www.grandstream.com UCM6300A UCM6302A UCM6304A UCM6308A Analog Telephone FXS Ports 2 RJ11 ports 4 RJ11 ports 8 RJ11 ports All ports have lifeline capability in case of power outage
Grandstream UCM6304A IP PBX System Ghana|Grandstream UCM6304A
Grandstream UCM6308A IP PBX System Ghana|Grandstream UCM6308A
Unified Communication & Collaboration Solution UCM6300 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 API available for third-party integrations, including CRM and PMS platforms 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 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 Free Wave App allows easy voice & Instant Messaging (IM) communications using desktops, Web, and Android/ iOS devices UCM RemoteConnect Automated NAT firewall traversal service facilitates secure remote connections Enhanced reliability with support for Hot Standby HighAvailability 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 www.grandstream.com Based on Asterisk* version 16 open source telephony operating system UCM6300A UCM6302A UCM6304A UCM6308A None 2 RJ11 ports Analog Telephone FXS Ports All ports have lifeline capability in case of power outage 4 RJ11 ports 8 RJ11 ports None 2 RJ11 ports PSTN Line FXO Ports All ports have lifeline capability in case of power outage 4 RJ11 ports 8 RJ11 ports 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*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 LED Indicators None Power 1/2, FXS, FXO, LAN, WAN, Heartbeat LCD Display 320x240 color LCD with touch screen for Shortcut Keys and Scroll Bar 128x32 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 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 & Mass provisioning using AES encrypted XML configuration file, auto-discovery & auto-provisioning of Grandstream IP endpoints via Plug-and-Play ZeroConfig (DHCP Option 66 multicast SIP SUBSCRIBE mDNS), eventlist 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 2x DC 12V Power Jack Input: 100~240VAC, 50/60Hz;Output: DC 12V, 2A Dimensions 270mm(L) x 175mm(W) x 36mm(H) 485mm(L) x 187.2mm(W) x 46.2mm(H) 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 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 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/ workload, in-queue announcement Customizable Auto Attendant Up to 5 layers of IVR (Interactive Voice Response) in multiple languages 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 Maximum Attendees of 3 meeting rooms and up to 50 Conference Bridges parties 5 meeting rooms and up to 75 parties 7 meeting rooms and up to 120 9 meeting rooms and up to 150 parties parties Free; Available for desktop (Windows 10+, Mac OS 10+), web (Firefox and Chrome Browsers) and mobile (Android & iOS), allows users Wave App 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 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 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, 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 *Asterisk is a Registered Trademark of Digium, Inc. www.grandstream.com 5.2021.033-Heights(TM) PDF Optimization Shell 5.9.1.5 (http://www -tools.com)