Vee Pee Infomatric Private Limited

The HT814 delivers powerful VoIP technology and routing capabilities to home and office environments, and allows users to successfully connect their analog devices to a manageable, robustnetwork. Built using Grandstream’s market-leading SIP ATA/gateway technology, with millions of units successfully deployed worldwide, this powerful ATA features exceptional voice quality in various applications and environments. The HT814 comes with 4 easy-to-use FXS ports, an integrated Gigabit NAT router, state-of-the-art encryption with a unique security certificate per unit, automated provisioning for volume deployment and device management and outstanding network performance.

• Supports 2 SIP profiles through 4 FXS ports and dual Gigabit ports
• Includes a built-in NAT router which can handle routing speeds up to 100MBps
• TLS and SRTP security encryption technology to protect calls and accounts
• Automated provisioning options include TR-069 and XML config files
• Supports 3-way voice conferencing
• Failover SIP server automatically switches to secondary server if main server loses connection
• Supports T.38 Fax for creating Fax-over-IP
• Supports a wide range of caller ID formats
• Use with Grandstream’s UCM series of IP PBXs for Zero Configuration provisioning

Matrix VoIP to T1/E1 PRI Gateway SETU VTEP :

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 codes 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.

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

VoIP Port :

DAG2500-48/64/72S FXS Analog VoIP Gateway :

The DAG2500-48/64/72S FXS is an analog gateway that offers smooth transmission between analog phones and VoIP networks. This intuitive gateway is able to offer the standard R21 interface for 48/64/72S ports in 1 U size. With this port, users will be able to connect their fax, analog phone, and PBX through the standard VoIP networks. The DAG2500 also comes with support for standard SIP protocol with high-end compatibility with IMS/NGN systems as well as SIP-based IP telephony platforms. Users with small and mid-sized businesses, multi-region environments, and call centres can choose this device for their daily VoIP service needs.

DAG2500-48/64/72S FXS: The reliable gateway for VoIP networks :

The DAG2500’s gateway unit is able to offer an open-standard support for SIP with IPv4 and IPv6 network protocol support as well. It comes with a cabling length of up to 3 KM. With Broadsoft/Elastix certification, this gateway offers full compatibility with leading SIP-based IP telephony platforms as well as NGN/IMS systems.

With the DAG2500-48/64/72S FXS Analog VoIP unit, comfortable noise generation is acknowledgeable. Moreover, the device works with silence suppression mechanism to keep the voice crisp and clear. VAD or voice activity detection is also a signature quality of this device which promotes echo cancellation as well. An adaptive jitter buffer is also featured in the device with programmable gain control.

Other important aspects about the device:

Power supply needed in the DAG2500-48/64/72S is 100-240V AC which runs at a frequency between 50 to 60 Hz. The power consumption of the device is 75 W. During usage, the temperature needed for operating the device is set between 0 C – 45 C. Storage temperature when the device is not in use is set at -20 C to 80 C. The DAG2500 unit is set at a humidity level from 10% to 90% with no condensing aspects. This unit comes in dimensions 440mm x 280mm x 44mm with 1 U port size. The weight of the device is 4.0 kg with CE and FCC compliance.

The DAG2500-48/64/72S FXS is perfect for seamless transmission to keep the analog system running. This device has been specifically designed for analog systems like telephones, fax machines etc.

The HT818 is a powerful 8-port VoIP gateway with 8 FXS ports and an integrated Gigabit NAT router. Built for users looking for a strong analog-to-VoIP converter, it features Grandstream’s market-leading SIP ATA/gateway technology with millions of units successfully deployed worldwide. This powerful gateway carries exceptional voice quality in various application environments, strong encryption with unique security certificate per unit, automated provisioning for volume deployment and device management, and outstanding network performance for enterprise use.

• Supports 2 SIP profiles and 8 FXS ports
• Strong AES encryption with security certificate per unit
• Automated & secure provisioning options using TR069
• 3-way voice conferencing per port
• Exceptional voice quality with wide-band HD codec
• Supports T.38 Fax for reliable Fax-over-IP
• Supports dual Gigabit network ports
• High performance NAT router

The HT813 is an analog telephone adapter that features 1 analog telephone FXS port and 1 PSTN line FXO port in order to offer backup lifeline support using a PSTN line. The integration of a FXO and FXS port enables this hybrid ATA to support remote calling to and from the PSTN line. For added flexibility, the FXS port extends VoIP service to one analog device. Users can convert their analog technology to VoIP thanks to the HT813’s ultra-compact size, HD voice quality, advanced VoIP functionality, high-end security protection and multiple auto provisioning options. These advanced features also allow service providers to offer high quality IP service to customers looking to upgrade to VoIP

• Supports 2 SIP profiles through 1 FXS port and 1 FXO port
• Dual 100Mbps LAN and WAN ports
• Lifeline support (FXS port will be hard-relayed to FXO port) in case of power outage
• 3-way voice conferencing per port
• Automated & secure provisioning options using TR069
• Supports T.38 Fax for reliable Fax-over-IP
• Failover SIP server automatically switches to secondary server if main server loses connection
• Strong AES encryption with security certificate per unit

The HT812 delivers powerful VoIP technology and routing capabilities to home and office environments, and allows users to successfully connect their analog devices to a manageable and powerful VoIP network. Built using Grandstream’s market-leading SIP ATA/gateway technology, with millions of units successfully deployed worldwide, this powerful ATA features exceptional voice quality in various applications and environments. The HT812 comes with 2 easy-to-use FXS ports, an integrated Gigabit NAT router, state-of-the-art encryption with a unique security certificate per unit, automated provisioning for volume deployment and device management and outstanding network performance.

• Supports 2 SIP profiles through 2 FXS ports and dual Gigabit ports
• Includes a built-in NAT router which can handle routing speeds up to 100MBps
• TLS and SRTP security encryption technology to protect calls and accounts
• Automated provisioning options include TR-069 and XML config files
• Supports 3-way voice conferencing
• Failover SIP server automatically switches to secondary server if main server loses connection
• Supports T.38 Fax for creating Fax-over-IP
• Supports a wide range of caller ID formats
• Use with Grandstream’s UCM series of IP PBXs for Zero Configuration provisioning

The HT802 delivers powerful VoIP technology and routing capabilities to home and office environments, and allows users to successfully connect their analog devices to a manageable and powerful VoIP network. Built upon Grandstream’s market-leading SIP ATA/gateway technology, with millions of units successfully deployed worldwide, this powerful ATA features exceptional voice quality in various applications and environments. The HT802 comes with 2 easy-to-use FXS ports, state-of-the-art encryption with a unique security certificate per unit, automated provisioning for volume deployment and device management and outstanding network performance.

• Supports 2 SIP profiles through 2 FXS ports and a single 10/100Mbps port
• TLS and SRTP security encryption technology to protect calls and accounts
• Automated provisioning options include TR-069 and XML config files
• Supports 3-way voice conferencing
• Failover SIP server automatically switches to secondary server if main server loses connection
• Supports T.38 Fax for creating Fax-over-IP
• Supports a wide range of caller ID formats
• Use with Grandstream’s UCM series of IP PBXs for Zero Configuration provisioning

The HT801 delivers powerful VoIP technology and routing capabilities to home and office environments, and allows users to successfully connect their analog devices to a manageable and powerful VoIP network. Built upon Grandstream’s market-leading SIP ATA/gateway technology, with millions of units successfully deployed worldwide, this powerful ATA features exceptional voice quality in various applications and environments. The HT801 comes with 1 easy-to-use FXS ports, state-of-the-art encryption with a unique security certificate per unit, automated provisioning for volume deployment and device management and outstanding network performance.

• Supports 1 SIP profile through a single FXS port and a single 10/100Mbps port
• TLS and SRTP security encryption technology to protect calls and accounts
• Automated provisioning options include TR-069 and XML config files
• Supports 3-way voice conferencing
• Failover SIP server automatically switches to secondary server if main server loses connection
• Supports T.38 Fax for creating Fax-over-IP
• Supports a wide range of caller ID formats
• Use with Grandstream’s UCM series of IP PBXs for Zero Configuration provisioning
• Supports advanced telephony features, including call transfer, call forward, call-waiting, do not disturb, message waiting indication, multilanguage prompts, flexible dial plan and more