SIPivr 6800GS
互動式語音伺服器
  • Up-to 120 Universal VOIP Channels
  • G.711, G.729A, G.723.1
  • Support Drag and Drop Call Flow Editor
  • Real Time Status/Variable debugger
  • Fully Web Management Interface

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SIPIVR 6800GS brought you a fully web user interfaced  Value Added Service Creation Application Server. By using easy drag and drug web interface, you can create your owned VOIP application or value added service very quick time to market. With built-in rich pre-designed  components, the developer can create their service without paying attention to the complexities of programming. Also the real time debugger makes developer very easy to debug and trace the call flow status.


網頁圖形化介面語音流程管理

Online Demo
Title
Description
Create a Sample Flow Call Flow Create / Edit / Save
Run the Sample Flow Runner Load / run
Debug the Sample Flow Debug a call flow
Create the Startup project How to use project and load it

Selected Features
  • Up-to 120 Universal VOIP Channels
  • SIP RFC 3261 Compliance
  • Fully Web Management Interface
  • Audio Codec G.711, G.729A, G.723.1*
  • Drag and Drop Call Flow Editor
  • Real Time Status/Variable debugger
  • Rich-set of predefined components:
    • Basic flow components
    • IVR components
    • Database components
    • Flow control components
    • RADIUS components
    • Channel components
    • HTTP Access Components
    • External Customized components
  • Support Call Hold and Transfer
  • Support in-Band and out-of-Band DTMF relay
  • Support Database Connection Pools
  • Support Internal/External Job Push & Retrieve
  • Support Internal/External Hook Function Calls
  • Free Text Math Expression with rich functions
  • Hitless Call Flow Update
  • Optimized Developing Platform

* G.723.1 is an optional for 6800S

Application Examples
  • Audio Broadcasting/Announcement Service
  • Hosted IVR
  • Prepaid Calling Card Service
  • Universal Call Back System
  • IP Centrex Service
  • Voice Mail System
  • Outbound Dialer System
  • VOIP Call Center

Applications