Adaptive delta pulse coded modulation. A method of data compression. One of the CODECS supported by the Asterisk PBX. The codec which implements this in Asterisk is known as g726aal2. The audio quality of this codec is superior to GSM, but inferior to uLAW.

ARIB

Analog Radio Interface Board. This is one of the supported of radio interfaces used with app_rpt.

A complete PBX implemented in software conceived and designed by Mark Spencer of Digium Inc. and others. Known as a "soft pbx" as most of its functions are implemented in software.

authentication

A process used by app_rpt to validate incoming connections from other nodes

autopatch

A method of initiating a connection from a radio user to a telephone number.

A device with a T1 interface which de-multiplexes the T1 data into multiple analog DS0 telephone ports.

An asterisk channel driver which allows modified USB sound fobs to be used with app_rpt.

Continuous Tone Coded Squelch System. A set of standardized sub-audio tones used to prevent a receiver from opening due to noise, or transmissions not intended for it.

A DTMF sequence received by a radio receiver on a node and acted upon which causes a change in the operating characteristics of that node.

A mode of operation where DTMF commands are not acted upon locally and instead they are sent to remote node to be decoded and acted upon.

Carrier Operated Relay. This is an archaic term. It is really a "carrier detect" signal. This is usually available from some point in the squelch circuit of the receiver.

A 6db per octave rolloff throughout the entire audio passband of 300-3000Hz. Used as a noise reduction technique for narrow band FM communications.

Dial Pulse Out. A method of signalling used on an old analog DID trunk. In the days of analog telephony, when a caller on the Public Switched Telephone Network placed a call to one of your DID numbers, the CO would use dial pulse signalling to the last 4-5 digits of the telephone number so that your PBX could direct the call to the correct extension.

When the called party answered on the PBX, the PBX would reverse the line polarity to so that the phone company could start billing for the call. With regard to app_rpt, we use the line polarity reversal feature of DPO signalling to tell a half-duplex radio to key and unkey its transmitter.

Direct Inward Dialing. In reference to Asterisk and app_rpt. This is a voip channel from an ITSP which is linked to one or more telephone numbers in the Public Switched Telephone Network. When someone calls your DID number assigned by your ITSP, the call is redirected over the Internet to your Asterisk server. DID's are used with with Asterisk and app_rpt to implement a reverse autopatch.

Audio which comes from the discriminator of a narrow band FM Radio This is also what is referred to as Flat Receive Audio

Digital Signal Processing. Asterisk uses DSP code for DTMF detection, chan_usbradio.c used DSP code for CTCSS encoding and decoding, pre-emphasis, limiting, de-emphasis, and noise squelch.

"Ear and Mouth" Signalling. This is a form of signalling which was used on analog microwave systems and inter-office telephone company trunks. In reference to app_rpt, this form of signalling is used by the radio channels on a Quad Radio PCI Card.

Processed Transmit Audio with pre-emphasis, soft limiting, splatter filtering applied previously and/or Receive Audio without de-emphasis. Flat receive audio usually comes from the discriminator output of a narrow band FM radio, and Flat Transmit Audio is usually inserted on the high side of the deviation pot, or channel element audio input.

Foreign Exchange Station. A type of analog telephone interface. The FXS interface provides talk battery and ringing and is used to connect telephone sets to a PBX

Foreign Exchange Office. A type of analog telephone interface. The FXO interface accepts battery and ringing and is used to connect a PBX to a telephone line going to the phone companies central office.

An audio system where both parties can talk at the same time, and where a receiver and transmitter are operating on a node simulataneously as a repeater.

Global System for Mobile communications. In reference to Asterisk and app_rpt, a type of CODEC used to compress and decompress audio. This codec uses the least amount of bandwidth.

An audio system where only one party can talk at a time, and where a receiver and transmitter on a node are not capable of operating simultaneously. This is usually used when a remote base is implemented.

Inter Asterisk Exchange. Used to connect Asterisk Servers to each other. A TCP/IP protocol using UDP datagrams on port 4569.

Internet Telephone Service Provider. An entity which allows VOIP traffic to be connected to the Public Switched Telephone Network and/or to other VOIP users. Traffic from the VOIP user to the Public Switched Telephone Network is called termination, and traffic from the Public Telephone Network to the VOIP user is called origination. ITSP's may provide origination only, termination only, or both origination and termination services.

A access point in a network of Asterisk systems running app_rpt usually having a receiver and transmitter interfaced. A node can connect to any other node, and also forward radio transmissions through itself to other nodes. All nodes have a unique address assigned by a node assignment authority

An organization which coordinates the assignment of app_rpt node numbers. An example would be http://allstarlink.org

A filter in the transmit audio processing chain which has a 6 db per octave roll-up from 300-3000Hz. It is used as a noise reduction technique for narrow band FM communications.

Push To Talk. This is a control signal from the radio interface to the transmitter which causes the transmitter to transmit.

A system consisting of a receiver and transmitter capable of simultaneously receiving and transmitting (i.e in full duplex).

An interface which allows app_rpt to connect to one or more radios. There are 3 types of radio interfaces for app_rpt which are currently supported. The Analog Radio Interface (ARIB), the Quad Radio PCI Card, and the Midufued USB Sound Fob.

Audio from from the receiver to the radio interface. May be either flat audio or speaker audio.

A function which lets a person calling on a telephone to access the radio system and talk to users on the radio system.

A limiting circuit used to limit the transmit deviation to within the channel bandwidth.

Receive audio which has been through de-emphasis. This is not necessarily taken from the speaker terminals of

the radio, it can be taken from a lower level output stage in the radio, but it must be after de-emphasis.

A filter in the transmit audio processing chain used to attenuate the high frequency components generated by the splatter filter.

A PCI card used with the Asterisk PBX and the Zaptel drivers which provides one or more T1 spans.

Audio which comes from the radio interface and goes to the transmitter microphone, or transmitter modulator.

User I/O. These are logic level signals used on the Quad Radio PCI Card to control frequency agile radios and to allow an external CTCSS decoder to be used.

A codec used in Asterisk which does minimal compression, but has the best sound quality, but it uses more bandwidth than ADPCM or GSM.

Zapata Telephony. A set of hardware interface cards and drivers compatible with the Asterisk PBX.