FONスクリプトAPI(FON Script API)は無償提供されていますが、Fonalityのサポート対象外になっています。Fonalityのサポートエンジニアは、あなたや第三者が書いた独自のAGIスクリプトに関して、追加開発、修正、トラブルシューティングを行なうことはありません。このページに書かれている内容や、一般的なAGIに関する質問などは受け付けておりませんのでご了承ください。
The FON Script Interface allows software developers and system integrators to extend the functionality of their PBXtra CCE systems by allowing them to create customized extensions to the PBXtra system using a standardized API. This facility allows the creation of custom IVR applications that will impress your friends, and probably get you promoted ;-).
One example of a FON Script application is a store locater. This program would prompt the caller to key in his zip code, and look up a list of stores in a cross reference table of a database. Once the correct store is identified the caller is then automatically connected with the correct store.
You must have the PBXtra Call Center Edition of the Fonality software in order to use this feature. In CCE it will be shown in your call menu as "Run Script".
Exmaple:
my_script.agi
Example for a script listening on a port:
fon://ip.add.re.ss:port
Note: If you are running PBXtra Core version 1.2.14-fon-o or newer, you can also pass HTTP style arguments to the script like this:
my_script.agi?extension=${EXTEN}
You can look for them in your script using the environment variables.
If you have connected your FON Interface Script to your call menu, you can dial "0" from any of your extensions to reach your main menu, and subsequently reach the AGI for testing. You may also dial in from an external line.
This example Perl script is available under the GNU GPLv2 or higher License. It demonstrates basic interaction with the PBXtra Core software from your server using the STDOUT file handle. It also shows how to get digit input from your callers.
#!/usr/bin/perl use strict; # Load the strict pragma to enforce good program form use Socket; # Load the socket library to let us speak TCP/IP use Carp; # Load an error handling library use IO::Handle; # Overload the I/O functions with more appropriate ones my $port = 4573; # TCP Port number to listen on $|=1; # Activate auto-flush to disable text buffering in perl # Setup Variables my %AGI; # Init a hash structure to be used for AGI environment my $tests = 0; # A counter for the number of tests that have been run my $fail = 0; # A counter for failure conditions my $pass = 0; # A counter for success conditions ############################################################################## # checkresult: A function to check the return status of a command to # detect any errors. # # Arguments: $res = A result from an AGI command # Returns: none, sets $pass and $fail counters ############################################################################## sub checkresult { my ($res) = @_; my $retval; $tests++; chomp $res; if ($res =~ /^200/) { $res =~ /result=(-?\d+)/; if (!length($1)) { print STDERR "FAIL ($res)\n"; $fail++; } else { print STDERR "PASS ($1)\n"; $pass++; } } else { print STDERR "FAIL (unexpected result '$res')\n"; $fail++; } } # Set up a TCP/IP socket, and bind/listen to it for new connections # from the PXBtra Core software (the client) socket(SERVER, PF_INET, SOCK_STREAM, 0); setsockopt(SERVER, SOL_SOCKET, SO_REUSEADDR, pack("l", 1)); bind(SERVER, sockaddr_in($port, INADDR_ANY)) || die("can't bind\n"); listen(SERVER, SOMAXCONN); # Start the FastAGI/FON request loop for(;;) { # We will wait for a connection. The accept() call will block until # a connection arrives. While processing, connections line up. my $raddr = accept(CLIENT, SERVER); # Get the next connection my ($s, $p) = sockaddr_in($raddr); # CLIENT->autoflush(1); # Disable text buffering client connection while(<CLIENT>) { # Read all the text sent from the client to the server chomp; # Eliminate any newlines on the end of the text last unless length($_); # If the line is empty, stop reading if (/^agi_(\w+)\:\s+(.*)$/) { # Look for variables starting with agi_ # Set AGI hash keyed by variable name to the value found $AGI{$1} = $2; # $1 is match from first set of parens above # $2 is match from second set of parens... } } # Print out a summary of the AGI environment. # Visible in asterisk -r console. print STDERR "AGI Environment Dump from $s:$p --\n"; foreach my $i (sort keys %AGI) { print STDERR " -- $i = $AGI{$i}\n"; } # Begin the first test. This plays a sound file, checks the result, and # counts the result. print STDERR "1. Testing 'sendfile'..."; # Tell PBXtra Core (client) to play a sound file to the caller print CLIENT "STREAM FILE beep \"\"\n"; my $result = <CLIENT>; # Read data from the PBXtra Core (client) &checkresult($result); # Check the output for success/failure # Ask PBXtra Core to read a number to the caller print STDERR "2. Testing 'saynumber'..."; print CLIENT "SAY NUMBER 192837465 \"\"\n"; my $result = <CLIENT>; &checkresult($result); # Ask PBXtra Core to read a number to wait 1 second for a digit print STDERR "3. Testing 'waitdtmf'..."; print CLIENT "WAIT FOR DIGIT 1000\n"; # Wait 1 sec (1000 ms) for digit my $result = <CLIENT>; &checkresult($result); # Ask PBXtra Core to record sound in a file print STDERR "4. Testing 'record'..."; print CLIENT "RECORD FILE testagi gsm 1234 3000\n"; my $result = <CLIENT>; &checkresult($result); # Play back the file we recorded print STDERR "5. Testing 'record' playback..."; print CLIENT "STREAM FILE testagi \"\"\n"; my $result = <CLIENT>; &checkresult($result); close(CLIENT); print STDERR "================== Complete ======================\n"; print STDERR "$tests tests completed, $pass passed, $fail failed\n"; print STDERR "==================================================\n"; }
The FON interface is 100% interface compatible with FastAGI. You can use any FastAGI development library you wish. The FON Interface for PBXtra Core improves efficiency over AGI, and allows for continued execution of your call menu if you ever have a problem with your FON Interface Script.
You may write FON Interface Scripts in any language you wish. They must be able to read input on STDIN, write output on STDOUT, and must be able to implement a simple request loop. See the sample script above for a working implementation. Technically speaking the FON Interface Script is a TCP/IP server program. Here are some other programming aids:
Note that each call will occupy your FastAGI server for its duration until you set a context and priority sequence number and exit, allowing the call to continue to flow through your system as indicated. To support multiple callers at once you will want to employ a pre-fork multi-process server or a multi-threaded server to allow multiple calls at once. For Perl, consider the Net::Server::Prefork library.
If you decide to multi-thread, and you will have a high call volume, be careful. The default stack size is set to 2MB, and you may quickly soak up a lot of memory, and waste a lot of system resources allocating new threads. Consider using a thread pool so you can create all the threads you need at startup.
NOTE: For performance and stability reasons, please do not consider loading MySQL or any other database engine on the PBXtra server directly. If your application will use a database, then run it on a separate server and connect to it over a socket connection.
The following environment variables will be available to your FON Interface Script upon execution. Note that the values will be set dynamically based on the channel information on your inbound call.
Purpose Answer channel if not already in answer state.
Returns -1 on channel failure, or 0 if successful.
Purpose Cause the channel to automatically hangup at <time> seconds in the future. If <time> is 0 then the auto-hangup feature is disabled on this channel.
Returns 0
Note If the channel is hungup prior to <time> seconds, this setting has no effect.
Purpose Return the status of the specified channel. If no channel name is specified, return the status of the current channel.
Returns -1 There is no channel that matches the given <channelname> 0 Channel is down and available 1 Channel is down, but reserved 2 Channel is off hook 3 Digits (or equivalent) have been dialed 4 Line is ringing 5 Remote end is ringing 6 Line is up 7 Line is busy
Examples CHANNEL STATUS Return the status of the current channel.
CHANNEL STATUS Zap/9-1 Return the status of channel Zap/9-1
Note The <channelname> to use is the same as the channel names reported by the Asterisk console 'show channels' command.
Purpose Executes the specified Asterisk <application> with given <options>.
Returns Whatever the application returns, or -2 on failure to find the application.
The following list details all of the applications that can be invoked within the PBXtra Core software using the EXEC interface command.
-- Info about application 'AbsoluteTimeout' -- [Synopsis] Set absolute maximum time of call [Description] AbsoluteTimeout(seconds): This application will set the absolute maximum amount of time permitted for a call. A setting of 0 disables the timeout. AbsoluteTimeout has been deprecated in favor of Set(TIMEOUT(absolute)=timeout)
-- Info about application 'AddQueueMember' -- [Synopsis] Dynamically adds queue members [Description] AddQueueMember(queuename[|interface[|penalty[|options]]]): 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)
-- Info about application 'ADSIProg' -- [Synopsis] Load Asterisk ADSI Scripts into phone [Description] ADSIProg(script): This application programs an ADSI Phone with the given script. If nothing is specified, the default script (asterisk.adsi) is used.
-- Info about application 'AGI' -- [Synopsis] Executes an AGI compliant application [Description] [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. Returns -1 on hangup (except for DeadAGI) or if application requested hangup, or 0 on non-hangup exit. Using 'EAGI' provides enhanced AGI, with incoming audio available out of band on file descriptor 3 Use the CLI command 'show agi' to list available agi commands
-- Info about application 'AlarmReceiver' -- [Synopsis] Provide support for receving alarm reports from a burglar or fire alarm panel [Description] 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.
-- Info about application 'Answer' -- [Synopsis] Answer a channel if ringing [Description] Answer([delay]): If the call has not been answered, this application will answer it. Otherwise, it has no effect on the call. If a delay is specified, Asterisk will wait this number of milliseconds before answering the call.
-- Info about application 'AppendCDRUserField' -- [Synopsis] Append to the CDR user field [Description] [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().
-- Info about application 'Authenticate' -- [Synopsis] Authenticate a user [Description] Authenticate(password[|options]): 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)
-- Info about application 'BackGround' -- [Synopsis] Play a file while awaiting extension [Description] 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 explicity 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
-- Info about application 'BackgroundDetect' -- [Synopsis] Background a file with talk detect [Description] 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.
-- Info about application 'Busy' -- [Synopsis] Indicate the Busy condition [Description] 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.
-- Info about application 'ChangeMonitor' -- [Synopsis] Change monitoring filename of a channel [Description] 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.
-= Info about application 'ChanInUse' =- [Synopsis] Checks to see if channel is on a call. [Description] ChanInUse(Location[|[!]Context]): Checks to see if the specific channel is on a call, or optionally on a call in a specific Context. If Context is negated with "!", checks if the channel is on a call *not* in the specified Context. Jumps to n+101 priority if it is on a call as specified. Returns -1 if there is an error. Example: ChanInUse(SIP/0004F2123456|neoagent)
-- Info about application 'ChanIsAvail' -- [Synopsis] Check channel availability [Description] 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
-- Info about application 'ChanSpy' -- [Synopsis] Listen to the audio of an active channel [Description] ChanSpy([chanprefix][|options]): This application is used to listen to the audio from an active 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 'grp'. q - Don't play a beep when beginning to spy on a channel. 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.
-- Info about application 'CheckGroup' -- [Synopsis] Check the channel count of a group against a limit [Description] Usage: CheckGroup(max[@category][|options]) Checks that the current number of total channels in the current channel's group does not exceed 'max'. If the number does not exceed 'max', we continue to the next step. The option string may contain zero of the following character: 'j' -- jump to n+101 priority if the number does in fact exceed max, and priority n+101 exists. Execuation then continues at that step, otherwise -1 is returned. This application sets the following channel variable upon successful completion: CHECKGROUPSTATUS The status of the check that the current channel's group does not exceed 'max'. It's value is one of OK | OVERMAX
-- Info about application 'Congestion' -- [Synopsis] Indicate the Congestion condition [Description] Congestion([timeout]): This application will indicate the congenstion 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.
-- Info about application 'ControlPlayback' -- [Synopsis] Play a file with fast forward and rewind [Description] 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
-- Info about application 'Curl' -- [Synopsis] Load an external URL [Description] Curl(URL[|postdata]): This application will request the specified URL. It is mainly used for signalling external applications of an event. Parameters: URL - This is the external URL to request. postdata - This information will be treated as POST data. This application will set the following variable: CURL - This variable will contain the resulting page. This application has been deprecated in favor of the CURL function.
-- Info about application 'Cut' -- [Synopsis] Splits a variable's contents using the specified delimiter [Description] Cut(newvar=varname,delimiter,fieldspec): This applicaiton will split the contents of a variable based on the given delimeter and store the result in a new variable. Parameters: newvar - new variable created from result string varname - variable you want cut delimiter - defaults to '-' fieldspec - number of the field you want (1-based offset) may also be specified as a range (with -) or group of ranges and fields (with &) This application has been deprecated in favor of the CUT function.
-- Info about application 'DateTime' -- [Synopsis] Says a specified time in a custom format [Description] 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"
-- Info about application 'DBdel' -- [Synopsis] Delete a key from the database [Description] DBdel(family/key): This applicaiton will delete a key from the Asterisk database.
-- Info about application 'DBdeltree' -- [Synopsis] Delete a family or keytree from the database [Description] DBdeltree(family[/keytree]): This application will delete a family or keytree from the Asterisk database
-- Info about application 'DBget' -- [Synopsis] Retrieve a value from the database [Description] DBget(varname=family/key[|options]): This application will retrieve a value from the Asterisk database and store it in the given variable. Options: j - Jump to priority n+101 if the requested family/key isn't found. This application sets the following channel variable upon completion: DBGETSTATUS - This variable will contain the status of the attempt FOUND | NOTFOUND This application has been deprecated in favor of the DB function.
-- Info about application 'DBput' -- [Synopsis] Store a value in the database [Description] DBput(family/key=value): This application will store the given value in the specified location in the Asterisk database. This application has been deprecated in favor of the DB function.
-- Info about application 'DeadAGI' -- [Synopsis] Executes AGI on a hungup channel [Description] [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. Returns -1 on hangup (except for DeadAGI) or if application requested hangup, or 0 on non-hangup exit. Using 'EAGI' provides enhanced AGI, with incoming audio available out of band on file descriptor 3 Use the CLI command 'show agi' to list available agi commands ]
-- Info about application 'Dial' -- [Synopsis] Place a call and connect to the current channel [Description] Dial(Technology/resource[&Tech2/resource2...][|timeout][|options][|URL]): This applicaiton 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 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. 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. 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. 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.
-- Info about application 'Dictate' -- [Synopsis] Virtual Dictation Machine [Description] Dictate([<base_dir>]) Start dictation machine using optional base dir for files.
-- Info about application 'DigitTimeout' -- [Synopsis] Set maximum timeout between digits [Description] DigitTimeout(seconds): Set the maximum amount of time permitted between digits when the user is typing in an extension. When this timeout expires, after the user has started to type in an extension, the extension will be considered complete, and will be interpreted. Note that if an extension typed in is valid, it will not have to timeout to be tested, so typically at the expiry of this timeout, the extension will be considered invalid (and thus control would be passed to the 'i' extension, or if it doesn't exist the call would be terminated). The default timeout is 5 seconds. DigitTimeout has been deprecated in favor of Set(TIMEOUT(digit)=timeout)
-- Info about application 'Directory' -- [Synopsis] Provide directory of voicemail extensions [Description] 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 applicaiton will immediate 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: f - Allow the caller to enter the first name of a user in the directory instead of using the last name.
-- Info about application 'DISA' -- [Synopsis] DISA (Direct Inward System Access) [Description] 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 allow 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.
-- Info about application 'DumpChan' -- [Synopsis] Dump Info About The Calling Channel [Description] 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.
-- Info about application 'DUNDiLookup' -- [Synopsis] Look up a number with DUNDi [Description] DUNDiLookup(number[|context[|options]]) Looks up a given number in the global context specified or in the reserved 'e164' context if not specified. Returns -1 if the channel is hungup during the lookup or 0 otherwise. On completion, the variable ${DUNDTECH} and ${DUNDDEST} will contain the technology and destination of the appropriate technology and destination to access the number. If no answer was found, and the priority n + 101 exists, execution will continue at that location. Note that this will only occur if the global priority jumping option is enabled in extensions.conf. If the 'b' option is specified, the internal DUNDi cache will by bypassed.
-- Info about application 'EAGI' -- [Synopsis] Executes an EAGI compliant application [Description] [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. Returns -1 on hangup (except for DeadAGI) or if application requested hangup, or 0 on non-hangup exit. Using 'EAGI' provides enhanced AGI, with incoming audio available out of band on file descriptor 3 Use the CLI command 'show agi' to list available agi commands
-- Info about application 'Echo' -- [Synopsis] Echo audio read back to the user [Description] Echo(): Echo audio read from channel back to the channel. User can exit the application by either pressing the '#' key, or hanging up.
-- Info about application 'EndWhile' -- [Synopsis] End A While Loop [Description] Usage: EndWhile() Return to the previous called While
-- Info about application 'EnumLookup' -- [Synopsis] Lookup number in ENUM [Description] EnumLookup(exten[|option]): Looks up an extension via ENUM and sets the variable 'ENUM'. For VoIP URIs this variable will look like 'TECHNOLOGY/URI' with the appropriate technology. Currently, the enumservices SIP, H323, IAX, IAX2 and TEL are recognized. Returns status in the ENUMSTATUS channel variable: ERROR Failed to do a lookup <tech> Technology of the successful lookup: SIP, H323, IAX, IAX2 or TEL BADURI Got URI Asterisk does not understand. The option string may contain zero or the following character: 'j' -- jump to +101 priority if the lookup isn't successful. and jump to +51 priority on a TEL entry.
-- Info about application 'Eval' -- [Synopsis] Evaluates a string [Description] Usage: Eval(newvar=somestring) Normally Asterisk evaluates variables inline. But what if you want to store variable offsets in a database, to be evaluated later? Eval is the answer, by allowing a string to be evaluated twice in the dialplan, the first time as part of the normal dialplan, and the second using Eval.
-- Info about application 'Exec' -- [Synopsis] Executes internal application [Description] Usage: Exec(appname(arguments)) Allows an arbitrary application to be invoked even when not hardcoded into the dialplan. To invoke external applications see the application System. Returns whatever value the app returns or a non-zero value if the app cannot be found.
-- Info about application 'ExecIf' -- [Synopsis] Conditional exec [Description] 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.
-- Info about application 'ExecIfTime' -- [Synopsis] Conditional application execution based on the current time [Description] 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. Further information on the time speicification can be found in examples illustrating how to do time-based context includes in the dialplan.
-- Info about application 'ExternalIVR' -- [Synopsis] Interfaces with an external IVR application [Description] 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/README.externalivr for a protocol specification.
-- Info about application 'Festival' -- [Synopsis] Say text to the user [Description] 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)
-- Info about application 'Flash' -- [Synopsis] Flashes a Zap Trunk [Description] 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.
-- Info about application 'FlushQueueStats' -- [Synopsis] Flushes stats for specified queue [Description] FlushQueueStats(queuename): Flushes the stats for specified queue. Returns -1 if there is an error. Example: FlushQueueStats(techsupport)
-- Info about application 'ForkCDR' -- [Synopsis] Forks the Call Data Record [Description] 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.
-- Info about application 'GetCPEID' -- [Synopsis] Get ADSI CPE ID [Description] GetCPEID: Obtains and displays ADSI CPE ID and other information in order to properly setup zapata.conf for on-hook operations.
-- Info about application 'GetGroupCount' -- [Synopsis] Get the channel count of a group [Description] Usage: GetGroupCount([groupname][@category]) Calculates the group count for the specified group, or uses the current channel's group if not specifed (and non-empty). Stores result in GROUPCOUNT. Note: This application has been deprecated, please use the function GROUP_COUNT.
-- Info about application 'GetGroupMatchCount' -- [Synopsis] Get the channel count of all groups that match a pattern [Description] Usage: GetGroupMatchCount(groupmatch[@category]) Calculates the group count for all groups that match the specified pattern. Uses standard regular expression matching (see regex(7)). Stores result in GROUPCOUNT. Always returns 0. Note: This application has been deprecated, please use the function GROUP_MATCH_COUNT.
-- Info about application 'Gosub' -- [Synopsis] Jump to label, saving return address [Description] Gosub([[context|]exten|]priority) Jumps to the label specified, saving the return address.
-- Info about application 'GosubIf' -- [Synopsis] Jump to label, saving return address [Description] 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.
-- Info about application 'Goto' -- [Synopsis] Jump to a particular priority, extension, or context [Description] Goto([[context|]extension|]priority): This application will cause the calling channel to continue dialplan execution at the specified priority. If no specific extension, or extension and context, are specified, then this application will jump to the specified priority of the current extension. If the attempt to jump to another location in the dialplan is not successful, then the channel will continue at the next priority of the current extension.
-- Info about application 'GotoIf' -- [Synopsis] Conditional goto [Description] GotoIf(condition?[labeliftrue]:[labeliffalse]): This application will cause the calling channel to jump to the specified location in the dialplan based on the evaluation of the given condition. 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, but execution continues with the next priority in the dialplan.
-- Info about application 'GotoIfTime' -- [Synopsis] Conditional Goto based on the current time [Description] GotoIfTime(<times>|<weekdays>|<mdays>|<months>?[[context|]exten|]priority): This application will have the calling channel jump to the speicified location int the dialplan if the current time matches the given time specification. Further information on the time specification can be found in examples illustrating how to do time-based context includes in the dialplan.
-- Info about application 'Hangup' -- [Synopsis] Hang up the calling channel [Description] Hangup(): This application will hang up the calling channel.
-- Info about application 'HasNewVoicemail' -- [Synopsis] Conditionally branches to priority + 101 with the right options set [Description] 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>
-- Info about application 'HasVoicemail' -- [Synopsis] Conditionally branches to priority + 101 with the right options set [Description] 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>
-- Info about application 'IAX2Provision' -- [Synopsis] Provision a calling IAXy with a given template [Description] 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.
-- Info about application 'ICES' -- [Synopsis] Encode and stream using 'ices' [Description] 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).
-- Info about application 'ImportVar' -- [Synopsis] Import a variable from a channel into a new variable [Description] 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.
-- Info about application 'LookupBlacklist' -- [Synopsis] Look up Caller*ID name/number from blacklist database [Description] 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()
-- Info about application 'LookupCIDName' -- [Synopsis] Look up CallerID Name from local database [Description] 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.
-- Info about application 'Macro' -- [Synopsis] Macro Implementation [Description] 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.
-- Info about application 'MacroExit' -- [Synopsis] Exit From Macro [Description] 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.
-- Info about application 'MacroIf' -- [Synopsis] Conditional Macro Implementation [Description] 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()
-- Info about application 'MailboxExists' -- [Synopsis] Check to see if Voicemail mailbox exists [Description] 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.
-- Info about application 'Math' -- [Synopsis] Performs Mathematical Functions [Description] Math(returnvar,<number1><op><number 2> Perform floating point calculation on number 1 to number 2 and store the result in returnvar. Valid ops are: +,-,/,*,%,<,>,>=,<=,== and behave as their C equivalents. This application has been deprecated in favor of the MATH function.
-- Info about application 'MD5' -- [Synopsis] Calculate MD5 checksum [Description] MD5(<var>=<string>): Calculates a MD5 checksum on <string>. Returns hash value in a channel variable.
-- Info about application 'MD5Check' -- [Synopsis] Check MD5 checksum [Description] MD5Check(<md5hash>|<string>[|options]): Calculates a MD5 checksum on <string> and compares it with the hash. Returns 0 if <md5hash> is correct for <string>. The option string may contain zero or more of the following characters: 'j' -- jump to priority n+101 if the hash and string do not match This application sets the following channel variable upon completion: CHECKMD5STATUS The status of the MD5 check, one of the following MATCH | NOMATCH
-- Info about application 'MeetMe' -- [Synopsis] MeetMe conference bridge [Description] 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 'i' -- announce user join/leave 'm' -- set monitor only mode (Listen only, no talking) 'M' -- enable music on hold when the conference has a single caller '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.
-- Info about application 'MeetMeAdmin' -- [Synopsis] MeetMe conference Administration [Description] 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 conference 'M' -- Mute conference 'n' -- Unmute entire conference (except admin) 'N' -- Mute entire conference (except admin)
-- Info about application 'MeetMeCount' -- [Synopsis] MeetMe participant count [Description] 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.
-- Info about application 'Milliwatt' -- [Synopsis] Generate a Constant 1000Hz tone at 0dbm (mu-law) [Description] Milliwatt(): Generate a Constant 1000Hz tone at 0dbm (mu-law)
-- Info about application 'MixMonitor' -- [Synopsis] Record a call and mix the audio during the recording [Description] 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. 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.
-- Info about application 'Monitor' -- [Synopsis] Monitor a channel [Description] 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.
-- Info about application 'MP3Player' -- [Synopsis] Play an MP3 file or stream [Description] 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.
-- Info about application 'MusicOnHold' -- [Synopsis] Play Music On Hold indefinitely [Description] 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.
-- Info about application 'NBScat' -- [Synopsis] Play an NBS local stream [Description] NBScat: Executes nbscat to listen to the local NBS stream. User can exit by pressing any key .
-- Info about application 'NoCDR' -- [Synopsis] Tell Asterisk to not maintain a CDR for the current call [Description] NoCDR(): This application will tell Asterisk not to maintain a CDR for the current call.
-- Info about application 'NoOp' -- [Synopsis] Do Nothing [Description] 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.
-- Info about application 'Page' -- [Synopsis] Pages phones [Description] 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
-- Info about application 'Park' -- [Synopsis] Park yourself [Description] 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.
-- Info about application 'ParkAndAnnounce' -- [Synopsis] Park and Announce [Description] ParkAndAnnounce(announce:template|timeout|dial|[return_context]): Park a call into the parkinglot and announce the call over the console. announce template: colon separated list of files to announce, the word PARKED will be replaced by a say_digits of the ext the call is parked in 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=prio+1
-- Info about application 'ParkedCall' -- [Synopsis] Answer a parked call [Description] 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.
-- Info about application 'PauseQueueMember' -- [Synopsis] Pauses a queue member [Description] 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)
-- Info about application 'Pickup' -- [Synopsis] Directed Call Pickup [Description] Pickup(extension[@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.
-- Info about application 'Playback' -- [Synopsis] Play a file [Description] 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
-- Info about application 'PlayTones' -- [Synopsis] Play a tone list [Description] 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.
-- Info about application 'PrivacyManager' -- [Synopsis] Require phone number to be entered, if no CallerID sent [Description] 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
-- Info about application 'Progress' -- [Synopsis] Indicate progress [Description] Progress(): This application will request that in-band progress information be provided to the calling channel.
-- Info about application 'Queue' -- [Synopsis] Queue a call for a call queue [Description] Queue(queuename[|options[|URL][|announceoverride][|timeout]]): 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. '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 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
-- Info about application 'Random' -- [Synopsis] Conditionally branches, based upon a probability [Description] Random([probability]:[[context|]extension|]priority) probability := INTEGER in the range 1 to 100
-- Info about application 'Read' -- [Synopsis] Read a variable [Description] 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. 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 -- may be 'skip' to return immediately if the line is not up, or 'noanswer' 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 -- if greater than 0, that value will override the default timeout. Read should disconnect if the function fails or errors out.
-- Info about application 'ReadFile' -- [Synopsis] ReadFile(varname=file,length) [Description] ReadFile(varname=file,length) Varname - Result stored here. File - The name of the file to read. Length - Maximum number of characters to capture.
-- Info about application 'RealTime' -- [Synopsis] Realtime Data Lookup [Description] 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. e.g. prefix of 'var_' would make the column 'name' become the variable ${var_name}
-- Info about application 'RealTimeUpdate' -- [Synopsis] Realtime Data Rewrite [Description] 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>
-- Info about application 'Record' -- [Synopsis] Record to a file [Description] 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 instead of default '#' If filename contains '%d', these characters will be replaced with a number incremented by one each time the file is recorded. Use '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.
-- Info about application 'RemoveQueueMember' -- [Synopsis] Dynamically removes queue members [Description] 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)
-- Info about application 'ResetCDR' -- [Synopsis] Resets the Call Data Record [Description] 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.
-- Info about application 'ResponseTimeout' -- [Synopsis] Set maximum timeout awaiting response [Description] ResponseTimeout(seconds): This will set the maximum amount of time permitted to wait for an extension to dialed (see the WaitExten application), before the timeout occurs. If this timeout is reached, dialplan execution will continue at the 't' extension, if it exists. ResponseTimeout has been deprecated in favor of Set(TIMEOUT(response)=timeout)
-- Info about application 'RetryDial' -- [Synopsis] Place a call, retrying on failure allowing optional exit extension. [Description] 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.
-- Info about application 'Return' -- [Synopsis] Return from gosub routine [Description] Return() Jumps to the last label on the stack, removing it.
-- Info about application 'Ringing' -- [Synopsis] Indicate ringing tone [Description] Ringing(): This application will request that the channel indicate a ringing tone to the user.
-- Info about application 'SayAlpha' -- [Synopsis] Say Alpha [Description] SayAlpha(string): This application will play the sounds that correspond to the letters of the given string.
-- Info about application 'SayDigits' -- [Synopsis] Say Digits [Description] 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.
-- Info about application 'SayNumber' -- [Synopsis] Say Number [Description] 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.
-- Info about application 'SayPhonetic' -- [Synopsis] Say Phonetic [Description] SayPhonetic(string): This application will play the sounds from the phonetic alphabet that correspond to the letters in the given string.
-- Info about application 'SayUnixTime' -- [Synopsis] Says a specified time in a custom format [Description] 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"
-- Info about application 'SendDTMF' -- [Synopsis] Sends arbitrary DTMF digits [Description] 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.
-- Info about application 'SendImage' -- [Synopsis] Send an image file [Description] 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
-- Info about application 'SendText' -- [Synopsis] Send a Text Message [Description] 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
-- Info about application 'SendURL' -- [Synopsis] Send a URL [Description] 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 Clien 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 and will return -1 if the peer is unable to load the URL Old behaviour (deprecated): If the client does not support Asterisk "html" transport, and there exists a step with priority n + 101, then execution will continue at that step. Otherwise, execution will continue at the next priority level. SendURL only returns 0 if the URL was sent correctly or if the channel does not support HTML transport, and -1 otherwise.
-- Info about application 'Set' -- [Synopsis] Set channel variable(s) or function value(s) [Description] 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)
-- Info about application 'SetAccount' -- [Synopsis] Set the CDR Account Code [Description] SetAccount([account]): This application will set the channel account code for billing purposes. SetAccount has been deprecated in favor of the Set(CDR(accountcode)=account).
-- Info about application 'SetAMAFlags' -- [Synopsis] Set the AMA Flags [Description] SetAMAFlags([flag]): This channel will set the channel's AMA Flags for billing purposes.
-- Info about application 'SetCallerID' -- [Synopsis] Set CallerID [Description] SetCallerID(clid[|a]): Set Caller*ID on a call to a new value. Sets ANI as well if a flag is used.
-- Info about application 'SetCallerPres' -- [Synopsis] Set CallerID Presentation [Description] 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
-- Info about application 'SetCDRUserField' -- [Synopsis] Set the CDR user field [Description] [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().
-- Info about application 'SetCIDName' -- [Synopsis] Set CallerID Name [Description] SetCIDName(cname[|a]): Set Caller*ID Name on a call to a new value, while preserving the original Caller*ID number. This is useful for providing additional information to the called party. SetCIDName has been deprecated in favor of the function CALLERID(name)
-- Info about application 'SetCIDNum' -- [Synopsis] Set CallerID Number [Description] SetCIDNum(cnum[|a]): Set Caller*ID Number on a call to a new value, while preserving the original Caller*ID name. This is useful for providing additional information to the called party. Sets ANI as well if a flag is used. SetCIDNum has been deprecated in favor of the function CALLERID(number)
-- Info about application 'SetGlobalVar' -- [Synopsis] Set a global variable to a given value [Description] SetGlobalVar(variable=value): This application sets a given global variable to the specified value.
-- Info about application 'SetGroup' -- [Synopsis] Set the channel's group [Description] Usage: SetGroup(groupname[@category]) Sets the channel group to the specified value. Equivalent to Set(GROUP=group). Always returns 0.
-- Info about application 'SetLanguage' -- [Synopsis] Set the channel's preferred language [Description] SetLanguage(language): This will set the channel language to the given value. This information is used for the syntax in generation of numbers, and to choose a sound file in the given language, when it is available. For example, if language is set to 'fr' and the file 'demo-congrats' is requested to be played, if the file 'fr/demo-congrats' exists, then it will play that file. If not, it will play the normal 'demo-congrats'. For some language codes, SetLanguage also changes the syntax of some Asterisk functions, like SayNumber. SetLanguage has been deprecated in favor of Set(LANGUAGE()=language)
-- Info about application 'SetMusicOnHold' -- [Synopsis] Set default Music On Hold class [Description] 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.
-- Info about application 'SetRDNIS' -- [Synopsis] Set RDNIS Number [Description] SetRDNIS(cnum): Set RDNIS Number on a call to a new value. SetRDNIS has been deprecated in favor of the function CALLERID(rdnis)
-- Info about application 'SetTransferCapability' -- [Synopsis] Set ISDN Transfer Capability [Description] 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:
-- Info about application 'SetVar' -- [Synopsis] Set channel variable(s) [Description] SetVar(name1=value1|name2=value2|..[|options]): This application has been deprecated in favor of using the Set application.
-- Info about application 'SIPAddHeader' -- [Synopsis] Add a SIP header to the outbound call [Description] 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
-- Info about application 'SIPDtmfMode' -- [Synopsis] Change the dtmfmode for a SIP call [Description] SIPDtmfMode(inband|info|rfc2833): Changes the dtmfmode for a SIP call
-- Info about application 'SIPGetHeader' -- [Synopsis] Get a SIP header from an incoming call [Description] SIPGetHeader(var=headername[|options]): Sets a channel variable to the content of a SIP header Options: j - Jump to priority n+101 if the requested header isn't found. This application sets the following channel variable upon completion: SIPGETSTATUS - This variable will contain the status of the attempt FOUND | NOTFOUND This application has been deprecated in favor of the SIP_HEADER function.
-- Info about application 'SMS' -- [Synopsis] Communicates with SMS service centres and SMS capable analogue phones [Description] 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.
-- Info about application 'SoftHangup' -- [Synopsis] Soft Hangup Application [Description] 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
-- Info about application 'Sort' -- [Synopsis] Sorts a list of keywords and values [Description] Sort(newvar=key1:val1[,key2:val2[[...],keyN:valN]]): This application will sort the list provided in ascending order. The result will be stored in the specified variable name. This applicaiton has been deprecated in favor of the SORT function.
-- Info about application 'StackPop' -- [Synopsis] Remove one address from gosub stack [Description] StackPop() Removes last label on the stack, discarding it.
-- Info about application 'StartMusicOnHold' -- [Synopsis] Play Music On Hold [Description] 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.
-- Info about application 'StopMonitor' -- [Synopsis] Stop monitoring a channel [Description] StopMonitor Stops monitoring a channel. Has no effect if the channel is not monitored
-- Info about application 'StopMusicOnHold' -- [Synopsis] Stop Playing Music On Hold [Description] StopMusicOnHold: Stops playing music on hold.
-- Info about application 'StopPlayTones' -- [Synopsis] Stop playing a tone list [Description] Stop playing a tone list
-- Info about application 'System' -- [Synopsis] Execute a system command [Description] 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.
-- Info about application 'TestClient' -- [Synopsis] Execute Interface Test Client [Description] TestClient(testid): Executes test client with given testid. Results stored in /var/log/asterisk/testreports/<testid>-client.txt
-- Info about application 'TestServer' -- [Synopsis] Execute Interface Test Server [Description] TestServer(): Perform test server function and write call report. Results stored in /var/log/asterisk/testreports/<testid>-server.txt
-- Info about application 'Transfer' -- [Synopsis] Transfer caller to remote extension [Description] 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
-- Info about application 'TrySystem' -- [Synopsis] Try executing a system command [Description] 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.
-- Info about application 'TXTCIDName' -- [Synopsis] Lookup caller name from TXT record [Description] TXTCIDName(<CallerIDNumber>[|options]): Looks up a Caller Name via DNS and sets the variable 'TXTCIDNAME'. TXTCIDName will either be blank or return the value found in the TXT record in DNS. The option string may contain the following character: 'j' -- jump to n+101 priority if the lookup fails This application sets the following channel variable upon completion: TXTCIDNAMESTATUS The status of the lookup as a text string, one of SUCCESS | FAILED
-- Info about application 'UnpauseQueueMember' -- [Synopsis] Unpauses a queue member [Description] 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)
-- Info about application 'UserEvent' -- [Synopsis] Send an arbitrary event to the manager interface [Description] UserEvent(eventname[|body]): Sends an arbitrary event to the manager interface, with an optional body representing additional arguments. The format of the event will be: Event: UserEvent<specified event name> Channel: <channel name> Uniqueid: <call uniqueid> [body] If the body is not specified, only Event, Channel, and Uniqueid fields will be present. Returns 0.
-- Info about application 'Verbose' -- [Synopsis] Send arbitrary text to verbose output [Description] Verbose([<level>|]<message>) level must be an integer value. If not specified, defaults to 0.
-- Info about application 'VMAuthenticate' -- [Synopsis] Authenticate with Voicemail passwords [Description] 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.
-- Info about application 'VoiceMail' -- [Synopsis] Leave a Voicemail message [Description] 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.
-- Info about application 'VoiceMailMain' -- [Synopsis] Check Voicemail messages [Description] 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.
-- Info about application 'Wait' -- [Synopsis] Waits for some time [Description] 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.
-- Info about application 'WaitExten' -- [Synopsis] Waits for an extension to be entered [Description] 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.
-- Info about application 'WaitForRing' -- [Synopsis] Wait for Ring Application [Description] 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
-- Info about application 'WaitForSilence' -- [Synopsis] Waits for a specified amount of silence [Description] WaitForSilence(x[|y]) Wait for Silence: Waits for up to 'x' milliseconds of silence, 'y' times or 1 if omitted Set the channel variable WAITSTATUS with to one of these values:SILENCE - if silence of x ms was detectedTIMEOUT - if silence of x ms was not detected.Examples: - WaitForSilence(500|2) will wait for 1/2 second of silence, twice - WaitForSilence(1000) will wait for 1 second of silence, once
-- Info about application 'WaitMusicOnHold' -- [Synopsis] Wait, playing Music On Hold [Description] 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.
-- Info about application 'While' -- [Synopsis] Start A While Loop [Description] Usage: While(<expr>) Start a While Loop. Execution will return to this point when EndWhile is called until expr is no longer true.
-- Info about application 'Zapateller' -- [Synopsis] Block telemarketers with SIT [Description] 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
-- Info about application 'ZapBarge' -- [Synopsis] Barge in (monitor) Zap channel [Description] 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.
-- Info about application 'ZapRAS' -- [Synopsis] Executes Zaptel ISDN RAS application [Description] 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.
-- Info about application 'ZapScan' -- [Synopsis] Scan Zap channels to monitor calls [Description] 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.
Purpose Plays the given file and receives DTMF data. This is similar to STREAM FILE, but this command can accept and return many DTMF digits, while STREAM FILE returns immediately after the first DTMF digit is detected.
Returns If the command ended due to timeout then the result is of the form
200 Result=<digits> (timeout)
where <digits> will be zero or more ASCII characters depending on what the user pressed.
If the command ended because the maximum number of digits were entered then the result is of the form
200 Result=<digits>
and the number of digits returned will be equal to <max digits>.
In either case what you get are actual ASCII characters. For example if the user pressed the one key, the three key and then the star key, the result would be
200 Result=13* (timeout)
This differs from other commands with return DTMF as numbers representing ASCII characters.
Notes Don't give an extension with the filename.
Asterisk looks for the file to play in /var/lib/asterisk/sounds
If the user doesn't press any keys then the message plays, there is <timeout> milliseconds of silence then the command ends.
The user has the opportunity to press a key at any time during the message or the post-message silence. If the user presses a key while the message is playing, the message stops playing. When the first key is pressed a timer starts counting for <timeout> milliseconds. Every time the user presses another key the timer is restarted. The command ends when the counter goes to zero or the maximum number of digits is entered, whichever happens first.
If you don't specify a time out then a default timeout of 2000 is used following a pressed digit. If no digits are pressed then 6 seconds of silence follow the message.
If you want to specify <max digits> then you *must* specify a <timeout> as well.
If you don't specify <max digits> then the user can enter as many digits as they want.
Pressing the # key has the same effect as the timer running out: the command ends and any previously keyed digits are returned. A side effect of this is that there is no way to read a # key using this command.
Purpose Fetch the value of a variable.
Returns Returns 0 if the variable hasn't been set. Returns 1 followed by the value of the variable in parenthesis if it has been set.
Example SET VARIABLE Foo "This is a test" 200 result=1 GET VARIABLE Foo 200 result=1 (This is a test)
Purpose Hangup the specified channel. If no channel name is given, hang up the current channel.
Returns If the hangup was successful then the result is 200 result=1
If no channel matches the <channelname> you specified then the result is 200 result=-1
Examples HANGUP Hangup the current channel.
HANGUP Zap/9-1 Hangup channel Zap/9-1
Notes The <channelname> to use is the same as the channel names reported by the Asterisk console 'show channels' command.
With power comes responsibility. Hanging up channels other than your own isn't something that is done routinely. If you are not sure why you are doing so, then don't.
Purpose Receive a character of text from a connected channel. Waits up to <timeout> milliseconds for a character to arrive, or infinitely if <timeout> is zero.
Returns If a character is received, returns the ASCII value of the character as a decimal number. For example if the character 'A' is received the result would be
result=65
If the channel does not support text reception or if the no character arrives in <timeout> milliseconds then the result is
result=0 (timeout)
On error or failure the result is
result=-1
Note Most channels do not support the reception of text.
Purpose Record sound to a file until an acceptable DTMF digit is received or a specified amount of time has passed. Optionally the file BEEP is played before recording begins.
Returns The documentation in the code says on hangup the result is -1, however when I tried it the hangup result was
result=0 (hangup)
If an error occurs then the result is -1. This can happen, for example, if you ask for a non-existent format.
If the user presses an acceptable escape digit then the result is a number representing the ASCII digit pressed. For example if recording terminated because the user pressed the '2' key the result is
result=50 (dtmf)
Example RECORD FILE foo gsm 123 5000 beep Record sound in gsm format to file 'foo.gsm'. Play a beep before starting to record. Stop recording if user presses '1', '2' or '3', after five seconds of recording, or if the user hangs up.
Notes Don't put an extension on the filename; the filename extension will be created based on the <format> specified.
The file will be created in /var/lib/asterisk/sounds
<format> specifies what kind of file will be recorded. GSM is a commonly used format. To find out what other formats are supported start Asterisk with at a verbosity level of at least 2 (-vvc) and look for the messages that appear saying "== Registered file format <whatever>'. Most but not all registered formats can be used, for example, Asterisk can read but not write files in 'mp3' format.
If you don't want ANY digits to terminate recording then specify "" instead of a digit string. To change the above example so no digits terminate recording use RECORD FILE foo gsm "" 5000 beep
<timeout> is the maximum record time in milliseconds, or -1 for no timeout. When this document was written [Nov 2002] I was unable to get <timeout> to work; this command always kept recording until I pressed an escape digit or hung up, as if -1 had been specified for timeout. A patch to correct this has been submitted but has not yet shown up in the CVS tree.
Purpose Say the given digit string, returning early if any of the given DTMF escape digits are received on the channel. If no DTMF digits are to be received specify "" for <escape digits>.
Returns Zero if playback completes without a digit being received, or the ASCII numerical representation of the digit pressed, or -1 on error or hangup.
Example SAY DIGITS 123 78#
The digits 'one', 'two', 'three' are spoken. If the user presses the '7', '8' or '#' key the speaking stops and the command ends. If the user pressed no keys the result would be 200 result=0. If the user pressed the '#' key then the result would be 200 result=35.
Purpose Say the given number, returning early if any of the given DTMF escape digits are received on the channel. If no DTMF digits are to be accepted specify "" for <escape digits>.
Returns Zero if playback completes without a digit being received, or the ASCII numerical representation of the digit pressed, or -1 on error or hangup.
Example SAY NUMBER 123 789
The phrase 'one hundred twenty three' is spoken. If the user presses the '7', '8' or '9' key the speaking stops and the command ends. If the user pressed no keys the result would be 200 result=0. If the user pressed the '#' key then the result would be 200 result=35.
Purpose Send the specified image on a channel. The image name should not should not include the extension.
Returns Zero if the image is sent or if the channel does not support image transmission. Returns -1 only on error or hangup.
Notes Asterisk looks for the image in /var/lib/asterisk/images
Most channels do not support the transmission of images.
Purpose Send the given text to the connected channel.
Returns 0 if text is sent or if the channel does not support text transmission. Returns -1 only on error or hangup.
Example SEND TEXT "Hello world"
Note Most channels do not support transmission of text.
Purpose Changes the caller ID of the current channel
Returns Always returns 200 result=1
Example SET CALLERID "John Smith"<1234567>
Notes This command will let you take liberties with the <caller ID specification> but the format shown in the example above works well: the name enclosed in double quotes followed immediately by the number inside angle brackets. If there is no name then you can omit it.
If the name contains no spaces you can omit the double quotes around it.
The number must follow the name immediately; don't put a space between them.
The angle brackets around the number are necessary; if you omit them the number will be considered to be part of the name.
Purpose Sets the context for continuation upon exiting the application.
Returns Always returns 200 result=0.
Example SET CONTEXT demo
Notes Setting the context does NOT automatically reset the extension and the priority; if you want to start at the top of the new context you should set extension and priority yourself.
If you specify a non-existent context you receive no error indication (the result returned is still 'result=0') but you do get a warning message on the Asterisk console.
Purpose Set the extension to be used for continuation upon exiting the application.
Returns Always returns 200 result=0.
Example SET EXTENSION 23
Note Setting the extension does NOT automatically reset the priority. If you want to start with the first priority of the extension you should set the priority yourself.
If you specify a non-existent extension you receive no error indication (the result returned is still 'result=0') but you do get a warning message on the Asterisk console.
Purpose Set the priority to be used for continuation upon exiting the application.
Returns Always returns 200 result=0.
Example SET PRIORITY 5
Note If you specify a non-existent priority you receive no error indication of any sort: the result returned is still 'result=0' and no warning is issued on the Asterisk console.
Purpose Sets a variable to the specified value. The variables so created can later be used by later using ${<variablename>} in the dialplan.
Returns Always returns 200 result=1.
Example SET VARIABLE station zap/3
Creates a variable named 'station' with the value 'zap/3'.
Notes Unlike most of Asterisk, variable names are case sensitive. The names 'Foo' and 'foo' refer to two separate and distinct variables.
If the value being assigned contains spaces then put it inside double quotes.
If you want double quotes inside the value then you have to escape them. For example to create a variable CID whose value is "John Doe"<555-1212> you could use: SET VARIABLE CID "\"John Doe \"<555-1212>
Be aware that the language you are using may eat the backslash before it gets passed to Asterisk; you may have to use two backslashes or otherwise tell the language that, yes, you really do want a backslash in the string you are sending.
These variables live in the channel Asterisk creates when you pickup a phone and as such they are both local and temporary. Variables created in one channel can not be accessed by another channel. When you hang up the phone, the channel is deleted and any variables in that channel are deleted as well.
Purpose Play the given audio file, allowing playback to be interrupted by a DTMF digit. This command is similar to the GET DATA command but this command returns after the first DTMF digit has been pressed while GET DATA can accumulated any number of digits before returning.
Returns If playback finished with no acceptable digit being pressed the result is zero. If an acceptable digit was pressed the result is the decimal representation of the pressed digit. If the channel was disconnected or an error occurred the result is -1.
Example STREAM FILE welcome #
Plays the file 'welcome'. If the user presses the '#' key the playing stops and the command returns 200 result=35
Note Don't give an extension with the filename.
Asterisk looks for the file to play in /var/lib/asterisk/sounds
Use double quotes if the message should play completely. For example to play audio file 'welcome' without allowing interruption by digits use: STREAM FILE welcome ""
Purpose Enable or disable TDD transmission/reception on the current channel.
Returns 1 if successful or 0 if the channel is not TDD capable.
Example TDD MODE on
Note The argument <setting> can be 'on' or 'tdd' to enable tdd mode. It can also be 'mate' which apparently sets some unspecified tdd mode. If it is anything else ('off' for example) then tdd mode is disabled.
Purpose Sends <message> to the Asterisk console via the 'verbose' message system.
Returns Always returns 1
Example VERBOSE Hello 3
Sends the message "Hello" to the console if the current Asterisk verbosity level is set to 3 or greater.
Notes <level> is the verbosity level in the range 1 through 4.
If your message contains spaces, then enclose it in double quotes.
The Asterisk verbosity system works as follows. The Asterisk user gets to set the desired verbosity at startup time or later using the console 'set verbose' command. Messages are displayed on the console if their verbose level is less than or equal to desired verbosity set by the user. More important messages should have a low verbose level; less important messages should have a high verbose level.
Purpose Waits up to 'timeout' milliseconds for channel to receive a DTMF digit
Returns -1 on channel failure, 0 if no digit is received in timeout or the numerical value of the ascii of the digit received.
Note Use -1 for the timeout value if you want the call to wait indefinitely.
Example WAIT FOR DIGIT 3000
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