Talk-Show and Multi-Conference System on radio and TV

SYSTEL IP is a broadcast and multiconference telephony system that significantly reduces the cost of communications, significantly improves quality, increases flexibility and integration with each station’s or company’s own telephone system. It involves a very low investment, with a rapid return on investment.

Based on 4-wire digital matrix. Allows simultaneous interventions
Big savings when connecting to Internet telephony providers
Flexible and dynamic sharing of IP lines with up to 4 studios
IP audio inputs and outputs (Dante, AES 67 compliant)
Different control terminals: touch phone, App for PC
Huge installations, with dozens of studios, with IP switchboards



The base of the system is a 19″ rack format equipment, with 4 digital inputs and 4 outputs, 2 analogue inputs and 2 analogue outputs and 32 IP inputs and outputs DanteTM/AES67 protocol. 4 IP lines for operator telephone. The equipment behaves as a multi-line IP telephone with SIP protocol signalling. Compatible with IP PBX, SIP Trunking and virtual PBX. Supports analogue and ISDN lines through gateways.



For 8 simultaneous IP telephone lines (expandable to 16).



For 16 simultaneous IP telephone lines.


Control Applications

Systel IP Original 

Consists of a control application on PC and a conventional IP phone. Based on call queuing, usually for radio. Best choice in high productivity environments with distributed functionality in roles such as producers, controllers and presenters.

Systel IP TV

Similar in appearance to the Original, it is multiplex-based, to provide access to intercom systems, or technical outdoor intercommunication facilities, more commonly used for television.


The operating application integrated into the SYSTELSET+ touchscreen phone allows for very flexible operation and avoids the presence of a PC at the controls or in other confined spaces.

Technical Details

General Features


SIP communications protocol: compatible with VoIP trunkings, free PBX, SIP Phones such as Phoenix Pocket or Phoenix Lite, N/ACIP compliant Audiocodecs such as Phoenix Mercury, Phoenix Studio, Phoenix Venus or Phoenix ALIO and POTS, ISDN, E1 and T1 FXO.

Based on non-blocking digital switching matrix, all16 lines can be simultaneously live participating in a program with no loss of quality.

GPI/Os: 4 GPI, 4 GPO and power supply on each DB15 female connector. All
functions are replicated over TCP / IP in the control network.

Audio Specifications

– Analog inputs: input impedance: 20Kohm. Electronically balanced, professional line level.
– Nominal input level: +4 dBu. Max. input level: +24 dBu.
– Analog outputs: output impedance < 100 ohm. Electronically balanced, professional line level.
– Nominal output level: +4 dBu. Max. output level: +24 dBu.
– Digital inputs / outputs: AES / EBU interfaces, configurable as AES-3 or SPDIF. Inputs include SRC.
– AES 1 input can be used for external AES-11 synchronization.

Audio Processing

– Phone audio in G.711, G.726, G.729, 50Hz – 3KHz.
– High-Definition audio with G.722 algorithm: 50Hz – 7KHz.
– Echo cancellation. Automatic gain control.
– Independent, digital gain control for all inputs and outputs with an adjustment range of +/- 12 dB and muting.
– Automatic gain control for telephone returns.

Configuration software and control server


–  Assigns audio, IP phone and chat circuits to the different studios, univocally.
–  Renames circuits.
–  Defines and manages phone books, allowing the user to share, edit and copy them.
–  Defines PFL signals assigned to each studio.
–  Defines auxiliary and master signals assigned to each studio.
–  Configure the initial audio levels for each line and each study.
–  Configures the format of the client screens, defining the number of lines per program, console operation, and the use of one or two call queues.
–  SIP configuration for communication with an IP PBX, FXO gateway and external
(Internet) or internal (LAN or WAN) service providers.
–  Distinguish and protect with rights on activities the functionalities of different types of user.

SYSTEL IP Original and SYSTEL IP TV control clients


– Call establishment: by number dialling, with SIP identifiers, or from phone book
– Call establishment: by dialing IP phone numbers taking advantage of the IP
handset functionality.
– Optical tally and acoustic RING signal.
– Caller ID. Contact list matching. Adding of a temporary name.
– Pick up incoming calls manually and automatically.
– Define and manage phone books, either general or private to each program.
– Create and manage phone call schedules.
– Queue the on-air ready calls on one or several faders, allowing for their
re-ordering and dynamically checking them.
– Grant the VIP attribute to a call in order to keep it on a dedicated fader.
– Accept incoming calls, either manual or automatically.
– Register new contacts in the call book.
– Talk by means of the headset or microphone / headphone with the people at the
remote line end.
– Put calls on hold, while the caller can listen to the program.
– Put calls on-air so they can contribute to the program.
– Place several call on CUE on one or several Faders calls ready to be placed on-air,
allowing for the dynamic reordering of calls and TalkBack.
– Tag a call as VIP, thus making it exclusive to a fader.
– Changes input and return levels for every phone line in the studio.
– Changes input and return levels for every phone line in the studio and every
– Display the status of all the phone lines and where they are being routed to.
– Label calls. Chat among the different controllers assigned to the program (Only Original version). Black-list management. Call-barring.

SYSTELSET+ Control client


– Making calls: dialing numbers, SIP identiers or register in phone books and
– Making calls: dialing telephone numbers.
– Sending an optical and acoustic RING signal.
– Showing the caller’s ID or number, substituted by the local name in the phone book or a temporary name assigned from SYSTEL IP Classic application.
– Answering incoming calls automatically or manually.
– Registering new contacts in the phone book.
– Managing general and program-private phone phone books.
– Managing phone book schedules.
– Talking by means of micro-earphone, hands-free or micro-headphone with the person at the other end of the line.
– Leaving calls on-hold while listening to the program.
– Putting calls on air so they can contribute to the program.
– Queuing on one or several faders the calls that are ready to be put on air, allowing for its re-ordering and dynamic consulting them.
– Assigning the VIP attribute to a call so it can be kept on a dedicated fader.
– Changing the phone listening levels and input and return levels for each of the studio’s phone lines.
– Displaying the status of each phone lines and where are they being routed, while they are kept in that status.
– Distinguishing and protecting with rights the functionalities assigned to producers, operators and presenters.
– Managing black lists and incoming call barring.
– Working with typical European and USA naming and functions.
– Activating the Dump Mode, in order to hang up or not calls after they have been on-air.
– Activating Page Lines, in order to send warnings to all lines and receive simultaneous replies.
– Activating Auto Next, that puts the next call on-air after one has been hung up.
– Activating PickUp Incoming in order to automatically connect to the oldest call when picking up the handset.
– Activating AutoQueue, in order to automatically queue the call when hanging the handset.
– Activating Direct Dial, in order to skip the “select line” step to make a call.
– Activating Direct Next, that puts on air the calls even if they have not been attended previously.

SYSTEL IP16 “16-lines IP engine”


-Power supply. Universal 100-240 V. 50/60 Hz. 50 VA power supply.
– Silent operation: natural convection cooling.
– Weight: 4 Kg (8,8 lbs).
– Width: 482 mm (19“) 1U rack height = 44 mm. (1,75”).
-Depth: 356 mm. (14”).

Inputs and Outputs

-DB15 female audio multi-connectors. Two I/O each.
– 2 analog balanced inputs.
– 2 analog balanced outputs.
– 2 digital AES- EBU (AES3 or SPDIF) dual inputs.
– 2 digital AES- EBU (AES3 or SPDIF) dual output.
– 1 WAN IP port for 16 VoIP lines, plus 4 VoIP lines for control phones.
– 2 LAN IP ports for control and 32 AoIP inputs / outputs in redundant Dante / AES-67 format.
– 1 DB15 connector for 4 opto-coupled GPI and 4 GPO each one.

SYSTELSET+ IP phone with preloaded control application


– 7” multi-touch screen.
– 8 pre-programmed function keys.
– SYSTELSET+ pre-loaded, running on Android 5.1.1.
– 12-key phone keyboard.
– Dual Gigabit Ethernet port: 10/100/1000Mbps.
– Headphone connector: 1 x RJ9 (4P4C).
– Handset connector: 1 x RJ9 (4P4C).
– USB 2.0 port for wireless or USB headphones.
– HD Voice.
– Hands-free.
– External power supply: 100-240 VAC 5V. DC and PoE (IEEE 802.3af), class 3 (max 6W).
– Dimensions (W* D* H* T): 259.4mm * 235.2mm * 194.5mm * 42.6mm.
– Weight: 916 gr.

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