Reference11r1:Interfaces/FXO/Signaling

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The call signalling settings of the analogue FXO interfaces can be made here:

Disable: Disables the relevant analogue FXO interface.
Speech Bearer Capability: Calls on the relevant interface are transmitted with Audio Bearer Capability as standard.
A checked check box transmits calls from the relevant interface with Speech Bearer Capability. This only makes sense if only telephones are operated on the relevant interface (no fax machine or modem).
Central Office Tone detection: When the FXO initiates a call, normally a audible tone is expected to be received from the central office/local PABX side. The tone detection works independantly from any country setting. When the expected tone cannot be detected within 2.6sec, then the call will be cancelled. If this tone detection is disabled, the FXO will continue dialing DTMF tones of the configured number, with a fixed delay of 800ms after hook-off.
Terminate ringing calls which failed to reach its destination by offhook-onhook sequence: Calls that are ringing on the FXO port and for any kind of reason failed to reach its destination on the IP side normally have no chance to get informed of that miss. Instead the call would stay in ringing state on the FXO port for long time. This option allows the FXO to react on a failed call to IP with a short offhook-onhook sequence on the FXO port and therefore terminate the ringing call immediately.
Establish call on line polarity reversal: When the FXO port detects a reversed polarity and the FXO port is idle, a new call wil be established. Countries that use DTMF CallerId also often use a Polarity Reversal preceding the DTMF CLIP sequence and start with AC ringing afterwards. In detail the sequence is as follows : Line polarity reversed, 200ms pause, DTMF CallerId transmission, Line polarity normal, 1000ms pause, AC ringing, pause ... . For instance Sweden is one of those countries that use DTMF CallerId combined with Line polarity reversal call establishment. If this checkmark is unchecked here, the call will miss the DTMF CallerId sequence, when established with the AC ringing. Note that this option can also be used with FSK CallerId.
PSTN drives conversation state in reversed line polaritity: Calls from FXO->PSTN normally switch to conversation/connected state within 4seconds after dialing. If this option is checked AND the PSTN changes its line polarity while a connection to the dialed peer is established, then the FXO can recognize the reversed line polarity and switch to connected state prior to the elapse of 4seconds. Once in connected state, the FXO can detect line polarity switched back to normal polarity and assume a peer disconnect without having detected any kind of call-progress-tone (e.g. busy-tone). This is faster and in some countries the busy-tone at the end of a call is simply omitted, when polarity switching is used. If you use Establish call on line polarity reversal, be shure if your provider additionally uses polarity reversals in conversation/connected state or not. If your provider does, and you miss to check this option, this might lead to strange but short and self-ending re-calls after each hook-on. This comes from the PSTN's final switch to the normal line polarity, which then leads to a wrong call. When checked, the FXO will wait for the PSTN's line polarity change at the end of the call. This behaviour is common for PSTN in Sweden.
Volume: Sets the output volume to the relevant interface (from IP to FXO), in decibel (dB), between -32dB and +32dB. No value or the value 0 is equal to the factory settings.
Caller ID 1 standard: Selects the standard in which CallerId/CLIP is detected and decoded. CallerId carries the caller's phone number in case of an incoming/ringing call. This number is fed into the resulting voip call as the CGPN (Calling Party Number).
Note that for this reason incoming calls from the FXO port must use en-bloc delay dialling to allow for receipt of the CLI. This may be done by registering the Port with a Gateway type PBX object and set the Enblock Count option in this object or use a route with Force enblock set in the Gateway->Routes Menu.

FSK CallerId standards can contain additional information like Called Line Id or Date/Time information. These additional informations will be dropped. DTMF CallerId only contains the Calling Line Id.

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