- 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 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
Specs:
Analog Telephone FXS Ports: 4 RJ11 – All ports have lifeline capability in case of power outage; the number of ports can be expanded by peering with an FXS gateway
PSTN Line FXO Ports: 4 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: 2*USB 3.0, 1*SD card interface
LED Indicators: Power 1/2, FXS, FXO, LAN, WAN, Heartbeat
LCD Display: 128×32 dot matrix graphic LCD with DOWN and OK buttons
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.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
Universal Power Supply Input: 2x DC 12V Power Jack Input: 100~240VAC, 50/60Hz;Output: DC 12V, 2A
Dimensions: 485mm(L) x 187.2mm(W) x 46.2mm(H)
Weight: Unit Weight: 2490g; Package Weight: 3260g
Temperature & Humidity Operating: 32 – 113ºF / 0 ~ 45ºC, Humidity 10 – 90% (non-condensing) Storage: 14 – 140ºF / -10 ~ 60ºC, Humidity 10 – 90% (non-condensing)
Mounting: Rack mount & Desktop
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
Caller ID: Bellcore/Telcordia, ETSI-FSK, ETSI-DTMF, SIN 227 – BT, NTT
Polarity Reversal/Wink: Yes, with enable/disable option upon call establishment and termination
Call Center: Multiple configurable call queues, automatic call distribution (ACD) based on agent skills/availability/ work-load, in-queue announcement
Customizable Auto Attendant: Up to 5 layers of IVR (Interactive Voice Response) in multiple languages
Maximum Call Capacity: Users: 2000 Concurrent calls (G.711): 300 Max concurrent SRTP calls (G.711): 200
Maximum Attendees of Conference Bridges: 8 Video Conference rooms and
up to 60 parties with 1080p, assuming 4 video feeds + 1 screen sharing (H.264 & Opus) Voice Conference: Up to 200 parties (G.711)
Wave App: Free; Available for desktop (Windows 10+, Mac OS 10+), web (Firefox and Chrome Browsers) and mobile (Android & iOS), allows users to join UCMhosted meetings/conferences, communicate with other users/solutions and make/receive calls using SIP accounts registered to a UCM6300 series IP PBX
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 conference, video conference, eventlist, feature codes, busy camp-on/ call completion, voice control, post-meeting reports, virtual fax sending/receiving, email to fax
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
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, ETSI 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
