- 16 lines with up to 16 SIP accounts
- Built-in 1 mega-pixel CMOS tiltable camera for video calling with privacy wheel
- Runs on the Android 7.0 operating system
- Built-in Bluetooth for syncing with mobile devices and connecting Bluetooth headsets
- Dual-switched auto-sensing 10/100/1000Mbps network ports
- Integrated dualband Wi-Fi (2.4GHz & 5GHz)
- Built-in PoE/PoE+ for power and network connections
- Dual-mic HD speakerphone with noise reduction, advanced echo cancellation & excellent doubletalk performance
- 4-core 1.3GHz ARM Cortex A53 processor with 2GB RAM and 8GB eMMC Flash
- 5.0” (1280×720) capacitive 5-point touch screen HD TFT LCD
- Peripherals include HDMI-out, USB, headset jack, EHS
- 6-way audio conferencing & 3-way 720p 30fps HD video conferencing capability
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- 16 lines with up to 16 SIP accounts
- Built-in mega-pixel camera for video calling with privacy shutter
- Runs on Android 7.0 operating system
- Built-in Bluetooth for syncing with mobile devices and connecting Bluetooth headset
- Dual-switched auto-sensing 10/100/1000Mbps network ports
- Integrated dual-band WiFi (2.4GHz & 5GHz)
- Built-in PoE/PoE+ to power the device and give it a network connection
- Speakerphone with HD acoustic chamber, advanced echo cancellation & excellent double-talk performance
- 4-core 1.3GHz ARM Cortex A53 processor with 2GB RAM and 8GB eMMC Flash
- 7’’ (1024×600) capacitive 5-point touch screen TFT LCD
- TLS and SRTP security encryption technology to protect calls and accounts
- 7-way audio conferencing & 3-way 720p 30fps HD video conferencing capability
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- 16 lines with up to 16 SIP accounts
- Built-in 2 megapixel camera for video calling with privacy shutter
- Runs on the Android 7.x operating system
- Built-in Bluetooth for syncing with mobile devices and connecting Bluetooth headsets
- Dual-switched auto-sensing 10/100/1000Mbps network ports
- Integrated dualband Wi-Fi (2.4GHz & 5GHz)
- Built-in PoE/PoE+ for power and network connections
- Dual-mic HD speakerphone with advanced echo cancellation & excellent double-talk performance for any scenario
- 64-bit quad-core processor, 2GB RAM, and 16GB Flash
- 8’’ (1280x800) capacitive 10-point touch screen IPS LCD
- Peripherals include HDMI-in/out, USB, Micro SD, headset jack, EHS (Plantronics headsets)
- 7-way HD audio conferencing & 3-way 1080p 30fps HD video capability
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- 4 or 8 FXO PSTN ports
- 2 10/100 Mbps network ports
- 3-way voice conferencing
- Comprehensive codec support, caller ID, flexible dial plans and security protection
- Advanced security protection with SRTP
- Designed and tested for full interoperability with leading IP-PBXs, soft-switches and SIP-based environments
- Failover SIP server feature in case main SIP server goes down
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- 16/24/32 FXS ports, GXW4248 includes 2 50-pin Telco connectors
- 1 Gigabit network port
- 132x48 backlit graphic display with support for multiples languages
- 4 SIP server profiles per system, independent SIP account per port
- Designed and tested for full interoperability with leading IP-PBXs, soft-switches and SIP-based environments
- Advanced security protection with SRTP/TLS/HTTPS
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• Integrates digital PSTN and ISDN trunks with VoIP networks
• 1/2/4 Port E1/T1/J1 spans
• 30/60/120 concurrent calls
• Supports PRI, SS7 and MFC R2 digital signaling
• Supports all popular voice codecs, including Opus, G.722, G.729, iLBC, GSM-FR and more • Support T.38 Fax for creating Fax-Over-IP
• Dual Gigabit ports, configurable NAT router
• 2 USB ports, SD card slot (extra memory)
• Multi-language voice prompts
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- Connects and constantly monitors two UCM6510 together for high availability
- Smart failover solution that automatically switches to a hot-standby secondary UCM6510 if the primary one fails
- Up to 14 LED indicators showing real-time status of all of the telecom lines, network links, auxiliary devices, etc
- Gratuitous ARP forces SIP endpoints to refresh the MAC address of the new UCM6510 without interruptions
- Fast 10 to 50 second system switching time depending on the number of registered endpoints
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- 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
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- 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
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- 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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- 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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- 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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- 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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- 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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- 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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- 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
