SKU : UCM6301
Categories : IP-PBX ,  GrandStream , 
Brand : Grandstream
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The UCM6300 series allows businesses to build powerful and scalable unified communication and collaboration solutions. This series of IP PBXs provides 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 web and mobile, which provides a hub for collaborating 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.
Grandstream UCM6301 | |
Analog Telephone FXS Ports | 1 RJ11 Port (All ports have lifeline capability in case of power outage; number of ports can be expanded by peering with an FXS gateway) |
PSTN Line FXO Ports | 1 RJ11 Port (All ports have lifeline capability in case of power outage; number of ports can be expanded by peering with an FXO gateway) |
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 |
LED Indicators | None |
LCD Display | 320x240 color LCD with touch screen for Shortcut Keys and Scroll Bar |
Reset Switch | Yes, long press for factory reset and short press for reboot |
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 |
Video Codecs | H.265, H.264, H.263, H263+, VP8 |
QoS | Layer 2 QoS (802.1Q, 802.1p) and Layer 3 (ToS, DiffServ, MPLS) QoS |
API | Full API available for third-party platform and application integration |
Telephony Operating System | Based on Asterisk version 16 |
DTMF Methods | In-band audio, RFC2833, and SIP INFO |
Provisioning Protocol & Plug-and-Play | Mass provisioning using AES encrypted XML configuration file, auto-discovery & auto-provisioning of Grandstream IP endpoints via ZeroConfig (DHCP Option 66 multicast SIP SUBSCRIBE mDNS), eventlist between local and remote trunk |
Network Protocols | SIP, TCP/UDP/IP, RTP/RTCP, IAX, ICMP, ARP, DNS, DDNS, DHCP, NTP, TFTP, SSH, HTTP/HTTPS, PPPoE, STUN, SRTP, TLS, LDAP, HDLC, HDLC-ETH, PPP, Frame Relay (pending), IPv6, OpenVPN® |
Disconnect Methods | Busy/Congestion/Howl Tone, Polarity Reversal, Hook Flash Timing, Loop Current Disconnect |
Media Encryption | SRTP, TLS, HTTPS, SSH, 802.1X, ZRTP |
(Free shipping for orders of 10,000 baht or more | Product prices include 7% VAT | Special discounts available for project work)