KBIT VoIP Phone Solution – Asterisk FreePBX Full Featured Phone System

Brisbane’s VoIP Experts. Upgrade your VoIP Phone System with KBIT. Let Freedom Ring. Contact KBIT Consultant Now – Call us today! +61 7 3108 2911

KBIT Asterisk – Asterisk Open Source Communications Solutions

Feature Rich: Asterisk-based telephony solutions offer a rich and flexible feature set. Asterisk offers both classical PBX functionality and advanced features, and interoperates with traditional standards-based telephony systems and Voice over IP systems. Asterisk offers the advanced features that are often associated with large, high end (and high cost) proprietary PBXs.

 

KBIT FreePBX – Let Freedom Ring

Let Freedom Ring FreePBX is a web-based open source GUI (graphical user interface) that controls and manages Asterisk (PBX), an open source communication server. FreePBX is licensed under the GNU General Public License (GPL), an open source license. FreePBX can be installed manually or as part of the pre-configured FreePBX Distro that includes the system OS, Asterisk, FreePBX GUI and assorted dependencies.

KBIT VoIP Phone Solution

ADSI On-Screen Menu System Alarm Receiver
Append Message Authentication
Automated Attendant Blacklists
Blind Transfer Call Detail Records
Call Forward on Busy Call Forward on No Answer
Call Forward Variable Call Monitoring
Call Parking Call Queuing
Call Recording Call Retrieval
Call Routing (DID & ANI) Call Snooping
Call Transfer Call Waiting
Caller ID Caller ID Blocking
Caller ID on Call Waiting Calling Cards
Conference Bridging Database Store / Retrieve
Database Integration Dial by Name
Direct Inward System Access Distinctive Ring
Distributed Universal Number Discovery (DUNDi™) Do Not Disturb
E911 ENUM
Fax Transmit and Receive Flexible Extension Logic
Interactive Directory Listing Interactive Voice Response (IVR)
Local and Remote Call Agents Macros
Music On Hold Music On Transfer
SMS Messaging Spell / Say
Streaming Hold Music Supervised Transfer
Talk Detection Text-to-Speech (via Festival)
Three-way Calling Time and Date
Transcoding Trunking
VoIP Gateways Voicemail

Google Talk
H.323
IAX™ (Inter-Asterisk eXchange)
Jingle/XMPP
MGCP (Media Gateway Control Protocol
SCCP (Cisco® Skinny®)
SIP (Session Initiation Protocol)
UNIStim

ADPCM
CELT (pass through)
G.711 (A-Law & μ-Law)
G.719 (pass through)
G.722
G.722.1 licensed from Polycom®
G.722.1 Annex C licensed from Polycom®
G.723.1 (pass through)
G.726
G.729a
GSM
iLBC
Linear
LPC-10
Speex
SILK

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