UCM6300A
- UCM6300A hỗ trợ 250 máy lẻ và 50 cuộc gọi đồng thời
- Hỗ trợ lên đến 250 người dùng và lên đến 50 cuộc gọi đồng thời
- Hỗ trợ 3 cổng mạng Gigabit có PoE+.
- Hỗ trợ chức năng Switch và Router cấp DHCP.
- Hỗ trợ Siptrunk kết nối các nhà mạng VNPT, Viettel, FPT trực tiếp qua đường internet/ cáp quang Cung cấp cấu hình không của điểm cuối Grandstream SIP
- Hỗ trợ cấu hình và quản lý thông qua phần mềm quản lý Cloud.
Ms Nhan:
- Kinh Doanh
Mr Minh: - Kinh Doanh
Mr Luân: - Kinh Doanh
WhatsApp: - Tư Vấn Giải Pháp
Facebook: - Hỗ Trợ Đơn Hàng
Sales Engineer: - 0989 544 978
Hotline: 0933 719 613
- Kinh Doanh
Mr Minh:
- Kinh Doanh
Mr Luân:
- Kinh DoanhWhatsApp:
- Tư Vấn Giải PhápFacebook:
- Hỗ Trợ Đơn HàngSales Engineer:
- 0989 544 978×
UCM6300A Audio Series Unified Communication & Collaboration Solution
UCM6300A 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 250 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 Grandstream UCM6300A series provides a powerful business communication platform for any organization.
- Supports up to 250 users and up to 50 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
- Free Wave App allows easy voice & Instant Messaging (IM) communications using desktops, Web, and Android/ iOS devices
- 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
- Enhanced reliability with support for Hot Standby High-Availability 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
- Based on Asterisk* version 16 open source telephony operating system
UCM6300A Datasheet
| Analog Telephone FXS Ports | None All ports have lifeline capability in case of power outage |
|---|---|
| PSTN Line FXO Ports | None All ports have lifeline capability in case of power outage |
| 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 |
| LED Indicators | None |
| LCD Display | 320×240 color LCD with touch screen for Shortcut Keys and Scroll Bar |
| 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 & 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 | 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 |
| Dimensions | 270mm(L) x 175mm(W) x 36mm(H |
| Weight | Unit Weight: 705g; Package Weight: 1131g |
| 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 |
| 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 |
| Maximum Attendees of Conference Bridges | 3 meeting rooms and up to 50 parties |
| 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 |
| 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 |
| Internet Protocol Standards | RFC 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 |
| 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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