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IP Telephony Solutions (VoIP) > IP PBX Solutions

Introducing a phone system the way it should be!

VoiceConX offers The Open Source PBX based on Asterisk with rich flexible Call Features at a great price (Starting Price is $995) with little or no maintenance. An Asterisk Phone System offers both classical PBX functionality and advanced features, and operates using a traditional standard telephone connection or using SIP Trunks. Using an Asterisk's based phone system, small businesses, Entrepreneurs, and Medical and Law offices are able to use a Phone System with functions similar to a high dollar phone system at a fraction of the cost (Starting Price is $995) and features that are easy to use!

Call Features

  • Automated Attendant
  • Blacklists
  • Blind Transfer
  • Call Detail Records
  • Call Forward on Busy
  • Call Forward on No Answer
  • Call Forward Variable
  • Call Parking
  • Call Queuing
  • Call Recording
  • Call Retrieval
  • Call Routing (DID & ANI)
  • Call Transfer
  • Call Waiting
  • Caller ID
  • Caller ID Blocking
  • Caller ID on Call Waiting
  • Conference Bridging
  • Dial by Name
  • Direct Inward System Access
  • Distinctive Ring
  • Do Not Disturb
  • E911
  • ENUM
  • Fax to email Service Available
  • Flexible Extension Logic
  • Interactive Directory Listing
  • Interactive Voice Response (IVR)
  • Local and Remote Call Agents
  • Music On Hold
  • Music On Transfer
  • Privacy
  • Remote Call Pickup
  • Remote Office Support
  • Roaming Extensions
  • Route by Caller ID
  • Supervised Transfer
  • Three-way Calling
  • Time and Date
  • Trunking
  • VoIP Gateways
  • Voicemail
    - Visual Indicator for Message Waiting
    - Stutter Dialtone for Message Waiting
    - Voicemail to email
    - Voicemail Groups
    - Web Voicemail Interface
    - Computer-Telephony Integration
  • AGI (Asterisk Gateway Interface)
  • Graphical Call Manager
  • Outbound Call Spooling
Scalability
  • TDMoE (Time Division Multiplex over Ethernet)
    - Allows direct connection of Asterisk PBX
    - Zero latency
    - Uses commodity Ethernet hardware
  • Voice-over IP
    - Allows for integration of physically separate installations
    - Uses commonly deployed data connections
    - Allows a unified dialplan across multiple offices
Codecs
  • ADPCM
  • G.711 (A-Law & µ-Law)
  • G.729 (through purchase of commercial license through Digium)
Protocols
  • IAX™ (Inter-Asterisk Exchange)
  • H.323
  • SIP (Session Initiation Protocol)
  • MGCP (Media Gateway Control Protocol
  • SCCP (Cisco® Skinny®)
Traditional Telephony Interoperability
  • E&M
  • E&M Wink
  • Feature Group D
  • FXS
  • FXO
  • GR-303
  • Loopstart
  • Groundstart
  • Kewlstart
  • MF and DTMF support
  • Robbed-bit Signaling (RBS) Types
  • MFC-R2 (Not supported by Digium. However, a patch is available at Voip-info.org)
PRI Protocols
  • 4ESS
  • BRI (ISDN4Linux)
  • DMS100
  • EuroISDN
  • Lucent 5E
  • National ISDN2
  • NFAS

To learn more about how VoiceConX can help your business take advantage of the services we offer, please contact us today.

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