Grandstream UCM6301 IP PBX
The Grandstream UCM6301 IP PBX is a powerful unified communications system designed for small to medium-sized businesses that require voice, conferencing, security, and centralized management—all in a compact and reliable format. Supporting up to 500 SIP extensions and 75 concurrent calls, the UCM6301 integrates traditional telephony with IP voice solutions, including analog line support, robust conferencing capabilities, advanced security, and seamless integration with mobile and desktop apps. Built on the Asterisk 16 platform, it provides license-free access to enterprise-grade features such as IVR, call recording, and remote collaboration.
Product Features
Scalable Voice Platform
Supports up to 500 SIP user extensions and 75 concurrent SIP calls, ideal for office environments with growing voice communication needs.
Hybrid Connectivity with Analog Support
Includes one FXO port for PSTN trunk lines and one FXS port for analog devices, ensuring seamless integration with legacy telephony systems.
Integrated Audio and Video Conferencing
Enables up to 75-party audio conferences and 12-party video meetings across two rooms. Supports internal bridge creation for secure and efficient collaboration.
Built-in Touchscreen Interface
Features a 320×240 color LCD touchscreen for quick access to system functions, diagnostics, and user status monitoring directly on the device.
Secure and Resilient Communication
Provides enterprise-grade security protocols including secure boot, TLS, SRTP, HTTPS, and firewall protection with fail2ban intrusion detection.
Centralized Management and Remote Access
Compatible with centralized device management tools and supports remote provisioning, backup, and monitoring. Offers app-based access via Grandstream’s mobile and desktop applications.
Zero-Touch Endpoint Configuration
Includes automatic provisioning and detection of Grandstream endpoints through ZeroConfig, supporting DHCP Option 66, multicast, and mDNS.
Open-Source Telephony Core
Powered by Asterisk 16 for advanced telephony features such as multi-level IVR, voicemail-to-email, call queues, call detail records, and call routing.
Product Specifications
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Supports up to 500 SIP user extensions
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Handles up to 75 concurrent SIP calls
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1 FXO port for PSTN trunk connection
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1 FXS port for analog phone/fax device
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Built-in support for 75-party audio conferencing
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Video conferencing support: 2 rooms with 12 participants
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3 auto-sensing Gigabit Ethernet ports with PoE+
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320 × 240 color LCD touchscreen display
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1 USB 3.0 port for external storage
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1 SD card slot for backup and call recording
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Audio codecs: Opus, G.711 A/u-law, G.722, G.722.1, G.723.1, G.726-32, G.729A/B, iLBC, GSM
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Video codecs: H.264, H.263, H.263+, H.265, VP8
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T.38 fax over IP support
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Dual firmware image with recovery support
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Secure boot and system authentication
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Built-in NAT router and firewall
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High availability and failover support
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Web-based GUI for management and configuration
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Power input: 100–240 VAC, 50/60Hz
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Power output: 12V DC, 1.5A or via PoE+
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Dimensions: 270 mm × 175 mm × 36 mm
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Weight: 705 grams
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Operating temperature: 0°C to 45°C
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Humidity: 10% to 90% non-condensing
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Wall-mount or desktop installation




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