UCM6300 Series – Grandstream Unified Communication IP PBX System for Voice & Video Collaboration, Ideal for Medium to Large Enterprises

Attribute:

SKU : UCM6301

Categories : IP-PBX GrandStream

Share

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.

Key Features

  • Supports up to 3000 users and up to 450 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 for 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*

Grandstream UCM6301
Analog Telephone FXS Ports1 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 Ports1 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 InterfacesThree self-adaptive Gigabit ports (switched, routed or dual mode) with PoE+
NAT RouterYes (supports router mode and switch mode)
Peripheral Ports1*USB 3.0, 1*SD card interface
LED IndicatorsNone
LCD Display320x240 color LCD with touch screen for Shortcut Keys and Scroll Bar
Reset SwitchYes, long press for factory reset and short press for reboot
Voice-over-Packet CapabilitiesLEC 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 CodecsOpus, 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 CodecsH.265, H.264, H.263, H263+, VP8
QoSLayer 2 QoS (802.1Q, 802.1p) and Layer 3 (ToS, DiffServ, MPLS) QoS
APIFull API available for third-party platform and application integration
Telephony Operating SystemBased on Asterisk version 16
DTMF MethodsIn-band audio, RFC2833, and SIP INFO
Provisioning Protocol & Plug-and-PlayMass 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 ProtocolsSIP, 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 MethodsBusy/Congestion/Howl Tone, Polarity Reversal, Hook Flash Timing, Loop Current Disconnect
Media EncryptionSRTP, 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)

Powered by MakeWebEasy.com
This website uses cookies for best user experience, to find out more you can go to our Privacy Policy  and  Cookies Policy