πππππππ πππππGrandstream UCM6308 IP PBX 6000 Users 400 Concurrent Calls Video Conferencing βββββββββββββββββββββββββββββββββββββββββββββββββββββπππππππ ππππππππGrandstream 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. βββββββββββββββββββββββββββββββββββββββββββββββββββββπππππππ ππππIP Phones βββββββββββββββββββββββββββββββββββββββββββββββββββββπππ π πππππππ 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 βββββββββββββββββββββββββββββββββββββββββββββββββββββπππππππππ ππππππ ππππππππ 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 βββββββββββββββββββββββββββββββββββββββββββββββββββββππππππππ πππ / ππππππππππππ Business voice communications Office, reception, and institutional environments Integration with compatible SIP/VoIP systems
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