UCM6304A.jpg
UCM6304A.jpg UCM6304A.jpg

Grandstream UCM6304A Audio Series IP PBX

Κωδικός προϊόντος:
UCM6304A
Κατασκευαστής Grandstream
Τιμή πώλησης: 605,00 €(Περιλαμβάνει 24% ΦΠΑ)
Τιμή χωρίς ΦΠΑ: 487,90 €
Διαθεσιμότητα:

Κόστος Μεταφορικών:
  • Περιγραφή
  • Χαρακτηριστικά
  • Λήψεις
  • Γνώμες πελατών
Grandstream UCM6304A Audio Series IP PBX

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.

Features:
• Supports up to 1500 users and up to 200 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

Εγγυήσεις - Επιστροφές:
Επιστροφή Χρημάτων: Επισκευή - Αντικατάσταση:
20 ημέρες 24 μήνες
Specifications
Analog Telephone FXS Ports 4 RJ11 Ports
All ports have lifeline capability in case of power outage
PSTN Line FXO Ports 4 RJ11 Ports
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 2*USB 3.0, 1*SD card interface
LED Indicators None
LCD Display 320x240 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 1x DC 12V Power Jack
Input: 100~240VAC, 50/60Hz;
Output: DC 12V, 2A
Dimensions 270mm(L) x 175mm(W) x 36mm(H)
Weight Unit Weight: 775g;
Package Weight: 1621g
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: 1000
Concurrent calls (G.711): 150
Max concurrent SRTP calls (G.711): 120
Maximum Attendees of Conference Bridges 7 meeting rooms and up to 120 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
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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