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UCM6300A IP PBX

UCM6300A IP PBX

£139.78 GBP
Sale price: £139.78 GBP Regular price: £0.00 GBP
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  • Users: Supports up to 250 users
  • Conference Rooms: 3 meeting rooms
In stock, 8 units
EAN: 6947273703549.00
SKU: UCM6300A
Vendor: Grandstream
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  • Analog Telephone FXS Ports: None
    All ports have lifeline capability in case of power outage
  • PSTN Line FXO Ports: None
    All ports have lifeline capability in case of power outage
  • 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 x USB 3.0, 1 x SD card interface
  • LED Indicators: None
  • LCD Display: 320×240 colour 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
  • 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, RFC4733, 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), event list between local and remote trunk
  • 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, 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
  • Universal Power Supply: Input: 100 ~ 240VAC, 50/60Hz; Output: DC 12V, 1.5A
  • Dimensions: 270mm (L) x 175mm (W) x 36mm (H)
  • Weight: Unit Weight: 705g; Package Weight: 1131g
  • Temperature & Humidity:

    Operating: 32–113ºF / 0–45ºC, Humidity 10–90% (non-condensing)
    Storage: 14–140ºF / -10–60ºC, Humidity 10–90% (non-condensing)
  • Mounting: Wall mount & Desktop
  • Multi-Language Support:
  • UCM6300A IP PBX
  • Quick Installation Guide
  • power supply
  • Ethernet cable

Datasheet


Overview

The UCM6300 Audio series allows businesses to build powerful and scalable unified communication and collaboration solutions. This series of IP PBXs provide a platform that unifies fundamental business communications needs, including voice, instant messaging (IM), voice meetings, audio web meetings, data, analytics, mobility, facility access, intercoms and more. The UCM6300 Audio Series supports up to 1500 users and includes a built-in instant messaging (IM), voice/web conferencing platform, and the free Wave App that allows users to communicate and collaborate from desktops, mobile devices, IP phones, and other SIP endpoints. It supports UCM RemoteConnect cloud service for remote users to offer a best-in-class hybrid platform that combines the control of an on-premise IP PBX with the remote access and system manageability of a cloud solution. By offering a high-end unified communications and collaboration solution packed with a suite of mobility, security, instant messaging, voice, conferencing and collaboration tools, the UCM6300 Audio series provides a powerful business communication platform for any organization

Technical Features
  • Supports up to 1500 users and up to 200 concurrent calls
  • Zero configuration provisioning of Grandstream SIP endpoints
  • Built-in Instant Messaging (IM), Audio Conferencing & Web Meetings platform that supports access from computers, mobile devices, and SIP endpoints
  • Free Wave App allows easy voice & Instant Messaging (IM) communications using desktops, Web, and Android/iOS devices
  • 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 NAT router
  • Automated NAT firewall traversal service facilitates secure remote connections
  • Enhanced reliability with support for Hot Standby High-Availability and local dual deployment
  • Supports Full-Band Opus voice 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