Phone In 6
(Available July 2012, contact Sonifex for details)
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Top View |
Iso Front View |
Iso Rear View |
Rear View |
PI-6R Remote Control Unit.
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Iso Front View |
Iso Rear View |
Front View |
Rear View |
PI-6C Base Console Unit.
Phone In 6
The Phone In 6 is a call control centre for phone-in radio shows. It is capable of interfacing to 2, 4, or 6 phone lines by fitting up to 3 modules, where each module can have 2 x PSTN lines or 1 x ISDN BRI line. The ISDN basic rate interface allows 2 calls to be handled, 1 on each B channel. The PSTN module has 2 interfaces each using a modern digital hybrid interface, which is settable by software to handle a variety of PSTN and PABX systems with varying impedances, call connection & disconnection tones.
The unit comprises 2 boxes: The PI-6C 1U rack mounted Base Console (BC) unit which has all the audio & telephony connections together with the metering and complex controls.
Secondly, there is the PI-6R Remote Control Panel (RCP) which connects via Ethernet based cabling to the main unit and allows the user to see the status of each caller, to receive an incoming call and to subsequently route it to the mixing desk. It also includes connectors for a headset that enable remote dialling out and the screening of incoming calls.
The Base Console has 3 module positions on the rear panel into which either of the PI-PSTN2 (2 x analogue telephone lines) or PI-ISDN2 (2 x ISDN B channels) modules can be fitted. The Base Console is capable of connecting internal DSP and hybrid circuits to any 2 lines and routing these out to a studio mixing desk.
The Phone In 6 uses an echo-cancellation DSP (digital signal processing) algorithm and impedance matching to give around 70dB cancellation, which provides excellent separation of caller & line send audio and elimination of feedback, distortion & echo on the incoming calls. This is close to being the best performance possible on a telephone line & uses the same enhanced echo cancellation algorithms as used on the DHY-03 telephone hybrid, the best performing telephone hybrid in the world. The DSP, which uses 24bit variables and at least 48bit math algorithms, can also monitor the call progress tones for various disconnection arrangements. The audio data is supplied to the DSP from analogue inputs via 24bit ADCs.
Phone In 6 Operation
The Base Console is connected to the incoming lines; to a music-on-hold source; to 1 or 2 channels on a mixing desk and to the Remote Control Panel. The Base Console accepts an incoming call and indicates the ring to the Remote Control Panel. The call is either automatically answered by the unit and supplied with music-on-hold, or is answered manually by the user. The user has the following options:
- Send the caller to the mixing desk by assigning it to an on-air channel.
- Place the caller on-hold and send them the music-on-hold signal .
- Drop the call.
During the initial call control there is limited processing being performed by the unit, but once the call is assigned to an on-air channel, the audio is passed through to the DSP to provide the highest quality possible for the station output.
DSP Features Of The Phone In 6 System
- Echo-cancellation which cleans the telephone signal to remove any echoes from the analogue portions of the call and ensures that the mixing desk is only sent the caller audio.
- AGC (automatic gain control) & a Noise Gate which sets the level of the caller audio to a suitable level for the mixing desk and also reduces the noise floor when the caller is silent.
- Ducking so when both caller and presenter are talking together the DSP can automatically reduce the level of the caller so that the presenter is always able to talk over the caller, e.g. to gracefully end a heated call;
- In conference mode the DSP mixes the other caller audio into the sent signal so that each caller can hear the station output and the other caller in the conference call.
Conferencing Calls & Use of the Cleanfeed/Mix-Minus
The Phone In 6 handles 2 live calls at once and normally each call is connected to a separate telco channel on a mixing desk. Each channel receives the caller audio and allows the presenter to talk to the caller off-air (usually via the pre fade listen (PFL) or Cue selection on the mixer), or to place the caller to air. In both cases the mixing desk outputs audio back to the caller via the Phone In 6 unit – this is usually the signal from the presenter’s microphone or a special mix of the station output.
The unit is configured in 2 ways:
Normal Mode where 2 calls can be handled independently, each with its own caller output and cleanfeed input.
Conference Mode where up to 2 calls are mixed within the unit and are presented as a single audio stream on the output with a single cleanfeed input.
The cleanfeed, also known as mix-minus, is a specific mix from the desk/mixer that contains all of the radio station output except for the caller himself or herself. In Conference Mode the unit provides a mix of both callers and the cleanfeed is a mix without either caller. The unit mixes in the other caller for each signal sent out to the callers.
In Normal Mode the unit should be connected to 2 telco channels that each generate their own cleanfeed output. If the mixing desk is able to generate a cleanfeed without either caller from the 2 independent caller outputs, then the unit can accept a single input and add in the other caller for the appropriate caller output the same as is performed in Conference Mode – this is known as Shared Mode.
Call Handling - Self-op & Call Screening Modes
The Phone In 6 allows radio shows to be organised in 2 main ways:
- Self-op where the show is entirely controlled by the presenter.
- Call-screening where an assistant or producer deals with incoming calls manually and then places the calls to the presenter via the on-air selections.
Self-op Mode
In Self-op Mode the unit can automatically answer calls and place them in a queue where they are supplied with the music on hold signal. When the presenter is ready, they can accept the next caller to air. The queuing system can be set to use 1 or 2 hybrid channels, though if the mixing desk has only one telco module available we advise that you do not use conferencing and queue to a single channel only. If 2 channels are available and you configure it to use only 1 hybrid for the queue, then the second channel is available for manual intervention, for example to save a call when the presenter needs further caller details as a prize winner, or to make outgoing calls.
Auto-Answer & Auto-Drop Calls
Calls are auto-answered and by designating one of the GPIO remote input ports to receive a ‘drop call’ command when the channel is switched off, the call is dropped once the conversation with the presenter has finished. The calls can be shuffled up so that line 1 is designated as the on-air 1 caller, then on-air 2, then unanswered calls in their queue order. Or the ON-AIR 1 and ON-AIR 2 (if used) buttons show the current calls to the mixing desk and the next call flashes its LINE HOLD button yellow.
Call Screening Mode
In Call-screening Mode an assistant/producer can answer calls and then the caller can be put on-hold and supplied with the Music On Hold signal until the presenter is ready to take the call. The call-screener communicates to the presenter by the normal talkback interface on the mixing desk. The procedure to manually handle calls is described below
Receiving a Call
When the unit receives a call, the LINE HOLD button flashes red for that line. The call handler should press the button to accept the call, which stops the flashing, illuminating the button green and routes it to the Music On Hold signal.
If the call handler is ready to talk to the caller straight away, then pressing the LOCAL button followed by the LINE HOLD button accepts the call, lights both buttons yellow and routes it to the call handler’s headset. The call can then be put on hold by pressing the LOCAL button again, which clears the LOCAL button and makes the LINE HOLD button go green.
Routing a Call
To route a call to the mixing desk, press the ON-AIR button associated with that line. The ON-AIR button illuminates green and the audio to and from the caller is routed through the DSP hybrid and to/from the mixing desk. The unit can be programmed to send a signal to the mixing desk from the GPIO port to allow the telco channel on the mixer to indicate that a call is present.
If the hybrid is already in use, or the unit is in conference mode and 2 calls are already being handled, then the ON-AIR button flashes red briefly and returns to the off state. To remove a call from the hybrid press & hold the ON-AIR button for a second and the caller is returned to the Music On Hold state
Dropping a Call
To drop a call, simply press & hold the LINE HOLD button for a second and the call is dropped.
Making a Call
To make a call press the LOCAL button followed by an unused line’s LINE HOLD button. The headset is now connected to the line and the call handler should hear the dial tone. Use the keyboard to direct dial a call using DTMF dialling or use the hash (#) key to use the phonebook entries or ## to repeat the last used number.
Phone In 6 Configuration
The Phone In 6 comprises 2 units, a Base Console (BC) and a Remote Control Panel (RCP) communicating through a TCP/IP network. The units each need a TCP/IP address to communicate and also need to be paired to work together. The configuration is performed entirely at the RCP where the TCP/IP address is set in static mode or set to use a local DHCP server. The RCP can then search for base units attached to the system. It may find a number of connected BC devices and the serial number and state of each unit’s connectivity is indicated. Choose the appropriate BC and designate it as the pair for the RCP. This unit can then also be configured for its TCP/IP address.
The system can then be set up to operate in the manner required using the menu system on the RCP. Some of the setup is specific to the actual install, e.g. country type for the hybrid settings, ring cadences & disconnect tones supplied by the network provider. Other settings can be program specific, e.g. call screening/self-op mode. The latter settings can change with the presenter, or as a particular show changes, so multiple sets of these parameters can be saved into permanent memory within the system which the user can easily store and change by using the * key and a single digit number to update those parameters.
The RCP has the ability to make calls to unused lines and there is a Phonebook that allows the user to create 99 preset numbers which are easily accessed by using the # key followed by the 2 digit number. Also a double # key calls the last dialled number.
