FiberMe IP PBX Audio Series FCM630A

A free of license powerful audio unified communication & collaboration solution for any organization, the FCM630A Audio series provides a high-end unified communications solution packed with an ecosystem of mobility, security, voice and collaboration tools. Enterprise features come with a free license to meet all your business need and develop your unified communication and collaboration.
Manufacturer: FIBERME
Old price: 12750.00 LE
12000.00 LE


The FCM630A 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 FCM630A Audio Series supports up to 250 users and includes a built-in instant messaging (IM), voice/web conferencing platform, IP phones, and other SIP endpoints. 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 FCM630A Audio series provides a powerful business communication platform for any organization.

Main Features

  • Supports up to 250 users and up to 50 concurrent calls
  • Zero configuration provisioning of FIBERME FAP SIP endpoints
  • Built-in Instant Messaging (IM), Audio Conferencing & Web Meetings platform that supports access from computers, mobile devices, and SIP endpoints
  • 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
  • Based on Asterisk* version 16 open source telephony operating system

Advanced Features

  • Office Time.
  • Voicemail.
  • Voicemail to Email.
  • Concurrent Regiestration.
  • Call Forward.
  • Call Follow-me.
  • Calling preverage per extension.
  • Call Pickup Group.
  • Call Ring Group.
  • Advanced incoming call routing.
  • Advanced outgoing call routing.
  • Paging and Intercom.
  • DISA.
  • Speed Dial.
  • Call Back.
  • Fax Server.
  • Fax to Email.
  • Email to Fax.
  • API Support.
  • AMI Support.
  • CRM/PMS Integration
  • Audio Conference Bridges.
  • Dial by Name.
  • Announcements.
  • LDAP Integration.
  • Static Defense.
  • Dynamic Defense.
  • Fail2Ban.
  • High Avilability.
  • User Groups.
  • User Portal.
  • Schedule Backup.
  • System Cleanup.
  • Zero Config and Auto Provission.
  • QueueMetrics Integration.
  • Call Center Features.
    • IVR.
    • Call Queue.
    • Verviual Queue.
    • Queue Announcements.
    • Custom Music On Hold.
    • Call Recoring.
    • Call Reporting.
    • Live Call Monitoring.
    • Queue Managers.
    • Call Statistics.
    • Call Spy.
    • Queue Login & Logout.