VX Prime

Multi-Studio Voip system. Delivery includes VX Producer Software and Broadcast Bionics xScreen Lite Softwares. Dual Redundant Power Supply. Up to 8 hybrids with up to 48 active calls
Actuellement indisponible
Acheter Prix catalogue
CHF 6'590.00


Matériel de démo, état de neuf / Vorführgerät, neuer Zustand

The heart of any VX system is the Engine. The fixed-capacity VX Prime+ system is powered by a 1RU rack-mount Engine with enormous processing power. In fact, the VX Prime+ Engine provides all the call control and audio processing needed for your entire on-air phone system.

With VX Prime+, you are equipped with 8 high-performance VoIP hybrids, to support multiple lines of concurrent on-air phones for two to four studios (depending on configuration).

Each VX Prime+ Engine features two Gigabit Ethernet ports, a high-density, cost-effective interface to both telephone lines and studio audio via proven Livewire Audio over IP (AoIP). VX systems are web-based, so remote control and configuration are easy—engineers can work from any place they can get online.

Call workflow for VX users is sophisticated and flexible. Lines may be readily shared among studios; the Web interface allows easy assignment of lines to “shows,” which can then be selected by users on VSet phone controllers and console drop-in modules. Each studio may be configured with its own Program-on-Hold as well.

The processing power of the VX Engine provides sophisticated DSP hybrids for every line, allowing multiple calls to be conferenced and aired simultaneously with excellent quality. The hybrids are equipped with a rich processing toolbox to make caller audio sound its best, no matter what kind of line or phone the caller uses.

Caller audio benefits from Smart AGC coupled with famous Telos three-band adaptive Digital Dynamic EQ and a three-band adaptive spectral processor. Call ducking and host override are part of the VX audio toolkit as well.

You’ll notice that there are no audio I/O or telco ports on VX Engines themselves. That’s because they’re meant for fast connection to Livewire AoIP systems; using Livewire, all I/O is handled via Ethernet. The Livewire network supports a wide variety of peripherals such as Axia audio consoles, VSet phones, PC-based screener applications, console-integrated controllers, and more. SIP servers and telecom providers connect through a dedicated WAN Ethernet jack for routing simplicity and easy maintenance.

For traditional phone services, VX works seamlessly with open-source Asterisk SIP servers, and most SIP PBXs. Telos VX experts speak fluent Asterisk, and are ready to assist you in specifying and configuring an installation to suit your studio’s requirements. VX also works with standard telco gateways to connect to T1/E1, ISDN, and POTS providers. And, if you already have a VoIP-based PBX or SIP endpoint service, VX systems can work with those as well.


  • 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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de 8H à 17H
021 925 30 00