, AMI, B8ZS
• Operating System: Linux Kernel: 2.4 to Latest
• Designed with voice platforms including:- Asterisk: 1.4 to Latest, Free SWITCH : > 1.2.7, Yate: Under Dev

• Allow up to 96 (T1) or 120 (E1) channels
• Carrier-grade optional hardware echo cancellation
• Work with open source voice platforms: Asterisk: 1.4 to Latest, FreeSWITCH : > 1.2.7, Yate: Under Dev
• 2U rack/server mountable
• Line coding: B8ZS, HDB3, AMI
• Easy to install

• Support E1/T1
• Support Asterisk
• 30 Channel
• Support ISDN protocol
• Support soft eco cancelation
• PC Connectivity
• Compatible with ARIA & All other Asterisk-based Call center software

• IVRS
• Call Recording
• Call Center
• Conference Bridge

• E1/T1 port data and voice card
• Available up to 1, 2, 4 and 8 ports
• 30 to 240 simultaneous voice calls
• Available in PCI Express in 2U form factor
• Includes on-board diagnostic tool kit
• Support most commercially available servers and motherboards
• PBX Support: FreePBX,Asterisk, FreeSWITCH, PBXact etc.
• Support Windows and Linux
• Supports Asterisk, FreePBX, proprietary PBX, IVR and more
• Protocol Support: CAS, MFC/R2, PRI, SS7, HDLC, Frame Relay, X.25, Transparent bit-stream
• Carrier-Grade hardware Echo Cancellation options

• High performance and cost-efficient digital telephony interface cards
• Selectable T1, E1 or J1 Mode
• Half-length, half-height, digital card
• T1/J1 or 30 E1 simultaneous calls
• RJ48 Interface Port
• Calling Card Platforms
• PCI-Express (TE131 and TE133), PCI (TE132 and TE134
• Protocol support includes: ISDN PRI, Robbed-Bit, CAS
• Risk-free ESP guarantee
• 5 year warranty
• Compatible with all versions of Asterisk using the DAHDI driver framework
• Available ports: 1, 2, 4, 8
• Not compatible with Switchvox E-Series appliances

• VoIP trunk gateway
• Friendly GUI
• Simple configuration
• Strong handling capacity
• Voice codec option G.711A/U,G.723.1 (option), G.729A/B(option), iLBC(option)
• Support ISDN PRI, SS7 and R2 signaling
• Support echo cancellation, DJB, VNG, VAD and Qos
• Good compatibility, support Asterisk, Elastix, Freeswitchand Small and medium IPPBX platform

• ISDN Interface 4*E1/T1, G.703, 120RJ-48c
• RS232 ConsoleRJ-45, 115200bps
• Network InterfaceWAN and LAN, 2 * 10/100M Base-TX, RJ-45

• Comfort Noise Generation (CNG)
• Voice Activity Detection (VAD)
• Adaptive (Dynamic) Jitter Buffer (DJB)
• HTTP/Telnet configuration
• Firmware upgrade by TFTP/Web
• DTMF mode RFC 2833, SIP INFO and INBAND
• T.38/ Pass-Through FAX over IP