Unified Communication & Collaboration Solution UCM6300A Audio Series
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Description
Unified Communication & Collaboration Solution
UCM6300A Audio Series
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.
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
(IM) communications using desktops, Web, and Android/ iOS devices
and PMS platforms
certificate and random default password to protect calls and accounts
PoE+ and support NAT router
secure remote connections
Enhanced reliability with support for Hot Standby High- Availability and local dual deploymen
Supports Full-Band Opus voice codec,jitter resilience up to 50% packet loss
monitoring
operating system
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UCM6300A |
UCM6302A |
UCM6304A |
UCM6308A |
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Analog Telephone FXS Ports |
None |
2 RJ11 ports |
4 RJ11 ports |
8 RJ11 ports |
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All ports have lifeline capability in case of power outage |
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PSTN Line FXO Ports |
None |
2 RJ11 ports |
4 RJ11 ports |
8 RJ11 ports |
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All ports have lifeline capability in case of power outage |
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Network Interfaces |
Three self-adaptive Gigabit ports (switched, routed or dual mode) with PoE+ |
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NAT Router |
Yes (supports router mode and switch mode) |
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Peripheral Ports |
1*USB 3.0, 1*SD card interface |
1*USB 2.0, 1*USB 3.0, 1*SD card interface |
2*USB 3.0, 1*SD card interface |
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LED Indicators |
None |
Power 1/2, FXS, FXO, LAN, WAN, Heartbeat |
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LCD Display |
320×240 color LCD with touch screen for Shortcut Keys and Scroll Bar |
128×32 dot matrix graphic LCD with DOWN and OK buttons |
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Reset Switch |
Yes, long press for factory reset and short press for reboot |
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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 |
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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 |
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QoS |
Layer 2 QoS (802.1Q, 802.1p) and Layer 3 (ToS, DiffServ, MPLS) QoS |
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API |
Full API available for third-party platform and application integration |
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Telephony Operating System |
Based on Asterisk version 16 |
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DTMF Methods |
In-band audio, RFC4733, and SIP INFO |
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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 |
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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® |
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Disconnect Methods |
Busy/Congestion/Howl Tone, Polarity Reversal, Hook Flash Timing, Loop Current Disconnect |
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Media Encryption |
SRTP, TLS, HTTPS, SSH, 802.1X |
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Universal Power Supply |
Input: 100 ~ 240VAC, 50/60Hz; Output: DC 12V, 1.5A |
1x DC 12V Power Jack Input: 100~240VAC, 50/60Hz;Output: DC 12V, 2A |
2x DC 12V Power Jack Input: 100~240VAC, 50/60Hz;Output: DC 12V, 2A |
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Dimensions |
270mm(L) x 175mm(W) x 36mm(H) |
485mm(L) x 187.2mm(W) x 46.2mm(H) |
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Weight |
Unit Weight: 705g; Package Weight: 1131g |
Unit Weight: 725g; Package Weight: 1221g |
Unit Weight: 775g; Package Weight: 1621g |
Unit Weight: 2538g; Package Weight: 3463g |
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Temperature & Humidity |
Operating: 32 – 113ºF / 0 ~ 45ºC, Humidity 10 – 90% (non-condensing) Storage: 14 – 140ºF / -10 ~ 60ºC, Humidity 10 – 90% (non-condensing) |
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Mounting |
Wall mount & Desktop |
Rack mount & Desktop |
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Multi-Language Support |
-Web UI: English, Simplified Chinese, Traditional Chinese, Spanish, French, Portuguese, German, Russian, Italian, Polish, Czech, Turkish -Customizable IVR/voice prompts: English, Chinese, British English, German, Spanish, Greek, French, Italian, Dutch, Polish, Portuguese, Russian, Swedish, Turkish, Hebrew, Arabic, Nederlands -Customizable language pack to support any other languages |
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Caller ID |
Bellcore/Telcordia, ETSI-FSK, ETSI-DTMF, SIN 227 – BT, NTT |
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Polarity Reversal/Wink |
Yes, with enable/disable option upon call establishment and termination |
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Call Center |
Multiple configurable call queues, automatic call distribution (ACD) based on agent skills/availability/ workload, in-queue announcement |
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Customizable Auto Attendant |
Up to 5 layers of IVR (Interactive Voice Response) in multiple languages |
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Maximum Call Capacity |
Users: 250 Concurrent calls (G.711): 50 Max concurrent SRTP calls (G.711): 50 |
Users: 500 Concurrent calls (G.711): 75 Max concurrent SRTP calls (G.711): 75 |
Users: 1000 Concurrent calls (G.711): 150 Max concurrent SRTP calls (G.711): 120 |
Users: 1500 Concurrent calls (G.711): 200 Max concurrent SRTP calls (G.711): 150 |
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Maximum Attendees of Conference Bridges |
3 meeting rooms and up to 50 parties |
5 meeting rooms and up to 75 parties |
7 meeting rooms and up to 120 parties |
9 meeting rooms and up to 150 parties |
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Wave App |
Free; Available for desktop (Windows 10+, Mac OS 10+), web (Firefox and Chrome Browsers) and mobile (Android & iOS), allows users to join UCM-hosted meetings, communicate with other users/solutions and make/receive calls using SIP accounts registered to a UCM6300 Audio series IP PBX |
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Call Features |
Call park, call forward, call transfer, call waiting, caller ID, call record, call history, ringtone, IVR, music on hold, call routes, DID, DOD, DND, DISA, ring group, ring simultaneously, time schedule, PIN groups, call queue, pickup group, paging/intercom, voicemail, call wakeup, SCA, BLF, voicemail to email, fax to email, speed dial, call back, dial by name, emergency call, call follow me, blacklist/whitelist, voice meeting, eventlist, feature codes, busy camp-on/ call completion, voice control |
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Firmware Upgrade |
Supported by Grandstream Device Management System (GDMS), a zero-touch cloud provisioning and management system, It provides a centralized interface to provision, manage, monitor and troubleshoot Grandstream products |
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Compliance |
FCC: Part 15 (CFR 47) Class B, Part 68 CE: EN 55032, EN 55035, EN 61000-3-2, EN 61000-3-3, EN 62368.1, ES 203 021, ITU-T K.21 IC: ICES-003, CS-03 Part I Issue 9 RCM: AS/NZS CISPR 32, AS/NZS 62368.1, AS/CA S002, AS/CA S003.1/.2 Power adapter: UL 60950-1 or UL 62368-1 |
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