GrandStream UCM6308A

SKU: GS-UCM6308A
DIMENSIONS: 270mm(L) x 175mm(W) x 43mm(H)
MSRP: USD 1499

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.

Analog Telephone FXS Ports8 RJ11 ports
PSTN Line FXO Ports8 RJ11 ports
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 1*USB 2.0, 1*USB 3.0, 1*SD 2*USB 3.0, 1*SD card interface card interface
LED IndicatorsPower 1/2, FXS, FXO, LAN, WAN, Heartbeat
LCD Display128x32 dot matrix graphic LCD with DOWN and OK buttons
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
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, RFC4733, 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 ProtocolsTCP/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 MethodsBusy/Congestion/Howl Tone, Polarity Reversal, Hook Flash Timing, Loop Current Disconnect
Media EncryptionSRTP, TLS, HTTPS, SSH, 802.1X
Universal Power Supply2x DC 12V Power Jack Input: 100~240VAC, 50/60Hz;Output: DC 12V, 2A
Dimensions485mm(L) x 187.2mm(W) x 46.2mm(H)
WeightUnit Weight: 2538g; Package Weight: 3463g
Temperature & HumidityOperating: 32 - 113oF / 0 ~ 45oC, Humidity 10 - 90% (non-condensing) Storage: 14 - 140oF / -10 ~ 60oC, Humidity 10 - 90% (non-condensing)
MountingRack mount & Desktop
Multi-Language SupportWeb 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 -Customizable language pack to support any other languages
Caller IDBellcore/Telcordia, ETSI-FSK, ETSI-DTMF, SIN 227 – BT, NTT
Polarity Reversal/WinkYes, with enable/disable option upon call establishment and termination
Call CenterMultiple configurable call queues, automatic call distribution (ACD) based on agent skills/availability/ workload, in-queue announcement
Customizable Auto AttendantUp to 5 layers of IVR (Interactive Voice Response) in multiple languages
Maximum Call CapacityUsers: 1500 Concurrent calls (G.711): 200 Max concurrent SRTP calls (G.711): 150
Maximum Attendees of Conference Bridges9 meeting rooms and up to 150 parties
Wave AppFree; 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
Call FeaturesCall 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
Firmware UpgradeSupported 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
Internet Protocol StandardsRFC 3261, RFC 3262, RFC 3263, RFC 3264, RFC 3515, RFC 3311, RFC 4028. RFC 2976, RFC 3842, RFC 3892, RFC 3428, RFC 4733, RFC 4566, RFC 2617, RFC 3856, RFC 3711, RFC 5245, RFC 5389, RFC 5766, RFC 6347, RFC 6455, RFC 8860, RFC 4734, RFC 3665, RFC 3323, RFC 3550
ComplianceFCC: 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
Manufacturers Warranty - 24 Months from purchase date with proof of sale. Extended warranty available at extra cost up to a total of 3 years.