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GRANDSTREAM UCM6300A IP AUDIO SERIES PBX

GRANDSTREAM UCM6300A IP AUDIO SERIES PBX
GRANDSTREAM UCM6300A IP AUDIO SERIES PBX
  • Price: 75,000৳
  • Regular Price: 80,250৳
  • Brand: Grandstream
  • Product ID : 21478
  • Model: UCM6300A
  • Key Features

    • Color: Black
    • Based on Asterisk version 16
    • In-band audio, RFC4733, and SIP INFO
    • 1*USB 3.0, 1*SD card interface
    Main Features
    Protocols/StandardsNetwork 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®
    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
    LanguagesMulti-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
    DisplayLCD Display: 320×240 color LCD with touch screen for Shortcut Keys and Scroll Bar
    Voice Codecs and CapabilitiesVoice-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
    API I 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
    PortsPeripheral Ports: 1*USB 3.0, 1*SD card interface
    QoSLayer 2 QoS (802.1Q, 802.1p) and Layer 3 (ToS, DiffServ, MPLS) QoS
    TemperatureTemperature & Humidity:
    Operating: 32 – 113ºF / 0 ~ 45ºC
    Humidity 10 – 90% (non-condensing)
    Storage: 14 – 140ºF / -10 ~ 60ºC
    Humidity 10 – 90% (non-condensing)
    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
    Telephony Features
    Wall MountingWall mount & Desktop
    Physical Specification
    Dimensionss 270mm(L) x 175mm(W) x 36mm(H)
    WeightUnit Weight: 705g;
    Package Weight: 1131g
    interfaceNetwork Interfaces: s Three self-adaptive Gigabit ports (switched, routed or dual mode) with PoE+
    Other Features
    OthersNAT Route: Yes (supports router mode and switch mode)
    Reset Switch: Yes, long press for factory reset and short press for reboot
    isconnect 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
    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
    Description

    GRANDSTREAM UCM6300A IP AUDIO SERIES PBX

    The GRANDSTREAM UCM6300A IP Audio Series PBX is a robust and feature-rich communication solution designed to meet the diverse needs of modern businesses. With its advanced telephony features and versatile protocols, it offers seamless integration into various network environments. The device supports a wide range of network protocols, ensuring compatibility and reliability in data transmission. Its multi-language support and customizable IVR/voice prompts enable businesses to cater to diverse customer bases with ease. The intuitive LCD display with touch screen functionality enhances user experience by providing quick access to shortcut keys and scroll bars. The UCM6300A boasts superior voice codecs and capabilities, ensuring high-quality audio transmission even in challenging network conditions. Its full API support allows for seamless integration with third-party platforms and applications, expanding its functionality and adaptability. With mass provisioning capabilities and plug-and-play functionality, deployment and management of Grandstream IP endpoints become efficient and hassle-free. The device also offers robust security features such as media encryption and universal power supply, ensuring data integrity and reliability. In terms of call management, the UCM6300A provides a comprehensive suite of features including call park, call transfer, call waiting, caller ID, call recording, and more. It supports up to 250 users and 50 concurrent calls, making it suitable for medium to large-scale enterprises. Furthermore, the device offers firmware upgrade support through the Grandstream Device Management System (GDMS), enabling centralized provisioning, management, monitoring, and troubleshooting of Grandstream products. Overall, the GRANDSTREAM UCM6300A IP Audio Series PBX combines advanced functionality, reliability, and scalability to deliver a comprehensive communication solution for businesses seeking enhanced productivity and efficiency in their operations.

    GRANDSTREAM UCM6300A IP AUDIO SERIES PBX FEATURES

    • Telephony Features: The UCM6300A supports both wall mount and desktop installation options, offering flexibility in deployment.
    • Protocols/Standards: It adheres to a wide array of network protocols and internet standards, ensuring seamless communication and interoperability across networks.
    • Multi-Language Support: With support for various languages in the web UI and customizable IVR/voice prompts, businesses can cater to diverse linguistic preferences and customer bases.
    • LCD Display: The device features a 320×240 color LCD with touch screen functionality, facilitating easy navigation and access to shortcut keys and scroll bars.
    • Voice Codecs and Capabilities: It incorporates advanced voice-over-packet capabilities, ensuring high-quality audio transmission and supporting a wide range of voice and fax codecs.
    • Telephony Operating System: Based on Asterisk version 16, the system provides a stable and feature-rich telephony environment.
    • Provisioning Protocol & Plug-and-Play: The UCM6300A supports mass provisioning using encrypted XML configuration files and offers auto-discovery and auto-provisioning of Grandstream IP endpoints, simplifying setup and management.
    • QoS (Quality of Service): It provides Layer 2 and Layer 3 QoS support, ensuring optimal performance and prioritization of voice traffic.
    • Temperature & Humidity: The device operates within specified temperature and humidity ranges, ensuring reliability and longevity even in challenging environmental conditions.
    • Compliance: It meets various regulatory standards such as FCC, CE, IC, and RCM, ensuring compliance with international requirements.
    • Physical Specification: The compact design and lightweight construction make it suitable for various installation environments.
    • Other Features: Additional features include NAT routing, media encryption, universal power supply, caller ID support, call center capabilities, and integration with the Wave App for seamless communication across devices and platforms.

    These features collectively make the GRANDSTREAM UCM6300A IP Audio Series PBX a comprehensive and reliable communication solution for businesses of all sizes.

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