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Grandstream UCM6308A Audio Series IP PBX

Grandstream

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Key Features

  • Up to 1500 users
  • Up to 200 concurrent calls
  • 9 meeting rooms

  Grandstream UCM6300 Audio Series Datasheet (PDF)
  Grandstream UCM6304/8 & UCM6308A Quick Installation Guide (PDF)
  Grandstream UCM630xA Series User Manual (PDF)

$810.59

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$2,140.57 for 3 item(s)

Grandstream UCM6308A Audio Series IP PBX

Grandstream UCM6308A Audio Series IP PBX

The Grandstream UCM6308A IP PBX offers businesses a scalable solution for unified communications and collaboration. This IP PBX series enables integration of all essential communication channels—voice, video calls, video conferencing, video surveillance, network conferencing, data transmission, analysis, mobility, facility access, and more—into a single, centralised network.

With support for up to 1,500 users, the UCM6300 Audio series includes built-in web and video conferencing capabilities, allowing employees to connect seamlessly via desktop computers, mobile devices, GVC series devices, and IP phones. For added flexibility, it integrates with the UCM6300 ecosystem, providing a hybrid solution that combines local IP PBX management with secure, cloud-based remote access.

Key Features of the Grandstream UCM6308A Audio Series IP PBX:

  • Supports up to 200 concurrent calls and 1,500 users
  • Zero-configuration setup for Grandstream SIP endpoints
  • Built-in conferencing platform with support for Wave app, desktop, and SIP endpoints
  • Based on open-source Asterisk Version 16 telephony operating system
  • Fully compatible with GDMS for cloud-based configuration, management, and monitoring
  • Three gigabit auto-sensing RJ45 network ports with PoE+ support and integrated NAT router
  • Automated NAT firewall traversal service for secure remote access
  • Advanced security features, including secure boot, unique certificates, and random default passwords for enhanced call and account security
  • Wave app available for Android, iOS, Chrome, and Firefox, enabling easy communication with all UCM6300 users and solutions
  • API availability for integration with third-party platforms, such as CRM and PMS

The UCM6308A series is an ideal choice for enterprises seeking a powerful, versatile platform with comprehensive communication, security, and collaboration capabilities.

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

  • 128×32 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), event list 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
  • 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: 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 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, event list, 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
  • 8 RJ11 Port
  • All ports have lifeline capability in case of power outage

PSTN Line FXO Ports

  • 8 RJ11 Port
  • 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

  • 128×32 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

Video Codecs

  • H.264, H.263, H263+, H.265, VP8

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, RFC2833, 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: 2550g
  • Package Weight: 3320g

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
  • Customizable IVR/voice prompts: English, Chinese, British English, German, Spanish, Greek, French, Italian, Dutch, Polish, Portuguese, Russian, Swedish, Turkish, Hebrew, Arabic, Netherlands
  • 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/ work-load, in-queue announcement

Customizable Auto Attendant

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

Maximum Call Capacity

  • Users: 3000
  • Concurrent calls (G.711): 450
  • Max concurrent SRTP calls (G.711): 300

Maximum Attendees Of Conference Bridges

  • 8 Video Conference rooms and up to 60 parties with 1080p, assuming 4 video feeds + 1 screen sharing (H.264 & G.711)
  • Voice Conference: Up to 300 parties (G.711)

Wave Mobile App

  • Allows Android & iOS users to join UCM-hosted meetings & communicate with other users/solutions registered to the UCM6300

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 conference, video conference, 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, ETSI 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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