[svn-commits] mvanbaak: branch group/appdocsxml r129119 - /team/group/appdocsxml/
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Tue Jul 8 15:20:53 CDT 2008
Author: mvanbaak
Date: Tue Jul 8 15:20:53 2008
New Revision: 129119
URL: http://svn.digium.com/view/asterisk?view=rev&rev=129119
Log:
temp add the generated file.
This should not be merged, and I'll prolly remove it again soon
Added:
team/group/appdocsxml/documentation.xml (with props)
Added: team/group/appdocsxml/documentation.xml
URL: http://svn.digium.com/view/asterisk/team/group/appdocsxml/documentation.xml?view=auto&rev=129119
==============================================================================
--- team/group/appdocsxml/documentation.xml (added)
+++ team/group/appdocsxml/documentation.xml Tue Jul 8 15:20:53 2008
@@ -1,0 +1,665 @@
+<?xml version="1.0" encoding="UTF-8"?>
+<docs>
+ <application name="AMD" language="en">
+ <synopsis>
+ Attempt to detect answering machines
+ </synopsis>
+ <description>
+ This application attempts to detect answering machines at the beginning
+ of outbound calls. Simply call this application after the call
+ has been answered (outbound only, of course).
+ When loaded, AMD reads amd.conf and uses the parameters specified as
+ default values. Those default values get overwritten when the calling AMD
+ with parameters.
+ </description>
+ <variable name="AMDSTATUS">
+ This is the status of the answering machine detection
+ <value name="MACHINE" />
+ <value name="HUMAN" />
+ <value name="NOTSURE" />
+ <value name="HANGUP" />
+ </variable>
+ <variable name="AMDCAUSE">
+ Indicates the cause that led to the conclusion
+ <value name="TOOLONG">
+ Total Time
+ </value>
+ <value name="INITIALSILENCE">
+ Silence Duration - Initial Silence
+ </value>
+ <value name="HUMAN">
+ Silence Duration - afterGreetingSilence
+ </value>
+ <value name="LONGGREETING">
+ Voice Duration - Greeting
+ </value>
+ <value name="MAXWORDLENGTH">
+ Word Count - maximum number of words
+ </value>
+ </variable>
+ <option name="initialSilence">
+ Is maximum initial silence duration before greeting.
+ If this is exceeded set as MACHINE
+ </option>
+ <option name="greeting">
+ is the maximum length of a greeting.
+ If this is exceeded set as MACHINE
+ </option>
+ <option name="afterGreetingSilence">
+ Is the silence after detecting a greeting.
+ If this is exceeded set as HUMAN
+ </option>
+ <option name="totalAnalysis Time">
+ Is the maximum time allowed for the algorithm
+ to decide HUMAN or MACHINE
+ </option>
+ <option name="miniumWordLength">
+ Is the minimum duration of Voice considered to be a word
+ </option>
+ <option name="betweenWordSilence">
+ Is the minimum duration of silence after a word to
+ consider the audio that follows to be a new word
+ </option>
+ <option name="maximumNumberOfWords">
+ Is the maximum number of words in a greeting
+ If this is exceeded set as MACHINE
+ </option>
+ <option name="silenceThreshold">
+ How long do we consider silence
+ </option>
+ <option name="maximumWordLength">
+ Is the maximum duration of a word to accept.
+ If exceeded set as MACHINE
+ </option>
+ </application>
+
+ <application name="ChanIsAvail" language="en">
+ <synopsis>
+ Check channel availability
+ </synopsis>
+ <description>
+ This application will check to see if any of the specified channels are available.
+ </description>
+ <variable name="AVAILCHAN">
+ The name of the available channel, if one exists
+ </variable>
+ <variable name="AVAILORIGCHAN">
+ The canonical channel name that was used to create the channel
+ </variable>
+ <variable name="AVAILSTATUS">
+ The status code for the available channel
+ </variable>
+ <option name="a">
+ Check for all available channels, not only the first one
+ </option>
+ <option name="s">
+ Consider the channel unavailable if the channel is in use at all
+ </option>
+ <option name="t" implies="s">
+ Simply checks if specified channels exist in the channel list
+ </option>
+ </application>
+ <application name="ChanIsAvail" language="nl">
+ <synopsis>
+ Kontroleer beschikbaarheid van een kanaal
+ </synopsis>
+ <description>
+ Deze applicatie kontroleerd of een kanaal beschikbaar is.
+ </description>
+ <variable name="AVAILCHAN">
+ De naam van het beschikbare kanaal, indien voorhanden
+ </variable>
+ <variable name="AVAILORIGCHAN">
+ De volledige naam die gebruikt is om het kanaal op te zetten
+ </variable>
+ <variable name="AVAILSTATUS">
+ De status code van het beschikbare kanaal
+ </variable>
+ <option name="a">
+ Kontroleer alle kanalen ipv alleen het eerste kanaal
+ </option>
+ <option name="s">
+ Als het kanaal in gebruik is, neem aan dat het niet beschikbaar is
+ </option>
+ <option name="t" implies="s">
+ Kontroleer alleen of het kanaal in de kanaallijst bestaat
+ </option>
+ </application>
+
+ <application name="Dial" language="en">
+ <synopsis>
+ Place a call and connect to the current channel.
+ </synopsis>
+ <description>
+ This application will place calls to one or more specified channels. As soon
+ as one of the requested channels answers, the originating channel will be
+ answered, if it has not already been answered. These two channels will then
+ be active in a bridged call. All other channels that were requested will then
+ be hung up.
+
+ Unless there is a timeout specified, the Dial application will wait
+ indefinitely until one of the called channels answers, the user hangs up, or
+ if all of the called channels are busy or unavailable. Dialplan executing will
+ continue if no requested channels can be called, or if the timeout expires.
+ This application will report normal termination if the originating channel
+ hangs up, or if the call is bridged and either of the parties in the bridge
+ ends the call.
+
+ If the OUTBOUND_GROUP variable is set, all peer channels created by this
+ application will be put into that group (as in Set(GROUP()=...).
+ If the OUTBOUND_GROUP_ONCE variable is set, all peer channels created by this
+ application will be put into that group (as in Set(GROUP()=...). Unlike OUTBOUND_GROUP,
+ however, the variable will be unset after use.
+ </description>
+ <variable name="DIALEDTIME">
+ This is the time from dialing a channel until when it is disconnected.
+ </variable>
+ <variable name="ANSWEREDTIME">
+ This is the amount of time for actual call.
+ </variable>
+ <variable name="DIALSTATUS">
+ This is the status of the call
+ <value name="CHANUNAVAIL" />
+ <value name="CONGESTION" />
+ <value name="NOANSWER" />
+ <value name="BUSY" />
+ <value name="ANSWER" />
+ <value name="CANCEL" />
+ <value name="DONTCALL">
+ For the Privacy and Screening Modes.
+ Will be set if the called party chooses to send the calling partey to the 'Go Away' script.
+ </value>
+ <value name="TORTURE">
+ For the Privacy and Screening Modes.
+ Will be set if the called party chooses to send the calling partey to the 'torture' script.
+ </value>
+ <value name="INVALIDARGS" />
+ </variable>
+ <option name="Technology/Resource" required="true" argsep="&">
+ <argument name="Technology2/Resource2">
+ Optional extra 'devices' to dial.
+ If you need more then one enter them like this:
+ Technology2/Resource2&Technology3/Resourse3&.....
+ </argument>
+ Device to dial
+ </option>
+ <option name="A">
+ <argument name="x" required="true">
+ The file to play to the called party
+ </argument>
+ Play an announcement to the called party, using 'x' as the file
+ </option>
+ <option name="C">
+ Reset the CDR for this call.
+ </option>
+ <option name="c">
+ If DIAL cancels this call, always set the flag to tell the channel
+ driver that the call is answered elsewhere.
+ </option>
+ <option name="d">
+ Allow the calling user to dial a 1 digit extension while waiting for
+ a call to be answered. Exit to that extension if it exists in the
+ current context, or the context defined in the EXITCONTEXT variable,
+ if it exists.
+ </option>
+ <option name="D" argsep=":">
+ <argument name="called" />
+ <argument name="calling" />
+ Send the specified DTMF strings *after* the called\n
+ party has answered, but before the call gets bridged. The 'called'
+ DTMF string is sent to the called party, and the 'calling' DTMF
+ string is sent to the calling party. Both parameters can be used
+ alone.
+ </option>
+ <option name="e">
+ execute the 'h' extension for peer after the call ends
+ </option>
+ <option name="f">
+ Force the callerid of the *calling* channel to be set as the
+ extension associated with the channel using a dialplan 'hint'.
+ For example, some PSTNs do not allow CallerID to be set to anything
+ other than the number assigned to the caller.
+ </option>
+ <option name="F" argsep="^">
+ <argument name="context" />
+ <argument name="exten" />
+ <argument name="pri" required="true" />
+ When the caller hangs up, transfer the called party
+ to the specified context and extension and continue execution.
+ </option>
+ <option name="g">
+ Proceed with dialplan execution at the current extension if the
+ destination channel hangs up.
+ </option>
+ <option name="G" argsep="^">
+ <argument name="context" />
+ <argument name="exten" />
+ <argument name="pri" required="true" />
+ If the call is answered, transfer the calling party to
+ the specified priority and the called party to the specified priority+1.
+ Optionally, an extension, or extension and context may be specified.
+ Otherwise, the current extension is used. You cannot use any additional
+ action post answer options in conjunction with this option.
+ </option>
+ <option name="h">
+ Allow the called party to hang up by sending the '*' DTMF digit.
+ </option>
+ <option name="H">
+ Allow the calling party to hang up by hitting the '*' DTMF digit.
+ </option>
+ <option name="i">
+ Asterisk will ignore any forwarding requests it may receive on this
+ dial attempt.
+ </option>
+ <option name="k">
+ Allow the called party to enable parking of the call by sending
+ the DTMF sequence defined for call parking in features.conf.
+ </option>
+ <option name="K">
+ Allow the calling party to enable parking of the call by sending
+ the DTMF sequence defined for call parking in features.conf.
+ </option>
+ <option name="L" args="x,y,z" argsep=":">
+ <argument name="x" required="true">
+ Maximum calltime in miliseconds
+ </argument>
+ <argument name="y" />
+ <argument name="z" />
+ Limit the call to 'x' ms. Play a warning when 'y' ms are
+ left. Repeat the warning every 'z' ms.
+ <variable name="LIMIT_PLAYAUDIO_CALLER">
+ <value name="yes" default="true" />
+ <value name="no" />
+ Play sounds to the caller.
+ </variable>
+ <variable name="LIMIT_PLAYAUDIO_CALLEE">
+ <value name="yes" />
+ <value name="no" />
+ Play sounds to the callee.
+ </variable>
+ <variable name="LIMIT_TIMEOUT_FILE">
+ <value name="filename">
+ If not set, the time remaining will be said.
+ </value>
+ File to play when time is up.
+ </variable>
+ <variable name="LIMIT_CONNECT_FILE">
+ <value name="filename">
+ If not set, the time remaining will be said.
+ </value>
+ File to play when call begins.
+ </variable>
+ <variable name="LIMIT_WARNING_FILE">
+ <value name="filename">
+ If not set, the time remaining will be said.
+ </value>
+ File to play as warning if 'y' is defined.
+ </variable>
+ </option>
+ <option name="m">
+ <argument name="class" />
+ Provide hold music to the calling party until a requested
+ channel answers. A specific MusicOnHold class can be
+ specified.
+ </option>
+ <option name="M" args="x,arg" argsep="^">
+ <argument name="x" required="true">
+ Macro name that should be executed.
+ </argument>
+ <argument name="arg">
+ Macro arguments seperated by ^
+ </argument>
+ Execute the Macro for the *called* channel before connecting
+ to the calling channel. Arguments can be specified to the Macro
+ using '^' as a delimiter. The Macro can set the variable
+ MACRO_RESULT to specify the following actions after the Macro is
+ finished executing.
+ <variable name="MACRO_RESULT">
+ If set, this action will be taken after the macro finished executing.
+ <value name="ABORT">
+ Hangup both legs of the call.
+ </value>
+ <value name="CONGESTION">
+ Behave as if line congestion was encountered.
+ </value>
+ <value name="BUSY">
+ Behave as if a busy signal was encountered.
+ </value>
+ <value name="CONTINUE">
+ Hangup the called party and allow the calling party to continue dialplan execution at the next priority.
+ </value>
+ <value name="GOTO:<context>^<exten>^<priority>">
+ Transfer the call to the specified priority. Optionally, an extension, or extension and priority can be specified.
+ </value>
+ </variable>
+ You cannot use any additional action post answer options in conjunction
+ with this option. Also, pbx services are not run on the peer (called) channel,
+ so you will not be able to set timeouts via the TIMEOUT() function in this macro.
+ </option>
+ <option name="n">
+ This option is a modifier for the screen/privacy mode. It specifies
+ that no introductions are to be saved in the priv-callerintros
+ directory.
+ </option>
+ <option name="N">
+ This option is a modifier for the screen/privacy mode. It specifies
+ that if callerID is present, do not screen the call.
+ </option>
+ <option name="o">
+ Specify that the CallerID that was present on the *calling* channel
+ be set as the CallerID on the *called* channel. This was the
+ behavior of Asterisk 1.0 and earlier.
+ </option>
+ <option name="O">
+ <argument name="x" />
+ "Operator Services" mode (Zaptel channel to Zaptel channel
+ only, if specified on non-Zaptel interface, it will be ignored).
+ When the destination answers (presumably an operator services
+ station), the originator no longer has control of their line.
+ They may hang up, but the switch will not release their line
+ until the destination party hangs up (the operator). Specified
+ without an arg, or with 1 as an arg, the originator hanging up
+ will cause the phone to ring back immediately. With a 2 specified,
+ when the "operator" flashes the trunk, it will ring their phone
+ back.
+ </option>
+ <option name="p">
+ This option enables screening mode. This is basically Privacy mode
+ without memory.
+ </option>
+ <option name="P">
+ <argument name="x" />
+ Enable privacy mode. Use 'x' as the family/key in the database if
+ it is provided. The current extension is used if a database
+ family/key is not specified.
+ </option>
+ <option name="r">
+ Indicate ringing to the calling party. Pass no audio to the calling
+ party until the called channel has answered.
+ </option>
+ <option name="S">
+ <argument name="x" required="true" />
+ Hang up the call after 'x' seconds *after* the called party has
+ answered the call.
+ </option>
+ <option name="t">
+ Allow the called party to transfer the calling party by sending the
+ DTMF sequence defined in features.conf.
+ </option>
+ <option name="T">
+ Allow the calling party to transfer the called party by sending the
+ DTMF sequence defined in features.conf.
+ </option>
+ <option name="U" argsep="^">
+ <argument name="x" required="true">
+ routine to execute via Gosub
+ </argument>
+ <argument name="arg">
+ Arguments for the Gosub routine
+ </argument>
+ Execute via Gosub the routine 'x' for the *called* channel before connecting
+ to the calling channel. Arguments can be specified to the Gosub
+ using '^' as a delimiter. The Gosub routine can set the variable
+ GOSUB_RESULT to specify the following actions after the Gosub returns.
+ <variable name="GOSUB_RESULT">
+ <value name="ABORT">
+ Hangup both legs of the call.
+ </value>
+ <value name="CONGESTION">
+ Behave as if line congestion was encountered.
+ </value>
+ <value name="BUSY">
+ Behave as if a busy signal was encountered.
+ </value>
+ <value name="CONTINUE">
+ Hangup the called party and allow the calling party
+ to continue dialplan execution at the next priority.
+ </value>
+ <value name="GOTO:<context>^<exten>^<priority>">
+ Transfer the call to the
+ specified priority. Optionally, an extension, or
+ extension and priority can be specified.
+ </value>
+ </variable>
+ You cannot use any additional action post answer options in conjunction
+ with this option. Also, pbx services are not run on the peer (called) channel,
+ so you will not be able to set timeouts via the TIMEOUT() function in this routine.
+ </option>
+ <option name="w">
+ Allow the called party to enable recording of the call by sending
+ the DTMF sequence defined for one-touch recording in features.conf.
+ </option>
+ <option name="W">
+ Allow the calling party to enable recording of the call by sending
+ the DTMF sequence defined for one-touch recording in features.conf.
+ </option>
+ <option name="x">
+ Allow the called party to enable recording of the call by sending
+ the DTMF sequence defined for one-touch automixmonitor in features.conf
+ </option>
+ <option name="X">
+ Allow the calling party to enable recording of the call by sending
+ the DTMF sequence defined for one-touch automixmonitor in features.conf
+ </option>
+ <option name="URL">
+ The optional URL will be sent to the called party if the channel supports it.
+ </option>
+ </application>
+ <application name="RetryDial" language="en">
+ <synopsis>
+ Place a call, retrying on failure allowing an optional exit extension.
+ </synopsis>
+ <description>
+ This application will attempt to place a call using the normal Dial application.
+ If no channel can be reached, the 'announce' file will be played.
+ Then, it will wait 'sleep' number of seconds before retrying the call.
+ After 'retries' number of attempts, the calling channel will continue at the next priority in the dialplan.
+ If the 'retries' setting is set to 0, this application will retry endlessly.
+ While waiting to retry a call, a 1 digit extension may be dialed. If that
+ extension exists in either the context defined in ${EXITCONTEXT} or the current
+ one, The call will jump to that extension immediately.
+ The 'dialargs' are specified in the same format that arguments are provided
+ to the Dial application.
+ </description>
+ <option name="announce" required="true">
+ Filename of sound that will be played when no channel can be reached
+ </option>
+ <option name="sleep" required="true">
+ Number of seconds to wait after a dialattempt failed before a new attempt is made
+ </option>
+ <option name="retries">
+ Number of retries. When this is reached flow will continue at the next priority in the dialplan
+ </option>
+ <option name="dialargs" required="true">
+ Some format as arguments providet to the Dial application
+ </option>
+ </application>
+ <application name="RetryDial" language="es">
+ <synopsis>
+ (THIS SHOULD BE SOME SPANISH !!!!!!!!!!!!!!!)
+ No hablo Espanol
+ </synopsis>
+ <description>
+ Si signor, No Hablo Espanol
+ </description>
+ </application>
+ <application name="Echo" language="en">
+ <synopsis>
+ Echo audio, video, DTMF back to the calling party
+ </synopsis>
+ <description>
+ Echos back any audio, video or DTMF frames read from the calling
+ channel back to itself. Note: If '#' detected application exits
+ </description>
+ </application>
+ <application name="SendFAX" language="en">
+ <synopsis>
+ Send a Fax
+ </synopsis>
+ <description>
+ Send a given TIFF file to the channel as a FAX.
+ </description>
+ <variable name="LOCALSTATIONID">
+ To identify itself to the remote end
+ </variable>
+ <variable name="LOCALHEADERINFO">
+ To generate a header line on each page
+ </variable>
+ <variable name="FAXSTATUS">
+ <value name="SUCCESS"/>
+ <value name="FAILED"/>
+ </variable>
+ <variable name="FAXERROR">
+ Cause of failure
+ </variable>
+ <variable name="REMOTESTATIONID">
+ The CSID of the remote side
+ </variable>
+ <variable name="FAXPAGES">
+ Number of pages sent
+ </variable>
+ <variable name="FAXBITRATE">
+ Transmission rate
+ </variable>
+ <variable name="FAXRESOLUTION">
+ Resolution of sent fax
+ </variable>
+ <option name="filename" required="true">
+ Filename of TIFF file to fax
+ </option>
+ <option name="a">
+ Makes the application behave as the answering machine
+ (Default behavior is as calling machine)
+ </option>
+ <return>
+ <value name="0">Success</value>
+ <value name="1">Failure</value>
+ </return>
+ </application>
+ <application name="JACK" language="en">
+ <synopsis>
+ Jack Audio Connection Kit
+ </synopsis>
+ <description>
+ When executing this application, two jack ports will be created;
+ one input and one output. Other applications can be hooked up to
+ these ports to access audio coming from, or being send to the channel.
+ </description>
+ <option name="s">
+ <variable name="name" required="true">
+ Connect to the specified jack server name
+ </variable>
+ </option>
+ <option name="i">
+ <variable name="name" required="true">
+ Connect the output port that gets created to the specified jack input port
+ </variable>
+ </option>
+ <option name="o">
+ <variable name="name" required="true">
+ Connect the input port that gets created to the specified jack output port
+ </variable>
+ </option>
+ <option name="c">
+ <variable name="name" required="true">
+ By default, Asterisk will use the channel name for the jack client name.
+ Use this option to specify a custom client name.
+ </variable>
+ </option>
+ </application>
+ <application name="Morsecode" language="en">
+ <synopsis>
+ Plays morse code
+ </synopsis>
+ <description>
+ Plays the Morse code equivalent of the passed string.
+ </description>
+ <variable name="MORSEDITLEN">
+ Use this value in (ms) for length of dit
+ </variable>
+ <variable name="MORSETONE">
+ The pitch of the tone in (Hz), default is 800
+ </variable>
+ <option name="string" required="true">
+ String to playback as morse code to channel
+ </option>
+ </application>
+ <application name="Page" language="en">
+ <synopsis>
+ Page series of phones
+ </synopsis>
+ <description>
+ Places outbound calls to the given technology / resource and dumps
+ them into a conference bridge as muted participants. The original
+ caller is dumped into conference as a speaker and the room is
+ destroyed when the original callers leaves.
+ </description>
+ <option name="Technology/Resource" required="true" argsep="&">
+ <argument name="Technology2/Resource2">
+ Optional extra 'devices' to dial.
+ If you need more then one enter them like this:
+ Technology2/Resource2&Technology3/Resourse3&.....
+ </argument>
+ Device to dial
+ </option>
+ <option name="d">
+ Full duplex audio
+ </option>
+ <option name="q">
+ Quiet, do not play beep to caller
+ </option>
+ <option name="r">
+ record the page into file (meetme option 'r')
+ </option>
+ <option name="s">
+ Only dial channel if devicestate says its 'notinuse'
+ </option>
+ </application>
+ <application name="SendDTMF" language="en">
+ <synopsis>
+ Sends arbitrary DTMF digits
+ </synopsis>
+ <description>
+ DTMF digits sent to a channel with half second pause
+ It will pass all digits or terminate if it encounters an error.
+ </description>
+ <option name="digits" required="true">
+ List of digits 0-9,*#,abcd
+ </option>
+ <option name="timeout_ms">
+ Amount of time to wait in ms between tones
+ </option>
+ </application>
+ <function name="SHELL" language="en">
+ <synopsis>
+ Executes a command as if you were at a shell.
+ </synopsis>
+ <description>
+ Returns the value from a system command
+ Example: Set(foo=${SHELL(echo \bar\)})
+ Note: When using the SHELL() dialplan function, your \SHELL\ is /bin/sh,
+ which may differ as to the underlying shell, depending upon your production
+ platform. Also keep in mind that if you are using a common path, you should
+ be mindful of race conditions that could result from two calls running
+ SHELL() simultaneously.
+ </description>
+ <option name="command" required="true">
+ This is the argument to the function, the command you want to pass to the shell.
+ </option>
+ </function>
+ <application name="Answer" language="en">
+ <synopsis>
+ Answer a channel if ringing.
+ </synopsis>
+ <description>
+ Answer([delay]): If the call has not been answered, this application will
+ answer it. Otherwise, it has no effect on the call. If a delay is specified,
+ Asterisk will wait this number of milliseconds before returning to
+ the dialplan after answering the call.
+ </description>
+ <option name="delay" required="false">
+ Delay that Asterisk will wait in milliseconds before returning to the dialplan after answering the call.
+ </option>
+ </application>
+
+</docs>
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