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Audio Codecs


Comrex Phone  6-Line Syatem

From: Comrex
  • Weight:3.7 kg

  • Shipping Dimensions:48.26 x 29.21x8.89 cm


-Send Input 1/AES3: XLR female, 0 dBu nominal, balanced, bridging, 20 kΩ impedance, Analog/Digital switchable
-Send Input 2: XLR female
-Caller 1 & 2 Outputs: XLR male, 0 dBu nominal, balanced, 50 Ω impedance
-Hold 1 Input: XLR female, 0 dBu nominal, balanced, bridging, 20 kΩ impedance
-Hold 2 Input: XLR female, 0 dBu nominal, balanced, bridging, 20 kΩ impedance

At the core of STAC VIP is a sophisticated engine that can process various types of incoming calls. Comrex STAC VIP can process up to 6 incoming VoIP calls on a single DSL line utilizing G.729 audio compression. By employing a single DSL instead of multiple POTS or ISDN phone lines, users can save a whole lot of money.

VoIP (Voice-over-IP) has taken the telephone industry by storm because of its great flexibility coupled with extremely low cost base and STAC-VIP puts Broadcasters in the position to benefit from this change. Comrex STAC-VIP smoothly integrates legacy PSTN/POTS lines with VoIP technology to deliver a new way to manage telephone calls for talk shows, interviews and contests. Comrex STAC-VIP can take traditional POTS calls but breaks new ground by handling calls from "HD Voice-capable" telephones and Smartphone apps. Complete with the Comrex STAC IP Call Screening with CLI Caller Identification and Call Logging as standard via its browser Control Interface, the Comrex STAC-VIP Caller Management system integrates with VoIP PBX systems and leads the charge in migrating old-fashioned on-air telephones to dramatically-clearer wideband audio quality.

Instead of connecting to standard PSTN lines, the STAC-VIP operates using VoIP connectivity and can conference up to 12 calls plus host on-air at the same time. Based-on the industry-standard STAC-6 and STAC-12 (and the legacy TS-612) it can handle either 6 or 12 "lines" (which can also be from PSTN subscribers via a SIP Gateway) and provides call management and screening capabilities over a networked browser interface as well as via the traditional hardware control surface and handset. Breaking new ground by handling calls from HD Voice-Capable telephones, Smartphone apps and the latest variable bitrate and low delay OPUS audio format.


-AES3 Out: XLR male, Discreet AES Output
-Aux Out: XLR male
-Dual USB host ports: Host Port 2.0
-Primary Ethernet Port: 1,000Base-T
-Secondary Ethernet Port: 1,000Base-T
-Contact Closure: 9 pin D-sub male
-Serial Port (unused): 9 pin D-sub female
-VGA: 15 pin D-sub female, high density
-Internal supply: Auto Adjusting 100/240 VAC 50/60 Hz
-Power Consumption: 30W nominal
-Caller to Send Separation: >60dB