VX Specs At A Glance

The first VoIP telephone system designed and built specifically for broadcasting.
Works with POTS

T1/E1

ISDN and SIP Trunking telco services for maximum flexibility and cost-savings.
Standards-based SIP/IP interface integrates with most VoIP-based PBX systems to allow transfers

line sharing 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. Connection of up to 100 control devices (software or hardware) is possible.
Modular

scalable system can be easily expanded to manage a network of up to 20 studios

each with a dedicated Program-On-Hold input – truly a “whole-plant” solution for on-air phones.
System capacity of up to 48 standard phone lines; supports up to 250 SIP numbers.
Up to 16 hybrids

with as many as 48 active calls (up to 4 per hybrid)

may be placed on-air concurrently.
Each call 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 to any radio console or other broadcast equipment using available Analog

AES/EBU and GPIO interfaces. Audio interfaces 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.
Drop-in modules can integrate VX phone control directly into your mixing consoles.
Included VX Producer screening software with built-in soft-phone allows a “phone” connection on any networked PC. Integrated recorder/editor simplifies recording of off-air conversations.
Clear

clean caller audio from fifth-generation Telos Adaptive Hybrid technology

including Digital Dynamic EQ

AGC

adjustable caller ducking

and send- and receive-audio dynamics processing by Omnia. Wideband acoustic echo cancellation from Fraunhofer IIS completely eliminates open-speaker feedback.
Support for G.722 codec enables high-fidelity phone calls from SIP clients.