UCaaS Platform

As businesses evolve and modernize, transition to remote work environments and
seek to leverage the scalability and reach of cloud-based communication technology, our Unified Communications  platform provides an all-in-one solution that equips organizations of any size with the innovative and reliable features they need.

This page provides an exhaustive breakdown of the platform’s features and capabilities This extended feature list is continuously updated as new features are released and existing features are enhanced — let us know if there’s something you need but can’t find. Click on the feature below to expand the title and learn more!

Hosted PBX (Private Branch Exchange) is the modern version of the traditional telephone closet. It puts the “brains” of a phone closet into professionally maintained data centers, transforms business communications and improves productivity without purchasing expensive, complex phone system software and hardware that depreciates and becomes outdated. Voice over IP (VoIP) refers to cloud-based phone systems that are accessible through a high-quality broadband connection. VoIP is more cost- effective and scalable than traditional phone systems. Our platform’s Hosted PBX & VoIP offers:

Call Conferencing

Conference Bridges
3-Way Conference Call

Call Management

Attended Transfer
Auto Attendant Answering
Automatic Call Distribution (ACD)
Barge
Burstable Virtual Call Paths
Busy Call Forwarding
Call Forwarding
Call Hold
Call Park
Call Queue
Call Recording
Call Routing Time Frames
Call Waiting Indicator
Caller ID
Caller ID Blocking
Caller ID Routing
Cloud Extensions
Direct Inward Dialing (DID)
Direct Inward System Access (DISA)
Directed Call Pickup
Disable Outbound Dialing
Do Not Disturb
Find Me (Digital Assistant)
Forward Calls Locally or Remotely (via Phone or Web)
Incoming Call Blocking
Incoming Call Identification
Incoming Caller ID Routing
Incoming Privacy Screening
Listen Live
Live Person Answering
Multicast Paging
Multicast Paging
No Answer Call Forwarding
Office Intercom
One Button Redial
1-6 Digit Extension Dialing
Outbound Dialing Rules
Outgoing Call Blocking
Premium Call Recording (Add On Subscription Required)
Ring Groups
Shared Virtual Call Paths
Speed Dial
Unattended Transfer
Voicemail
Voicemail Box Monitoring
Visual Voicemail
Voicemail to Text (Add On Subscription Required)

Music On Hold

Commercials on Hold (by Phone Number)
Music on Hold (Custom or Default)

Origination and Termination

Domestic Origination
Domestic Terminations
E911 Support
Endpoint Templates
International Termination
Shared Line Appearance
Toll Free Numbers

Reporting

Activity Reports
Call Detail Records
Call Traffic by Extension
Call Volume Graphs

Virtual Auto Attendants

Multiple Top-level Auto Attendants
Sub-level Auto Attendants
Top-level Auto Attendants (Always On or Time-based)

SIP Trunking

A Session Initiation Protocol (SIP) Trunking solution is a virtual phone line that utilizes a Broadband connection for access, offering up to 40% savings over traditional telephone lines. SIP Trunking enables the ability to plan for peak concurrent call utilization across the enterprise versus on a per-location basis, delivers immediate cost savings and provides a flexible and scalable business communications solution. Features include:

Standard SIP Trunking
Enhanced SIP Trunking

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