Reference10:Interfaces/FXO/Signaling

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.
Alert Tone detection:	After the configured number has been dialed, the FXO normally waits for a audible response from the dialed side that can be identified as an alerting tone. The proper detection and analysis of this tone takes up to a few seconds. If you additionally enable the Assume alert option, the detection is simplified and therefore reduced to a simple tone detection. This means that an alert will be regarded as detected na matter what tone or rhythm is provided. This is much faster than the exact analysis and normally has no negative effect. When an alert is detected, the voip>-FXO->peer connection is switched through (connected), so that the voip initiator of the call now can hear the alerting tone from peer side. Disabling Alert Tone detection will directly connect the call after dialing is complete, which is much faster.
Assume Alert:	see previous topic for details. Assume Alert has no effect when Alert Tone detection is disabled.
Drop line on polarity reversal:	When the FXO port detects a reversed polarity, the current connection will be terminated. Polarity reversals must must be longer than 100ms to be detecable.
CallerID 1 standard:	Selects the standard in which CallerID (also known as CLIP) is detected and decoded. What is of interest here is 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 the FXO port has to be registerd to the PBX as gateway, and has to be configured with Force enblock in the Gateway->Routes Menu. You can change the value of 4000ms to shorter values, but the time must be enough to collect the digits of the longest possible number. This heavily depends on the typical number length - estimate 135ms for each DTMF digit + 1300ms overhead + duration of the first ring pulse. 4000ms will cover 18 DTMF digits if the first ring pulse is very short. Note that for most countries 2 invisible digits are added. This delay time is normally used if no CallerID can be detected at all. However, if FSK or DTMF digits are detected, this time will normally reduce drastically. The voip call then starts with this short delay in which the CallerID/CLIP information is received. The FSK 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'.