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VirtualPBX

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XVB – Voice Application Server / HostedPBX solution / HostedIVR solution based on asterisk ( v.3, build 8645 ).

The virtual-pbx application is designed for processing incoming/outgoing calls in an isolated environment (numbered plan, routing calls, phones, cdrs, web gui and so forth. ) for multiple users.

Virtual-pbx application include:

  • Processing incoming / outgoing calls.
  • Completely isolated environment for different users ( incoming / outgoing routes, dial plan, sip-endpoints, web-interface, cdrs, call-recordings, etc ).
  • Full control of users ( disk quota, the number of phones, the number of messages, the number of concurrent calls, etc ).
  • Custom user greetings support.
  • Email notifications.
  • Multiple language voice prompts.
  • Text To Speech ( TTS ) for multiple languages.
  • ‘Simple’ or ‘Expert’ mode for configuration.
  • Custom music on hold (MOH) for each user.
  • Multiple language WEB interface ( screenshots )
  • xml/json API support.
  • Flexible customisation for system voice messages.
  • Managing voice mail via phone or WEB interface.
  • User specified time zones support.
  • Call transfer support ( blind transfer, with return of unsuccessful call ).
  • Smart call pickup.
  • White / Black lists support for each IVR item.
  • PhoneBook with speed dial feature.
  • Journal configuration changes.
  • CDR support ( search by caller, called, date of call, etc.. )
  • XML backup / restore configuration.
  • Multiple roles within a single IVR account.
  • Privated / Shared DIDs support.
  • SQL reports.
  • Management API.
  • Call routing based on Google.Calendar events. *
  • Support Google Analytics for calls trekking.
  • Radius accounting.
  • Background music for Find-Me / Queue calls.
  • Fail2ban integration.
  • Support for wideband / HD audio codec ( g722 ).
  • Email/Web interface Branding.
  • Recording outgoing calls ( auto / on demand ).
  • Full DTMF history for each call.
  • Support presettings for SIP providers.
  • Support Multi-tenant asterisk with Kamailio as sip registrar server / load balancer.
  • Easy to update.
  • speedyAGI server ( greater stability && high performance )
  • FMC – Fixed Mobile Convergence / One touch call transfer without breaking the call.
  • Google-chart for extended statistics for: DialOut, Queue, CallBlast, Polling, etc
  • WEB logon / logoff for the agents of the queue.
  • Support for perl plugins.
  • Time limits for outbound calls based on route.
  • Schedule for routes.
  • IP protection to register phones.
  • Administrator / Partner / User interfaces
  • CallerID for phone / route / peer
  • Click2Call
  • Sip domains for incoming calls
  • DND / CallForward for local phones
  • Call trace for admin interface
  • Schedule for user routes
  • Easy to CRM integration
  • Several types of extensions:
    • Playback ( with time tags feature ) / Auto Attendant
    • VoiceMail
    • DialOut with Find-Me feautures.
    • Company Directory
    • Schedule
    • WEB-Request ( sync / async mode )
    • Chat-Room ( with WEB broadcasting via IceCast and call recording )
    • Fax2Email
    • DTMF2Email
    • Polling (Voting)
    • DISA
    • Podcast2Phone gate
    • ACD Queues
    • CallBack
    • DateTime
    • Fax on Demand / web fax
    • Bulletin board system
    • CallBlast ( Predictive Dialer, with schedule, call limits, custom environment for each call, and many other features. )
    • WEB Variables
    • Stored DB variables
    • Goto If
    • Alarm Clock ( Telephone Reminder / Wake Up )
    • Google Calendar – Shedule: checks status busy / free. *
    • RoboTEXT with support money, date, time, phonenumber, percent, degree formats
    • MP3 Streaming
    • Call Parking
    • GoogleCalendar – Events: reads events via TTS + confirmation / cancellation of events. *
    • Paging / Intercom
    • Play DTMF Tones
    • MultiDialout

start using right now – just download vmware image and run it in VMWare Player !

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