Grandstream Audio IP PBX voice system support 4 FXO Port , 4 FXS Port

DescriptionThe 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.FeaturesSupports up to 1500 users and up to 200 concurrent callsZero configuration provisioning of Grandstream SIP endpointsBuilt-in Instant Messaging (IM), Audio Conferencing & Web Meetings platform that supports                                                                            access from computers, mobile devices, and SIP endpointsFree Wave App allows easy voice & Instant Messaging (IM) communications using desktops,   Web, and Android/ iOS devicesAPI available for third-party integrations, including CRM and PMS platformsAdvanced security protection with secure boot, unique certificate and random default password  to protect calls and accountsThree Gigabit auto-sensing RJ45 network ports with integrated PoE and support NAT routerAutomated NAT firewall traversal service facilitates secure remote connectionsEnhanced reliability with support for Hot Standby High-Availability and local dual deploymentSupports Full-Band Opus voice codec, jitter resilience up to 50% packet lossCompatible with GDMS for cloud setup, management, and monitoringBased on Asterisk* version 16 open source telephony operating systemDatasheet Show More Show Less