Grandstream UCM6308A Audio IP PBX — 1500 Users, 300 Concurrent Calls, 3 Gigabit Ports, PoE+, Audio/Web Conferencing, Wave App, Asterisk-Based UC Platform

The Grandstream UCM6308A is the flagship Audio IP PBX in the UCM6300 Audio Series, supporting up to 1500 users and 300 concurrent calls. Grandstream UCM6308A it is  offers a centralized, on-premise and hybrid cloud-enabled Unified Communication & Collaboration platform for enterprise-grade voice, IM, audio/web conferencing, and mobility.

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    SKU: FGINPG5FC73P Categories: , Brand:
    Description

    The UCM6308A from Grandstream delivers enterprise-level Unified Communication & Collaboration (UCC) in a secure, scalable, and feature-rich IP PBX appliance. Grandstream UCM6308A is designed for larger organizations and growing enterprises, it supports up to 1500 users and 300 concurrent calls, with built-in tools for instant messaging, audio/web conferencing, and mobile/desktop collaboration via the free Wave App.

    This Grandstream UCM6308A integrates voice, instant messaging, conferencing, mobility, facility access, and intercom functionality in one appliance. It offers seamless local and remote user connectivity via the UCM RemoteConnect cloud service without complex VPN or NAT configurations.

    Grandstream UCM6308A comes with Asterisk 16 at its core, the UCM6308A ensures advanced call routing, customizable multi-language IVRs, call queues with skills-based ACD, and robust security features like TLS, SRTP, secure boot, and random default passwords.

    The Grandstream UCM6308A features 3 Gigabit network ports with PoE+, USB 3.0 and SD card interfaces, a color LCD touchscreen interface, and is managed via an intuitive web UI or cloud-based Grandstream Device Management System (GDMS).

    🎯 Key Features:

    • Supports up to 1500 users and 300 concurrent calls

    • 3 Gigabit network ports (switched, routed, or dual mode) with PoE+

    • Integrated NAT router with firewall and automated NAT traversal

    • Built-in audio/web conferencing: 3 conference bridges, 50 participants each

    • Instant Messaging & Wave App for desktop, web, and mobile (Android/iOS)

    • Zero-config provisioning for Grandstream SIP endpoints

    • Advanced security: Secure boot, random passwords, TLS, SRTP, HTTPS, SSH

    • API for third-party integrations (CRM, PMS, etc.)

    • Codec support: Opus, G.711, G.722, G.726, G.729, iLBC, GSM, T.38

    • QoS support at Layer 2 and Layer 3

    • UCM RemoteConnect cloud service for secure, remote access

    • Up to 5-layer multi-language IVR and customizable call routing

    • Call center capabilities: call queues, ACD, in-queue announcements

    • Up to 300 SRTP-encrypted concurrent calls

    • High-Availability Hot Standby support

    • 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
    About Grandstream

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    Explore Our Grandstream Product Range

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    Product Details

    Feature UCM6308A
    Analog Telephone FXS Ports 8 x RJ11 ports (all with lifeline capability in case of power outage)156
    PSTN Line FXO Ports 8 x RJ11 ports (all with lifeline capability in case of power outage)156
    Network Interfaces Three self-adaptive Gigabit ports (switched, routed, or dual mode) with PoE+156
    NAT Router Yes (supports router mode and switch mode)1
    Peripheral Ports 2 x USB 3.0, 1 x SD card interface156
    LED Indicators Power 1/2, FXS, FXO, LAN, WAN, Heartbeat1256
    LCD Display 128x32 dot matrix graphic LCD with DOWN and OK buttons1256
    Reset Switch Yes, long press for factory reset, short press for reboot15
    Voice-over-Packet Capabilities LEC with NLP, 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 loss15
    Voice and Fax Codecs Opus, G.711 A-law/U-law, G.722, G.722.1, G.722.1C, G.723.1 (5.3K/6.3K), G.726-32, G.729A/B, iLBC, GSM; T.38157
    QoS Layer 2 QoS (802.1Q, 802.1p) and Layer 3 (ToS, DiffServ, MPLS) QoS15
    API Full API available for third-party platform and application integration2
    Telephony Operating System Based on Asterisk version 1623
    DTMF Methods In-band audio, RFC4733, SIP INFO1
    Provisioning Protocols & Plug-and-Play Mass provisioning via AES encrypted XML, auto-discovery & auto-provisioning of Grandstream IP endpoints via ZeroConfig (DHCP Option 66, multicast SIP SUBSCRIBE, mDNS), eventlist between local and remote trunk12
    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®12
    Disconnect Methods Busy/Congestion/Howl Tone, Polarity Reversal, Hook Flash Timing, Loop Current Disconnect1
    Media Encryption SRTP, TLS, HTTPS, SSH, 802.1X12
    Power Supply Input: 100–240VAC, 50/60Hz; Output: DC 12V, 2A56
    Dimensions 485mm (L) x 187.2mm (W) x 46.2mm (H)57
    Weight Unit: 2538g; Package: 3463g5
    Temperature & Humidity Operating: 0–45ºC, 10–90% (non-condensing); Storage: -10–60ºC, 10–90% (non-condensing)5
    Mounting Rack mount & desktop57
    Multi-Language Support Web UI: English, Simplified/Traditional Chinese, Spanish, French, Portuguese, German, Russian, Italian, Polish, Czech, Turkish Customizable IVR: English, Chinese, British English, German, Spanish, Greek, French, Italian, Dutch, Polish, Portuguese, Russian, Swedish, Turkish, Hebrew, Arabic, Nederlands1
    Caller ID Bellcore/Telcordia, ETSI-FSK, ETSI-DTMF, SIN 227 – BT, NTT1
    Polarity Reversal/Wink Yes, with enable/disable option1
    Call Center Multiple configurable call queues, ACD based on agent skills/availability/workload, in-queue announcement6
    Customizable Auto Attendant Up to 5 layers of IVR in multiple languages26
    Maximum Call Capacity Up to 1500 users, 200 concurrent calls (G.711), 150 max concurrent SRTP calls246
    Conference Bridges 9 meeting rooms, up to 150 parties246
    Wave App Free app for desktop, web, and mobile (Android/iOS) for meetings and SIP calls24
    Call Features Call park, forward, transfer, waiting, caller ID, record, history, ringtone, IVR, music on hold, call routes, DID, DOD, DND, DISA, ring group, time schedule, PIN groups, call queue, pickup group, paging/intercom, voicemail, SCA, BLF, voicemail to email, fax to email, speed dial, call back, dial by name, emergency call, follow me, blacklist/whitelist, voice meeting, eventlist, feature codes, busy camp-on/call completion, voice control12
    Firmware Upgrade Supported by Grandstream Device Management System (GDMS) for centralized provisioning and management23
    Compliance FCC, CE, IC, RCM, UL, RoHS

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