Grandstream UCM6308A Audio Series IP PBX
Review: 5 - "A masterpiece of literature" by , written on May 4, 20020
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Grandstream UCM6308A Audio Series IP PBX

Available:In Stock
£1,196.40
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Grandstream UCM6308A Audio Series IP PBX

The Grandstream UCM6308A 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 end points. 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 withthe 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 andcollaboration tools, the UCM6300 Audio series provides a powerful business communication platform for any organisation.

Grandstream UCM6308A Key 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
Grandstream UCM6308A - Technical Specifications

Analog Telephone FXS Ports

  • 8 RJ11 ports
  • All ports have lifeline capability in case of power outage

PSTN Line FXO Ports

  • 8 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

  • Power 1/2, FXS, FXO, LAN, WAN,Heartbeat

LCD Display

  • 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 & 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

    • 2x DC 12V Power Jack Input: 100~240VAC, 50/60Hz;Output: DC 12V, 2A

    Dimensions

    • 485mm(L) x 187.2mm(W) x 46.2mm(H)

    Weight

    • 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

    • Rack mount & Desktop

    Multi-Language Support

    • Web UI: English, Simplified Chinese, Traditional Chinese, Spanish, French, Portuguese, German, Russian, Italian, Polish, Czech, Turkish 
    • Customisable IVR/voice prompts: English, Chinese, British English, German, Spanish, Greek, French, Italian, Dutch, Polish, Portuguese, Russian, Swedish, Turkish, Hebrew, Arabic, Nederlands 
    • Customisable 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

    Customisable Auto Attendant

    • Up to 5 layers of IVR (Interactive Voice Response) in multiple languages

    Maximum Call Capacity

    • Users: 1500 
    • Concurrent calls (G.711): 200 
    • Max concurrent SRTP calls (G.711): 150

    Maximum Attendees of Conference Bridges

    • 9 meeting rooms and up to 150 parties

    Wave Mobile App

    • Free; Available for desktop (Windows 10+, Mac OS 10+), web (Firefox and Chrome Browsers) and mobile (Android & iOS), allows usersto 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, callwakeup, 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 centralised 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

    Grandstream UCM6308A Audio Series IP PBX

    The Grandstream UCM6308A 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 end points. 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 withthe 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 andcollaboration tools, the UCM6300 Audio series provides a powerful business communication platform for any organisation.

    Grandstream UCM6308A Key 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
    Grandstream UCM6308A - Technical Specifications

    Analog Telephone FXS Ports

    • 8 RJ11 ports
    • All ports have lifeline capability in case of power outage

    PSTN Line FXO Ports

    • 8 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

    • Power 1/2, FXS, FXO, LAN, WAN,Heartbeat

    LCD Display

    • 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 & 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

      • 2x DC 12V Power Jack Input: 100~240VAC, 50/60Hz;Output: DC 12V, 2A

      Dimensions

      • 485mm(L) x 187.2mm(W) x 46.2mm(H)

      Weight

      • 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

      • Rack mount & Desktop

      Multi-Language Support

      • Web UI: English, Simplified Chinese, Traditional Chinese, Spanish, French, Portuguese, German, Russian, Italian, Polish, Czech, Turkish 
      • Customisable IVR/voice prompts: English, Chinese, British English, German, Spanish, Greek, French, Italian, Dutch, Polish, Portuguese, Russian, Swedish, Turkish, Hebrew, Arabic, Nederlands 
      • Customisable 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

      Customisable Auto Attendant

      • Up to 5 layers of IVR (Interactive Voice Response) in multiple languages

      Maximum Call Capacity

      • Users: 1500 
      • Concurrent calls (G.711): 200 
      • Max concurrent SRTP calls (G.711): 150

      Maximum Attendees of Conference Bridges

      • 9 meeting rooms and up to 150 parties

      Wave Mobile App

      • Free; Available for desktop (Windows 10+, Mac OS 10+), web (Firefox and Chrome Browsers) and mobile (Android & iOS), allows usersto 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, callwakeup, 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 centralised 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