Grandstream IP PBX & Gateways

Grandstream UCM IP PBXs provide a platform that unifies all business communication on one centralized network, including voice, video calling, video conferencing, video surveillance, web meetings, data, analytics, mobility, facility access, intercoms and more.  Grandstream UCMs range from small and medium business deployments to small and medium enterprise solutions. They offer an extensive set of features in an easy-to-manage solution.

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  • Grandstream

    Grandstream – GAC2570 7″ ANDROID10 CONFERENCE PHONE

    0 out of 5

    The GAC2570 is a full-duplex conferencing device featuring high-quality voice communication for meeting rooms, huddle areas, boardrooms, and other conferencing spaces. The device features an exceptional speaker with MMAD (Multichannel Microphone Array Design) to deliver a natural and immersive conferencing experience to both local and remote participants.

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  • Grandstream

    Grandstream DP760 DECT Repeater

    0 out of 5

    Grandstream DP760 is a powerful wideband HD DECT repeater that auto associates with Grandstream’s DP750/DP752 DECT Base Stations to offer extended mobility to business and residential users.

    • Up to 2 concurrent HD calls
    • Extends an additional range of 300 meters outdoors and 50 meters indoors for added mobility
    • Automatic or manual association to DP750/DP752 base stations for easy use
    • Supports Plug-n-Play features like auto association, auto region detection and seamless call handover
    • Automated provisioning options include XML config files
    • Ethernet connection provides PoE features
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  • Grandstream

    Grandstream HT801 NBN Ready Analog to VoIP converter.

    0 out of 5

    Designed for users looking to connect their analog devices to NBN /VoIP network at home or in the office. The HT801 is a powerful analog telephone adapter that is easily deployable and manageable.

    • Supports connection of 1 Analog Device
    • Supports 3 Way voice conference
    • Supports Call Forward, Call Waiting,  Message Waiting Indication.
    • TLS and SRTP security encryption technology to protect calls and accounts
    • Supports T.38 Fax Over IP
    • Supports wide range of Caller ID Formats
    • Failover SIP server automatically switches to secondary server if main server loses connection

     

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  • Grandstream

    Grandstream HT802 NBN Ready Analog to VoIP converter.

    0 out of 5

    Designed for users looking to connect their analog devices to NBN /VoIP network at home or in the office. The HT802 is a powerful analog telephone adapter that is easily deployable and manageable.

    • Supports connection of 2 Analog Devices
    • Supports 3 Way voice conference
    • Supports Call Forward, Call Waiting,  Message Waiting Indication.
    • TLS and SRTP security encryption technology to protect calls and accounts
    • Supports T.38 Fax Over IP
    • Supports wide range of Caller ID Formats
    • Failover SIP server automatically switches to secondary server if main server loses connection

     

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  • Grandstream

    Grandstream UCM 6300 IP Phone Systems with FXO / FXS Ports

    0 out of 5

    A powerful unified communication & collaboration solution for any organization, the UCM6300 series provides a high-end unified communications solution packed with an ecosystem of mobility, security, video and collaboration tools.

    • 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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