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History: Chat and Video Conference

Preview of version: 13

Chat and Video Conference

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WikiSuite PBX Phone system, Chat & Video Conference features are built on enterprise level industry leading open source technologies. Cross-platform telephony platform is designed to route and interconnect popular communication protocols (XMPP + WebRTC + SIP) using audio, video, text or any other form of media.

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FusionPBX, freeSWITCH, Kaltura

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Community
353 contribueurs travaillent actuellement sur 34 requêtes d’améliorations, et de nouvellles fonctionnalités.

Participez aux discussions dans la section Communauté.

Features

Possible Uses

  • Rating & Routing Server
  • Transcoding B2BUA
  • IVR & Announcement Server
  • Conference Server
  • Voicemail Server
  • SBC (Session Border Controller)
  • Basic Topology Hiding Session Border Controller
  • DAHDI, Khomp, PIKA, Rhino, Sangoma and Xorcom Hardware Support
  • Fax server
  • And, of course, a PBX


Key Features

  • WebRTC support
  • Centralized User/Domain Directory (directory.xml)
  • Nano Second CDR granularity
  • Call recording (In Stereo caller/callee left/right)
  • High Performance Multi-Threaded Core engine
  • Configuration via cURL to your HTTP server (mod_xml_curl).
  • XML Config files for easy parsing.
  • Protocol Agnostic
  • ZRTP support for transparent RTP based key exchange and encryption
  • Configurable RFC 2833 Payload type
  • Inband DTMF generation and detection.
  • Software based Conference (no hardware requirement)
  • and more...

Voicemail

  • Multitenancy - Enterprise/Carrier configuration
  • Time of Day Greetings
  • Urgent Message Tagging
  • E-mail Delivery
  • Playback and Rerecord messages before delivery.
  • Keys are templates so you can rearrange to fit your needs.
  • Callback support from inside voicemail.
  • Podcast of Voicemail (RSS)
  • Message Waiting Indicator (MWI)


Conference

  • Software based Conferencing without any hardware requirements.
  • Wideband conferences.
  • Multiple on-demand or scheduled conferences with entry/exit announcements
  • Play files into the conference or a single member.
  • Relationships
  • TTS integration
  • Transfers
  • Outbound Calling
  • Configurable Key Lay
  • Volume, Gain and Energy level per call.
  • Bridge to Conference transition
  • Multi Party outbound dialing.
  • RFC 4579 SIP CC Conferencing for UAs
  • Automatic or on-demand recording

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History

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Information Version
M. Bilal Siddiq 15
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Marc Laporte 14
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M. Bilal Siddiq 13
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M. Bilal Siddiq 12
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M. Bilal Siddiq 11
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M. Bilal Siddiq 10
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M. Bilal Siddiq 9
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M. Bilal Siddiq 8
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M. Bilal Siddiq 7
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M. Bilal Siddiq 6
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M. Bilal Siddiq 5
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M. Bilal Siddiq 4
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M. Bilal Siddiq 3
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M. Bilal Siddiq 2
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Marc Laporte 1
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