COMREX VH2 two-line VoIP hybrid for individual broadcast or flawless conferencing

2,765.00 incl. GR VAT

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Hybrid phone two VoIP telephone lines. Supports telephone lines G.711 & G.722. Has ability checks the phone lines via VoIP devices phone. Easy installation via web based environment. In the area of I / O features two analog inputs and two outputs to XLR form configurable and AES / EBU. audio input “on-hold” in order […]

 

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Hybrid phone two VoIP telephone lines. Supports telephone lines G.711 & G.722. Has ability checks the phone lines via VoIP devices phone. Easy installation via web based environment. In the area of I / O features two analog inputs and two outputs to XLR form configurable and AES / EBU. audio input “on-hold” in order to provide music on hold, AGC and network port 10/100/1000.

Any studio that processes phone calls needs hardware to interface with phone lines. This device, traditionally called a Hybrid, filters, separates and provides gain adjustment and call control to more easily allow for recording or broadcasting “phoners”.

As telephone companies are shifting from traditional phone lines to Voice-over-IP (VoIP) systems, the devices that performed this job in the past are becoming outmoded. Broadcasters need a VoIP hybrid to ensure on-air and recorded phone calls sound as good as possible.

A dual-line hybrid, VH2 connects two VoIP lines to a studio for individual broadcast or flawless conferencing. VH2 prevents echo and distortion, and automatically adjusts caller audio to a uniform level, leaving you with a result that’s clean and clear. Plus, VH2 uses only VoIP phone lines, saving you money and increasing functionality. VH2 can even connect to many VoIP PBX systems

  • Prevents echo and other artifacts
  • Supports normal phone calls (G.711) and wideband phone calls (G.722)
  • Allows for caller ID, outgoing calls, screening, and post-air management using companion VoIP phone set
  • Set up via easy-to-use web based configuration page

Easy Hookup

VH2 can be configured in several ways to be compatible in environments where the studio has different telephone connection arrangements. Dual or single input and outputs can be selected, and AES3* or analog audio I/O can be chosen. VH2 can be configured for callers to hear each other, or be isolated depending on the needs of the studio. *Supports 48 KHz sampling rate only for AES3.

On-Hold Inputs

VH2 offers a pair of audio inputs for callers “on-hold”. This allows for listeners to hear your programming while they are waiting to be put “on-air”.

Consistent Audio Levels

Selectable automatic gain control (AGC) maintains a uniform audio output, even when the caller signal varies widely. Also, selectable caller ducking lowers incoming caller audio, so local talent remains in control.

Companion Phone

When configured with its companion phone (the Polycom VVX 201 IP phone), VH2 does even more. Calls can be answered on the handset and easily transferred back and forth to VH2, just like a traditional telephone hybrid. And the companion phone supports caller ID and outbound calling.

Status Indication

VH2 is outfitted with front panel controls and status indicators so it can be used out-of-the-box. Front panel buttons can also be remoted via the rear panel connector, so your console buttons can trigger its functions.

Audio Connections

  • Caller audio out on balanced XLR-M output
  • Send audio in on balanced XLR-F input (clip +20dBu). Switchable to AES3 I/O (48KHz sampling rate only for AES3).
  • On-hold audio in on ¼” TRS jack input (clip +20dBu)

Other Connections

  • 10/1000 Ethernet port
  • Contact closures
  • 9 pin mini DIN
  • Serial port on 8 pin mini DIN
  • Power in on 4 pin mini DIN
  • Universal external power supply +24VDC
  • Compliant with worldwide regulations, including FCC, CSA, and CE

When you want to present, broadcast, or record a telephone conversation, you need a device to process the phone call and present it to the console, as well as to separate “send” audio from the “receive” audio on the call. If send and receive audio aren’t isolated, it will result in an echoey, muddy sound – not to mention, annoyed listeners.

As many major markets are shifting from traditional phone lines to Voice-over-IP (VoIP) systems, the digital hybrids that could have performed this job in the past are becoming outmoded. Radio stations need a VoIP hybrid to ensure on-air and recorded phone calls sound beautiful.

A dual-line hybrid, VH2 connects two VoIP lines for individual broadcast or flawless conferencing. VH2 prevents echo and distortion, and automatically adjusts caller audio to a uniform level, leaving you with a result that’s clean and clear. Plus, VH2 uses VoIP phone lines, saving you money and increasing functionality. VH2 can even connect to many VoIP PBX systems.

Main Features:
Audio Processing and Performance:
• Prevents echo and other artifacts
• G.722 codec support for wideband calls. Also supports G.711
• Receive filter reduces telephone line noise
• Selectable automatic gain control (AGC) maintains a consistent audio output, even when the caller signal varies widely
• Selectable caller ducking lowers incoming caller audio so local talent remains in control of the conversation
• Can be configured to automatically answer and disconnect incoming calls

Operation:
• Easily segue from caller-to-caller
• Separately selectable single-ring auto-answer function for assisted or unattended operation
• Handy front panel controls and status indication
• When used with companion VoIP telephone, calls can be answered on handset and easily transferred back and forth to VH2
• Hybrid on/off controls and status remotable via web or contact closures
• Send and caller level indication
• Easy call conferencing
• Dual “On-Hold” audio inputs to send program to callers on hold
• Auto-Switching External Power Supply
• Compliant with worldwide regulations, including FCC, CSA and CE

Audio Configuration:
• Configure for separate caller outs of single caller mix
• Configure for separate send feeds or single
• Pro level, balanced audio I/O in XLR
• Selectable AES3 I/O

IP Features:
• Web-based configuration for the VoIP phone line setup, making it easy to adjust settings remotely from a browser
• Transfer calls back and forth to many PBXs, or use optional companion extension VoIP phone
• Ability to engage or drop or dial calls via web page
• Companion phone – easy to move calls between handset and hybrid with the touch of a button

Weight 1.00 kg

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COMREX VH2 two-line VoIP hybrid for individual broadcast or flawless conferencing removeDBX 266XS COMPRESSOR/GATE 2CH removeAxel Technology MACROTEL7 Telephone Hybrid removeORBAN OPTIMOD-FM5500 2-5 BAND AUDIO SIGNAL PROCESSOR removeSAMSON S-COM 4 Four Channel Compressor-Gate Processor removeTOA L-1102 2 CHANNEL LIMITER-LEVELLER remove
NameCOMREX VH2 two-line VoIP hybrid for individual broadcast or flawless conferencing removeDBX 266XS COMPRESSOR/GATE 2CH removeAxel Technology MACROTEL7 Telephone Hybrid removeORBAN OPTIMOD-FM5500 2-5 BAND AUDIO SIGNAL PROCESSOR removeSAMSON S-COM 4 Four Channel Compressor-Gate Processor removeTOA L-1102 2 CHANNEL LIMITER-LEVELLER remove
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SKUVH2DBX 266XSMACROTEL7OPTIMOD FM 5500SCOM 4L-1102
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Hybrid phone two VoIP telephone lines. Supports telephone lines G.711 & G.722. Has ability checks the phone lines via VoIP devices phone. Easy installation via web based environment. In the area of I / O features two analog inputs and two outputs to XLR form configurable and AES / EBU. audio input "on-hold" in order to provide music on hold, AGC and network port 10/100/1000.

Any studio that processes phone calls needs hardware to interface with phone lines. This device, traditionally called a Hybrid, filters, separates and provides gain adjustment and call control to more easily allow for recording or broadcasting "phoners”.

As telephone companies are shifting from traditional phone lines to Voice-over-IP (VoIP) systems, the devices that performed this job in the past are becoming outmoded. Broadcasters need a VoIP hybrid to ensure on-air and recorded phone calls sound as good as possible.

A dual-line hybrid, VH2 connects two VoIP lines to a studio for individual broadcast or flawless conferencing. VH2 prevents echo and distortion, and automatically adjusts caller audio to a uniform level, leaving you with a result that’s clean and clear. Plus, VH2 uses only VoIP phone lines, saving you money and increasing functionality. VH2 can even connect to many VoIP PBX systems

  • Prevents echo and other artifacts
  • Supports normal phone calls (G.711) and wideband phone calls (G.722)
  • Allows for caller ID, outgoing calls, screening, and post-air management using companion VoIP phone set
  • Set up via easy-to-use web based configuration page

Easy Hookup

VH2 can be configured in several ways to be compatible in environments where the studio has different telephone connection arrangements. Dual or single input and outputs can be selected, and AES3* or analog audio I/O can be chosen. VH2 can be configured for callers to hear each other, or be isolated depending on the needs of the studio. *Supports 48 KHz sampling rate only for AES3.

On-Hold Inputs

VH2 offers a pair of audio inputs for callers "on-hold”. This allows for listeners to hear your programming while they are waiting to be put "on-air”.

Consistent Audio Levels

Selectable automatic gain control (AGC) maintains a uniform audio output, even when the caller signal varies widely. Also, selectable caller ducking lowers incoming caller audio, so local talent remains in control.

Companion Phone

When configured with its companion phone (the Polycom VVX 201 IP phone), VH2 does even more. Calls can be answered on the handset and easily transferred back and forth to VH2, just like a traditional telephone hybrid. And the companion phone supports caller ID and outbound calling.

Status Indication

VH2 is outfitted with front panel controls and status indicators so it can be used out-of-the-box. Front panel buttons can also be remoted via the rear panel connector, so your console buttons can trigger its functions.

Audio Connections
  • Caller audio out on balanced XLR-M output
  • Send audio in on balanced XLR-F input (clip +20dBu). Switchable to AES3 I/O (48KHz sampling rate only for AES3).
  • On-hold audio in on ¼” TRS jack input (clip +20dBu)

Other Connections
  • 10/1000 Ethernet port
  • Contact closures
  • 9 pin mini DIN
  • Serial port on 8 pin mini DIN
  • Power in on 4 pin mini DIN
  • Universal external power supply +24VDC
  • Compliant with worldwide regulations, including FCC, CSA, and CE

When you want to present, broadcast, or record a telephone conversation, you need a device to process the phone call and present it to the console, as well as to separate "send” audio from the "receive” audio on the call. If send and receive audio aren’t isolated, it will result in an echoey, muddy sound - not to mention, annoyed listeners.

As many major markets are shifting from traditional phone lines to Voice-over-IP (VoIP) systems, the digital hybrids that could have performed this job in the past are becoming outmoded. Radio stations need a VoIP hybrid to ensure on-air and recorded phone calls sound beautiful.

A dual-line hybrid, VH2 connects two VoIP lines for individual broadcast or flawless conferencing. VH2 prevents echo and distortion, and automatically adjusts caller audio to a uniform level, leaving you with a result that’s clean and clear. Plus, VH2 uses VoIP phone lines, saving you money and increasing functionality. VH2 can even connect to many VoIP PBX systems.

Main Features:
Audio Processing and Performance:
• Prevents echo and other artifacts
• G.722 codec support for wideband calls. Also supports G.711
• Receive filter reduces telephone line noise
• Selectable automatic gain control (AGC) maintains a consistent audio output, even when the caller signal varies widely
• Selectable caller ducking lowers incoming caller audio so local talent remains in control of the conversation
• Can be configured to automatically answer and disconnect incoming calls

Operation:
• Easily segue from caller-to-caller
• Separately selectable single-ring auto-answer function for assisted or unattended operation
• Handy front panel controls and status indication
• When used with companion VoIP telephone, calls can be answered on handset and easily transferred back and forth to VH2
• Hybrid on/off controls and status remotable via web or contact closures
• Send and caller level indication
• Easy call conferencing
• Dual "On-Hold” audio inputs to send program to callers on hold
• Auto-Switching External Power Supply
• Compliant with worldwide regulations, including FCC, CSA and CE

Audio Configuration:
• Configure for separate caller outs of single caller mix
• Configure for separate send feeds or single
• Pro level, balanced audio I/O in XLR
• Selectable AES3 I/O

IP Features:
• Web-based configuration for the VoIP phone line setup, making it easy to adjust settings remotely from a browser
• Transfer calls back and forth to many PBXs, or use optional companion extension VoIP phone
• Ability to engage or drop or dial calls via web page
• Companion phone - easy to move calls between handset and hybrid with the touch of a button

Bring a more professional sound to your mix Adding a dbx® 266xs Compressor/Limiter/Gate to your live sound rig or studio gives you more dynamic control to help create a more polished, professional sound. Having compression in your audio chain gives you the ability to smooth out uneven levels, add sustain to guitars and fatten up your drums. It also makes it easy to bring vocals to the front of your mix - adding greater clarity and making them stand out from the surrounding instruments. dbx knows compressors...after all we invented them! The 266xs is the latest in a long line of the world's most successful compressors from the inventors of the technology. Its patented Overeasy® compression technology provides smooth and musical performance while the AutoDynamic™ attack and release controls, found only on dbx compressors, puts great sound within easy reach. The 266xs can operate in stereo or dual-mono modes, has true RMS power summing and features quality XLR and 1/4" TRS inputs and outputs. It cuts no corners on visual feedback with gain reduction metering and easy-to-read backlit switches. Features
  • Error proof operation to smooth uneven levels, add sustain to guitars, fatten drums or tighten up mixes
  • New gate timing algorithms ensure the smoothest release characteristics
  • Program-adaptive expander/gates
  • Great sounding dynamics control for any type of program material
  • Separate precision LED displays for gain reduction, compression threshold and gate threshold allow quick, accurate setup
  • Stereo or dual-mono operation
  • Balanced inputs and outputs on 1/4" TRS and XLR connectors
  • Side Chain insert
  • Classic dbx® "Auto" mode
Specifications
Input Connectors 1/4" TRS and female XLR (pin 2 hot)
Input Type Electronically balanced/unbalanced, RF filtered
Input Impedance 40kΩ balanced/unbalanced
Max Input >+22dBu Balanced or Unbalanced
Output Connectors 1/4" TRS, female XLR (pin 2 hot)
Output Type Impedance-balanced/unbalanced, RF filtered
Output Impedance +4dBu: 100Ω balanced, 50Ω unbalanced; -10dBu: 1kΩ balanced, 500Ω unbalanced
Max Output >+21dBu balanced/unbalanced into 2kΩ or greater; >+18dBm balanced/unbalanced (into 600Ω)
Sidechain 1/4" TRS Phone, Normalled: Ring = Output (send); tip = Input (return)
Sidechain Impedance Tip = >10kΩ (Input), Ring = 2kΩ (Output)
Sidechain Max Input Level Tip = >+22dBu (Input)
Sidechain Max Output Level Ring = >+20dBu (Output)
Compressor Threshold Range -40dBu to +20dBu
Compressor Threshold Characteristic Selectable OverEasy® or hard knee
Compressor Ratio 1:1 to Infinity:1
Compressor Attack Time Scalable Program-Dependent AutoDynamic™
Compressor Release Time Scalable Program-Dependent AutoDynamic™
Expander/Gate Threshold Range -60dBu to +15dBu
Expander/Gate Ratio 1:1 to 4:1
Expander/Gate Attack Time

GENERAL DATA
Power Supply 220 / 110V 25 VA
Dimension 434x351x44mm (1 rack unit)
Weight » 4 Kg
MEETING INTERFACE IN / OUT
Connector DB 9 female
Nominal in / out audio level 0 dBm
Input impedance 600 W / 10 KW (selectable)
Output impedance 100 W
* nominal line level: - 6 dBm
** it can vary depending on characteristics of telephone line
All measurements are intended at 1 kHz.
AUDIO PROGRAM IN / OUT
Connectors XLR, electr. balanced
Input impedance 600 W / 10K W (selectable)
Output level * - ¥ ¸ + 16 dBm
Output impedance 100 W
Noise on Receive output 2-Wire separation £ 25 dB**
4-Wire separation £ 70 dB
2-WIRE SECTION
Connector RJ11
Nominal input level - 6 dBm
Nominal output level - 6 dBm
Compensation mode Electronic, transf. decoupled
Impedance 600 W
4-WIRE SECTION
Connector Terminal block
Impedance 600 W
Nominal TX level 0 dBm
Nominal RX level 0 dBm

OPTIMOD-FM 5500 puts coveted five-band and two-band OPTIMOD processing into a single rack unit package and brings it to you at the most affordable price ever.

Quality sound is what 5500 is all about-sound that attracts audiences by providing a polished, professional presentation regardless of format and source material. Exceptional versatility allows you to adjust the processor's audio texture to brand your audio, knowing that the resulting signature sound will remain consistent, cut-to-cut and source-to-source. Branding builds businesses and no other processors have the consistency to brand your sound like OPTIMOD.

With the 5500, your signature sound is just a preset away. An easy, one-knob Less/More adjustment allows you to customize any factory preset, trading cleanliness against processing artifacts according to the requirements of your market and competitive environment. Full Control gives you the versatility to customize your audio further. And, if you're a hard-core processing expert, you can explore Advanced Control to tweak presets at the same level as Orban's factory programmers. This versatility makes the 5500 a superb choice for any format. Its five-band processing is ideal for any pop music format (even the most competitive and aggressive CHR), while phase-linear two-band processing yields ultratransparent sound for classical, classic jazz, and fine arts formats. Regardless of your choice, 5500's optimized technology ensures unusually high average modulation and coverage for a given level of subjective quality. Unlike many lesser processors, the 5500 handles speech particularly well. It's always clean, even when you process for loudness.

If you're concerned about latency because you need to feed live talent headphones off air, be assured that the 5500's ultra-low-latency (5 ms delay) processing will keep the most finicky talent happy. Or use optimum latency (15 ms delay) processing for the most competitive sound with delay that's still low enough to satisfy most any talent.

Versatility doesn't stop with sound.

The 5500 can also be used as a superb stand-alone stereo encoder with latency as low as 2 ms and full overshoot limiting in both the left/right and composite baseband domains.

When used in this mode, the 5500 must be driven (usually via an STL) by a full-featured FM audio processor (like Orban's 8600) that incorporates pre-emphasis aware HF limiting and peak control. In both modes, the 5500's stereo encoder helps deliver a transmitted signal that's always immaculately clean and perfectly peak limited, with full spectral protection of subcarriers and RDS/RBDS regardless of the amount of composite limiting.

It is the ideal choice for network broadcasters who process with Orban's flagship OPTIMOD-FM 8600 at the network origination point and who need a processor at every transmitter to eliminate STL overshoots (using the 5500's stand-alone stereo encoder mode) and/or to process local insertions while also eliminating network STL overshoots (using the 5500's audio processor / stereo encoder mode). Moreover, the 5500's two modes make it easy for large government and network broadcasters to manage its inventory of spares because any 5500 can be used as a stereo encoder with or without audio processing.

Available in both modes, the built-in, defeat-able ITU-BS412 multiplex power controller allows the 5500's output to meet even the most stringent European government regulations.

A 10 MHz frequency reference input allows the stereo pilot tone frequency to be locked to GPS or another high-accuracy frequency standard. This improves the performance of single-frequency networks in areas where coverage of the transmitters overlaps.

The 5500's built-in stereo encoder, AES/EBU digital inputs and outputs, and analog I/O permit hasslefree interfacing to any broadcast plant, whether the 5500 is located at the studio or the transmitter. Tight band limiting to 15 kHz means you can use any uncompressed digital STL to pass 5500- processed audio from studio to transmitter without compromising on-air loudness - there's no need to use STL's having 44.1 or 48 kHz sample-rate.

The stereo encoder's stereo sub-channel modulator can operate in normal double sideband mode and in an experimental compatible single sideband mode (SSB/VSB) that is offered to enable users to compare and assess the two modes.

Analog Fallback to Digital control that allows Silence Sense to switch the active input from Analog to Digital if silence is detected in the analog input signal but not on the digital input signal. This function works vice versa as well on both analog and digital AES input. The silence sense parameters apply to both simultaneously and both detectors are available to drive the 5500's tally outputs and sending SNMP Traps/Alerts.

If you want to locate the 5500 away from the studio, you'll be pleased by its three separate remote control ports - GPI contact closures, RS232 serial and built-in Ethernet for TCP/IP networks. The serial and Ethernet ports are supported by the supplied 5500 PC Remote Control application. This Windows® 2000/XP/Vista/7/8 application allows you to do even more with the 5500 than you can do through its front panel, making remote control a pleasure.

5500 PC Remote software allows you to access all 5500 features and allows you to archive and restore presets, automation lists, and system setups (containing I/O levels, digital word lengths, GPI functional assignments, etc.)

Built-in clock-based automation lets you automatically day-part the processing. You can control many other 5500 operating parameters too.

The 5500's feature set fully exploits the processor's DSP and computer-based control architecture. To ensure absolute accuracy, you can automatically synchronize the clock to an Internet timeserver. It has a cool-running, energy-efficient switching power supply and uses the latest dual-core DSP chip technology from Freescale Semiconductor.

 

The S com 4 is a compact and versatile single-rack space device that provides
four channels of high quality dynamics processing with an Expander/Gate and
Compressor/Limiter on each channel.
The four channels operate independently or in stereo pairs. Its multi-segment LED
metering displays input/output level as well as gain reduction. The Expander/Gate
section features a continuously variable Threshold control as well as a switch for
fast or slow release times.
The Compressor/Limiter section includes variable Threshold, Ratio and Output levels.
S com 4's Enhancer switch restores high frequencies that are sometimes diminished
by heavy compression. This wide-ranging combination of features makes the S com
4 an efficient and versatile audio tool for a wide variety of applications. Its clean and
quiet audio characteristics make it ideal where high sonic integrity is imperative. 
 
FEATURES:

4 Channel Compressor/Limiter, Expander/Gate with Enhancer
Linkable in two Stereo pairs
Expander/Gate with variable Trigger control and switchable Fast/Slow Release
5 Segment LED meters for Input/Output levels and Gain Reduction on each channel
Advanced circuit design utilizing Low Noise Operational
Amplifiers and High Quality VCA's
Servo Balanced Inputs and Outputs on XLR and 1/4" connectors
SKD (Smart Knee Detector) circuit automatically switches from soft to hard knee
based on the level of Input signal applied
Individual Threshold and Ratio controls on each channel
Three-year extended warranty 

L-1102 Leveler/Limiter * Program dependent attack and release for transparent operation. * Noise gate to cut off system background noise when there is no signal. * In limiter mode sense signal is from the amplifier output. * Limits 4 ohm, 8 ohm, 25 V, 70 V and 100 V amplifier outputs. * 20 dB gain in signal leveler mode. * Security cover and rack-mount ears included. * Five year warranty.
Weight1.00 kg3.00 kg4.00 kg9.50 kg3.40 kg4.00 kg
DimensionsN/AN/AN/AN/AN/AN/A
Additional information
Weight 1.00 kg
Weight 3.00 kg
Weight 4.00 kg
Weight 9.50 kg
Weight 3.40 kg
Weight 4.00 kg