• Grandstream HT801 Analog Terminal Adapter (ATA)

    • 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
  • Grandstream HT813 Analogo Telephone Adapter UAE

    • 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
  • Grandstream HT814 ATA

    • 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
  • Grandstream HT818

    • 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
  • Grandstream HT841

    • 4 FXO
    • 1 FXS
    • 2 GigE PoE
    • High-performance NAT router
    • Lifeline support (FXS port will be hard-relayed to FXO port) in case of a 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 the main server loses connection
    • Strong AES encryption with security certificate per unit
  • Grandstream HT881

    • Supports 3 SIP profiles through 1 FXS port and 4/8 FXO ports
    • High-performance NAT router
    • Lifeline support (FXS port will be hard-relayed to FXO port) in case of a 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 the main server loses connection
    • Strong AES encryption with security certificate per unit
  • Grandstream IP Phone GS Dect Supported By DP720 Base Wireless

    • It will only work with the Grandstream DP750 base station already installed
    • 5 DP720 phones are compatible with each DP750
    • Supports up to 10 sip counts per phone
    • Supports 984.3 ft. outdoor and 164.0 ft. indoor range from the dp750 base station
    • Full HD audio on both speaker and headset
    • DECT encryption and authentication technology to protect calls and accounts
  • Grandstream IP Phone GS Dect Supported By DP722

    • Supports a range of up to 350 meters outdoors and 50 meters indoors when used with the DP752 (300 meters outdoors when used with DP750)
    • 1.8 inch (128×160) color LCD with 2 programmable soft keys
    • Offers 20 hours talk time and 250 hours standby time
    • Supports up to 10 SIP accounts and 10 lines per handset as well as 3-way conferencing
    • Push-to-talk via a configurable button
    • Seamless 1-touch door control with Grandstream’s GDS series of Facility Access devices
    • HD audio on the speakerphone, handset, and headset jack
    • Software and firmware updates over-the-air
  • Grandstream IP Phone GS Dect Supported By DP730

    • Supports a range of up to 400 meters outdoors and 50 meters indoors when used with the DP752 (300 meters outdoors when used with DP750)
    • 2.4 inch (240×320) color LCD with 3 programmable soft keys
    • Offers 40-hour talk time and 500-hour standby time
    • Supports up to 10 SIP accounts and 10 lines per handset as well as 3-way conferencing
  • Grandstream UCM6301 IP PBX

    • Enterprise-Grade Unified Communication Solution
    • IP PBXs allows businesses of all sizes to build powerful and scalable unified communication solutions in an easy-to-manage fashion with no licensing fees
    • Analog Telephone FXS Ports: 1 RJ11 Port
    • PSTN Line FXO Ports: 1 RJ11 Port
    • Users: 50
    • Concurrent calls: 15
    • Max concurrent SRTP calls: 15
    • Video Conference: None
    • Voice Conference: Up to 15
    • Three self-adaptive Gigabit ports (switched, routed or dual card mode) with PoE+
    • 1USB 3.0, 1SD card interface
    • 320×240 color LCD with touch screen for Shortcut Keys and Scroll Bar
  • Grandstream UCM6302 IP-PBX

    • Supports up to 3000 users and up to 450 concurrent calls
    • Zero configuration provisioning of Grandstream SIP endpoints
    • Built-in conferencing & meetings platform; supports desktop, Wave app, and SIP endpoints
    • Wave for Android, iOS, Chrome and Firefox browsers allows communication with all UCM6300 users & solutions
    • API available for third-party integrations, including CRM and PMS platforms
    • Advanced security protection with secure boot, unique certificate and random default password to protect calls and accounts
    • Three Gigabit auto-sensing RJ45 network ports with integrated PoE+ and support NAT router
    • Automated NAT firewall traversal service facilitates secure remote connections
    • Supports Full-Band Opus voice codec and H.264/H.263/ H.263+/H.265/VP8 video codec, jitter resilience up to 50% packet loss
    • Compatible with GDMS for cloud setup, management and monitoring
    • Based on Asterisk* version 16 open source telephony operating system

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