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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.
· Supports 250-1500 SIP users and 50-150 concurrent calls
· Zero configuration provisioning of Grandstream SIP endpoints
· 3 Gigabit RJ45 network ports with integrated PoE+ and support NAT router
· Automated NAT firewall traversal service
· Compatible with GDMS for cloud setup, management, and monitoring
· Based on Asterisk* version 16 open source telephony operating system
· Built-in Instant Messaging (IM), Audio Conferencing & Web Meetings platform that supports access from computers, mobile devices, and SIP endpoints
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| Models optional: |
UCM6300A : Support Max 250 SIP Users ,Max 50 Concurrent calls Max 50 concurrent SRTP calls ,200 SIP trunk Support 3 meeting rooms and up to 50 parties Without FXO/FXS port , 1*USB 3.0, 1*SD card interface | UCM6302A: Support Max 500 SIP Users ,Max 75 Concurrent calls Max 75 concurrent SRTP calls ,200 SIP trunk, Support 5 meeting rooms and up to 75 parties With 2FXO+2FXS port , 1*USB 2.0, 1*USB 3.0, 1*SDcard interface |
UCM6304A: Support Max 1000 SIP Users ,Max 150 Concurrent calls Max 120 concurrent SRTP calls ,200 SIP trunk Support 7 meeting rooms and up to 120parties With 4FXO+4FXS port , 2*USB 3.0, 1*SD card interface | UCM6308A: Support Max 1500 SIP Users ,Max 200 Concurrent calls Max 150 concurrent SRTP calls ,200 SIP trunk Support 9 meeting rooms and up to 150parties With 8FXO+8FXS port , 2*USB 3.0, 1*SD card interface |
UCM6300 Audio Series
Unified Communication & Collaboration Solution
This series of IP PBXs provide a platform that unifies all business communication on one centralized network, including voice,
instant messaging (IM), voice meetings, audio web meetings, data, analytics, mobility, facility access, intercoms and more.
· 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
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