VoIP Gateways

An easy-to-use 2 port ATA
The HT802 is a 2 port analog telephone adapter (ATA) that allows users to create a high-quality and manageable IP telephony solution for residential and office environments. Its ultra-compact size, voice quality, advanced VoIP functionality, security protection and auto provisioning options enable users to take advantage of VoIP on analog phones and enables service providers to offer high quality IP service. The HT802 is an ideal ATA for individual use and for large scale commercial IP voice deployments.

• 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
• Supports advanced telephony features, including call transfer, call forward, call-waiting, do not disturb, message waiting indication, multilanguage prompts, flexible dial plan and more

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.

The HA100 offers an automated failover solution for the UCM6510 IP PBX. When connecting between two UCM6510, the HA100 constantly monitors the operation status of both UCM6510 and automatically switches the system control (including all of the connected telecom lines, network links, auxiliary devices, and all of the SIP endpoints previously registered on the primary UCM6510) to the hot-standby secondary UCM6510 in the event that the primary UCM6510 fails. It can complete the entire system switch between 10 and 50 seconds depending on the number of registered SIP endpoints. Thanks to its smart monitoring and automated failover capability, the HA100 is an ideal high-availability solution for the UCM6510 to ensure maximum total system reliability and uptime.

The GXW4500 series are E1/T1 Digital VoIP Gateways that allow digital PSTN and ISDN trunks to be integrated with VoIP networks. By connecting the GXW4500 series with a VoIP network and a traditional PBX or E1/T1/J1 provider, businesses can drastically increase the amount of PSTN/ISDN trunks integrated with their VoIP network. The GXW4500 series offers three models that provide 1, 2 or 4 T1/E1/J1 spans and support 30, 60 or 120 concurrent calls to cater to the VoIP needs of large and medium sized enterprises.

The GXW4500 series supports PRI, SS7 and MFC R2 digital signaling protocols and nearly all major voice codecs including Opus, G.722, G.711, G.729, G.723, iLBC, G.726, GSM-FR, etc. This digital VoIP gateway series includes dual Gigabit network ports with a configurable integrated NAT router, two USB 3.0 ports, and one SD card interface. By adding echo cancellation, jitter buffer, T.38 fax support, TLS and SRTP security encryption as well as automated provisioning by HTTP/TFTP with XML config files, the GXW4500 series is able to cater to the VoIP needs of large and medium enterprises.

The GXW4500 series are E1/T1 Digital VoIP Gateways that allow digital PSTN and ISDN trunks to be integrated with VoIP networks. By connecting the GXW4500 series with a VoIP network and a traditional PBX or E1/ T1 provider, businesses can drastically increase the amount of PSTN/ISDN trunks integrated with their VoIP network. The GXW4500 series offers three models that provide 1, 2 or 4 E1/T1/J1 spans and support 30, 60 or 120 concurrent calls to cater to the VoIP needs of large and medium sized enterprises.

• Dual Gigabit autosensing RJ45 network ports with integrated NAT router
• TLS and SRTP security encryption technology to protect calls and accounts
• Supports a widerange of voice codecs, including G.722, G.729, iLBC, and more
• Supports T.38 Fax for creating Faxover-IP
• Supports multilanguage voice promtps
• Supports up to 120 concurrent calls