UCM6300 Series IP PBX | Unified Communications up to 3000 Users | Grandstream

Asterisk version 16, the Grandstream UCM6300 Series is a powerful and scalable audio‑focused unified communication platform. It centralizes business voice communication, including traditional telephony, VoIP calls, audio conferencing, voicemail, fax, intercoms, and call center features onto a single network.

Designed for pure audio telephony, the UCM6300 series supports up to 3000 users and up to 450 concurrent G.711 audio calls (UCM6308 model), with maximum concurrent SRTP encrypted audio calls reaching 300 on the same model.

Audio Codecs Supported:

• Opus (Full‑Band)

• 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

• T.38 for fax

Audio Quality & Reliability:

• LEC with NLP packetized voice protocol unit

• 128‑ms 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

Analog Audio Connectivity (FXS & FXO ports):

• UCM6301: 1 FXS port + 1 FXO port (RJ11)

• UCM6302: 2 FXS + 2 FXO

• UCM6304: 4 FXS + 4 FXO

• UCM6308: 8 FXS + 8 FXO

• All ports have lifeline capability during power outage

• Number of ports can be expanded by peering with FXS/FXO gateways

Audio Call Features (over 40 included):
Call park, call forward, call transfer, call waiting, caller ID, call record, call history, ringtone, IVR (up to 5 layers), music on hold, call routes, DID, DOD, DND, DISA, ring groups, simultaneous ring, time schedules, PIN groups, call queues, pickup groups, 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, audio conference (unlimited meeting rooms – up to 25 simultaneous public rooms on UCM6308, with up to 300 simultaneous participants combined in audio‑only mode), eventlist, feature codes, busy camp‑on/call completion, voice control, post‑meeting reports, virtual fax sending/receiving, email‑to‑fax.

Call Center Audio Features:
Multiple configurable call queues, automatic call distribution (ACD) based on agent skills/availability/workload, in‑queue announcements.

Audio Disconnect Methods:
Busy/congestion/howl tone, polarity reversal, hook flash timing, loop current disconnect.

DTMF Methods:
In‑band audio, RFC2833, and SIP INFO.

Audio Provisioning & 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.

Security for Audio:
SRTP, TLS, HTTPS, SSH, 802.1X, ZRTP for media encryption. Advanced security protection with secure boot, unique certificate, and random default password.

Hardware (Audio‑focused):

• Three self‑adaptive Gigabit ports with PoE+ and NAT router support

• Peripheral ports: USB 2.0, USB 3.0, SD card interface

• 320x240 color LCD with touch screen

• Reset switch (long press factory reset, short press reboot)

• Wall mount or desktop mounting

Audio Operating Environment:
Operating temperature: 32‑113°F / 0‑45°C, humidity 10‑90% (non‑condensing)

Compliance (Audio telephony standards):
FCC Part 68, CE (EN 55032, EN 55035), IC (CS‑003), RCM (AS/CA 5002, AS/CA 5003.1.2), Power adapter UL 60950‑1 or UL 62368‑1.

Audio 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®.