Grandstream UCM6301 IP PBX 1FXO 1FXS Appliance

The UCM6300 series allows businesses to build powerful and scalable unified communication and collaboration solutions. This series of 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. The UCM6300 series supports up to 3000 users and includes a built-in web meetings and video conferencing solution that allows employees to connect from the desktop, mobile, GVC series devices and IP phones. It can be paired with the UCM6300 ecosystem to offer a hybrid platform that combines the control of an on-premise IP PBX with the remote access of a cloud solution. The UCM6300 ecosystem consists of the Wave app for desktop, web and mobile, which provides a hub for collaborting remotely, and UCM RemoteConnect, a cloud NAT traversal service for ensuring secure remote connections. The UCM6300 series also offers cloud setup and management through GDMS and an API for integration with third-party platforms. By offering a high-end unified communications and collaboration solution packed with a suite of mobility, security, meeting and collaboration tools, the UCM6300 series provides a powerful platform for any organization.

Product Features:

• Supports up to 500 users and up to 75 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
• Supports video conferencing

Product Specifications:

• Analog Telephone FXS Port: 1 RJ11 Port
  
• PSTN Line FXO Ports: 1 RJ11 Port
  
• Network Interfaces: Three self-adaptive Gigabit ports (switched, routed or dual mode) with PoE+
  
• NAT Router: Yes (supports router mode and switch mode)
  
• Peripheral Ports: 1*USB 3.0, 1*SD card interface
  
• LCD Display: 320x240 color LCD with touch screen for Shortcut Keys and Scroll Bar
  
• Voice-over-Packet Capabilities: LEC with NLP Packetized Voice Protocol Unit, 128ms-tail-length carrier grade Line Echo Cancellation, Dynamic Jitter Buffer, Modem detection & auto-switch to G.711, NetEQ, FEC 2.0, jitter resilience up to 50% audio packet loss
  
• Voice and Fax Codecs: Opus, G.711 A-law/U-law, G.722, G722.1 G722.1C, G.723.1 5.3K/6.3K, G.726-32, G.729A/B, iLBC, GSM; T.38
  
• Maximum Call Capacity: Users: 500
  
• Concurrent calls (G.711): 75
• Max concurrent SRTP calls (G.711): 50
• Maximum Attendees of Conference Bridges: 4 Video Conference rooms and up to 20 parties with 1080p, assuming 4 video feeds + 1 screen sharing (H.264 & Opus) Voice Conference: Up to 75 parties (G.711)
  
• Operating: 32 - 113ºF / 0 ~ 45ºC, Humidity 10 - 90% (non-condensing)
  
• Storage: 14 - 140ºF / -10 ~ 60ºC, Humidity 10 - 90% (non-condensing)