Reference10:Interfaces/FXO/Signaling: Difference between revisions

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|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.
|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.
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|valign=top nowrap=true|'''Alert Tone detection:'''
|valign=top nowrap=true|'''Establish call on line polarity reversal:'''
|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.<br>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.<br>Disabling '''Alert Tone detection''' will directly connect the call after dialing is complete, which is much faster.
|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.
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|valign=top nowrap=true|'''Assume Alert:'''
|valign=top nowrap=true|'''PSTN drives conversation state in reversed line polaritity:'''
|see previous topic for details. '''Assume Alert''' has no effect when '''Alert Tone detection''' is disabled.
|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.
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|valign=top nowrap=true|'''Drop line on polarity reversal:'''
|valign=top nowrap=true|'''Volume:'''
|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.
|Sets the volume for the relevant interface, in decibel (dB), between -32dB and +32dB. No value or the value 0 is equal to the factory settings.
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|valign=top nowrap=true|'''CallerID 1 standard:'''
|valign=top nowrap=true|'''Caller ID 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).<br>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.
|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).<br>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.  
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.<br>
<!-- 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.
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'.
The 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.<br>-->
'''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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Latest revision as of 20:41, 22 October 2014


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.
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 volume for the relevant interface, 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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