INFINITY INF-VX-PRIME+ PN 2001-00510-000 VX PRIME+VoIP SYSTEM

Features:

• Includes support for AES67, giving broadcasters added flexibility of integrating VX into any AES67 network and compatible SMPTE 2110-30 networks, in addition to Livewire networks.
• HTML-5 control via Pathfinder Core PRO.
• SIP call-handling throughout—no internal conversion to analog call handling like some other so-called “VoIP” systems.
• Works directly with hosted VoIP or PBX services, and in conjunction with a PBX may support POTS, T1/E1, and ISDN BRI for maximum flexibility and cost savings.*
• Standards-based SIP interface integrates with all modern VoIP-based PBX systems to allow transfers, line-sharing, caller ID, 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 simplified studio infrastructure. Connection to hundreds of control devices (software or hardware) is possible.
• Each call receives a dedicated processing path 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 Telos Alliance AES/EBU, Mixed Signal, and GPIO xNodes. 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 Call 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 directly integrate VX phone control into your mixing consoles.
• Clear, clean caller audio from fifth-generation Telos Adaptive Telephony technology, including Digital Dynamic EQ, AGC, adjustable caller ducking, and audio dynamics processing by Omnia®.
• Wideband acoustic echo cancellation eliminates open-speaker feedback.
• Support for the G.722 codec enables high-fidelity phone calls from iPhone and Android SIP softphones using a SIP server.