Grandstream UCM6308 IP PBX 6000 Users 400 Concurrent Calls Video Conferencing

𝐏𝐑𝐎𝐃𝐔𝐂𝐓 𝐓𝐈𝐓𝐋𝐄
Grandstream UCM6308 IP PBX 6000 Users 400 Concurrent Calls Video Conferencing

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𝐏𝐑𝐎𝐃𝐔𝐂𝐓 𝐎𝐕𝐄𝐑𝐕𝐈𝐄𝐖
Grandstream UCM6308 enterprise IP PBX supports up to 6000 users and 400 concurrent calls with integrated voice, video conferencing, call center functions, SIP endpoint provisioning, API integration, and GDMS management.

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𝐏𝐑𝐎𝐃𝐔𝐂𝐓 𝐓𝐘𝐏𝐄
IP Phones

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𝐊𝐄𝐘 𝐅𝐄𝐀𝐓𝐔𝐑𝐄𝐒

• Supports up to 6000 users and 400 concurrent calls
• Zero-configuration provisioning of Grandstream SIP endpoints
• Built-in collaboration server with conferencing, chat, screen sharing, whiteboard, and recording
• Supports the Grandstream Wave desktop app
• Built-in call center suite and call queue functionality
• 5-level IVR and multi-language auto-attendant
• Secure boot, unique certificate, and random default password security features
• Three self-adaptive Gigabit network ports with integrated PoE+
• Supports NAT router and switch modes
• Full API available for compatible third-party platform and application integrations, including CRM and PMS
• Compatible with GDMS for cloud-based setup, management, and monitoring
• Supports Hot Standby High-Availability
• Supports Full-Band Opus voice codec
• Supports H.264, H.263, H.263+, H.265, and VP8 video codecs
• Eight analog FXS ports and eight PSTN FXO ports
• 320 x 240 color LCD touchscreen

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𝐓𝐄𝐂𝐇𝐍𝐈𝐂𝐀𝐋 𝐒𝐏𝐄𝐂𝐈𝐅𝐈𝐂𝐀𝐓𝐈𝐎𝐍𝐒

• Brand: Grandstream
• Model: UCM6308
• Maximum Users: 6000
• Maximum Concurrent Calls: 400
• Analog Telephone FXS Ports: 8 RJ11
• PSTN Line FXO Ports: 8 RJ11
• Network Interfaces: 3 self-adaptive Gigabit ports
• Network Modes: Switched, routed, or dual-card mode
• PoE+: Integrated
• NAT Router: Supports router mode and switch mode
• Peripheral Ports: 1 USB 2.0, 1 USB 3.0, 1 SD card interface
• Display: 320 x 240 color LCD touchscreen
• Reset Switch: Long press for factory reset; short press for reboot
• Telephone Operating System: Based on Asterisk version 16
• Voice Codecs: Opus, G.711 A-law/U-law, G.722, G.722.1, G.722.1C, G.723.1 5.3K/6.3K, G.726-32, G.729A/B, iLBC, GSM
• Fax: T.38
• Video Codecs: H.264, H.263, H.263+, H.265, VP8
• DTMF Methods: In-band audio, RFC2833, SIP INFO
• Layer 2 QoS: 802.1Q, 802.1p
• Layer 3 QoS: ToS, DiffServ, MPLS
• API: Full API for third-party platform and application integration
• Provisioning: AES-encrypted XML configuration, auto-discovery, and ZeroConfig auto-provisioning
• Voice Processing: 128ms-tail-length line echo cancellation, dynamic jitter buffer, modem detection with automatic G.711 switching, NetEQ, and FEC 2.0
• Audio Packet Loss Resilience: Up to 50%
• Network Protocols: TCP/UDP/IP, RTP/RTCP, ICMP, ARP, DNS, DDNS, DHCP, NTP, TFTP, SSH, HTTP/HTTPS, PPPoE, STUN, SRTP, TLS, LDAP, HDLC, HDLC ETH, PPP, IPv6, OpenVPN
• Frame Relay: Pending
• Media Encryption: SRTP, TLS, HTTPS, SSH, 802.1X
• Disconnect Methods: Busy/Congestion/Howl Tone, Polarity Reversal, Hook Flash Timing, Loop Current Disconnect

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𝐈𝐍𝐓𝐄𝐍𝐃𝐄𝐃 𝐔𝐒𝐄 / 𝐀𝐏𝐏𝐋𝐈𝐂𝐀𝐓𝐈𝐎𝐍𝐒

• Business voice communications
• Office, reception, and institutional environments
• Integration with compatible SIP/VoIP systems