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Grandstream Audio Series UCM6302A IP PBX Supporting 2x FXO
Grandstream

Grandstream Audio Series UCM6302A IP PBX Supporting 2x FXO

SKU: UCM6302A
$481.95inc GSTRRP $786.50
$438.14 ex GST
14 in stock · ships direct

Specifications

BrandGrandstream
Manufacturer SKUUCM6302A
CategoryPhones & VoIP
Price (ex GST)$438.14
Price (inc GST)$481.95
Availability14 available · drop-ship

<h1 style='color:#3e60ac;text-align:center;font-family:arial;margin: -5px 0px'>On Premise IP PBX (Audio Only) with 2 FXO ports, 2 FXS ports and supporting up to 500 users</h1> <h2 style='color:#46a8ca;text-align:center;font-family:arial;margin: -20px 0px'>UCM6302A</h2><div style='text-align:justify;'> The UCM6300 Audio series enables enterprises to create strong and scalable unified communication and collaboration solutions. This IP PBX Audio series provides a platform that integrates basic business communications needs such as voice, instant messaging (IM), voice meetings, audio web meetings, data, analytics, mobility, facility access, intercoms, and more. The UCM6302A can support up to 500 users and 100 concurrent calls and has a built-in instant messaging (IM), voice/web conferencing platform, and the free Wave App, which allows users to connect and collaborate from desktops, mobile devices, IP phones, and other SIP endpoints Features • Supports up to 500 users and up to 100 concurrent calls • Zero configuration provisioning of Grandstream SIP endpoints • Built-in Instant Messaging, Audio Conferencing & Web Meetings platform supporting access from computers, mobile devices, and SIP endpoints • Wave for Android, iOS, Chrome and Firefox browsers allows communication with all UCM6300 users & solutions • 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 • Supports Full-Band Opus voice codec and H.264/H.263/ H.263+/H.265/VP8 video 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 </div>

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