Features

HIGHLIGHTS

  • Build conversational speech interfaces on-the-fly
  • Stay with globally adopted standards of VXML & SIP
  • Seamlessly integrate with legacy / existing systems
  • Leverage built-in DTMF recognizer
  • Deploy on off-the-shelf hardware or on-cloud
  • Distributed architecture for carrier-grade reliability
  • Leverage existing apps, integrations & business logic
  • Secure interactions for PCI and HIPAA compliance
  • Centralize logging, reporting and analytics
  • Built in CTI support
Speech Support: Ready support for range of commercially available and open source MRCP v2 ASR & TTS engines. Extended interface to support next-gen HTTP based speech engines.
SIP Interface: InterpreXer™ leverages open source FreeSWITCH® to handle telephony interactions, like in/outbound call handling for VXML, call <transfer> (bridge / blind), SIP Register/Auth, CTI, etc. Seamlessly deploy within any SIP infrastructure or with any SIP provider globally.
Platform Flexibility: Deploy on desktop machines, server grade hardware on-prem or in clustered environment oncloud with standard CentOS 6.x, 7.x (x64) and Windows 2016 (x64). Also available as part of Phonologies’ SaaS offering.
Provisioning and Management: Simple web-based interface for telephony DiD provisioning, configuration of components, application mapping, SIP registrations, real time logging, etc.
Security & Compliance: All interactions are PCI compliant / HIPAA ready with support enabled for NAT and SIP Auth (ACL) at telephony layer and SSL for web service integrations.
Performance and Scalability: Distributed modules for telephony/SIP, dialog and media handling. Configurable SIP interface for deploying thousands of ports in a load balanced / HA environment with failure detection.
Virtualization: Leverage virtualized environments to deploy on-cloud, on-prem using standard operating systems.
Reporting & Logging: Out-of-the-box logging lets you track every state of your call / session to identify issues within the application, modules or connectors
Smarter logging: Out-of-the-box extension for smart analytics, CDR export to BI tools and monitoring with builtin alerting for pre-defined rules (via SMS/Voice/Email, webhooks to Slack, LogDNA and more).
Multi-tenant Support: ANI / DNIS based call provisioning and logging with custom extensions to push data, recordings, metrics to centralized infrastructure.
Complete Call Progress Analysis & Control: Total control over outbound call campaigns via web-services API to capture call progress information (session start, ringing, no/answer, busy, invalid number format, etc) in real time. Add-on beep detection feature to distinguish between human VS voicemail answers.
CTI Support: Leverage built-in CTI interface to trigger events and/or connections with leading contact center platforms. Execute inbound CTI events and custom application types within a VoiceXML session.

Architecture & Deployment

Technical Specifications

Application Protocols

  • VoiceXML 2.0 & 2.1
  • GrXML & SSML

Speech Technology

  • MRCP v2 for ASR & TTS
  • Google Speech (UniMRCP)
  • Nuance NSS 11
  • LumenVox 16

Signaling / Codecs

  • SIP / RTP (g711uLaw / g729)
  • MRCP v2 (g711uLaw)

Standards

  • HTTP/1.1 & HTTPS
  • ECMAScript

Custom Extensions

  • Invoke FreeSWITCH® apps via VXML
    (Fax, SMS, etc.)
  • Call recording

SIP Support

  • In/outbound call support
  • Call transfers / bridging
  • SIP registrar support
  • SIP proxy support
  • SIP authentication
  • Built-in NAT / IP translation
  • Configurable DTMF

Load Balancing

  • OpenSIPs (custom module)

Management GUI

  • Provisioning DID / SIP Extensions
  • Multiple VXML URI configuration
  • Configuration of Speech Engines
  • Real time Logs / System monitoring

VXML App development IDE

  • Java based (custom module)

Operating System & Hardware

  • Virtual Machines
  • Linux CentOS 6.x, 7.x (x64)
  • Win2016
  • Intel Quad Core 3Ghz Proc
    + 10GB HD
    + 8GB RAM