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VX Prime+ Broadcast VOIP Phone System

Model: Telos VX Prime+

  • A true VoIP telephone system designed and built specifically for broadcasting; VX Prime+ is ideal for small to medium studios with 2 to 4 studios.
  • Includes support for AES67, giving broadcasters added flexibility of integrating VX Prime+ into any AES67 network, in addition to our own Axia Livewire network.
  • SIP call-handling throughout—no internal conversion to analog call handling like some other so-called “VoIP” systems.
  • Standards-based SIP interface integrates with Asterisk open-source SIP phone servers and most VoIP-based PBX systems to allow transfers and common telco services for business and studio phones.
  • Standard Ethernet backbone provides a common transport path for both studio audio and telecom needs, resulting in cost savings and a simplified studio infrastructure.
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Telos VX® talk-show systems are the world’s first true VoIP-based broadcast phone systems and have been proven to deliver the power of VoIP to the broadcast studio like no other. The Telos VX Prime+, with built-in support for AES67, is the next evolution of Telos VX VoIP phone systems in a powerful new 1RU hardware unit. Additionally, support for the G.722 voice codec ensures the highest quality calls from supported mobile devices. With capacity of 8 fixed hybrids/faders, VX Prime+ is ideal for facilities with 2 to 4 studios. (For larger facilities, check out VX Enterprise with up to 120-hybrid capacity.)

  • A true VoIP telephone system designed and built specifically for broadcasting; VX Prime+ is ideal for small to medium studios with 2 to 4 studios.
  • Includes support for AES67, giving broadcasters added flexibility of integrating VX Prime+ into any AES67 network, in addition to our own Axia Livewire network.
  • SIP call-handling throughout—no internal conversion to analog call handling like some other so-called “VoIP” systems.
  • Standards-based SIP interface integrates with Asterisk open-source SIP phone servers and most VoIP-based PBX systems to allow transfers and common telco services for business and studio phones.
  • Standard Ethernet backbone provides a common transport path for both studio audio and telecom needs, resulting in cost savings and a simplified studio infrastructure.
  • System capacity of 8 hybrids. Each call placed on the air receives a dedicated hybrid for unmatched clarity and superior conferencing.
  • Native Livewire integration—one connection integrates caller audio, program-on-hold, mix-minus and logic directly into Axia AoIP consoles and networks.
  • Connect VX systems to any third-party radio console or other broadcast equipment using available Telos Alliance Mixed Signal, AES/EBU and GPIO xNodes. xNodes feature 48 kHz sampling rate and studio-grade 24-bit A/D converters with 256x oversampling.
  • Powerful dynamic line management enables instant reallocation of call-in lines to studios requiring increased capacity.
  • VSet phone controllers with full-color LCD displays and Telos Status Symbols present producers and talent with a rich graphical information display. Each VSet features its own address book and call log.
  • The “Drop-in” Vset Call Controller™ modules can integrate VX phone control directly into your mixing consoles.
  • XScreen Lite screening software included.
  • Clear, clean caller audio from 5th-generation Telos Adaptive Hybrid technology, including Digital Dynamic EQ, AGC, adjustable caller ducking and send- and receive-audio dynamics processing by Omnia.
  • Support for G.722 codec enables high-fidelity phone calls from iPhone and Android SIP softphones using an SIP server.
  • Wideband acoustic echo cancellation from Fraunhofer IIS completely eliminates open-speaker feedback.
  • Works with POTS, T1/E1, ISDN and SIP Trunking telco services for maximum flexibility and cost savings, via Asterisk servers.
 
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