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| author | Russell Bryant <russell@russellbryant.com> | 2007-03-15 22:29:45 +0000 |
|---|---|---|
| committer | Russell Bryant <russell@russellbryant.com> | 2007-03-15 22:29:45 +0000 |
| commit | 1cf3a12047bb358a1645892454a43dbf611253a7 (patch) | |
| tree | 7e44cd7d9a574a5cc64239ef2ef9132c63946efd /doc/ast_appdocs.tex | |
| parent | 4787adb4e9add47c2dbfac819810577483a484cb (diff) | |
Merged revisions 58931 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r58931 | russell | 2007-03-15 17:25:12 -0500 (Thu, 15 Mar 2007) | 13 lines
Merge changes from svn/asterisk/team/russell/LaTeX_docs.
* Convert most of the doc directory into a single LaTeX formatted document
so that we can generate a PDF, HTML, or other formats from this
information.
* Add a CLI command to dump the application documentation into LaTeX format
which will only be include if the configure script is run with
--enable-dev-mode.
* The PDF turned out to be close to 1 MB, so it is not included. However, you
can simply run "make asterisk.pdf" to generate it yourself. We may include
it in release tarballs or have automatically generated ones on the web site,
but that has yet to be decided.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@58932 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Diffstat (limited to 'doc/ast_appdocs.tex')
| -rw-r--r-- | doc/ast_appdocs.tex | 3225 |
1 files changed, 3225 insertions, 0 deletions
diff --git a/doc/ast_appdocs.tex b/doc/ast_appdocs.tex new file mode 100644 index 000000000..7d59d80a1 --- /dev/null +++ b/doc/ast_appdocs.tex @@ -0,0 +1,3225 @@ +% This file is automatically generated. Any manual edits will be lost. +\section{AddQueueMember} +\subsection{Synopsis} +\begin{verbatim} +Dynamically adds queue members +\end{verbatim} +\subsection{Description} +\begin{verbatim} + AddQueueMember(queuename[|interface[|penalty[|options[|membername]]]]): +Dynamically adds interface to an existing queue. +If the interface is already in the queue and there exists an n+101 priority +then it will then jump to this priority. Otherwise it will return an error +The option string may contain zero or more of the following characters: + 'j' -- jump to +101 priority when appropriate. + This application sets the following channel variable upon completion: + AQMSTATUS The status of the attempt to add a queue member as a + text string, one of + ADDED | MEMBERALREADY | NOSUCHQUEUE +Example: AddQueueMember(techsupport|SIP/3000) + +\end{verbatim} + + +\section{ADSIProg} +\subsection{Synopsis} +\begin{verbatim} +Load Asterisk ADSI Scripts into phone +\end{verbatim} +\subsection{Description} +\begin{verbatim} + ADSIProg(script): This application programs an ADSI Phone with the given +script. If nothing is specified, the default script (asterisk.adsi) is used. + +\end{verbatim} + + +\section{AgentCallbackLogin} +\subsection{Synopsis} +\begin{verbatim} +Call agent callback login +\end{verbatim} +\subsection{Description} +\begin{verbatim} + AgentCallbackLogin([AgentNo][|[options][|[exten]@context]]): +Asks the agent to login to the system with callback. +The agent's callback extension is called (optionally with the specified +context). +The option string may contain zero or more of the following characters: + 's' -- silent login - do not announce the login ok segment agent logged in/off + +\end{verbatim} + + +\section{AgentLogin} +\subsection{Synopsis} +\begin{verbatim} +Call agent login +\end{verbatim} +\subsection{Description} +\begin{verbatim} + AgentLogin([AgentNo][|options]): +Asks the agent to login to the system. Always returns -1. While +logged in, the agent can receive calls and will hear a 'beep' +when a new call comes in. The agent can dump the call by pressing +the star key. +The option string may contain zero or more of the following characters: + 's' -- silent login - do not announce the login ok segment after agent logged in/off + +\end{verbatim} + + +\section{AgentMonitorOutgoing} +\subsection{Synopsis} +\begin{verbatim} +Record agent's outgoing call +\end{verbatim} +\subsection{Description} +\begin{verbatim} + AgentMonitorOutgoing([options]): +Tries to figure out the id of the agent who is placing outgoing call based on +comparison of the callerid of the current interface and the global variable +placed by the AgentCallbackLogin application. That's why it should be used only +with the AgentCallbackLogin app. Uses the monitoring functions in chan_agent +instead of Monitor application. That have to be configured in the agents.conf file. + +Return value: +Normally the app returns 0 unless the options are passed. Also if the callerid or +the agentid are not specified it'll look for n+101 priority. + +Options: + 'd' - make the app return -1 if there is an error condition and there is + no extension n+101 + 'c' - change the CDR so that the source of the call is 'Agent/agent_id' + 'n' - don't generate the warnings when there is no callerid or the + agentid is not known. + It's handy if you want to have one context for agent and non-agent calls. + +\end{verbatim} + + +\section{AGI} +\subsection{Synopsis} +\begin{verbatim} +Executes an AGI compliant application +\end{verbatim} +\subsection{Description} +\begin{verbatim} + [E|Dead]AGI(command|args): Executes an Asterisk Gateway Interface compliant +program on a channel. AGI allows Asterisk to launch external programs +written in any language to control a telephony channel, play audio, +read DTMF digits, etc. by communicating with the AGI protocol on stdin +and stdout. + This channel will stop dialplan execution on hangup inside of this +application, except when using DeadAGI. Otherwise, dialplan execution +will continue normally. + A locally executed AGI script will receive SIGHUP on hangup from the channel +except when using DeadAGI. This can be disabled by setting the AGISIGHUP channel +variable to "no" before executing the AGI application. + Using 'EAGI' provides enhanced AGI, with incoming audio available out of band +on file descriptor 3 + + Use the CLI command 'agi show' to list available agi commands + This application sets the following channel variable upon completion: + AGISTATUS The status of the attempt to the run the AGI script + text string, one of SUCCESS | FAILED | HANGUP + +\end{verbatim} + + +\section{AlarmReceiver} +\subsection{Synopsis} +\begin{verbatim} +Provide support for receiving alarm reports from a burglar or fire alarm panel +\end{verbatim} +\subsection{Description} +\begin{verbatim} + AlarmReceiver(): Only 1 signalling format is supported at this time: Ademco +Contact ID. This application should be called whenever there is an alarm +panel calling in to dump its events. The application will handshake with the +alarm panel, and receive events, validate them, handshake them, and store them +until the panel hangs up. Once the panel hangs up, the application will run the +system command specified by the eventcmd setting in alarmreceiver.conf and pipe +the events to the standard input of the application. The configuration file also +contains settings for DTMF timing, and for the loudness of the acknowledgement +tones. + +\end{verbatim} + + +\section{AMD} +\subsection{Synopsis} +\begin{verbatim} +Attempts to detect answering machines +\end{verbatim} +\subsection{Description} +\begin{verbatim} + AMD([initialSilence][|greeting][|afterGreetingSilence][|totalAnalysisTime] + [|minimumWordLength][|betweenWordsSilence][|maximumNumberOfWords] + [|silenceThreshold]) + 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 calling AMD + with parameters. +- 'initialSilence' is the maximum silence duration before the greeting. If + exceeded then MACHINE. +- 'greeting' is the maximum length of a greeting. If exceeded then MACHINE. +- 'afterGreetingSilence' is the silence after detecting a greeting. + If exceeded then HUMAN. +- 'totalAnalysisTime' is the maximum time allowed for the algorithm to decide + on a HUMAN or MACHINE. +- 'minimumWordLength'is the minimum duration of Voice to considered as a word. +- 'betweenWordsSilence' is the minimum duration of silence after a word to + consider the audio that follows as a new word. +- 'maximumNumberOfWords'is the maximum number of words in the greeting. + If exceeded then MACHINE. +- 'silenceThreshold' is the silence threshold. +This application sets the following channel variable upon completion: + AMDSTATUS - This is the status of the answering machine detection. + Possible values are: + MACHINE | HUMAN | NOTSURE | HANGUP + AMDCAUSE - Indicates the cause that led to the conclusion. + Possible values are: + TOOLONG-<%d total_time> + INITIALSILENCE-<%d silenceDuration>-<%d initialSilence> + HUMAN-<%d silenceDuration>-<%d afterGreetingSilence> + MAXWORDS-<%d wordsCount>-<%d maximumNumberOfWords> + LONGGREETING-<%d voiceDuration>-<%d greeting> + +\end{verbatim} + + +\section{Answer} +\subsection{Synopsis} +\begin{verbatim} +Answer a channel if ringing +\end{verbatim} +\subsection{Description} +\begin{verbatim} + 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. + +\end{verbatim} + + +\section{AppendCDRUserField} +\subsection{Synopsis} +\begin{verbatim} +Append to the CDR user field +\end{verbatim} +\subsection{Description} +\begin{verbatim} +[Synopsis] +AppendCDRUserField(value) + +[Description] +AppendCDRUserField(value): Append value to the CDR user field + The Call Data Record (CDR) user field is an extra field you + can use for data not stored anywhere else in the record. + CDR records can be used for billing or storing other arbitrary data + (I.E. telephone survey responses) + Also see SetCDRUserField(). + +\end{verbatim} + + +\section{Authenticate} +\subsection{Synopsis} +\begin{verbatim} +Authenticate a user +\end{verbatim} +\subsection{Description} +\begin{verbatim} + Authenticate(password[|options[|maxdigits]]): This application asks the caller +to enter a given password in order to continue dialplan execution. If the password +begins with the '/' character, it is interpreted as a file which contains a list of +valid passwords, listed 1 password per line in the file. + When using a database key, the value associated with the key can be anything. +Users have three attempts to authenticate before the channel is hung up. If the +passsword is invalid, the 'j' option is specified, and priority n+101 exists, +dialplan execution will continnue at this location. + Options: + a - Set the channels' account code to the password that is entered + d - Interpret the given path as database key, not a literal file + j - Support jumping to n+101 if authentication fails + m - Interpret the given path as a file which contains a list of account + codes and password hashes delimited with ':', listed one per line in + the file. When one of the passwords is matched, the channel will have + its account code set to the corresponding account code in the file. + r - Remove the database key upon successful entry (valid with 'd' only) + maxdigits - maximum acceptable number of digits. Stops reading after + maxdigits have been entered (without requiring the user to + press the '#' key). + Defaults to 0 - no limit - wait for the user press the '#' key. + +\end{verbatim} + + +\section{BackGround} +\subsection{Synopsis} +\begin{verbatim} +Play an audio file while waiting for digits of an extension to go to. +\end{verbatim} +\subsection{Description} +\begin{verbatim} + Background(filename1[&filename2...][|options[|langoverride][|context]]): +This application will play the given list of files while waiting for an +extension to be dialed by the calling channel. To continue waiting for digits +after this application has finished playing files, the WaitExten application +should be used. The 'langoverride' option explicitly specifies which language +to attempt to use for the requested sound files. If a 'context' is specified, +this is the dialplan context that this application will use when exiting to a +dialed extension. If one of the requested sound files does not exist, call processing will be +terminated. + Options: + s - Causes the playback of the message to be skipped + if the channel is not in the 'up' state (i.e. it + hasn't been answered yet). If this happens, the + application will return immediately. + n - Don't answer the channel before playing the files. + m - Only break if a digit hit matches a one digit + extension in the destination context. + +\end{verbatim} + + +\section{BackgroundDetect} +\subsection{Synopsis} +\begin{verbatim} +Background a file with talk detect +\end{verbatim} +\subsection{Description} +\begin{verbatim} + BackgroundDetect(filename[|sil[|min|[max]]]): Plays back a given +filename, waiting for interruption from a given digit (the digit must +start the beginning of a valid extension, or it will be ignored). +During the playback of the file, audio is monitored in the receive +direction, and if a period of non-silence which is greater than 'min' ms +yet less than 'max' ms is followed by silence for at least 'sil' ms then +the audio playback is aborted and processing jumps to the 'talk' extension +if available. If unspecified, sil, min, and max default to 1000, 100, and +infinity respectively. + +\end{verbatim} + + +\section{Busy} +\subsection{Synopsis} +\begin{verbatim} +Indicate the Busy condition +\end{verbatim} +\subsection{Description} +\begin{verbatim} + Busy([timeout]): This application will indicate the busy condition to +the calling channel. If the optional timeout is specified, the calling channel +will be hung up after the specified number of seconds. Otherwise, this +application will wait until the calling channel hangs up. + +\end{verbatim} + + +\section{ChangeMonitor} +\subsection{Synopsis} +\begin{verbatim} +Change monitoring filename of a channel +\end{verbatim} +\subsection{Description} +\begin{verbatim} +ChangeMonitor(filename_base) +Changes monitoring filename of a channel. Has no effect if the channel is not monitored +The argument is the new filename base to use for monitoring this channel. + +\end{verbatim} + + +\section{ChanIsAvail} +\subsection{Synopsis} +\begin{verbatim} +Check channel availability +\end{verbatim} +\subsection{Description} +\begin{verbatim} + ChanIsAvail(Technology/resource[&Technology2/resource2...][|options]): +This application will check to see if any of the specified channels are +available. The following variables will be set by this application: + ${AVAILCHAN} - the name of the available channel, if one exists + ${AVAILORIGCHAN} - the canonical channel name that was used to create the channel + ${AVAILSTATUS} - the status code for the available channel + Options: + s - Consider the channel unavailable if the channel is in use at all + j - Support jumping to priority n+101 if no channel is available + +\end{verbatim} + + +\section{ChannelRedirect} +\subsection{Synopsis} +\begin{verbatim} +Redirects given channel to a dialplan target. +\end{verbatim} +\subsection{Description} +\begin{verbatim} +ChannelRedirect(channel|[[context|]extension|]priority): + Sends the specified channel to the specified extension priority + +\end{verbatim} + + +\section{ChanSpy} +\subsection{Synopsis} +\begin{verbatim} +Listen to a channel, and optionally whisper into it +\end{verbatim} +\subsection{Description} +\begin{verbatim} + ChanSpy([chanprefix][|options]): This application is used to listen to the +audio from an Asterisk channel. This includes the audio coming in and +out of the channel being spied on. If the 'chanprefix' parameter is specified, +only channels beginning with this string will be spied upon. + While spying, the following actions may be performed: + - Dialing # cycles the volume level. + - Dialing * will stop spying and look for another channel to spy on. + - Dialing a series of digits followed by # builds a channel name to append + to 'chanprefix'. For example, executing ChanSpy(Agent) and then dialing + the digits '1234#' while spying will begin spying on the channel + 'Agent/1234'. + Options: + b - Only spy on channels involved in a bridged call. + g(grp) - Match only channels where their ${SPYGROUP} variable is set to + contain 'grp' in an optional : delimited list. + q - Don't play a beep when beginning to spy on a channel, or speak the + selected channel name. + r[(basename)] - Record the session to the monitor spool directory. An + optional base for the filename may be specified. The + default is 'chanspy'. + v([value]) - Adjust the initial volume in the range from -4 to 4. A + negative value refers to a quieter setting. + w - Enable 'whisper' mode, so the spying channel can talk to + the spied-on channel. + W - Enable 'private whisper' mode, so the spying channel can + talk to the spied-on channel but cannot listen to that + channel. + +\end{verbatim} + + +\section{Congestion} +\subsection{Synopsis} +\begin{verbatim} +Indicate the Congestion condition +\end{verbatim} +\subsection{Description} +\begin{verbatim} + Congestion([timeout]): This application will indicate the congestion +condition to the calling channel. If the optional timeout is specified, the +calling channel will be hung up after the specified number of seconds. +Otherwise, this application will wait until the calling channel hangs up. + +\end{verbatim} + + +\section{ContinueWhile} +\subsection{Synopsis} +\begin{verbatim} +Restart a While loop +\end{verbatim} +\subsection{Description} +\begin{verbatim} +Usage: ContinueWhile() +Returns to the top of the while loop and re-evaluates the conditional. + +\end{verbatim} + + +\section{ControlPlayback} +\subsection{Synopsis} +\begin{verbatim} +Play a file with fast forward and rewind +\end{verbatim} +\subsection{Description} +\begin{verbatim} + ControlPlayback(file[|skipms[|ff[|rew[|stop[|pause[|restart|options]]]]]]]): +This application will play back the given filename. By default, the '*' key +can be used to rewind, and the '#' key can be used to fast-forward. +Parameters: + skipms - This is number of milliseconds to skip when rewinding or + fast-forwarding. + ff - Fast-forward when this DTMF digit is received. + rew - Rewind when this DTMF digit is received. + stop - Stop playback when this DTMF digit is received. + pause - Pause playback when this DTMF digit is received. + restart - Restart playback when this DTMF digit is received. +Options: + j - Jump to priority n+101 if the requested file is not found. +This application sets the following channel variable upon completion: + CPLAYBACKSTATUS - This variable contains the status of the attempt as a text + string, one of: SUCCESS | USERSTOPPED | ERROR + +\end{verbatim} + + +\section{DateTime} +\subsection{Synopsis} +\begin{verbatim} +Says a specified time in a custom format +\end{verbatim} +\subsection{Description} +\begin{verbatim} +DateTime([unixtime][|[timezone][|format]]) + unixtime: time, in seconds since Jan 1, 1970. May be negative. + defaults to now. + timezone: timezone, see /usr/share/zoneinfo for a list. + defaults to machine default. + format: a format the time is to be said in. See voicemail.conf. + defaults to "ABdY 'digits/at' IMp" + +\end{verbatim} + + +\section{DBdel} +\subsection{Synopsis} +\begin{verbatim} +Delete a key from the database +\end{verbatim} +\subsection{Description} +\begin{verbatim} + DBdel(family/key): This application will delete a key from the Asterisk +database. + This application has been DEPRECATED in favor of the DB_DELETE function. + +\end{verbatim} + + +\section{DBdeltree} +\subsection{Synopsis} +\begin{verbatim} +Delete a family or keytree from the database +\end{verbatim} +\subsection{Description} +\begin{verbatim} + DBdeltree(family[/keytree]): This application will delete a family or keytree +from the Asterisk database + +\end{verbatim} + + +\section{DeadAGI} +\subsection{Synopsis} +\begin{verbatim} +Executes AGI on a hungup channel +\end{verbatim} +\subsection{Description} +\begin{verbatim} + [E|Dead]AGI(command|args): Executes an Asterisk Gateway Interface compliant +program on a channel. AGI allows Asterisk to launch external programs +written in any language to control a telephony channel, play audio, +read DTMF digits, etc. by communicating with the AGI protocol on stdin +and stdout. + This channel will stop dialplan execution on hangup inside of this +application, except when using DeadAGI. Otherwise, dialplan execution +will continue normally. + A locally executed AGI script will receive SIGHUP on hangup from the channel +except when using DeadAGI. This can be disabled by setting the AGISIGHUP channel +variable to "no" before executing the AGI application. + Using 'EAGI' provides enhanced AGI, with incoming audio available out of band +on file descriptor 3 + + Use the CLI command 'agi show' to list available agi commands + This application sets the following channel variable upon completion: + AGISTATUS The status of the attempt to the run the AGI script + text string, one of SUCCESS | FAILED | HANGUP + +\end{verbatim} + + +\section{Dial} +\subsection{Synopsis} +\begin{verbatim} +Place a call and connect to the current channel +\end{verbatim} +\subsection{Description} +\begin{verbatim} + Dial(Technology/resource[&Tech2/resource2...][|timeout][|options][|URL]): +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 sets the following channel variables upon completion: + DIALEDTIME - This is the time from dialing a channel until when it + is disconnected. + ANSWEREDTIME - This is the amount of time for actual call. + DIALSTATUS - This is the status of the call: + CHANUNAVAIL | CONGESTION | NOANSWER | BUSY | ANSWER | CANCEL + DONTCALL | TORTURE | INVALIDARGS + For the Privacy and Screening Modes, the DIALSTATUS variable will be set to +DONTCALL if the called party chooses to send the calling party to the 'Go Away' +script. The DIALSTATUS variable will be set to TORTURE if the called party +wants to send the caller to the 'torture' script. + 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. + The optional URL will be sent to the called party if the channel supports it. + If the OUTBOUND_GROUP variable is set, all peer channels created by this +application will be put into that group (as in Set(GROUP()=...). + + Options: + A(x) - Play an announcement to the called party, using 'x' as the file. + C - Reset the CDR for this call. + 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. + D([called][:calling]) - Send the specified DTMF strings *after* the called + 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. + 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. + g - Proceed with dialplan execution at the current extension if the + destination channel hangs up. + G(context^exten^pri) - 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. + h - Allow the called party to hang up by sending the '*' DTMF digit. + H - Allow the calling party to hang up by hitting the '*' DTMF digit. + i - Asterisk will ignore any forwarding requests it may receive on this + dial attempt. + j - Jump to priority n+101 if all of the requested channels were busy. + L(x[:y][:z]) - Limit the call to 'x' ms. Play a warning when 'y' ms are + left. Repeat the warning every 'z' ms. The following special + variables can be used with this option: + * LIMIT_PLAYAUDIO_CALLER yes|no (default yes) + Play sounds to the caller. + * LIMIT_PLAYAUDIO_CALLEE yes|no + Play sounds to the callee. + * LIMIT_TIMEOUT_FILE File to play when time is up. + * LIMIT_CONNECT_FILE File to play when call begins. + * LIMIT_WARNING_FILE File to play as warning if 'y' is defined. + The default is to say the time remaining. + m([class]) - Provide hold music to the calling party until a requested + channel answers. A specific MusicOnHold class can be + specified. + M(x[^arg]) - Execute the Macro for the *called* channel before connecting + to the calling channel. Arguments can be specified to the Macro + using '^' as a delimeter. The Macro can set the variable + MACRO_RESULT to specify the following actions after the Macro is + finished executing. + * ABORT Hangup both legs of the call. + * CONGESTION Behave as if line congestion was encountered. + * BUSY Behave as if a busy signal was encountered. This will also + have the application jump to priority n+101 if the + 'j' option is set. + * CONTINUE Hangup the called party and allow the calling party + to continue dialplan execution at the next priority. + * GOTO:<context>^<exten>^<priority> - Transfer the call to the + specified priority. Optionally, an extension, or + extension and priority can be specified. + 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. + 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. + N - This option is a modifier for the screen/privacy mode. It specifies + that if callerID is present, do not screen the call. + 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. + O([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. + p - This option enables screening mode. This is basically Privacy mode + without memory. + P([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. + r - Indicate ringing to the calling party. Pass no audio to the calling + party until the called channel has answered. + S(x) - Hang up the call after 'x' seconds *after* the called party has + answered the call. + t - Allow the called party to transfer the calling party by sending the + DTMF sequence defined in features.conf. + T - Allow the calling party to transfer the called party by sending the + DTMF sequence defined in features.conf. + w - Allow the called party to enable recording of the call by sending + the DTMF sequence defined for one-touch recording in features.conf. + W - Allow the calling party to enable recording of the call by sending + the DTMF sequence defined for one-touch recording in features.conf. + k - Allow the called party to enable parking of the call by sending + the DTMF sequence defined for call parking in features.conf. + K - Allow the calling party to enable parking of the call by sending + the DTMF sequence defined for call parking in features.conf. + +\end{verbatim} + + +\section{Dictate} +\subsection{Synopsis} +\begin{verbatim} +Virtual Dictation Machine +\end{verbatim} +\subsection{Description} +\begin{verbatim} + Dictate([<base_dir>[|<filename>]]) +Start dictation machine using optional base dir for files. + +\end{verbatim} + + +\section{Directory} +\subsection{Synopsis} +\begin{verbatim} +Provide directory of voicemail extensions +\end{verbatim} +\subsection{Description} +\begin{verbatim} + Directory(vm-context[|dial-context[|options]]): This application will present +the calling channel with a directory of extensions from which they can search +by name. The list of names and corresponding extensions is retrieved from the +voicemail configuration file, voicemail.conf. + This application will immediately exit if one of the following DTMF digits are +received and the extension to jump to exists: + 0 - Jump to the 'o' extension, if it exists. + * - Jump to the 'a' extension, if it exists. + + Parameters: + vm-context - This is the context within voicemail.conf to use for the + Directory. + dial-context - This is the dialplan context to use when looking for an + extension that the user has selected, or when jumping to the + 'o' or 'a' extension. + + Options: + e - In addition to the name, also read the extension number to the + caller before presenting dialing options. + f - Allow the caller to enter the first name of a user in the directory + instead of using the last name. + +\end{verbatim} + + +\section{DISA} +\subsection{Synopsis} +\begin{verbatim} +DISA (Direct Inward System Access) +\end{verbatim} +\subsection{Description} +\begin{verbatim} +DISA(<numeric passcode>[|<context>]) or DISA(<filename>) +The DISA, Direct Inward System Access, application allows someone from +outside the telephone switch (PBX) to obtain an "internal" system +dialtone and to place calls from it as if they were placing a call from +within the switch. +DISA plays a dialtone. The user enters their numeric passcode, followed by +the pound sign (#). If the passcode is correct, the user is then given +system dialtone on which a call may be placed. Obviously, this type +of access has SERIOUS security implications, and GREAT care must be +taken NOT to compromise your security. + +There is a possibility of accessing DISA without password. Simply +exchange your password with "no-password". + + Example: exten => s,1,DISA(no-password|local) + +Be aware that using this compromises the security of your PBX. + +The arguments to this application (in extensions.conf) allow either +specification of a single global passcode (that everyone uses), or +individual passcodes contained in a file. It also allows specification +of the context on which the user will be dialing. If no context is +specified, the DISA application defaults the context to "disa". +Presumably a normal system will have a special context set up +for DISA use with some or a lot of restrictions. + +The file that contains the passcodes (if used) allows specification +of either just a passcode (defaulting to the "disa" context, or +passcode|context on each line of the file. The file may contain blank +lines, or comments starting with "#" or ";". In addition, the +above arguments may have |new-callerid-string appended to them, to +specify a new (different) callerid to be used for this call, for +example: numeric-passcode|context|"My Phone" <(234) 123-4567> or +full-pathname-of-passcode-file|"My Phone" <(234) 123-4567>. Last +but not least, |mailbox[@context] may be appended, which will cause +a stutter-dialtone (indication "dialrecall") to be used, if the +specified mailbox contains any new messages, for example: +numeric-passcode|context||1234 (w/a changing callerid). Note that +in the case of specifying the numeric-passcode, the context must be +specified if the callerid is specified also. + +If login is successful, the application looks up the dialed number in +the specified (or default) context, and executes it if found. +If the user enters an invalid extension and extension "i" (invalid) +exists in the context, it will be used. Also, if you set the 5th argument +to 'NOANSWER', the DISA application will not answer initially. + +\end{verbatim} + + +\section{DumpChan} +\subsection{Synopsis} +\begin{verbatim} +Dump Info About The Calling Channel +\end{verbatim} +\subsection{Description} +\begin{verbatim} + DumpChan([<min_verbose_level>]) +Displays information on channel and listing of all channel +variables. If min_verbose_level is specified, output is only +displayed when the verbose level is currently set to that number +or greater. + +\end{verbatim} + + +\section{EAGI} +\subsection{Synopsis} +\begin{verbatim} +Executes an EAGI compliant application +\end{verbatim} +\subsection{Description} +\begin{verbatim} + [E|Dead]AGI(command|args): Executes an Asterisk Gateway Interface compliant +program on a channel. AGI allows Asterisk to launch external programs +written in any language to control a telephony channel, play audio, +read DTMF digits, etc. by communicating with the AGI protocol on stdin +and stdout. + This channel will stop dialplan execution on hangup inside of this +application, except when using DeadAGI. Otherwise, dialplan execution +will continue normally. + A locally executed AGI script will receive SIGHUP on hangup from the channel +except when using DeadAGI. This can be disabled by setting the AGISIGHUP channel +variable to "no" before executing the AGI application. + Using 'EAGI' provides enhanced AGI, with incoming audio available out of band +on file descriptor 3 + + Use the CLI command 'agi show' to list available agi commands + This application sets the following channel variable upon completion: + AGISTATUS The status of the attempt to the run the AGI script + text string, one of SUCCESS | FAILED | HANGUP + +\end{verbatim} + + +\section{Echo} +\subsection{Synopsis} +\begin{verbatim} +Echo audio, video, or DTMF back to the calling party +\end{verbatim} +\subsection{Description} +\begin{verbatim} + Echo(): This application will echo any audio, video, or DTMF frames read from +the calling channel back to itself. If the DTMF digit '#' is received, the +application will exit. + +\end{verbatim} + + +\section{EndWhile} +\subsection{Synopsis} +\begin{verbatim} +End a while loop +\end{verbatim} +\subsection{Description} +\begin{verbatim} +Usage: EndWhile() +Return to the previous called While + +\end{verbatim} + + +\section{Exec} +\subsection{Synopsis} +\begin{verbatim} +Executes dialplan application +\end{verbatim} +\subsection{Description} +\begin{verbatim} +Usage: Exec(appname(arguments)) + Allows an arbitrary application to be invoked even when not +hardcoded into the dialplan. If the underlying application +terminates the dialplan, or if the application cannot be found, +Exec will terminate the dialplan. + To invoke external applications, see the application System. + If you would like to catch any error instead, see TryExec. + +\end{verbatim} + + +\section{ExecIf} +\subsection{Synopsis} +\begin{verbatim} +Executes dialplan application, conditionally +\end{verbatim} +\subsection{Description} +\begin{verbatim} +Usage: ExecIF (<expr>|<app>|<data>) +If <expr> is true, execute and return the result of <app>(<data>). +If <expr> is true, but <app> is not found, then the application +will return a non-zero value. + +\end{verbatim} + + +\section{ExecIfTime} +\subsection{Synopsis} +\begin{verbatim} +Conditional application execution based on the current time +\end{verbatim} +\subsection{Description} +\begin{verbatim} + ExecIfTime(<times>|<weekdays>|<mdays>|<months>?appname[|appargs]): +This application will execute the specified dialplan application, with optional +arguments, if the current time matches the given time specification. + +\end{verbatim} + + +\section{ExitWhile} +\subsection{Synopsis} +\begin{verbatim} +End a While loop +\end{verbatim} +\subsection{Description} +\begin{verbatim} +Usage: ExitWhile() +Exits a While loop, whether or not the conditional has been satisfied. + +\end{verbatim} + + +\section{ExtenSpy} +\subsection{Synopsis} +\begin{verbatim} +Listen to a channel, and optionally whisper into it +\end{verbatim} +\subsection{Description} +\begin{verbatim} + ExtenSpy(exten[@context][|options]): This application is used to listen to the +audio from an Asterisk channel. This includes the audio coming in and +out of the channel being spied on. Only channels created by outgoing calls for the +specified extension will be selected for spying. If the optional context is not +supplied, the current channel's context will be used. + While spying, the following actions may be performed: + - Dialing # cycles the volume level. + - Dialing * will stop spying and look for another channel to spy on. + Options: + b - Only spy on channels involved in a bridged call. + g(grp) - Match only channels where their ${SPYGROUP} variable is set to + contain 'grp' in an optional : delimited list. + q - Don't play a beep when beginning to spy on a channel, or speak the + selected channel name. + r[(basename)] - Record the session to the monitor spool directory. An + optional base for the filename may be specified. The + default is 'chanspy'. + v([value]) - Adjust the initial volume in the range from -4 to 4. A + negative value refers to a quieter setting. + w - Enable 'whisper' mode, so the spying channel can talk to + the spied-on channel. + W - Enable 'private whisper' mode, so the spying channel can + talk to the spied-on channel but cannot listen to that + channel. + +\end{verbatim} + + +\section{ExternalIVR} +\subsection{Synopsis} +\begin{verbatim} +Interfaces with an external IVR application +\end{verbatim} +\subsection{Description} +\begin{verbatim} + ExternalIVR(command[|arg[|arg...]]): Forks an process to run the supplied command, +and starts a generator on the channel. The generator's play list is +controlled by the external application, which can add and clear entries +via simple commands issued over its stdout. The external application +will receive all DTMF events received on the channel, and notification +if the channel is hung up. The application will not be forcibly terminated +when the channel is hung up. +See doc/externalivr.txt for a protocol specification. + +\end{verbatim} + + +\section{Festival} +\subsection{Synopsis} +\begin{verbatim} +Say text to the user +\end{verbatim} +\subsection{Description} +\begin{verbatim} + Festival(text[|intkeys]): Connect to Festival, send the argument, get back the waveform,play it to the user, allowing any given interrupt keys to immediately terminate and return +the value, or 'any' to allow any number back (useful in dialplan) + +\end{verbatim} + + +\section{Flash} +\subsection{Synopsis} +\begin{verbatim} +Flashes a Zap Trunk +\end{verbatim} +\subsection{Description} +\begin{verbatim} + Flash(): Sends a flash on a zap trunk. This is only a hack for +people who want to perform transfers and such via AGI and is generally +quite useless oths application will only work on Zap trunks. + +\end{verbatim} + + +\section{FollowMe} +\subsection{Synopsis} +\begin{verbatim} +Find-Me/Follow-Me application +\end{verbatim} +\subsection{Description} +\begin{verbatim} + FollowMe(followmeid|options): +This application performs Find-Me/Follow-Me functionality for the caller +as defined in the profile matching the <followmeid> parameter in +followme.conf. If the specified <followmeid> profile doesn't exist in +followme.conf, execution will be returned to the dialplan and call +execution will continue at the next priority. + + Options: + s - Playback the incoming status message prior to starting the follow-me step(s) + a - Record the caller's name so it can be announced to the callee on each step + n - Playback the unreachable status message if we've run out of steps to reach the + or the callee has elected not to be reachable. +Returns -1 on hangup + +\end{verbatim} + + +\section{ForkCDR} +\subsection{Synopsis} +\begin{verbatim} +Forks the Call Data Record +\end{verbatim} +\subsection{Description} +\begin{verbatim} + ForkCDR([options]): Causes the Call Data Record to fork an additional +cdr record starting from the time of the fork call +If the option 'v' is passed all cdr variables will be passed along also. + +\end{verbatim} + + +\section{GetCPEID} +\subsection{Synopsis} +\begin{verbatim} +Get ADSI CPE ID +\end{verbatim} +\subsection{Description} +\begin{verbatim} + GetCPEID: Obtains and displays ADSI CPE ID and other information in order +to properly setup zapata.conf for on-hook operations. + +\end{verbatim} + + +\section{Gosub} +\subsection{Synopsis} +\begin{verbatim} +Jump to label, saving return address +\end{verbatim} +\subsection{Description} +\begin{verbatim} +Gosub([[context|]exten|]priority) + Jumps to the label specified, saving the return address. + +\end{verbatim} + + +\section{GosubIf} +\subsection{Synopsis} +\begin{verbatim} +Conditionally jump to label, saving return address +\end{verbatim} +\subsection{Description} +\begin{verbatim} +GosubIf(condition?labeliftrue[:labeliffalse]) + If the condition is true, then jump to labeliftrue. If false, jumps to +labeliffalse, if specified. In either case, a jump saves the return point +in the dialplan, to be returned to with a Return. + +\end{verbatim} + + +\section{Goto} +\subsection{Synopsis} +\begin{verbatim} +Jump to a particular priority, extension, or context +\end{verbatim} +\subsection{Description} +\begin{verbatim} + Goto([[context|]extension|]priority): This application will set the current +context, extension, and priority in the channel structure. After it completes, the +pbx engine will continue dialplan execution at the specified location. +If no specific extension, or extension and context, are specified, then this +application will just set the specified priority of the current extension. + At least a priority is required as an argument, or the goto will return a -1, +and the channel and call will be terminated. + If the location that is put into the channel information is bogus, and asterisk cannot +find that location in the dialplan, +then the execution engine will try to find and execute the code in the 'i' (invalid) +extension in the current context. If that does not exist, it will try to execute the +'h' extension. If either or neither the 'h' or 'i' extensions have been defined, the +channel is hung up, and the execution of instructions on the channel is terminated. +What this means is that, for example, you specify a context that does not exist, then +it will not be possible to find the 'h' or 'i' extensions, and the call will terminate! + +\end{verbatim} + + +\section{GotoIf} +\subsection{Synopsis} +\begin{verbatim} +Conditional goto +\end{verbatim} +\subsection{Description} +\begin{verbatim} + GotoIf(condition?[labeliftrue]:[labeliffalse]): This application will set the current +context, extension, and priority in the channel structure based on the evaluation of +the given condition. After this application completes, the +pbx engine will continue dialplan execution at the specified location in the dialplan. +The channel will continue at +'labeliftrue' if the condition is true, or 'labeliffalse' if the condition is +false. The labels are specified with the same syntax as used within the Goto +application. If the label chosen by the condition is omitted, no jump is +performed, and the execution passes to the next instruction. +If the target location is bogus, and does not exist, the execution engine will try +to find and execute the code in the 'i' (invalid) +extension in the current context. If that does not exist, it will try to execute the +'h' extension. If either or neither the 'h' or 'i' extensions have been defined, the +channel is hung up, and the execution of instructions on the channel is terminated. +Remember that this command can set the current context, and if the context specified +does not exist, then it will not be able to find any 'h' or 'i' extensions there, and +the channel and call will both be terminated! + +\end{verbatim} + + +\section{GotoIfTime} +\subsection{Synopsis} +\begin{verbatim} +Conditional Goto based on the current time +\end{verbatim} +\subsection{Description} +\begin{verbatim} + GotoIfTime(<times>|<weekdays>|<mdays>|<months>?[[context|]exten|]priority): +This application will set the context, extension, and priority in the channel structure +if the current time matches the given time specification. Otherwise, nothing is done. +Further information on the time specification can be found in examples +illustrating how to do time-based context includes in the dialplan. +If the target jump location is bogus, the same actions would be taken as for Goto. + +\end{verbatim} + + +\section{Hangup} +\subsection{Synopsis} +\begin{verbatim} +Hang up the calling channel +\end{verbatim} +\subsection{Description} +\begin{verbatim} + Hangup([causecode]): This application will hang up the calling channel. +If a causecode is given the channel's hangup cause will be set to the given +value. + +\end{verbatim} + + +\section{HasNewVoicemail} +\subsection{Synopsis} +\begin{verbatim} +Conditionally branches to priority + 101 with the right options set +\end{verbatim} +\subsection{Description} +\begin{verbatim} +HasNewVoicemail(vmbox[/folder][@context][|varname[|options]]) +Assumes folder 'INBOX' if folder is not specified. Optionally sets <varname> to the number of messages +in that folder. + The option string may contain zero of the following character: + 'j' -- jump to priority n+101, if there is new voicemail in folder 'folder' or INBOX + This application sets the following channel variable upon completion: + HASVMSTATUS The result of the new voicemail check returned as a text string as follows + <# of messages in the folder, 0 for NONE> + +\end{verbatim} + + +\section{HasVoicemail} +\subsection{Synopsis} +\begin{verbatim} +Conditionally branches to priority + 101 with the right options set +\end{verbatim} +\subsection{Description} +\begin{verbatim} +HasVoicemail(vmbox[/folder][@context][|varname[|options]]) + Optionally sets <varname> to the number of messages in that folder. Assumes folder of INBOX if not specified. + The option string may contain zero or the following character: + 'j' -- jump to priority n+101, if there is voicemail in the folder indicated. + This application sets the following channel variable upon completion: + HASVMSTATUS The result of the voicemail check returned as a text string as follows + <# of messages in the folder, 0 for NONE> + +\end{verbatim} + + +\section{IAX2Provision} +\subsection{Synopsis} +\begin{verbatim} +Provision a calling IAXy with a given template +\end{verbatim} +\subsection{Description} +\begin{verbatim} + IAX2Provision([template]): Provisions the calling IAXy (assuming +the calling entity is in fact an IAXy) with the given template or +default if one is not specified. Returns -1 on error or 0 on success. + +\end{verbatim} + + +\section{ICES} +\subsection{Synopsis} +\begin{verbatim} +Encode and stream using 'ices' +\end{verbatim} +\subsection{Description} +\begin{verbatim} + ICES(config.xml) Streams to an icecast server using ices +(available separately). A configuration file must be supplied +for ices (see examples/asterisk-ices.conf). + +\end{verbatim} + + +\section{ImportVar} +\subsection{Synopsis} +\begin{verbatim} +Import a variable from a channel into a new variable +\end{verbatim} +\subsection{Description} +\begin{verbatim} + ImportVar(newvar=channelname|variable): This application imports a variable +from the specified channel (as opposed to the current one) and stores it as +a variable in the current channel (the channel that is calling this +application). Variables created by this application have the same inheritance +properties as those created with the Set application. See the documentation for +Set for more information. + +\end{verbatim} + + +\section{JabberSend} +\subsection{Synopsis} +\begin{verbatim} +JabberSend(jabber,screenname,message) +\end{verbatim} +\subsection{Description} +\begin{verbatim} +JabberSend(Jabber,ScreenName,Message) + Jabber - Client or transport Asterisk uses to connect to Jabber + ScreenName - User Name to message. + Message - Message to be sent to the buddy + +\end{verbatim} + + +\section{JabberStatus} +\subsection{Synopsis} +\begin{verbatim} +JabberStatus(Jabber,ScreenName,Variable) +\end{verbatim} +\subsection{Description} +\begin{verbatim} +JabberStatus(Jabber,ScreenName,Variable) + Jabber - Client or transport Asterisk uses to connect to Jabber + ScreenName - User Name to retrieve status from. + Variable - Variable to store presence in will be 1-6. + In order, Online, Chatty, Away, XAway, DND, Offline + If not in roster variable will = 7 + +\end{verbatim} + + +\section{Log} +\subsection{Synopsis} +\begin{verbatim} +Send arbitrary text to a selected log level +\end{verbatim} +\subsection{Description} +\begin{verbatim} +Log(<level>|<message>) + level must be one of ERROR, WARNING, NOTICE, DEBUG, VERBOSE, DTMF + +\end{verbatim} + + +\section{LookupBlacklist} +\subsection{Synopsis} +\begin{verbatim} +Look up Caller*ID name/number from blacklist database +\end{verbatim} +\subsection{Description} +\begin{verbatim} + LookupBlacklist(options): Looks up the Caller*ID number on the active +channel in the Asterisk database (family 'blacklist'). +The option string may contain the following character: + 'j' -- jump to n+101 priority if the number/name is found in the blacklist +This application sets the following channel variable upon completion: + LOOKUPBLSTATUS The status of the Blacklist lookup as a text string, one of + FOUND | NOTFOUND +Example: exten => 1234,1,LookupBlacklist() + +This application is deprecated and may be removed from a future release. +Please use the dialplan function BLACKLIST() instead. + +\end{verbatim} + + +\section{LookupCIDName} +\subsection{Synopsis} +\begin{verbatim} +Look up CallerID Name from local database +\end{verbatim} +\subsection{Description} +\begin{verbatim} + LookupCIDName: Looks up the Caller*ID number on the active +channel in the Asterisk database (family 'cidname') and sets the +Caller*ID name. Does nothing if no Caller*ID was received on the +channel. This is useful if you do not subscribe to Caller*ID +name delivery, or if you want to change the names on some incoming +calls. + +LookupCIDName is deprecated. Please use ${DB(cidname/${CALLERID(num)})} +instead. + +\end{verbatim} + + +\section{Macro} +\subsection{Synopsis} +\begin{verbatim} +Macro Implementation +\end{verbatim} +\subsection{Description} +\begin{verbatim} + Macro(macroname|arg1|arg2...): Executes a macro using the context +'macro-<macroname>', jumping to the 's' extension of that context and +executing each step, then returning when the steps end. +The calling extension, context, and priority are stored in ${MACRO_EXTEN}, +${MACRO_CONTEXT} and ${MACRO_PRIORITY} respectively. Arguments become +${ARG1}, ${ARG2}, etc in the macro context. +If you Goto out of the Macro context, the Macro will terminate and control +will be returned at the location of the Goto. +If ${MACRO_OFFSET} is set at termination, Macro will attempt to continue +at priority MACRO_OFFSET + N + 1 if such a step exists, and N + 1 otherwise. +WARNING: Because of the way Macro is implemented (it executes the priorities + contained within it via sub-engine), and a fixed per-thread + memory stack allowance, macros are limited to 7 levels + of nesting (macro calling macro calling macro, etc.); It + may be possible that stack-intensive applications in deeply nested macros + could cause asterisk to crash earlier than this limit. + +\end{verbatim} + + +\section{MacroExclusive} +\subsection{Synopsis} +\begin{verbatim} +Exclusive Macro Implementation +\end{verbatim} +\subsection{Description} +\begin{verbatim} + MacroExclusive(macroname|arg1|arg2...): +Executes macro defined in the context 'macro-macroname' +Only one call at a time may run the macro. +(we'll wait if another call is busy executing in the Macro) +Arguments and return values as in application Macro() + +\end{verbatim} + + +\section{MacroExit} +\subsection{Synopsis} +\begin{verbatim} +Exit From Macro +\end{verbatim} +\subsection{Description} +\begin{verbatim} + MacroExit(): +Causes the currently running macro to exit as if it had +ended normally by running out of priorities to execute. +If used outside a macro, will likely cause unexpected +behavior. + +\end{verbatim} + + +\section{MacroIf} +\subsection{Synopsis} +\begin{verbatim} +Conditional Macro Implementation +\end{verbatim} +\subsection{Description} +\begin{verbatim} + MacroIf(<expr>?macroname_a[|arg1][:macroname_b[|arg1]]) +Executes macro defined in <macroname_a> if <expr> is true +(otherwise <macroname_b> if provided) +Arguments and return values as in application macro() + +\end{verbatim} + + +\section{MailboxExists} +\subsection{Synopsis} +\begin{verbatim} +Check to see if Voicemail mailbox exists +\end{verbatim} +\subsection{Description} +\begin{verbatim} + MailboxExists(mailbox[@context][|options]): Check to see if the specified +mailbox exists. If no voicemail context is specified, the 'default' context +will be used. + This application will set the following channel variable upon completion: + VMBOXEXISTSSTATUS - This will contain the status of the execution of the + MailboxExists application. Possible values include: + SUCCESS | FAILED + + Options: + j - Jump to priority n+101 if the mailbox is found. + +\end{verbatim} + + +\section{MeetMe} +\subsection{Synopsis} +\begin{verbatim} +MeetMe conference bridge +\end{verbatim} +\subsection{Description} +\begin{verbatim} + MeetMe([confno][,[options][,pin]]): Enters the user into a specified MeetMe +conference. If the conference number is omitted, the user will be prompted +to enter one. User can exit the conference by hangup, or if the 'p' option +is specified, by pressing '#'. +Please note: The Zaptel kernel modules and at least one hardware driver (or ztdummy) + must be present for conferencing to operate properly. In addition, the chan_zap + channel driver must be loaded for the 'i' and 'r' options to operate at all. + +The option string may contain zero or more of the following characters: + 'a' -- set admin mode + 'A' -- set marked mode + 'b' -- run AGI script specified in ${MEETME_AGI_BACKGROUND} + Default: conf-background.agi (Note: This does not work with + non-Zap channels in the same conference) + 'c' -- announce user(s) count on joining a conference + 'd' -- dynamically add conference + 'D' -- dynamically add conference, prompting for a PIN + 'e' -- select an empty conference + 'E' -- select an empty pinless conference + 'F' -- Pass DTMF through the conference. DTMF used to activate any + conference features will not be passed through. + 'i' -- announce user join/leave with review + 'I' -- announce user join/leave without review + 'l' -- set listen only mode (Listen only, no talking) + 'm' -- set initially muted + 'M' -- enable music on hold when the conference has a single caller + 'o' -- set talker optimization - treats talkers who aren't speaking as + being muted, meaning (a) No encode is done on transmission and + (b) Received audio that is not registered as talking is omitted + causing no buildup in background noise + 'p' -- allow user to exit the conference by pressing '#' + 'P' -- always prompt for the pin even if it is specified + 'q' -- quiet mode (don't play enter/leave sounds) + 'r' -- Record conference (records as ${MEETME_RECORDINGFILE} + using format ${MEETME_RECORDINGFORMAT}). Default filename is + meetme-conf-rec-${CONFNO}-${UNIQUEID} and the default format is + wav. + 's' -- Present menu (user or admin) when '*' is received ('send' to menu) + 't' -- set talk only mode. (Talk only, no listening) + 'T' -- set talker detection (sent to manager interface and meetme list) + 'w[(<secs>)]' + -- wait until the marked user enters the conference + 'x' -- close the conference when last marked user exits + 'X' -- allow user to exit the conference by entering a valid single + digit extension ${MEETME_EXIT_CONTEXT} or the current context + if that variable is not defined. + '1' -- do not play message when first person enters + +\end{verbatim} + + +\section{MeetMeAdmin} +\subsection{Synopsis} +\begin{verbatim} +MeetMe conference Administration +\end{verbatim} +\subsection{Description} +\begin{verbatim} + MeetMeAdmin(confno,command[,user]): Run admin command for conference + 'e' -- Eject last user that joined + 'k' -- Kick one user out of conference + 'K' -- Kick all users out of conference + 'l' -- Unlock conference + 'L' -- Lock conference + 'm' -- Unmute one user + 'M' -- Mute one user + 'n' -- Unmute all users in the conference + 'N' -- Mute all non-admin users in the conference + 'r' -- Reset one user's volume settings + 'R' -- Reset all users volume settings + 's' -- Lower entire conference speaking volume + 'S' -- Raise entire conference speaking volume + 't' -- Lower one user's talk volume + 'T' -- Raise one user's talk volume + 'u' -- Lower one user's listen volume + 'U' -- Raise one user's listen volume + 'v' -- Lower entire conference listening volume + 'V' -- Raise entire conference listening volume + +\end{verbatim} + + +\section{MeetMeCount} +\subsection{Synopsis} +\begin{verbatim} +MeetMe participant count +\end{verbatim} +\subsection{Description} +\begin{verbatim} + MeetMeCount(confno[|var]): Plays back the number of users in the specified +MeetMe conference. If var is specified, playback will be skipped and the value +will be returned in the variable. Upon app completion, MeetMeCount will hangup +the channel, unless priority n+1 exists, in which case priority progress will +continue. +A ZAPTEL INTERFACE MUST BE INSTALLED FOR CONFERENCING FUNCTIONALITY. + +\end{verbatim} + + +\section{Milliwatt} +\subsection{Synopsis} +\begin{verbatim} +Generate a Constant 1000Hz tone at 0dbm (mu-law) +\end{verbatim} +\subsection{Description} +\begin{verbatim} +Milliwatt(): Generate a Constant 1000Hz tone at 0dbm (mu-law) + +\end{verbatim} + + +\section{MixMonitor} +\subsection{Synopsis} +\begin{verbatim} +Record a call and mix the audio during the recording +\end{verbatim} +\subsection{Description} +\begin{verbatim} + MixMonitor(<file>.<ext>[|<options>[|<command>]]) + +Records the audio on the current channel to the specified file. +If the filename is an absolute path, uses that path, otherwise +creates the file in the configured monitoring directory from +asterisk.conf. + +Valid options: + a - Append to the file instead of overwriting it. + b - Only save audio to the file while the channel is bridged. + Note: Does not include conferences or sounds played to each bridged + party. + v(<x>) - Adjust the heard volume by a factor of <x> (range -4 to 4) + V(<x>) - Adjust the spoken volume by a factor of <x> (range -4 to 4) + W(<x>) - Adjust the both heard and spoken volumes by a factor of <x> + (range -4 to 4) + +<command> will be executed when the recording is over +Any strings matching ^{X} will be unescaped to ${X} and +all variables will be evaluated at that time. +The variable MIXMONITOR_FILENAME will contain the filename used to record. + +\end{verbatim} + + +\section{Monitor} +\subsection{Synopsis} +\begin{verbatim} +Monitor a channel +\end{verbatim} +\subsection{Description} +\begin{verbatim} +Monitor([file_format[:urlbase]|[fname_base]|[options]]): +Used to start monitoring a channel. The channel's input and output +voice packets are logged to files until the channel hangs up or +monitoring is stopped by the StopMonitor application. + file_format optional, if not set, defaults to "wav" + fname_base if set, changes the filename used to the one specified. + options: + m - when the recording ends mix the two leg files into one and + delete the two leg files. If the variable MONITOR_EXEC is set, the + application referenced in it will be executed instead of + soxmix and the raw leg files will NOT be deleted automatically. + soxmix or MONITOR_EXEC is handed 3 arguments, the two leg files + and a target mixed file name which is the same as the leg file names + only without the in/out designator. + If MONITOR_EXEC_ARGS is set, the contents will be passed on as + additional arguements to MONITOR_EXEC + Both MONITOR_EXEC and the Mix flag can be set from the + administrator interface + + b - Don't begin recording unless a call is bridged to another channel + +Returns -1 if monitor files can't be opened or if the channel is already +monitored, otherwise 0. + +\end{verbatim} + + +\section{Morsecode} +\subsection{Synopsis} +\begin{verbatim} +Plays morse code +\end{verbatim} +\subsection{Description} +\begin{verbatim} +Usage: Morsecode(<string>) +Plays the Morse code equivalent of the passed string. If the variable +MORSEDITLEN is set, it will use that value for the length (in ms) of the dit +(defaults to 80). Additionally, if MORSETONE is set, it will use that tone +(in Hz). The tone default is 800. + +\end{verbatim} + + +\section{MP3Player} +\subsection{Synopsis} +\begin{verbatim} +Play an MP3 file or stream +\end{verbatim} +\subsection{Description} +\begin{verbatim} + MP3Player(location) Executes mpg123 to play the given location, +which typically would be a filename or a URL. User can exit by pressing +any key on the dialpad, or by hanging up. +\end{verbatim} + + +\section{MusicOnHold} +\subsection{Synopsis} +\begin{verbatim} +Play Music On Hold indefinitely +\end{verbatim} +\subsection{Description} +\begin{verbatim} +MusicOnHold(class): Plays hold music specified by class. If omitted, the default +music source for the channel will be used. Set the default +class with the SetMusicOnHold() application. +Returns -1 on hangup. +Never returns otherwise. + +\end{verbatim} + + +\section{NBScat} +\subsection{Synopsis} +\begin{verbatim} +Play an NBS local stream +\end{verbatim} +\subsection{Description} +\begin{verbatim} + NBScat: Executes nbscat to listen to the local NBS stream. +User can exit by pressing any key +. +\end{verbatim} + + +\section{NoCDR} +\subsection{Synopsis} +\begin{verbatim} +Tell Asterisk to not maintain a CDR for the current call +\end{verbatim} +\subsection{Description} +\begin{verbatim} + NoCDR(): This application will tell Asterisk not to maintain a CDR for the +current call. + +\end{verbatim} + + +\section{NoOp} +\subsection{Synopsis} +\begin{verbatim} +Do Nothing +\end{verbatim} +\subsection{Description} +\begin{verbatim} + NoOp(): This applicatiion does nothing. However, it is useful for debugging +purposes. Any text that is provided as arguments to this application can be +viewed at the Asterisk CLI. This method can be used to see the evaluations of +variables or functions without having any effect. +\end{verbatim} + + +\section{Page} +\subsection{Synopsis} +\begin{verbatim} +Pages phones +\end{verbatim} +\subsection{Description} +\begin{verbatim} +Page(Technology/Resource&Technology2/Resource2[|options]) + Places outbound calls to the given technology / resource and dumps +them into a conference bridge as muted participants. The original +caller is dumped into the conference as a speaker and the room is +destroyed when the original caller leaves. Valid options are: + d - full duplex audio + q - quiet, do not play beep to caller + r - record the page into a file (see 'r' for app_meetme) + +\end{verbatim} + + +\section{Park} +\subsection{Synopsis} +\begin{verbatim} +Park yourself +\end{verbatim} +\subsection{Description} +\begin{verbatim} +Park():Used to park yourself (typically in combination with a supervised +transfer to know the parking space). This application is always +registered internally and does not need to be explicitly added +into the dialplan, although you should include the 'parkedcalls' +context (or the context specified in features.conf). + +If you set the PARKINGEXTEN variable to an extension in your +parking context, park() will park the call on that extension, unless +it already exists. In that case, execution will continue at next +priority. + +\end{verbatim} + + +\section{ParkAndAnnounce} +\subsection{Synopsis} +\begin{verbatim} +Park and Announce +\end{verbatim} +\subsection{Description} +\begin{verbatim} + ParkAndAnnounce(announce:template|timeout|dial|[return_context]): +Park a call into the parkinglot and announce the call to another channel. + +announce template: Colon-separated list of files to announce. The word PARKED + will be replaced by a say_digits of the extension in which + the call is parked. +timeout: Time in seconds before the call returns into the return + context. +dial: The app_dial style resource to call to make the + announcement. Console/dsp calls the console. +return_context: The goto-style label to jump the call back into after + timeout. Default <priority+1>. + +The variable ${PARKEDAT} will contain the parking extension into which the +call was placed. Use with the Local channel to allow the dialplan to make +use of this information. + +\end{verbatim} + + +\section{ParkedCall} +\subsection{Synopsis} +\begin{verbatim} +Answer a parked call +\end{verbatim} +\subsection{Description} +\begin{verbatim} +ParkedCall(exten):Used to connect to a parked call. This application is always +registered internally and does not need to be explicitly added +into the dialplan, although you should include the 'parkedcalls' +context. + +\end{verbatim} + + +\section{PauseMonitor} +\subsection{Synopsis} +\begin{verbatim} +Pause monitoring of a channel +\end{verbatim} +\subsection{Description} +\begin{verbatim} +PauseMonitor +Pauses monitoring of a channel until it is re-enabled by a call to UnpauseMonitor. + +\end{verbatim} + + +\section{PauseQueueMember} +\subsection{Synopsis} +\begin{verbatim} +Pauses a queue member +\end{verbatim} +\subsection{Description} +\begin{verbatim} + PauseQueueMember([queuename]|interface[|options]): +Pauses (blocks calls for) a queue member. +The given interface will be paused in the given queue. This prevents +any calls from being sent from the queue to the interface until it is +unpaused with UnpauseQueueMember or the manager interface. If no +queuename is given, the interface is paused in every queue it is a +member of. If the interface is not in the named queue, or if no queue +is given and the interface is not in any queue, it will jump to +priority n+101, if it exists and the appropriate options are set. +The application will fail if the interface is not found and no extension +to jump to exists. +The option string may contain zero or more of the following characters: + 'j' -- jump to +101 priority when appropriate. + This application sets the following channel variable upon completion: + PQMSTATUS The status of the attempt to pause a queue member as a + text string, one of + PAUSED | NOTFOUND +Example: PauseQueueMember(|SIP/3000) + +\end{verbatim} + + +\section{Pickup} +\subsection{Synopsis} +\begin{verbatim} +Directed Call Pickup +\end{verbatim} +\subsection{Description} +\begin{verbatim} + Pickup(extension[@context][&extension2@context...]): This application can pickup any ringing channel +that is calling the specified extension. If no context is specified, the current +context will be used. If you use the special string "PICKUPMARK" for the context parameter, for example +10@PICKUPMARK, this application tries to find a channel which has defined a channel variable with the same content +as "extension". +\end{verbatim} + + +\section{Playback} +\subsection{Synopsis} +\begin{verbatim} +Play a file +\end{verbatim} +\subsection{Description} +\begin{verbatim} + Playback(filename[&filename2...][|option]): Plays back given filenames (do not put +extension). Options may also be included following a pipe symbol. The 'skip' +option causes the playback of the message to be skipped if the channel +is not in the 'up' state (i.e. it hasn't been answered yet). If 'skip' is +specified, the application will return immediately should the channel not be +off hook. Otherwise, unless 'noanswer' is specified, the channel will +be answered before the sound is played. Not all channels support playing +messages while still on hook. If 'j' is specified, the application +will jump to priority n+101 if present when a file specified to be played +does not exist. +This application sets the following channel variable upon completion: + PLAYBACKSTATUS The status of the playback attempt as a text string, one of + SUCCESS | FAILED + +\end{verbatim} + + +\section{PlayTones} +\subsection{Synopsis} +\begin{verbatim} +Play a tone list +\end{verbatim} +\subsection{Description} +\begin{verbatim} +PlayTones(arg): Plays a tone list. Execution will continue with the next step immediately, +while the tones continue to play. +Arg is either the tone name defined in the indications.conf configuration file, or a directly +specified list of frequencies and durations. +See the sample indications.conf for a description of the specification of a tonelist. + +Use the StopPlayTones application to stop the tones playing. + +\end{verbatim} + + +\section{PrivacyManager} +\subsection{Synopsis} +\begin{verbatim} +Require phone number to be entered, if no CallerID sent +\end{verbatim} +\subsection{Description} +\begin{verbatim} + PrivacyManager([maxretries[|minlength[|options]]]): If no Caller*ID +is sent, PrivacyManager answers the channel and asks the caller to +enter their phone number. The caller is given 3 attempts to do so. +The application does nothing if Caller*ID was received on the channel. + Configuration file privacy.conf contains two variables: + maxretries default 3 -maximum number of attempts the caller is allowed + to input a callerid. + minlength default 10 -minimum allowable digits in the input callerid number. +If you don't want to use the config file and have an i/o operation with +every call, you can also specify maxretries and minlength as application +parameters. Doing so supercedes any values set in privacy.conf. +The option string may contain the following character: + 'j' -- jump to n+101 priority after <maxretries> failed attempts to collect + the minlength number of digits. +The application sets the following channel variable upon completion: +PRIVACYMGRSTATUS The status of the privacy manager's attempt to collect + a phone number from the user. A text string that is either: + SUCCESS | FAILED + +\end{verbatim} + + +\section{Progress} +\subsection{Synopsis} +\begin{verbatim} +Indicate progress +\end{verbatim} +\subsection{Description} +\begin{verbatim} + Progress(): This application will request that in-band progress information +be provided to the calling channel. + +\end{verbatim} + + +\section{Queue} +\subsection{Synopsis} +\begin{verbatim} +Queue a call for a call queue +\end{verbatim} +\subsection{Description} +\begin{verbatim} + Queue(queuename[|options[|URL][|announceoverride][|timeout][|AGI]): +Queues an incoming call in a particular call queue as defined in queues.conf. +This application will return to the dialplan if the queue does not exist, or +any of the join options cause the caller to not enter the queue. +The option string may contain zero or more of the following characters: + 'd' -- data-quality (modem) call (minimum delay). + 'h' -- allow callee to hang up by hitting *. + 'H' -- allow caller to hang up by hitting *. + 'n' -- no retries on the timeout; will exit this application and + go to the next step. + 'i' -- ignore call forward requests from queue members and do nothing + when they are requested. + 'r' -- ring instead of playing MOH + 't' -- allow the called user transfer the calling user + 'T' -- to allow the calling user to transfer the call. + 'w' -- allow the called user to write the conversation to disk via Monitor + 'W' -- allow the calling user to write the conversation to disk via Monitor + In addition to transferring the call, a call may be parked and then picked +up by another user. + The optional URL will be sent to the called party if the channel supports +it. + The optional AGI parameter will setup an AGI script to be executed on the +calling party's channel once they are connected to a queue member. + The timeout will cause the queue to fail out after a specified number of +seconds, checked between each queues.conf 'timeout' and 'retry' cycle. + This application sets the following channel variable upon completion: + QUEUESTATUS The status of the call as a text string, one of + TIMEOUT | FULL | JOINEMPTY | LEAVEEMPTY | JOINUNAVAIL | LEAVEUNAVAIL + +\end{verbatim} + + +\section{QueueLog} +\subsection{Synopsis} +\begin{verbatim} +Writes to the queue_log +\end{verbatim} +\subsection{Description} +\begin{verbatim} + QueueLog(queuename|uniqueid|agent|event[|additionalinfo]): +Allows you to write your own events into the queue log +Example: QueueLog(101|${UNIQUEID}|${AGENT}|WENTONBREAK|600) + +\end{verbatim} + + +\section{Random} +\subsection{Synopsis} +\begin{verbatim} +Conditionally branches, based upon a probability +\end{verbatim} +\subsection{Description} +\begin{verbatim} +Random([probability]:[[context|]extension|]priority) + probability := INTEGER in the range 1 to 100 +DEPRECATED: Use GotoIf($[${RAND(1,100)} > <number>]?<label>) + +\end{verbatim} + + +\section{Read} +\subsection{Synopsis} +\begin{verbatim} +Read a variable +\end{verbatim} +\subsection{Description} +\begin{verbatim} + Read(variable[|filename][|maxdigits][|option][|attempts][|timeout]) + +Reads a #-terminated string of digits a certain number of times from the +user in to the given variable. + filename -- file to play before reading digits or tone with option i + maxdigits -- maximum acceptable number of digits. Stops reading after + maxdigits have been entered (without requiring the user to + press the '#' key). + Defaults to 0 - no limit - wait for the user press the '#' key. + Any value below 0 means the same. Max accepted value is 255. + option -- options are 's' , 'i', 'n' + 's' to return immediately if the line is not up, + 'i' to play filename as an indication tone from your indications.conf + 'n' to read digits even if the line is not up. + attempts -- if greater than 1, that many attempts will be made in the + event no data is entered. + timeout -- An integer number of seconds to wait for a digit response. If greater + than 0, that value will override the default timeout. + +Read should disconnect if the function fails or errors out. + +\end{verbatim} + + +\section{ReadFile} +\subsection{Synopsis} +\begin{verbatim} +ReadFile(varname=file,length) +\end{verbatim} +\subsection{Description} +\begin{verbatim} +ReadFile(varname=file,length) + Varname - Result stored here. + File - The name of the file to read. + Length - Maximum number of characters to capture. + +\end{verbatim} + + +\section{RealTime} +\subsection{Synopsis} +\begin{verbatim} +Realtime Data Lookup +\end{verbatim} +\subsection{Description} +\begin{verbatim} +Use the RealTime config handler system to read data into channel variables. +RealTime(<family>|<colmatch>|<value>[|<prefix>]) + +All unique column names will be set as channel variables with optional prefix +to the name. For example, a prefix of 'var_' would make the column 'name' +become the variable ${var_name}. REALTIMECOUNT will be set with the number +of values read. + +\end{verbatim} + + +\section{RealTimeUpdate} +\subsection{Synopsis} +\begin{verbatim} +Realtime Data Rewrite +\end{verbatim} +\subsection{Description} +\begin{verbatim} +Use the RealTime config handler system to update a value +RealTimeUpdate(<family>|<colmatch>|<value>|<newcol>|<newval>) + +The column <newcol> in 'family' matching column <colmatch>=<value> will be +updated to <newval>. REALTIMECOUNT will be set with the number of rows +updated or -1 if an error occurs. + +\end{verbatim} + + +\section{Record} +\subsection{Synopsis} +\begin{verbatim} +Record to a file +\end{verbatim} +\subsection{Description} +\begin{verbatim} + Record(filename.format|silence[|maxduration][|options]) + +Records from the channel into a given filename. If the file exists it will +be overwritten. +- 'format' is the format of the file type to be recorded (wav, gsm, etc). +- 'silence' is the number of seconds of silence to allow before returning. +- 'maxduration' is the maximum recording duration in seconds. If missing +or 0 there is no maximum. +- 'options' may contain any of the following letters: + 'a' : append to existing recording rather than replacing + 'n' : do not answer, but record anyway if line not yet answered + 'q' : quiet (do not play a beep tone) + 's' : skip recording if the line is not yet answered + 't' : use alternate '*' terminator key (DTMF) instead of default '#' + 'x' : ignore all terminator keys (DTMF) and keep recording until hangup + +If filename contains '%d', these characters will be replaced with a number +incremented by one each time the file is recorded. A channel variable +named RECORDED_FILE will also be set, which contains the final filemname. + +Use 'core show file formats' to see the available formats on your system + +User can press '#' to terminate the recording and continue to the next priority. + +If the user should hangup during a recording, all data will be lost and the +application will teminate. + +\end{verbatim} + + +\section{RemoveQueueMember} +\subsection{Synopsis} +\begin{verbatim} +Dynamically removes queue members +\end{verbatim} +\subsection{Description} +\begin{verbatim} + RemoveQueueMember(queuename[|interface[|options]]): +Dynamically removes interface to an existing queue +If the interface is NOT in the queue and there exists an n+101 priority +then it will then jump to this priority. Otherwise it will return an error +The option string may contain zero or more of the following characters: + 'j' -- jump to +101 priority when appropriate. + This application sets the following channel variable upon completion: + RQMSTATUS The status of the attempt to remove a queue member as a + text string, one of + REMOVED | NOTINQUEUE | NOSUCHQUEUE +Example: RemoveQueueMember(techsupport|SIP/3000) + +\end{verbatim} + + +\section{ResetCDR} +\subsection{Synopsis} +\begin{verbatim} +Resets the Call Data Record +\end{verbatim} +\subsection{Description} +\begin{verbatim} + ResetCDR([options]): This application causes the Call Data Record to be +reset. + Options: + w -- Store the current CDR record before resetting it. + a -- Store any stacked records. + v -- Save CDR variables. + +\end{verbatim} + + +\section{RetryDial} +\subsection{Synopsis} +\begin{verbatim} +Place a call, retrying on failure allowing optional exit extension. +\end{verbatim} +\subsection{Description} +\begin{verbatim} + RetryDial(announce|sleep|retries|dialargs): 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 retying the call. After 'retires' 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. + +\end{verbatim} + + +\section{Return} +\subsection{Synopsis} +\begin{verbatim} +Return from gosub routine +\end{verbatim} +\subsection{Description} +\begin{verbatim} +Return() + Jumps to the last label on the stack, removing it. + +\end{verbatim} + + +\section{Ringing} +\subsection{Synopsis} +\begin{verbatim} +Indicate ringing tone +\end{verbatim} +\subsection{Description} +\begin{verbatim} + Ringing(): This application will request that the channel indicate a ringing +tone to the user. + +\end{verbatim} + + +\section{SayAlpha} +\subsection{Synopsis} +\begin{verbatim} +Say Alpha +\end{verbatim} +\subsection{Description} +\begin{verbatim} + SayAlpha(string): This application will play the sounds that correspond to +the letters of the given string. + +\end{verbatim} + + +\section{SayDigits} +\subsection{Synopsis} +\begin{verbatim} +Say Digits +\end{verbatim} +\subsection{Description} +\begin{verbatim} + SayDigits(digits): This application will play the sounds that correspond +to the digits of the given number. This will use the language that is currently +set for the channel. See the LANGUAGE function for more information on setting +the language for the channel. + +\end{verbatim} + + +\section{SayNumber} +\subsection{Synopsis} +\begin{verbatim} +Say Number +\end{verbatim} +\subsection{Description} +\begin{verbatim} + SayNumber(digits[,gender]): This application will play the sounds that +correspond to the given number. Optionally, a gender may be specified. +This will use the language that is currently set for the channel. See the +LANGUAGE function for more information on setting the language for the channel. + +\end{verbatim} + + +\section{SayPhonetic} +\subsection{Synopsis} +\begin{verbatim} +Say Phonetic +\end{verbatim} +\subsection{Description} +\begin{verbatim} + SayPhonetic(string): This application will play the sounds from the phonetic +alphabet that correspond to the letters in the given string. + +\end{verbatim} + + +\section{SayUnixTime} +\subsection{Synopsis} +\begin{verbatim} +Says a specified time in a custom format +\end{verbatim} +\subsection{Description} +\begin{verbatim} +SayUnixTime([unixtime][|[timezone][|format]]) + unixtime: time, in seconds since Jan 1, 1970. May be negative. + defaults to now. + timezone: timezone, see /usr/share/zoneinfo for a list. + defaults to machine default. + format: a format the time is to be said in. See voicemail.conf. + defaults to "ABdY 'digits/at' IMp" + +\end{verbatim} + + +\section{SendDTMF} +\subsection{Synopsis} +\begin{verbatim} +Sends arbitrary DTMF digits +\end{verbatim} +\subsection{Description} +\begin{verbatim} + SendDTMF(digits[|timeout_ms]): Sends DTMF digits on a channel. + Accepted digits: 0-9, *#abcd, w (.5s pause) + The application will either pass the assigned digits or terminate if it + encounters an error. + +\end{verbatim} + + +\section{SendImage} +\subsection{Synopsis} +\begin{verbatim} +Send an image file +\end{verbatim} +\subsection{Description} +\begin{verbatim} + SendImage(filename): Sends an image on a channel. +If the channel supports image transport but the image send +fails, the channel will be hung up. Otherwise, the dialplan +continues execution. +The option string may contain the following character: + 'j' -- jump to priority n+101 if the channel doesn't support image transport +This application sets the following channel variable upon completion: + SENDIMAGESTATUS The status is the result of the attempt as a text string, one of + OK | NOSUPPORT + +\end{verbatim} + + +\section{SendText} +\subsection{Synopsis} +\begin{verbatim} +Send a Text Message +\end{verbatim} +\subsection{Description} +\begin{verbatim} + SendText(text[|options]): Sends text to current channel (callee). +Result of transmission will be stored in the SENDTEXTSTATUS +channel variable: + SUCCESS Transmission succeeded + FAILURE Transmission failed + UNSUPPORTED Text transmission not supported by channel + +At this moment, text is supposed to be 7 bit ASCII in most channels. +The option string many contain the following character: +'j' -- jump to n+101 priority if the channel doesn't support + text transport + +\end{verbatim} + + +\section{SendURL} +\subsection{Synopsis} +\begin{verbatim} +Send a URL +\end{verbatim} +\subsection{Description} +\begin{verbatim} + SendURL(URL[|option]): Requests client go to URL (IAX2) or sends the +URL to the client (other channels). +Result is returned in the SENDURLSTATUS channel variable: + SUCCESS URL successfully sent to client + FAILURE Failed to send URL + NOLOAD Client failed to load URL (wait enabled) + UNSUPPORTED Channel does not support URL transport + +If the option 'wait' is specified, execution will wait for an +acknowledgement that the URL has been loaded before continuing + +If jumping is specified as an option (the 'j' flag), the client does not +support Asterisk "html" transport, and there exists a step with priority +n + 101, then execution will continue at that step. + +SendURL continues normally if the URL was sent correctly or if the channel +does not support HTML transport. Otherwise, the channel is hung up. + +\end{verbatim} + + +\section{Set} +\subsection{Synopsis} +\begin{verbatim} +Set channel variable(s) or function value(s) +\end{verbatim} +\subsection{Description} +\begin{verbatim} + Set(name1=value1|name2=value2|..[|options]) +This function can be used to set the value of channel variables or dialplan +functions. It will accept up to 24 name/value pairs. When setting variables, +if the variable name is prefixed with _, the variable will be inherited into +channels created from the current channel. If the variable name is prefixed +with __, the variable will be inherited into channels created from the current +channel and all children channels. + Options: + g - Set variable globally instead of on the channel + (applies only to variables, not functions) + +\end{verbatim} + + +\section{SetAMAFlags} +\subsection{Synopsis} +\begin{verbatim} +Set the AMA Flags +\end{verbatim} +\subsection{Description} +\begin{verbatim} + SetAMAFlags([flag]): This application will set the channel's AMA Flags for + billing purposes. + +\end{verbatim} + + +\section{SetCallerID} +\subsection{Synopsis} +\begin{verbatim} +Set CallerID +\end{verbatim} +\subsection{Description} +\begin{verbatim} + SetCallerID(clid[|a]): Set Caller*ID on a call to a new +value. Sets ANI as well if a flag is used. + +\end{verbatim} + + +\section{SetCallerPres} +\subsection{Synopsis} +\begin{verbatim} +Set CallerID Presentation +\end{verbatim} +\subsection{Description} +\begin{verbatim} + SetCallerPres(presentation): Set Caller*ID presentation on a call. + Valid presentations are: + + allowed_not_screened : Presentation Allowed, Not Screened + allowed_passed_screen : Presentation Allowed, Passed Screen + allowed_failed_screen : Presentation Allowed, Failed Screen + allowed : Presentation Allowed, Network Number + prohib_not_screened : Presentation Prohibited, Not Screened + prohib_passed_screen : Presentation Prohibited, Passed Screen + prohib_failed_screen : Presentation Prohibited, Failed Screen + prohib : Presentation Prohibited, Network Number + unavailable : Number Unavailable + + +\end{verbatim} + + +\section{SetCDRUserField} +\subsection{Synopsis} +\begin{verbatim} +Set the CDR user field +\end{verbatim} +\subsection{Description} +\begin{verbatim} +[Synopsis] +SetCDRUserField(value) + +[Description] +SetCDRUserField(value): Set the CDR 'user field' to value + The Call Data Record (CDR) user field is an extra field you + can use for data not stored anywhere else in the record. + CDR records can be used for billing or storing other arbitrary data + (I.E. telephone survey responses) + Also see AppendCDRUserField(). + +\end{verbatim} + + +\section{SetGlobalVar} +\subsection{Synopsis} +\begin{verbatim} +Set a global variable to a given value +\end{verbatim} +\subsection{Description} +\begin{verbatim} + SetGlobalVar(variable=value): This application sets a given global variable to +the specified value. + +\end{verbatim} + + +\section{SetMusicOnHold} +\subsection{Synopsis} +\begin{verbatim} +Set default Music On Hold class +\end{verbatim} +\subsection{Description} +\begin{verbatim} +SetMusicOnHold(class): Sets the default class for music on hold for a given channel. When +music on hold is activated, this class will be used to select which +music is played. + +\end{verbatim} + + +\section{SetTransferCapability} +\subsection{Synopsis} +\begin{verbatim} +Set ISDN Transfer Capability +\end{verbatim} +\subsection{Description} +\begin{verbatim} + SetTransferCapability(transfercapability): Set the ISDN Transfer +Capability of a call to a new value. +Valid Transfer Capabilities are: + + SPEECH : 0x00 - Speech (default, voice calls) + DIGITAL : 0x08 - Unrestricted digital information (data calls) + RESTRICTED_DIGITAL : 0x09 - Restricted digital information + 3K1AUDIO : 0x10 - 3.1kHz Audio (fax calls) + DIGITAL_W_TONES : 0x11 - Unrestricted digital information with tones/announcements + VIDEO : 0x18 - Video + + +\end{verbatim} + + +\section{SIPAddHeader} +\subsection{Synopsis} +\begin{verbatim} +Add a SIP header to the outbound call +\end{verbatim} +\subsection{Description} +\begin{verbatim} + SIPAddHeader(Header: Content) +Adds a header to a SIP call placed with DIAL. +Remember to user the X-header if you are adding non-standard SIP +headers, like "X-Asterisk-Accountcode:". Use this with care. +Adding the wrong headers may jeopardize the SIP dialog. +Always returns 0 + +\end{verbatim} + + +\section{SIPDtmfMode} +\subsection{Synopsis} +\begin{verbatim} +Change the dtmfmode for a SIP call +\end{verbatim} +\subsection{Description} +\begin{verbatim} +SIPDtmfMode(inband|info|rfc2833): Changes the dtmfmode for a SIP call + +\end{verbatim} + + +\section{SLAStation} +\subsection{Synopsis} +\begin{verbatim} +Shared Line Appearance Station +\end{verbatim} +\subsection{Description} +\begin{verbatim} + SLAStation(station): +This application should be executed by an SLA station. The argument depends +on how the call was initiated. If the phone was just taken off hook, then +the argument "station" should be just the station name. If the call was +initiated by pressing a line key, then the station name should be preceded +by an underscore and the trunk name associated with that line button. +For example: "station1_line1". +\end{verbatim} + + +\section{SLATrunk} +\subsection{Synopsis} +\begin{verbatim} +Shared Line Appearance Trunk +\end{verbatim} +\subsection{Description} +\begin{verbatim} + SLATrunk(trunk): +This application should be executed by an SLA trunk on an inbound call. +The channel calling this application should correspond to the SLA trunk +with the name "trunk" that is being passed as an argument. + +\end{verbatim} + + +\section{SMS} +\subsection{Synopsis} +\begin{verbatim} +Communicates with SMS service centres and SMS capable analogue phones +\end{verbatim} +\subsection{Description} +\begin{verbatim} + SMS(name|[a][s]): SMS handles exchange of SMS data with a call to/from SMS capabale +phone or SMS PSTN service center. Can send and/or receive SMS messages. +Works to ETSI ES 201 912 compatible with BT SMS PSTN service in UK +Typical usage is to use to handle called from the SMS service centre CLI, +or to set up a call using 'outgoing' or manager interface to connect +service centre to SMS() +name is the name of the queue used in /var/spool/asterisk/sms +Arguments: + a: answer, i.e. send initial FSK packet. + s: act as service centre talking to a phone. +Messages are processed as per text file message queues. +smsq (a separate software) is a command to generate message +queues and send messages. + +\end{verbatim} + + +\section{SoftHangup} +\subsection{Synopsis} +\begin{verbatim} +Soft Hangup Application +\end{verbatim} +\subsection{Description} +\begin{verbatim} + SoftHangup(Technology/resource|options) +Hangs up the requested channel. If there are no channels to hangup, +the application will report it. +- 'options' may contain the following letter: + 'a' : hang up all channels on a specified device instead of a single resource + +\end{verbatim} + + +\section{SpeechActivateGrammar} +\subsection{Synopsis} +\begin{verbatim} +Activate a Grammar +\end{verbatim} +\subsection{Description} +\begin{verbatim} +SpeechActivateGrammar(Grammar Name) +This activates the specified grammar to be recognized by the engine. A grammar tells the speech recognition engine what to recognize, +and how to portray it back to you in the dialplan. The grammar name is the only argument to this application. + +\end{verbatim} + + +\section{SpeechBackground} +\subsection{Synopsis} +\begin{verbatim} +Play a sound file and wait for speech to be recognized +\end{verbatim} +\subsection{Description} +\begin{verbatim} +SpeechBackground(Sound File|Timeout) +This application plays a sound file and waits for the person to speak. Once they start speaking playback of the file stops, and silence is heard. +Once they stop talking the processing sound is played to indicate the speech recognition engine is working. +Once results are available the application returns and results (score and text) are available using dialplan functions. +The first text and score are ${SPEECH_TEXT(0)} AND ${SPEECH_SCORE(0)} while the second are ${SPEECH_TEXT(1)} and ${SPEECH_SCORE(1)}. +The first argument is the sound file and the second is the timeout. Note the timeout will only start once the sound file has stopped playing. + +\end{verbatim} + + +\section{SpeechCreate} +\subsection{Synopsis} +\begin{verbatim} +Create a Speech Structure +\end{verbatim} +\subsection{Description} +\begin{verbatim} +SpeechCreate(engine name) +This application creates information to be used by all the other applications. It must be called before doing any speech recognition activities such as activating a grammar. +It takes the engine name to use as the argument, if not specified the default engine will be used. + +\end{verbatim} + + +\section{SpeechDeactivateGrammar} +\subsection{Synopsis} +\begin{verbatim} +Deactivate a Grammar +\end{verbatim} +\subsection{Description} +\begin{verbatim} +SpeechDeactivateGrammar(Grammar Name) +This deactivates the specified grammar so that it is no longer recognized. The only argument is the grammar name to deactivate. + +\end{verbatim} + + +\section{SpeechDestroy} +\subsection{Synopsis} +\begin{verbatim} +End speech recognition +\end{verbatim} +\subsection{Description} +\begin{verbatim} +SpeechDestroy() +This destroys the information used by all the other speech recognition applications. +If you call this application but end up wanting to recognize more speech, you must call SpeechCreate +again before calling any other application. It takes no arguments. + +\end{verbatim} + + +\section{SpeechLoadGrammar} +\subsection{Synopsis} +\begin{verbatim} +Load a Grammar +\end{verbatim} +\subsection{Description} +\begin{verbatim} +SpeechLoadGrammar(Grammar Name|Path) +Load a grammar only on the channel, not globally. +It takes the grammar name as first argument and path as second. + +\end{verbatim} + + +\section{SpeechProcessingSound} +\subsection{Synopsis} +\begin{verbatim} +Change background processing sound +\end{verbatim} +\subsection{Description} +\begin{verbatim} +SpeechProcessingSound(Sound File) +This changes the processing sound that SpeechBackground plays back when the speech recognition engine is processing and working to get results. +It takes the sound file as the only argument. + +\end{verbatim} + + +\section{SpeechStart} +\subsection{Synopsis} +\begin{verbatim} +Start recognizing voice in the audio stream +\end{verbatim} +\subsection{Description} +\begin{verbatim} +SpeechStart() +Tell the speech recognition engine that it should start trying to get results from audio being fed to it. This has no arguments. + +\end{verbatim} + + +\section{SpeechUnloadGrammar} +\subsection{Synopsis} +\begin{verbatim} +Unload a Grammar +\end{verbatim} +\subsection{Description} +\begin{verbatim} +SpeechUnloadGrammar(Grammar Name) +Unload a grammar. It takes the grammar name as the only argument. + +\end{verbatim} + + +\section{StackPop} +\subsection{Synopsis} +\begin{verbatim} +Remove one address from gosub stack +\end{verbatim} +\subsection{Description} +\begin{verbatim} +StackPop() + Removes last label on the stack, discarding it. + +\end{verbatim} + + +\section{StartMusicOnHold} +\subsection{Synopsis} +\begin{verbatim} +Play Music On Hold +\end{verbatim} +\subsection{Description} +\begin{verbatim} +StartMusicOnHold(class): Starts playing music on hold, uses default music class for channel. +Starts playing music specified by class. If omitted, the default +music source for the channel will be used. Always returns 0. + +\end{verbatim} + + +\section{StopMixMonitor} +\subsection{Synopsis} +\begin{verbatim} +Stop recording a call through MixMonitor +\end{verbatim} +\subsection{Description} +\begin{verbatim} + StopMixMonitor() + +Stops the audio recording that was started with a call to MixMonitor() +on the current channel. + +\end{verbatim} + + +\section{StopMonitor} +\subsection{Synopsis} +\begin{verbatim} +Stop monitoring a channel +\end{verbatim} +\subsection{Description} +\begin{verbatim} +StopMonitor +Stops monitoring a channel. Has no effect if the channel is not monitored + +\end{verbatim} + + +\section{StopMusicOnHold} +\subsection{Synopsis} +\begin{verbatim} +Stop Playing Music On Hold +\end{verbatim} +\subsection{Description} +\begin{verbatim} +StopMusicOnHold: Stops playing music on hold. + +\end{verbatim} + + +\section{StopPlayTones} +\subsection{Synopsis} +\begin{verbatim} +Stop playing a tone list +\end{verbatim} +\subsection{Description} +\begin{verbatim} +Stop playing a tone list +\end{verbatim} + + +\section{System} +\subsection{Synopsis} +\begin{verbatim} +Execute a system command +\end{verbatim} +\subsection{Description} +\begin{verbatim} + System(command): Executes a command by using system(). If the command +fails, the console should report a fallthrough. +Result of execution is returned in the SYSTEMSTATUS channel variable: + FAILURE Could not execute the specified command + SUCCESS Specified command successfully executed + +Old behaviour: +If the command itself executes but is in error, and if there exists +a priority n + 101, where 'n' is the priority of the current instance, +then the channel will be setup to continue at that priority level. +Note that this jump functionality has been deprecated and will only occur +if the global priority jumping option is enabled in extensions.conf. + +\end{verbatim} + + +\section{TestClient} +\subsection{Synopsis} +\begin{verbatim} +Execute Interface Test Client +\end{verbatim} +\subsection{Description} +\begin{verbatim} +TestClient(testid): Executes test client with given testid. +Results stored in /var/log/asterisk/testreports/<testid>-client.txt +\end{verbatim} + + +\section{TestServer} +\subsection{Synopsis} +\begin{verbatim} +Execute Interface Test Server +\end{verbatim} +\subsection{Description} +\begin{verbatim} +TestServer(): Perform test server function and write call report. +Results stored in /var/log/asterisk/testreports/<testid>-server.txt +\end{verbatim} + + +\section{Transfer} +\subsection{Synopsis} +\begin{verbatim} +Transfer caller to remote extension +\end{verbatim} +\subsection{Description} +\begin{verbatim} + Transfer([Tech/]dest[|options]): Requests the remote caller be transferred +to a given destination. If TECH (SIP, IAX2, LOCAL etc) is used, only +an incoming call with the same channel technology will be transfered. +Note that for SIP, if you transfer before call is setup, a 302 redirect +SIP message will be returned to the caller. + +The result of the application will be reported in the TRANSFERSTATUS +channel variable: + SUCCESS Transfer succeeded + FAILURE Transfer failed + UNSUPPORTED Transfer unsupported by channel driver +The option string many contain the following character: +'j' -- jump to n+101 priority if the channel transfer attempt + fails + +\end{verbatim} + + +\section{TryExec} +\subsection{Synopsis} +\begin{verbatim} +Executes dialplan application, always returning +\end{verbatim} +\subsection{Description} +\begin{verbatim} +Usage: TryExec(appname(arguments)) + Allows an arbitrary application to be invoked even when not +hardcoded into the dialplan. To invoke external applications +see the application System. Always returns to the dialplan. +The channel variable TRYSTATUS will be set to: + SUCCESS if the application returned zero + FAILED if the application returned non-zero + NOAPP if the application was not found or was not specified + +\end{verbatim} + + +\section{TrySystem} +\subsection{Synopsis} +\begin{verbatim} +Try executing a system command +\end{verbatim} +\subsection{Description} +\begin{verbatim} + TrySystem(command): Executes a command by using system(). +on any situation. +Result of execution is returned in the SYSTEMSTATUS channel variable: + FAILURE Could not execute the specified command + SUCCESS Specified command successfully executed + APPERROR Specified command successfully executed, but returned error code + +Old behaviour: +If the command itself executes but is in error, and if +there exists a priority n + 101, where 'n' is the priority of the current +instance, then the channel will be setup to continue at that +priority level. Otherwise, System will terminate. + +\end{verbatim} + + +\section{UnpauseMonitor} +\subsection{Synopsis} +\begin{verbatim} +Unpause monitoring of a channel +\end{verbatim} +\subsection{Description} +\begin{verbatim} +UnpauseMonitor +Unpauses monitoring of a channel on which monitoring had +previously been paused with PauseMonitor. + +\end{verbatim} + + +\section{UnpauseQueueMember} +\subsection{Synopsis} +\begin{verbatim} +Unpauses a queue member +\end{verbatim} +\subsection{Description} +\begin{verbatim} + UnpauseQueueMember([queuename]|interface[|options]): +Unpauses (resumes calls to) a queue member. +This is the counterpart to PauseQueueMember and operates exactly the +same way, except it unpauses instead of pausing the given interface. +The option string may contain zero or more of the following characters: + 'j' -- jump to +101 priority when appropriate. + This application sets the following channel variable upon completion: + UPQMSTATUS The status of the attempt to unpause a queue + member as a text string, one of + UNPAUSED | NOTFOUND +Example: UnpauseQueueMember(|SIP/3000) + +\end{verbatim} + + +\section{UserEvent} +\subsection{Synopsis} +\begin{verbatim} +Send an arbitrary event to the manager interface +\end{verbatim} +\subsection{Description} +\begin{verbatim} + UserEvent(eventname[|body]): Sends an arbitrary event to the manager +interface, with an optional body representing additional arguments. The +body may be specified as a | delimeted list of headers. Each additional +argument will be placed on a new line in the event. The format of the +event will be: + Event: UserEvent + UserEvent: <specified event name> + [body] +If no body is specified, only Event and UserEvent headers will be present. + +\end{verbatim} + + +\section{Verbose} +\subsection{Synopsis} +\begin{verbatim} +Send arbitrary text to verbose output +\end{verbatim} +\subsection{Description} +\begin{verbatim} +Verbose([<level>|]<message>) + level must be an integer value. If not specified, defaults to 0. + +\end{verbatim} + + +\section{VMAuthenticate} +\subsection{Synopsis} +\begin{verbatim} +Authenticate with Voicemail passwords +\end{verbatim} +\subsection{Description} +\begin{verbatim} + VMAuthenticate([mailbox][@context][|options]): This application behaves the +same way as the Authenticate application, but the passwords are taken from +voicemail.conf. + If the mailbox is specified, only that mailbox's password will be considered +valid. If the mailbox is not specified, the channel variable AUTH_MAILBOX will +be set with the authenticated mailbox. + + Options: + s - Skip playing the initial prompts. + +\end{verbatim} + + +\section{VoiceMail} +\subsection{Synopsis} +\begin{verbatim} +Leave a Voicemail message +\end{verbatim} +\subsection{Description} +\begin{verbatim} + VoiceMail(mailbox[@context][&mailbox[@context]][...][|options]): This +application allows the calling party to leave a message for the specified +list of mailboxes. When multiple mailboxes are specified, the greeting will +be taken from the first mailbox specified. Dialplan execution will stop if the +specified mailbox does not exist. + The Voicemail application will exit if any of the following DTMF digits are +received: + 0 - Jump to the 'o' extension in the current dialplan context. + * - Jump to the 'a' extension in the current dialplan context. + This application will set the following channel variable upon completion: + VMSTATUS - This indicates the status of the execution of the VoiceMail + application. The possible values are: + SUCCESS | USEREXIT | FAILED + + Options: + b - Play the 'busy' greeting to the calling party. + g(#) - Use the specified amount of gain when recording the voicemail + message. The units are whole-number decibels (dB). + s - Skip the playback of instructions for leaving a message to the + calling party. + u - Play the 'unavailable greeting. + j - Jump to priority n+101 if the mailbox is not found or some other + error occurs. + +\end{verbatim} + + +\section{VoiceMailMain} +\subsection{Synopsis} +\begin{verbatim} +Check Voicemail messages +\end{verbatim} +\subsection{Description} +\begin{verbatim} + VoiceMailMain([mailbox][@context][|options]): This application allows the +calling party to check voicemail messages. A specific mailbox, and optional +corresponding context, may be specified. If a mailbox is not provided, the +calling party will be prompted to enter one. If a context is not specified, +the 'default' context will be used. + + Options: + p - Consider the mailbox parameter as a prefix to the mailbox that + is entered by the caller. + g(#) - Use the specified amount of gain when recording a voicemail + message. The units are whole-number decibels (dB). + s - Skip checking the passcode for the mailbox. + a(#) - Skip folder prompt and go directly to folder specified. + Defaults to INBOX + +\end{verbatim} + + +\section{Wait} +\subsection{Synopsis} +\begin{verbatim} +Waits for some time +\end{verbatim} +\subsection{Description} +\begin{verbatim} + Wait(seconds): This application waits for a specified number of seconds. +Then, dialplan execution will continue at the next priority. + Note that the seconds can be passed with fractions of a second. For example, +'1.5' will ask the application to wait for 1.5 seconds. + +\end{verbatim} + + +\section{WaitExten} +\subsection{Synopsis} +\begin{verbatim} +Waits for an extension to be entered +\end{verbatim} +\subsection{Description} +\begin{verbatim} + WaitExten([seconds][|options]): This application waits for the user to enter +a new extension for a specified number of seconds. + Note that the seconds can be passed with fractions of a second. For example, +'1.5' will ask the application to wait for 1.5 seconds. + Options: + m[(x)] - Provide music on hold to the caller while waiting for an extension. + Optionally, specify the class for music on hold within parenthesis. + +\end{verbatim} + + +\section{WaitForRing} +\subsection{Synopsis} +\begin{verbatim} +Wait for Ring Application +\end{verbatim} +\subsection{Description} +\begin{verbatim} + WaitForRing(timeout) +Returns 0 after waiting at least timeout seconds. and +only after the next ring has completed. Returns 0 on +success or -1 on hangup + +\end{verbatim} + + +\section{WaitForSilence} +\subsection{Synopsis} +\begin{verbatim} +Waits for a specified amount of silence +\end{verbatim} +\subsection{Description} +\begin{verbatim} + WaitForSilence(silencerequired[|iterations][|timeout]) +Wait for Silence: Waits for up to 'silencerequired' +milliseconds of silence, 'iterations' times or once if omitted. +An optional timeout specified the number of seconds to return +after, even if we do not receive the specified amount of silence. +Use 'timeout' with caution, as it may defeat the purpose of this +application, which is to wait indefinitely until silence is detected +on the line. This is particularly useful for reverse-911-type +call broadcast applications where you need to wait for an answering +machine to complete its spiel before playing a message. +The timeout parameter is specified only to avoid an infinite loop in +cases where silence is never achieved. Typically you will want to +include two or more calls to WaitForSilence when dealing with an answering +machine; first waiting for the spiel to finish, then waiting for the beep, etc. + +Examples: + - WaitForSilence(500|2) will wait for 1/2 second of silence, twice + - WaitForSilence(1000) will wait for 1 second of silence, once + - WaitForSilence(300|3|10) will wait for 300ms silence, 3 times, + and returns after 10 sec, even if silence is not detected + +Sets the channel variable WAITSTATUS with to one of these values: +SILENCE - if exited with silence detected +TIMEOUT - if exited without silence detected after timeout + +\end{verbatim} + + +\section{WaitMusicOnHold} +\subsection{Synopsis} +\begin{verbatim} +Wait, playing Music On Hold +\end{verbatim} +\subsection{Description} +\begin{verbatim} +WaitMusicOnHold(delay): Plays hold music specified number of seconds. Returns 0 when +done, or -1 on hangup. If no hold music is available, the delay will +still occur with no sound. + +\end{verbatim} + + +\section{While} +\subsection{Synopsis} +\begin{verbatim} +Start a while loop +\end{verbatim} +\subsection{Description} +\begin{verbatim} +Usage: While(<expr>) +Start a While Loop. Execution will return to this point when +EndWhile is called until expr is no longer true. + +\end{verbatim} + + +\section{Zapateller} +\subsection{Synopsis} +\begin{verbatim} +Block telemarketers with SIT +\end{verbatim} +\subsection{Description} +\begin{verbatim} + Zapateller(options): Generates special information tone to block +telemarketers from calling you. Options is a pipe-delimited list of +options. The following options are available: +'answer' causes the line to be answered before playing the tone, +'nocallerid' causes Zapateller to only play the tone if there +is no callerid information available. Options should be separated by | +characters + +\end{verbatim} + + +\section{ZapBarge} +\subsection{Synopsis} +\begin{verbatim} +Barge in (monitor) Zap channel +\end{verbatim} +\subsection{Description} +\begin{verbatim} + ZapBarge([channel]): Barges in on a specified zap +channel or prompts if one is not specified. Returns +-1 when caller user hangs up and is independent of the +state of the channel being monitored. +\end{verbatim} + + +\section{ZapRAS} +\subsection{Synopsis} +\begin{verbatim} +Executes Zaptel ISDN RAS application +\end{verbatim} +\subsection{Description} +\begin{verbatim} + ZapRAS(args): Executes a RAS server using pppd on the given channel. +The channel must be a clear channel (i.e. PRI source) and a Zaptel +channel to be able to use this function (No modem emulation is included). +Your pppd must be patched to be zaptel aware. Arguments should be +separated by | characters. + +\end{verbatim} + + +\section{ZapScan} +\subsection{Synopsis} +\begin{verbatim} +Scan Zap channels to monitor calls +\end{verbatim} +\subsection{Description} +\begin{verbatim} + ZapScan([group]) allows a call center manager to monitor Zap channels in +a convenient way. Use '#' to select the next channel and use '*' to exit +Limit scanning to a channel GROUP by setting the option group argument. + +\end{verbatim} + + |