Fiberme FCM630A VoIP PBX

Fiberme FCM630A VoIP PBX

Description

FIBERME FCM630A VoIP PBX

A free office 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 developy our unified communication and collaboration.

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 AudioSeries 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.

MainFeatures

• Supports up to 250 users and up to 50 concurrent calls
• Zero configuration provisioning of FIBERME FAP SIP endpoints
• Built-inInstant 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
• ThreeGigabit 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
• SupportsFull-Band Opus voice codec, jitter resilience up to 50% packet loss
• Based onAsterisk* version 16 open source telephony operating system

AdvancedFeatures

• Office Time.
• Voicemail.
• Voicemail toEmail.
• Concurrent Registration.
• Call Forward.
• CallFollow-me.
• Calling peerageper extension.
• Call PickupGroup.
• Call RingGroup.
• Advanced incoming call routing.
• Advanced outgoing call routing.
• Paging andIntercom.
• DISA.
• Speed Dial.
• Call Back.
•  Fax Server.
• Fax toEmail.
• Email toFax.
• API Support.
• AMI Support.
• CRM/PMSIntegration
•  Audio Conference Bridges.
• Dial byName.
• Announcements.
• LDAPIntegration.
• StaticDefense.
• DynamicDefense.
• Fail2Ban.
•  High Availability.
• User Groups.
• User Portal.
• ScheduleBackup.
• SystemCleanup.
• Zero Config and Auto Provision.
• QueueMetricsIntegration.
• Call CenterFeatures.
• IVR.
• Call Queue.
• Verviual Queue.
• QueueAnnouncements.
• Custom MusicOn Hold.
• Call Recording.
• CallReporting.
• Live CallMonitoring.
• QueueManagers.
• CallStatistics.
• Call Spy.
• Queue Login& Logout.