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Vxi*

The most advanced IVR plugin for Asterisk

To download our software for a free evaluation, we require you to be a registered user. If you have technical support questions please visit our community.

Built on Open Standards to protect your investment

Open Standard VoiceXML

Vxi* is fully compliant with the W3C’s VoiceXML 2.1 specification and provide some extended tags like <transfert> or <object>. We keep all interpreter very standard to allow you a fast migration from any other platforms.

Speech Recognition

Vxi* includes several speech recognition (ASR) connectors with Asterisk native API or MRCP v1 and v2 through uniMRCP. You can run advanced speech enabled services thanks to VoiceXML.

Text-to-Speech

Vxi* supports most Text-to-Speech (TTS) engine thru an advanced, universal HTTP open source interface that helps developers manage multi-vendor TTS engines in the same or different server(s).

Text-to-Video

Vxi* supports a Text-to-Video (TTV) extended feature for video services enabling to generate a real-time video content from a text in order to print a message or dynamic menu on your video phone screen.

Speaker Identification Verification

Vxi* can works with most phone biometrics existing APIs in the market. It includes a simple VoiceXML interface to make any speaker identification and verification at phone.

Voice and Video Convergent

Vxi* is the first voice and video convergent platform for Asterisk.It works with voice calls as well as video calls to deploy today amazing next generation communication services.

The first IVR system thought for every business

Reliable

ReliableWe understand that your business objective is to manage communications without worries. We built Vxi* to deliver exceptional reliability so you can confidently build your business on our software.

Economical

EconomicalWith Vxi* you have a choice of licensing editions that make it easy and affordable to put the industry's premier IVR / IVVR to work for you. No one else can match the value of Vxi* software suite.

Extensible

ExtensibleAn extensive set of native modules let you easily extend the Vxi* functionality to fit perfectly with your applications and environment. Vxi* is also a foundation for valuable Xtras* to take your phone service to new heights.

Manageable

ManageableVxi* offers extensive management and administrative functionality with standards-based programming interfaces, comprehensive logging and Asterisk compatibility with many off-the-shelf tools and management systems.

Standard

StandardVxi* is based on many telephony standards like VoiceXML, SIP, RTMP, MRCP, H323,…. Vxi* does not require any particular skills in terms of architecture or proprietary programming.

Scalable

ScalableVxi* software performance is in a category by itself. With the Vxi* platform you can deliver up to 120 or 240 calls from a single server or easily scale up to multi-server bare metal or cloud environment.

Efficient

EfficientVxi* is a powerful multi-threaded software built in C that delivers up to 60 or 120 calls of per-core  for on-demand inbound or outbound communications. The bottom line – you can count on Vxi* to deliver.

 Smooth Telecom Technology, not Rocket Science

  • Autoattendant
  • Mobile TV
  • Video Sharing
  • Video eLearning
  • Video Surveillance
  • Voice and Video Messaging
  • QoS Surveys Calls
  • Voice and Video Portals
  • Televoting
  • Video dating
  • Voice and Video Self Services
  • Predictive Dialing
  • Emergency Notification
  • Voice and Video Broadcast
  • VoiceXML V2.0 compliant
  • VoiceXML V2.1 tags extended (like <transfer>,…)
  • Call Control functions/variables for Asterisk
  • High performance C built
  • Base on OpenVXI voice browser template
  • Application for Asterisk PBX (app_vxml.so)
  • Binaries available for Linux OS 32bit and 64bit
  • VoiceXML accounts management
  • Support plugging objects with the VoiceXML tag, SDK with API available
  • Video calling over SIP / H323 / 3G-324m / RTMP (prompt and record)
  • VoIP / SS7 / ISDN Networks
  • Fax over VoiceXML (send and receive)
  • Video silence parameter with VoiceXML syntax
  • Text-to-Speech (TTS) connectors
  • Automatic-Speech-Recognition (ASR) connectors
  • MRCP v1 and v2 Client for ASR (multi-grammar allowed)
  • Text-to-Video (TTV) with HTTP connector
  • Voice-Silent Detection (VSD)
  • Support RTSP-URI with <audio> voice and video
  • Control VCR function for WAV clips with property “control” (* and # keys)
  • VoiceXML 2.1
  • MRCP v1 and v2
  • SRGS 1.0
  • SSML 1.0
  • RTSP
  • RTMP
  • HTTP and HTTPS
  • Simultaneous support of PSTN and IP calls
  • Simultaneous support of audio and video
  • Seamless migration of PSTN to IMS
  • Flexible configurations supporting audio only, audio with video
  • Outbound and/or inbound call control through Asterisk
  • Audio playback support, with VCR controls
  • Video transcoding through optional Video Transcoder server
  • Audio conferencing support for contact-center-like functions
  • Capacity upgrade through software license
  • VoiceXML 2.0 compliant and some additional tags supported
  • VoiceXML scripts are cached
  • Detailed logging with full information level of the voice browser
  • SIP interface allows load balancing and redundant configurations
  • VoiceXML server control of speech server through Asterisk API
  • VoiceXML server control through uniMRCP v1 and v2
  • Application-selected ASR
  • Synthesized prompts cached for improved performance (hypercache)
  • Prompt engine provides low-latency streaming of synthesized speech-to-bearer channels
  • Application selected TTS
  • Synthesized prompts are cached for improved performance and reduce TTS usage
  • Uses SSML tags to improve speech articulation for TTS engines SSML compliant
  • HTTP technology allowing multi-server resource sharing between different VXI* systems.
  • G.711 A-law/μ -law, AMR-NB, G.723.1, G.726 @ 32 kbps (DSP), G.729ab (Digium License)
  • H.263, H.263+, H.264, MPEG-4 , H263 Sorenson (Flash/RTMP)
  • On-demand, any-to-any, real-time video transcoding supported through VXI* video transcoder
  • DTMF detection and generation
  • Echo cancellation through Asterisk
  • Voice Activity Detection
  • Audio transcoding
  • Audio ringback tones over VoiceXML
  • VCR controls (audio and video)
  • Conferencing support through Asterisk control
  • Echo cancellation through Asterisk control
  • Coaching mode
  • Call recording
  • Music-on-hold
  • Video support through VoiceXML
  • Optional embedded 3G-324M gateway (Xtras* Video IP/3G)
  • H.264, MPEG-4, H.263 with any-to-any video transcoding
  • Video rate & size adaptation (43Kb to 384Kb, QCIF/CIF)
  • Video refresh (RFC 5168)
  • Integrated with ASR and TTS support
  • Separate audio or video sources
  • Video conferencing up to 12 channels
  • Background Recording all channels
  • Simultaneous video play and record (video karaoke)
  • Streaming video with RTSP connection sharing (Xtras* Video IP/3G)
  • Provisioned through CLI* Shell Management Console
  • DNIS to application URI mapping
  • Loquendo
  • Lumenvox (Native API)
  • Nuance
  • Sestek
  • Verbio (Native API)
  • Vestec (Native API)
  • Voice Interaction (Native API)
  • MRCP v1 and v2 compatible engines
  • Acapela
  • Cepstral
  • Google Text-to-Speech
  • Ivona
  • Loquendo
  • Nuance
  • Neospeech
  • Microsoft Text-to-Speech
  • Sestek
  • Verbio
  • Voxygen (Baratinoo)
  • Voice Interaction (Audimus)
  • Flite / eSpeak / MBROLA *free TTS
  • Yantra Software
  • Helix Mobile Server (Real)
  • Darwin Streaming Server (Apple)
  • .raw 8kHz 16-bit (PCM) mono
  • .gsm 8kHz 16-bit mono
  • .wav 8kHz 16-bit (PCM) mono
  • .mp3 (MPEG audio layer 3)
  • .3gp (324M video)
  • .tiff (TIFF image file format)
  • And any other Asterisk’s audio file format supported
  • FILE:// local or remote
  • HTTP(S):// local or remote, with caching per header
  • RTSP:// local or remote, streaming audio or video
  • RFC 3550/3551 (RTP)
  • RFC 2326 (RTSP)
  • RFC 2833 (DTMF)
  • RFC 3267/IF2 (AMR)
  • RFC 2190 (H.263)
  • RFC 2429 (H.263+)
  • RFC 3016 (MPEG-4)
  • RFC 3984 (H.264)
  • RFC 5552 (SIP-VoiceXML) • RTMP
  • Simultaneous PSTN and VoIP support with integrated PSTN-SIP gateway
  • Gigabit Ethernet (redundant recommended)
  • T1/E1 Through Asterisk compliant boards
  • ISDN Through Asterisk compliant boards
  • ISUP/SS7 Through Asterisk compliant boards
  • ISUP/SIGTRAN Through Asterisk compliant boards
  • SIP (internetworking with ISDN/ISUP/BICC)
  • RTMP Flash
  • H323 for voice and video
  • T30 FAX
  • VoiceXML SIP | PSTN — 480 ports (Voice)
  • VoiceXML SIP | PSTN — 240 ports (Video)
  • Capacity upgrade through software license
  • Capacity depends on codec, frame rate, bit rate, and size.
  • For example: Up to 180 streaming ports of H.263 to H.264 (QCIF) transcoding

Examples:

  • HP DL360 G6
  • HP DL160 G6
  • HP DL320
  • HP DL120
  • DELL Poweredge series
  • Fujitsu Primergy series
  • Any other Linux Server compliant hardware, min 2Gb or 4Gb RAM
  • Asterisk 1.4.X 32bit / 64bit
  • Asterisk 1.6.X 32bit / 64bit
  • Asterisk 1.8.X 32bit / 64bit
  • Linux Debian 7.0, 6.0, 5.0 32bit / 64bit
  • Linux CentOS 6.x, 5.x 32bit / 64bit
  • Any other compliant distribution like Ubuntu, Readhat, Mandriva, Slackware, Fedora…

Ready to purchase?

If you have questions about buying our software, or want to make sure you understand the process before you start your project, give us a call. Fill out our simple application. We’ll take it from there.
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