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IP PBXs

A powerful yet easy to manage voice, video, data, and mobile communications platform have never been more achievable than when anchored with Grandstream’s series of UCM IP PBXs. The UCM series of IP PBXs offer an extensive set of unified communication features in an easy-to-manage on premise solution with no licensing fees, upgrade fees or costs per features. Our UCMs range from small and medium business deployments to small and medium enterprise solutions

IP VOICE TELEPHONY

As business and their scale differ, so do the IP Voice Telephony needs and requirements. Keeping in mind this factor, Customization or customized solutions is at the core of our offering. Explore Grandstream’s product that along with streamlining business communication, also increases the efficiency of workload, accessibility to communication system more flexible, and deployment of such easily. Constructing the perfect communication network for your business is our motto.

IP VIDEO TELEPHONY

Let the advanced video and audio communications create a rich powerful multimedia experience for users. Grandstream’s GXV3200 series of IP Video Phones with AndroidTM are cutting-edge video phones offers tablet-like functionality and web access right on your desktop.

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    WP850 WiFi IP Phone

    • Excellent data capture performance on 1D and 2D barcodes, Support NFC for data communication, Supports fingerprint recognition
    • 7-inch HD screen, support multi-touch capacitive, cover with 2.5D Corning 3rd generation glass
    • Dual-band WiFi with efficient antenna design and fast roaming support
    • Rechargeable 4800mAh battery, 12-hour talk time, 200-hour standby; USB Type-C port, supports fast charging
    • IP67 dustproof and waterproof, Drop-safe from 1.2-meter height.
    • Bluetooth for pairing headsets and mobile devices
    • HD voice design with AEC and Noise Shield Technology
    • 6 SIP accounts, 6 lines
    • Ergonomic design, 12.6mm slim
    • Android 9.0 OS, supports custom Android apps
     
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  • WP820 WiFi IP Phone

    • Dual-band WiFi with efficient antenna design and advanced roaming support
    • Bluetooth for syncing headsets and mobile devices (contacts and call transferring)
    • HD voice & dual MIC design with AEC and Noise Shield Technology
    • Rechargeable 1500mAh battery, 7.5 hour talk time, 150-hour standby
    • Accelerometer and configurable button for push-to-talk, panic and other related functions
    • Micro USB port and 3.5mm headset jack
    • 2 SIP accounts, 2 lines
    • Supports custom Android apps that fit the phone’s screen/keys
     
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  • WP810 WiFi IP Phone

    • Dual-band Wi-Fi with efficient antenna design and advanced roaming support
    • 2 SIP accounts, 2 lines
    • HD voice & dual MIC design with AEC and Noise Shield Technology
    • Rechargeable 1500mAh battery, 6-hour talk time, 120-hour standby
    • Configurable button for push-to-talk
    • Micro USB port and 3.5mm headset jack
     
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  • UCM6510 IP PBX

    • Supports up to 2000 SIP endpoint registrations, up to 200 concurrent calls and up to 64 conference attendees
    • 1GHz quad-core Cortex A9 processor
    • 1GB DDR3 Ram, 32GB Flash
    • 1 Integrated T1/E1/J1 interface, 2PSTN trunk FXO ports, 2 analog telephone/Fax FXS ports with lifeline capability
    • Gigabit network ports with Integrates PoE, USB, SD card, integrated NAT router
    • Comprehensive security protection using SRTP, TLS and HTTPS with hardware encryption accelerator
    • Quickly setup and provision Grandstream endpoints using the Auto-Discovery and Zero Config feature within the product’s web user interface
       
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  • UCM6308

    • UCM6308 supports up to 3000 users and 450 concurrent calls
    • Auto Discovery and Zero Configuration of Grandstream SIP endpoints
    • Integrated 8 PSTN trunk FXO ports, 8 analog telephone FXS ports with lifeline capability and up to 50 SIP trunk accounts
    • Three Gigabit auto-sensing RJ45 network ports with integrated PoE+ and support NAT router, USB, SD card etc.
    • Supports up to a 5-level IVR (Interactive Voice Response)
    • Built-in call recordings server; recordings accessible via web user interface
    • Built-in Call Detail Records (CDR) for tracking phone usage by line, date, etc.
    • Supports multi-language auto-attendant and call queue to efficiently handle incoming calls
    • Strongest possible security protection using SRTP, TLS and HTTPS encryption
    • Supports any SIP video endpoint that uses the H.264, H.263 or H.263+, H.265 codecs
    • Compatible with GDMS for cloud setup, management and monitoring
    • Automated NAT firewall traversal service facilitates secure remote connections
       
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  • UCM6304

    • UCM6304 supports up to 2000 users and 300 concurrent calls
    • Auto Discovery and Zero Configuration of Grandstream SIP endpoints
    • Integrated 4 PSTN trunk FXO ports, 4 analog telephone FXS ports with lifeline capability and up to 50 SIP trunk accounts
    • Three Gigabit auto-sensing RJ45 network ports with integrated PoE+ and support NAT router, USB, SD card etc.
    • Supports up to a 5-level IVR (Interactive Voice Response)
    • Built-in call recordings server; recordings accessible via web user interface
    • Built-in Call Detail Records (CDR) for tracking phone usage by line, date, etc.
    • Supports multi-language auto-attendant and call queue to efficiently handle incoming calls
    • Strongest possible security protection using SRTP, TLS and HTTPS encryption
    • Supports any SIP video endpoint that uses the H.264, H.263 or H.263+, H.265 codecs
    • Compatible with GDMS for cloud setup, management and monitoring
    • Automated NAT firewall traversal service facilitates secure remote connections
       
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  • UCM6302

    • Auto Discovery and Zero Configuration of Grandstream SIP endpoints
    • Integrated 2 PSTN trunk FXO ports, 2 analog telephone FXS ports with lifeline capability and up to 50 SIP trunk accounts
    • Three Gigabit auto-sensing RJ45 network ports with integrated PoE+ and support NAT router, USB, SD card etc.
    • Supports up to a 5-level IVR (Interactive Voice Response)
    • Built-in call recordings server; recordings accessible via web user interface
    • Built-in Call Detail Records (CDR) for tracking phone usage by line, date, etc.
    • Supports multi-language auto-attendant and call queue to efficiently handle incoming calls
    • Strongest possible security protection using SRTP, TLS and HTTPS encryption
    • Supports any SIP video endpoint that uses the H.264, H.263 or H.263+, H.265 codecs
    • Compatible with GDMS for cloud setup, management and monitoring
    • Automated NAT firewall traversal service facilitates secure remote connections
       
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  • UCM6301

    • UCM6301support up to 500 users and 75 concurrent calls
    • Auto Discovery and Zero Configuration of Grandstream SIP endpoints
    • Integrated 1 PSTN trunk FXO ports, 1 analog telephone FXS ports with lifeline capability and up to 50 SIP trunk accounts
    • Three Gigabit auto-sensing RJ45 network ports with integrated PoE+ and support NAT router, USB, SD card etc.
    • Supports up to a 5-level IVR (Interactive Voice Response)
    • Built-in call recordings server; recordings accessible via web user interface
    • Built-in Call Detail Records (CDR) for tracking phone usage by line, date, etc.
    • Supports multi-language auto-attendant and call queue to efficiently handle incoming calls
    • Strongest possible security protection using SRTP, TLS and HTTPS encryption
    • Supports any SIP video endpoint that uses the H.264, H.263 or H.263+, H.265 codecs
    • Compatible with GDMS for cloud setup, management and monitoring
    • Automated NAT firewall traversal service facilitates secure remote connections
       
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  • UCM6300 Series [New Arrival]

      Supports from 500 to 3000 users Supports from 75 to 450 concurrent calls Zero configuration provisioning of Grandstream SIP endpoints Security protection using SRTP, TLS and HTTPS encryption technology to protect calls and accounts Three Gigabit auto-sensing RJ45 network ports with integrated PoE+ and support NAT router API available for third-party integrations, including CRM and PMS platforms Built-in video conferencing platform with support for Grandstream Wave desktop app Support for wideband HD codecs OPUS & G.722; jitter resilience up to 50% audio packet loss Built-in call center suite and call queue for efficient call volume management Integrated Call Detail Records and call recording server 5-level IVR and multi-language auto-attendant to efficiently handle incoming calls Compatible with GDMS for cloud setup, management and monitoring Supports any SIP video endpoint that using the H.264, H.263 and H.263+ and VP8 codecs
       
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  • UCM6200 Series

    • UCM6202 and UCM6204 support up to 500 users and 50/75 concurrent calls, UCM6208 supports up to 800 users and 100 concurrent calls
    • Auto Discovery and Zero Configuration of Grandstream SIP endpoints
    • Integrated 2/4/8 PSTN trunk FXO ports, 2 analog telephone FXS ports with lifeline capability and up to 50 SIP trunk accounts
    • Gigabit network ports with Integrates PoE, USB, SD card
    • Supports up to a 5-level IVR (Interactive Voice Response)
    • Built-in call recordings server; recordings accessible via web user interface
    • Built-in Call Detail Records (CDR) for tracking phone usage by line, date, etc.
    • Supports multi-language auto-attendant and call queue to efficiently handle incoming calls
    • Strongest possible security protection using SRTP, TLS and HTTPS encryption
    • Supports any SIP video endpoint that uses the H.264, H.263 or H.263+ codecs
       
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  • IPVT10 – Enterprise Video Conferencing Server

    • Support for up to 300 participants and 120 video feeds per session; up to 10 simultaneous sessions
    • Video and audio recording with 500GB local storage
    • 1080p HD at 30fps via H.264/VP8 for real-time video and screen sharing
    • Advanced meeting controls, • Flexible scheduling, customizable registration options, follow-up email options, meeting reports and more
    • Access from PC/Mac, mobile devices, video conferencing systems, video phones, PSTN trunk, or SIP PBX
    • HTTPS and WSS/DTLSSRTP encryption for WebRTC, TLS/SRTP encryption for SIP
    • Advanced anti-jitter algorithm to sustain smooth audio & video against up to 30% packet loss
    • Live broadcast using Facebook/YouTube Live features
       
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  • IPVideoTalk

    • 50/100/150 Participants.
    • .• 49 Video Feeds at a time. • 720P/1080p Full HD Video & HD Audio • Unlimited 1-1 Call • Unlimited Number of Meetings • Access from Any Device • (Windows, Mac, Android ™ or iOS) • Minimum 70Min at Free Plan and Up to 6 Hours Per Group Meeting at paid Version • Screen Sharing, Chats, • Reports/Analytics, Recording • 1-Click Meetings, No Client Downloads • YouTube and Facebook Live Integration • 512MB/2GB/5GB Cloud Recording/Storage • VoIP Dial-In, Toll Dial-In from Land Line or Cell Phone • Group Accounts Administration & Billing • Unlimited Webinars
       
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  • HT818 Analog Telephone Adapter [ ATA ]

    • 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
     
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  • HT814 Analog Telephone Adapter [ 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
     
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  • HT813 Analog Telephone Adapter [ ATA ]

    • Supports 2 SIP profiles through 1 FXS port and 1 FXO port
    • Dual 100Mbps LAN and WAN ports
    • Lifeline support (FXS port will be hardrelayed 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 Faxover-IP
    • Failover SIP server automatically switches to secondary server if main server loses connection
    • Strong AES encryption with security certificate per unit
     
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  • HT812 Analog Telephone Adapter [ ATA ]

    • 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
     
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