Das Asterisk Buch von Stefan Wintermeyer

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= 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)

Differenz zum internen Hilfetext in Asterisk 1.2: 

- keine -

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= 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.

Differenz zum internen Hilfetext in Asterisk 1.2: 

- keine - 

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= Info about application 'AgentCallbackLogin' =- 

[Synopsis]
Call agent callback login

[Description]
  AgentCallbackLogin([AgentNo][|[options][|[exten]@context]]):
Asks the agent to login to the system with callback.
The agent's callback extension is called (optionally with the specified
context).
The option string may contain zero or more of the following characters:
      's' -- silent login - do not announce the login ok segment agent logged in/off

Differenz zum internen Hilfetext in Asterisk 1.2: 

- keine - 

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= Info about application 'AgentLogin' =- 

[Synopsis]
Call agent login

[Description]
  AgentLogin([AgentNo][|options]):
Asks the agent to login to the system.  Always returns -1.  While
logged in, the agent can receive calls and will hear a 'beep'
when a new call comes in. The agent can dump the call by pressing
the star key.
The option string may contain zero or more of the following characters:
      's' -- silent login - do not announce the login ok segment after agent logged in/off

Differenz zum internen Hilfetext in Asterisk 1.2: 

- keine - 

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= Info about application 'AgentMonitorOutgoing' =- 

[Synopsis]
Record agent's outgoing call

[Description]
  AgentMonitorOutgoing([options]):
Tries to figure out the id of the agent who is placing outgoing call based on
comparison of the callerid of the current interface and the global variable 
placed by the AgentCallbackLogin application. That's why it should be used only
with the AgentCallbackLogin app. Uses the monitoring functions in chan_agent 
instead of Monitor application. That have to be configured in the agents.conf file.

Return value:
Normally the app returns 0 unless the options are passed. Also if the callerid or
the agentid are not specified it'll look for n+101 priority.

Options:
 'd' - make the app return -1 if there is an error condition and there is
       no extension n+101
 'c' - change the CDR so that the source of the call is 'Agent/agent_id'
 'n' - don't generate the warnings when there is no callerid or the
       agentid is not known.
             It's handy if you want to have one context for agent and non-agent calls.

Differenz zum internen Hilfetext in Asterisk 1.2: 

- keine - 

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= 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.
  This channel will stop dialplan execution on hangup inside of this
application, except when using DeadAGI.  Otherwise, dialplan execution
will continue normally.
  A locally executed AGI script will receive SIGHUP on hangup from the channel
except when using DeadAGI. This can be disabled by setting the AGISIGHUP channel
variable to "no" before executing the AGI application.
  Using 'EAGI' provides enhanced AGI, with incoming audio available out of band
on file descriptor 3

  Use the CLI command 'agi show' to list available agi commands
  This application sets the following channel variable upon completion:
     AGISTATUS      The status of the attempt to the run the AGI script
                    text string, one of SUCCESS | FAILED | HANGUP

Differenz zum internen Hilfetext in Asterisk 1.2: 

13,19c13,15
<   This channel will stop dialplan execution on hangup inside of this
< application, except when using DeadAGI.  Otherwise, dialplan execution
< will continue normally.
<   A locally executed AGI script will receive SIGHUP on hangup from the channel
< except when using DeadAGI. This can be disabled by setting the AGISIGHUP channel
< variable to "no" before executing the AGI application.
<   Using 'EAGI' provides enhanced AGI, with incoming audio available out of band
---
> 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
22,25c18
<   Use the CLI command 'agi show' to list available agi commands
<   This application sets the following channel variable upon completion:
<      AGISTATUS      The status of the attempt to the run the AGI script
<                     text string, one of SUCCESS | FAILED | HANGUP
---
> Use the CLI command 'show agi' to list available agi commands

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= Info about application 'AlarmReceiver' =- 

[Synopsis]
Provide support for receiving 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.

Differenz zum internen Hilfetext in Asterisk 1.2: 

5c5
< Provide support for receiving alarm reports from a burglar or fire alarm panel
---
> Provide support for receving alarm reports from a burglar or fire alarm panel

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= Info about application 'AMD' =- 

[Synopsis]
Attempts to detect answering machines

[Description]
  AMD([initialSilence][|greeting][|afterGreetingSilence][|totalAnalysisTime]
      [|minimumWordLength][|betweenWordsSilence][|maximumNumberOfWords]
      [|silenceThreshold])
  This application attempts to detect answering machines at the beginning
  of outbound calls.  Simply call this application after the call
  has been answered (outbound only, of course).
  When loaded, AMD reads amd.conf and uses the parameters specified as
  default values. Those default values get overwritten when calling AMD
  with parameters.
- 'initialSilence' is the maximum silence duration before the greeting. If
   exceeded then MACHINE.
- 'greeting' is the maximum length of a greeting. If exceeded then MACHINE.
- 'afterGreetingSilence' is the silence after detecting a greeting.
   If exceeded then HUMAN.
- 'totalAnalysisTime' is the maximum time allowed for the algorithm to decide
   on a HUMAN or MACHINE.
- 'minimumWordLength'is the minimum duration of Voice to considered as a word.
- 'betweenWordsSilence' is the minimum duration of silence after a word to 
   consider the audio that follows as a new word.
- 'maximumNumberOfWords'is the maximum number of words in the greeting. 
   If exceeded then MACHINE.
- 'silenceThreshold' is the silence threshold.
This application sets the following channel variable upon completion:
    AMDSTATUS - This is the status of the answering machine detection.
                Possible values are:
                MACHINE | HUMAN | NOTSURE | HANGUP
    AMDCAUSE - Indicates the cause that led to the conclusion.
               Possible values are:
               TOOLONG-<%d total_time>
               INITIALSILENCE-<%d silenceDuration>-<%d initialSilence>
               HUMAN-<%d silenceDuration>-<%d afterGreetingSilence>
               MAXWORDS-<%d wordsCount>-<%d maximumNumberOfWords>
               LONGGREETING-<%d voiceDuration>-<%d greeting>

Differenz zum internen Hilfetext in Asterisk 1.2: 

-- in Version 1.2 nicht vorhanden --

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= 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.

Differenz zum internen Hilfetext in Asterisk 1.2: 

- keine - 

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= 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().

Differenz zum internen Hilfetext in Asterisk 1.2: 

- keine - 

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= Info about application 'Authenticate' =- 

[Synopsis]
Authenticate a user

[Description]
  Authenticate(password[|options[|maxdigits]]): This application asks the caller
to enter a given password in order to continue dialplan execution. If the password
begins with the '/' character, it is interpreted as a file which contains a list of
valid passwords, listed 1 password per line in the file.
  When using a database key, the value associated with the key can be anything.
Users have three attempts to authenticate before the channel is hung up. If the
passsword is invalid, the 'j' option is specified, and priority n+101 exists,
dialplan execution will continnue at this location.
  Options:
     a - Set the channels' account code to the password that is entered
     d - Interpret the given path as database key, not a literal file
     j - Support jumping to n+101 if authentication fails
     m - Interpret the given path as a file which contains a list of account
         codes and password hashes delimited with ':', listed one per line in
         the file. When one of the passwords is matched, the channel will have
         its account code set to the corresponding account code in the file.
     r - Remove the database key upon successful entry (valid with 'd' only)
     maxdigits  - maximum acceptable number of digits. Stops reading after
         maxdigits have been entered (without requiring the user to
         press the '#' key).
         Defaults to 0 - no limit - wait for the user press the '#' key.

Differenz zum internen Hilfetext in Asterisk 1.2: 

8,10c8,10
<   Authenticate(password[|options[|maxdigits]]): This application asks the caller
< to enter a given password in order to continue dialplan execution. If the password
< begins with the '/' character, it is interpreted as a file which contains a list of
---
>   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
25,28d24
<      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.

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= Info about application 'BackGround' =- 

[Synopsis]
Play an audio file while waiting for digits of an extension to go to.

[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 explicitly specifies which language
to attempt to use for the requested sound files. If a 'context' is specified,
this is the dialplan context that this application will use when exiting to a
dialed extension.  If one of the requested sound files does not exist, call processing will be
terminated.
  Options:
    s - Causes the playback of the message to be skipped
          if the channel is not in the 'up' state (i.e. it
          hasn't been answered yet). If this happens, the
          application will return immediately.
    n - Don't answer the channel before playing the files.
    m - Only break if a digit hit matches a one digit
          extension in the destination context.

Differenz zum internen Hilfetext in Asterisk 1.2: 

5c5
< Play an audio file while waiting for digits of an extension to go to.
---
> Play a file while awaiting extension
12c12
< should be used. The 'langoverride' option explicitly specifies which language
---
> should be used. The 'langoverride' option explicity specifies which language
18c18
<     s - Causes the playback of the message to be skipped
---
>     s - causes the playback of the message to be skipped
20c20
<           hasn't been answered yet). If this happens, the
---
>           hasn't been answered yet.) If this happens, the
22,24c22,24
<     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.
---
>     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

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= 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.

Differenz zum internen Hilfetext in Asterisk 1.2: 

- keine - 

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= 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.

Differenz zum internen Hilfetext in Asterisk 1.2: 

- keine - 

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= 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.

Differenz zum internen Hilfetext in Asterisk 1.2: 

- keine - 

Dass ein Kanal verfügbar ist, bedeutet aber nicht automatisch, dass er auch frei ist oder dass das Endgerät einen Anruf akzeptieren würde. Das erfährt man durch ein Dial() zum Kanal.

Diese Anwendung arbeitet auf MGCP-Kanälen nicht ordnungsgemäß.

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= 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

Differenz zum internen Hilfetext in Asterisk 1.2: 

- keine - 

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= Info about application 'ChannelRedirect' =- 

[Synopsis]
Redirects given channel to a dialplan target.

[Description]
ChannelRedirect(channel|[[context|]extension|]priority):
  Sends the specified channel to the specified extension priority

Differenz zum internen Hilfetext in Asterisk 1.2: 

-- in Version 1.2 nicht vorhanden --

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= Info about application 'ChanSpy' =- 

[Synopsis]
Listen to a channel, and optionally whisper into it

[Description]
  ChanSpy([chanprefix][|options]): This application is used to listen to the
audio from an Asterisk channel. This includes the audio coming in and
out of the channel being spied on. If the 'chanprefix' parameter is specified,
only channels beginning with this string will be spied upon.
  While spying, the following actions may be performed:
    - Dialing # cycles the volume level.
    - Dialing * will stop spying and look for another channel to spy on.
    - Dialing a series of digits followed by # builds a channel name to append
      to 'chanprefix'. For example, executing ChanSpy(Agent) and then dialing
      the digits '1234#' while spying will begin spying on the channel
      'Agent/1234'.
  Options:
    b             - Only spy on channels involved in a bridged call.
    g(grp)        - Match only channels where their ${SPYGROUP} variable is set to
                    contain 'grp' in an optional : delimited list.
    q             - Don't play a beep when beginning to spy on a channel, or speak the
                    selected channel name.
    r[(basename)] - Record the session to the monitor spool directory. An
                    optional base for the filename may be specified. The
                    default is 'chanspy'.
    v([value])    - Adjust the initial volume in the range from -4 to 4. A
                    negative value refers to a quieter setting.
    w             - Enable 'whisper' mode, so the spying channel can talk to
                    the spied-on channel.
    W             - Enable 'private whisper' mode, so the spying channel can
                    talk to the spied-on channel but cannot listen to that
                    channel.

Differenz zum internen Hilfetext in Asterisk 1.2: 

5c5,6
< Listen to a channel, and optionally whisper into it
---
> Listen to the audio of an active channel
> 
9c10
< audio from an Asterisk channel. This includes the audio coming in and
---
> audio from an active Asterisk channel. This includes the audio coming in and
12c13
<   While spying, the following actions may be performed:
---
>   While Spying, the following actions may be performed:
17c18
<       the digits '1234#' while spying will begin spying on the channel
---
>       the digits '1234#' while spying will begin spying on the channel,
20,24c21,24
<     b             - Only spy on channels involved in a bridged call.
<     g(grp)        - Match only channels where their ${SPYGROUP} variable is set to
<                     contain 'grp' in an optional : delimited list.
<     q             - Don't play a beep when beginning to spy on a channel, or speak the
<                     selected channel name.
---
>     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.
28,34c28,29
<     v([value])    - Adjust the initial volume in the range from -4 to 4. A
<                     negative value refers to a quieter setting.
<     w             - Enable 'whisper' mode, so the spying channel can talk to
<                     the spied-on channel.
<     W             - Enable 'private whisper' mode, so the spying channel can
<                     talk to the spied-on channel but cannot listen to that
<                     channel.
---
>     v([value]) - Adjust the initial volume in the range from -4 to 4. A
>                  negative value refers to a quieter setting.

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= Info about application 'Congestion' =- 

[Synopsis]
Indicate the Congestion condition

[Description]
  Congestion([timeout]): This application will indicate the congestion
condition to the calling channel. If the optional timeout is specified, the
calling channel will be hung up after the specified number of seconds.
Otherwise, this application will wait until the calling channel hangs up.

Differenz zum internen Hilfetext in Asterisk 1.2: 

8c8
<   Congestion([timeout]): This application will indicate the congestion
---
>   Congestion([timeout]): This application will indicate the congenstion

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= Info about application 'ContinueWhile' =- 

[Synopsis]
Restart a While loop

[Description]
Usage:  ContinueWhile()
Returns to the top of the while loop and re-evaluates the conditional.

Differenz zum internen Hilfetext in Asterisk 1.2: 

-- nicht in Version 1.2. vorhanden --

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= 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

Differenz zum internen Hilfetext in Asterisk 1.2: 

- keine - 

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= Info about application 'DBdel' =- 

[Synopsis]
Delete a key from the database

[Description]
  DBdel(family/key): This application will delete a key from the Asterisk
database.
  This application has been DEPRECATED in favor of the DB_DELETE function.

Differenz zum internen Hilfetext in Asterisk 1.2: 

10d9
<   This application has been DEPRECATED in favor of the DB_DELETE function.

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= 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

Differenz zum internen Hilfetext in Asterisk 1.2: 

- keine - 

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= 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.
  This channel will stop dialplan execution on hangup inside of this
application, except when using DeadAGI.  Otherwise, dialplan execution
will continue normally.
  A locally executed AGI script will receive SIGHUP on hangup from the channel
except when using DeadAGI. This can be disabled by setting the AGISIGHUP channel
variable to "no" before executing the AGI application.
  Using 'EAGI' provides enhanced AGI, with incoming audio available out of band
on file descriptor 3

  Use the CLI command 'agi show' to list available agi commands
  This application sets the following channel variable upon completion:
     AGISTATUS      The status of the attempt to the run the AGI script
                    text string, one of SUCCESS | FAILED | HANGUP

Differenz zum internen Hilfetext in Asterisk 1.2: 

13,19c13,15
<   This channel will stop dialplan execution on hangup inside of this
< application, except when using DeadAGI.  Otherwise, dialplan execution
< will continue normally.
<   A locally executed AGI script will receive SIGHUP on hangup from the channel
< except when using DeadAGI. This can be disabled by setting the AGISIGHUP channel
< variable to "no" before executing the AGI application.
<   Using 'EAGI' provides enhanced AGI, with incoming audio available out of band
---
> 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
22,25c18
<   Use the CLI command 'agi show' to list available agi commands
<   This application sets the following channel variable upon completion:
<      AGISTATUS      The status of the attempt to the run the AGI script
<                     text string, one of SUCCESS | FAILED | HANGUP
---
> Use the CLI command 'show agi' to list available agi commands

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= Info about application 'Dial' =- 

[Synopsis]
Place a call and connect to the current channel

[Description]
  Dial(Technology/resource[&Tech2/resource2...][|timeout][|options][|URL]):
This application will place calls to one or more specified channels. As soon
as one of the requested channels answers, the originating channel will be
answered, if it has not already been answered. These two channels will then
be active in a bridged call. All other channels that were requested will then
be hung up.
  Unless there is a timeout specified, the Dial application will wait
indefinitely until one of the called channels answers, the user hangs up, or
if all of the called channels are busy or unavailable. Dialplan executing will
continue if no requested channels can be called, or if the timeout expires.

  This application sets the following channel variables upon completion:
    DIALEDTIME   - This is the time from dialing a channel until when it
                   is disconnected.
    ANSWEREDTIME - This is the amount of time for actual call.
    DIALSTATUS   - This is the status of the call:
                   CHANUNAVAIL | CONGESTION | NOANSWER | BUSY | ANSWER | CANCEL
                   DONTCALL | TORTURE
  For the Privacy and Screening Modes, the DIALSTATUS variable will be set to
DONTCALL if the called party chooses to send the calling party to the 'Go Away'
script. The DIALSTATUS variable will be set to TORTURE if the called party
wants to send the caller to the 'torture' script.
  This application will report normal termination if the originating channel
hangs up, or if the call is bridged and either of the parties in the bridge
ends the call.
  The optional URL will be sent to the called party if the channel supports it.
  If the OUTBOUND_GROUP variable is set, all peer channels created by this
application will be put into that group (as in Set(GROUP()=...).

  Options:
    A(x) - Play an announcement to the called party, using 'x' as the file.
    C    - Reset the CDR for this call.
    d    - Allow the calling user to dial a 1 digit extension while waiting for
           a call to be answered. Exit to that extension if it exists in the
           current context, or the context defined in the EXITCONTEXT variable,
           if it exists.
    D([called][:calling]) - Send the specified DTMF strings *after* the called
           party has answered, but before the call gets bridged. The 'called'
           DTMF string is sent to the called party, and the 'calling' DTMF
           string is sent to the calling party. Both parameters can be used
           alone.
    f    - Force the callerid of the *calling* channel to be set as the
           extension associated with the channel using a dialplan 'hint'.
           For example, some PSTNs do not allow CallerID to be set to anything
           other than the number assigned to the caller.
    g    - Proceed with dialplan execution at the current extension if the
           destination channel hangs up.
    G(context^exten^pri) - If the call is answered, transfer the calling party to
           the specified priority and the called party to the specified priority+1.
           Optionally, an extension, or extension and context may be specified. 
           Otherwise, the current extension is used. You cannot use any additional
           action post answer options in conjunction with this option.
    h    - Allow the called party to hang up by sending the '*' DTMF digit.
    H    - Allow the calling party to hang up by hitting the '*' DTMF digit.
    i    - Asterisk will ignore any forwarding requests it may receive on this
           dial attempt.
    j    - Jump to priority n+101 if all of the requested channels were busy.
    L(x[:y][:z]) - Limit the call to 'x' ms. Play a warning when 'y' ms are
           left. Repeat the warning every 'z' ms. The following special
           variables can be used with this option:
           * LIMIT_PLAYAUDIO_CALLER   yes|no (default yes)
                                      Play sounds to the caller.
           * LIMIT_PLAYAUDIO_CALLEE   yes|no
                                      Play sounds to the callee.
           * LIMIT_TIMEOUT_FILE       File to play when time is up.
           * LIMIT_CONNECT_FILE       File to play when call begins.
           * LIMIT_WARNING_FILE       File to play as warning if 'y' is defined.
                                      The default is to say the time remaining.
    m([class]) - Provide hold music to the calling party until a requested
           channel answers. A specific MusicOnHold class can be
           specified.
    M(x[^arg]) - Execute the Macro for the *called* channel before connecting
           to the calling channel. Arguments can be specified to the Macro
           using '^' as a delimeter. The Macro can set the variable
           MACRO_RESULT to specify the following actions after the Macro is
           finished executing.
           * ABORT        Hangup both legs of the call.
           * CONGESTION   Behave as if line congestion was encountered.
           * BUSY         Behave as if a busy signal was encountered. This will also
                          have the application jump to priority n+101 if the
                          'j' option is set.
           * CONTINUE     Hangup the called party and allow the calling party
                          to continue dialplan execution at the next priority.
           * GOTO:<context>^<xexten>^<priority> - Transfer the call to the
                          specified priority. Optionally, an extension, or
                          extension and priority can be specified.
           You cannot use any additional action post answer options in conjunction
           with this option. Also, pbx services are not run on the peer (called) channel,
           so you will not be able to set timeouts via the TIMEOUT() function in this macro.
    n    - This option is a modifier for the screen/privacy mode. It specifies
           that no introductions are to be saved in the priv-callerintros
           directory.
    N    - This option is a modifier for the screen/privacy mode. It specifies
           that if callerID is present, do not screen the call.
    o    - Specify that the CallerID that was present on the *calling* channel
           be set as the CallerID on the *called* channel. This was the
           behavior of Asterisk 1.0 and earlier.
    O([x]) - "Operator Services" mode (Zaptel channel to Zaptel channel
             only, if specified on non-Zaptel interface, it will be ignored).
             When the destination answers (presumably an operator services
             station), the originator no longer has control of their line.
             They may hang up, but the switch will not release their line
             until the destination party hangs up (the operator). Specified
             without an arg, or with 1 as an arg, the originator hanging up
             will cause the phone to ring back immediately. With a 2 specified,
             when the "operator" flashes the trunk, it will ring their phone
             back.
    p    - This option enables screening mode. This is basically Privacy mode
           without memory.
    P([x]) - Enable privacy mode. Use 'x' as the family/key in the database if
           it is provided. The current extension is used if a database
           family/key is not specified.
    r    - Indicate ringing to the calling party. Pass no audio to the calling
           party until the called channel has answered.
    S(x) - Hang up the call after 'x' seconds *after* the called party has
           answered the call.
    t    - Allow the called party to transfer the calling party by sending the
           DTMF sequence defined in features.conf.
    T    - Allow the calling party to transfer the called party by sending the
           DTMF sequence defined in features.conf.
    w    - Allow the called party to enable recording of the call by sending
           the DTMF sequence defined for one-touch recording in features.conf.
    W    - Allow the calling party to enable recording of the call by sending
           the DTMF sequence defined for one-touch recording in features.conf.
    k    - Allow the called party to enable parking of the call by sending
           the DTMF sequence defined for call parking in features.conf.
    K    - Allow the calling party to enable parking of the call by sending
           the DTMF sequence defined for call parking in features.conf.

Differenz zum internen Hilfetext in Asterisk 1.2: 

62,63d61
<     i    - Asterisk will ignore any forwarding requests it may receive on this
<            dial attempt.
95,96c93
<            with this option. Also, pbx services are not run on the peer (called) channel,
<            so you will not be able to set timeouts via the TIMEOUT() function in this macro.
---
>            with this option.
105,114d101
<     O([x]) - "Operator Services" mode (Zaptel channel to Zaptel channel
<              only, if specified on non-Zaptel interface, it will be ignored).
<              When the destination answers (presumably an operator services
<              station), the originator no longer has control of their line.
<              They may hang up, but the switch will not release their line
<              until the destination party hangs up (the operator). Specified
<              without an arg, or with 1 as an arg, the originator hanging up
<              will cause the phone to ring back immediately. With a 2 specified,
<              when the "operator" flashes the trunk, it will ring their phone
<              back.
132,135d118
<     k    - Allow the called party to enable parking of the call by sending
<            the DTMF sequence defined for call parking in features.conf.
<     K    - Allow the calling party to enable parking of the call by sending
<            the DTMF sequence defined for call parking in features.conf.

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= Info about application 'Dictate' =- 

[Synopsis]
Virtual Dictation Machine

[Description]
  Dictate([<base_dir>[|<filename>]])
Start dictation machine using optional base dir for files.

Differenz zum internen Hilfetext in Asterisk 1.2: 

8c8
<   Dictate([<base_dir>[|<filename>]])
---
>   Dictate([<base_dir>])

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= 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 application will immediately exit if one of the following DTMF digits are
received and the extension to jump to exists:
    0 - Jump to the 'o' extension, if it exists.
    * - Jump to the 'a' extension, if it exists.

  Parameters:
    vm-context   - This is the context within voicemail.conf to use for the
                   Directory.
    dial-context - This is the dialplan context to use when looking for an
                   extension that the user has selected, or when jumping to the
                   'o' or 'a' extension.

  Options:
    e - In addition to the name, also read the extension number to the
        caller before presenting dialing options.
    f - Allow the caller to enter the first name of a user in the directory
        instead of using the last name.

Differenz zum internen Hilfetext in Asterisk 1.2: 

25,26d24
<     e - In addition to the name, also read the extension number to the
<         caller before presenting dialing options.

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= 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 allows specification
of the context on which the user will be dialing. If no context is
specified, the DISA application defaults the context to "disa".
Presumably a normal system will have a special context set up
for DISA use with some or a lot of restrictions. 

The file that contains the passcodes (if used) allows specification
of either just a passcode (defaulting to the "disa" context, or
passcode|context on each line of the file. The file may contain blank
lines, or comments starting with "#" or ";". In addition, the
above arguments may have |new-callerid-string appended to them, to
specify a new (different) callerid to be used for this call, for
example: numeric-passcode|context|"My Phone" <(234) 123-4567> or 
full-pathname-of-passcode-file|"My Phone" <(234) 123-4567>.  Last
but not least, |mailbox[@context] may be appended, which will cause
a stutter-dialtone (indication "dialrecall") to be used, if the
specified mailbox contains any new messages, for example:
numeric-passcode|context||1234 (w/a changing callerid).  Note that
in the case of specifying the numeric-passcode, the context must be
specified if the callerid is specified also.

If login is successful, the application looks up the dialed number in
the specified (or default) context, and executes it if found.
If the user enters an invalid extension and extension "i" (invalid) 
exists in the context, it will be used. Also, if you set the 5th argument
to 'NOANSWER', the DISA application will not answer initially.

Differenz zum internen Hilfetext in Asterisk 1.2: 

8c8
< DISA(<numeric passcode>[|<context>]) or DISA(<filename>)
---
> DISA(<numeric passcode>[|<context>]) or disa(<filename>)
28c28
< individual passcodes contained in a file. It also allows specification
---
> individual passcodes contained in a file. It also allow specification
52,53c52
< exists in the context, it will be used. Also, if you set the 5th argument
< to 'NOANSWER', the DISA application will not answer initially.
---
> exists in the context, it will be used.

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= 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.

Differenz zum internen Hilfetext in Asterisk 1.2: 

- keine - 

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= 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.
  This channel will stop dialplan execution on hangup inside of this
application, except when using DeadAGI.  Otherwise, dialplan execution
will continue normally.
  A locally executed AGI script will receive SIGHUP on hangup from the channel
except when using DeadAGI. This can be disabled by setting the AGISIGHUP channel
variable to "no" before executing the AGI application.
  Using 'EAGI' provides enhanced AGI, with incoming audio available out of band
on file descriptor 3

  Use the CLI command 'agi show' to list available agi commands
  This application sets the following channel variable upon completion:
     AGISTATUS      The status of the attempt to the run the AGI script
                    text string, one of SUCCESS | FAILED | HANGUP

Differenz zum internen Hilfetext in Asterisk 1.2: 

13,19c13,15
<   This channel will stop dialplan execution on hangup inside of this
< application, except when using DeadAGI.  Otherwise, dialplan execution
< will continue normally.
<   A locally executed AGI script will receive SIGHUP on hangup from the channel
< except when using DeadAGI. This can be disabled by setting the AGISIGHUP channel
< variable to "no" before executing the AGI application.
<   Using 'EAGI' provides enhanced AGI, with incoming audio available out of band
---
> 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
22,25c18
<   Use the CLI command 'agi show' to list available agi commands
<   This application sets the following channel variable upon completion:
<      AGISTATUS      The status of the attempt to the run the AGI script
<                     text string, one of SUCCESS | FAILED | HANGUP
---
> Use the CLI command 'show agi' to list available agi commands

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= Info about application 'Echo' =- 

[Synopsis]
Echo audio, video, or DTMF back to the calling party

[Description]
  Echo(): This application will echo any audio, video, or DTMF frames read from
the calling channel back to itself. If the DTMF digit '#' is received, the
application will exit.

Differenz zum internen Hilfetext in Asterisk 1.2: 

5c5
< Echo audio, video, or DTMF back to the calling party
---
> Echo audio read back to the user
8,10c8,10
<   Echo(): This application will echo any audio, video, or DTMF frames read from
< the calling channel back to itself. If the DTMF digit '#' is received, the
< application will exit.
---
>   Echo():  Echo audio read from channel back to the channel. 
> User can exit the application by either pressing the '#' key, 
> or hanging up.

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= Info about application 'EndWhile' =- 

[Synopsis]
End a while loop

[Description]
Usage:  EndWhile()
Return to the previous called While

Differenz zum internen Hilfetext in Asterisk 1.2: 

5c5
< End a while loop
---
> End A While Loop
10a11
> 

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= Info about application 'Exec' =- 

[Synopsis]
Executes dialplan application

[Description]
Usage: Exec(appname(arguments))
  Allows an arbitrary application to be invoked even when not
hardcoded into the dialplan.  If the underlying application
terminates the dialplan, or if the application cannot be found,
Exec will terminate the dialplan.
  To invoke external applications, see the application System.
  If you would like to catch any error instead, see TryExec.

Differenz zum internen Hilfetext in Asterisk 1.2: 

5c5
< Executes dialplan application
---
> Executes internal application
10,14c10,12
< hardcoded into the dialplan.  If the underlying application
< terminates the dialplan, or if the application cannot be found,
< Exec will terminate the dialplan.
<   To invoke external applications, see the application System.
<   If you would like to catch any error instead, see TryExec.
---
> 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.

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= Info about application 'ExecIf' =- 

[Synopsis]
Executes dialplan application, conditionally

[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.

Differenz zum internen Hilfetext in Asterisk 1.2: 

5c5
< Executes dialplan application, conditionally
---
> Conditional exec
12d11
< 

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= 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.

Differenz zum internen Hilfetext in Asterisk 1.2: 

10c10,12
< arguments, if the current time matches the given time specification.
---
> 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.

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= Info about application 'ExitWhile' =- 

[Synopsis]
End a While loop

[Description]
Usage:  ExitWhile()
Exits a While loop, whether or not the conditional has been satisfied.

Differenz zum internen Hilfetext in Asterisk 1.2: 

-- in Version 1.2 nicht vorhanden --

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= Info about application 'ExtenSpy' =- 

[Synopsis]
Listen to a channel, and optionally whisper into it

[Description]
  ExtenSpy(exten[@context][|options]): This application is used to listen to the
audio from an Asterisk channel. This includes the audio coming in and
out of the channel being spied on. Only channels created by outgoing calls for the
specified extension will be selected for spying. If the optional context is not
supplied, the current channel's context will be used.
  While spying, the following actions may be performed:
    - Dialing # cycles the volume level.
    - Dialing * will stop spying and look for another channel to spy on.
  Options:
    b             - Only spy on channels involved in a bridged call.
    g(grp)        - Match only channels where their ${SPYGROUP} variable is set to
                    contain 'grp' in an optional : delimited list.
    q             - Don't play a beep when beginning to spy on a channel, or speak the
                    selected channel name.
    r[(basename)] - Record the session to the monitor spool directory. An
                    optional base for the filename may be specified. The
                    default is 'chanspy'.
    v([value])    - Adjust the initial volume in the range from -4 to 4. A
                    negative value refers to a quieter setting.
    w             - Enable 'whisper' mode, so the spying channel can talk to
                    the spied-on channel.
    W             - Enable 'private whisper' mode, so the spying channel can
                    talk to the spied-on channel but cannot listen to that
                    channel.

Differenz zum internen Hilfetext in Asterisk 1.2: 

-- in Version 1.2 nicht vorhanden --

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= 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/externalivr.txt for a protocol specification.

Differenz zum internen Hilfetext in Asterisk 1.2: 

15c15
< See doc/externalivr.txt for a protocol specification.
---
> See doc/README.externalivr for a protocol specification.

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= 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)

Differenz zum internen Hilfetext in Asterisk 1.2: 

- keine - 

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= 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.

Differenz zum internen Hilfetext in Asterisk 1.2: 

- keine - 

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= Info about application 'FollowMe' =- 

[Synopsis]
Find-Me/Follow-Me application

[Description]
  FollowMe(followmeid|options):
This application performs Find-Me/Follow-Me functionality for the caller
as defined in the profile matching the <followmeid> parameter in
followme.conf. If the specified <followmeid> profile doesn't exist in
followme.conf, execution will be returned to the dialplan and call
execution will continue at the next priority.

  Options:
    s    - Playback the incoming status message prior to starting the follow-me step(s)
    a    - Record the caller's name so it can be announced to the callee on each step
    n    - Playback the unreachable status message if we've run out of steps to reach the
           or the callee has elected not to be reachable.
Returns -1 on hangup

Differenz zum internen Hilfetext in Asterisk 1.2: 

-- in Version 1.2 nicht vorhanden --

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= 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.

Differenz zum internen Hilfetext in Asterisk 1.2: 

- keine - 

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= 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.

Differenz zum internen Hilfetext in Asterisk 1.2: 

- keine - 

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= 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.

Differenz zum internen Hilfetext in Asterisk 1.2: 

- keine - 

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= Info about application 'GosubIf' =- 

[Synopsis]
Conditionally 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.

Differenz zum internen Hilfetext in Asterisk 1.2: 

5c5
< Conditionally jump to label, saving return address
---
> Jump to label, saving return address

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= 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.

Differenz zum internen Hilfetext in Asterisk 1.2: 

- keine - 

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= 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.

Differenz zum internen Hilfetext in Asterisk 1.2: 

- keine - 

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= 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 specified location
in the dialplan if the current time matches the given time specification.

Differenz zum internen Hilfetext in Asterisk 1.2: 

9,10c9,12
< This application will have the calling channel jump to the specified location
< in the dialplan if the current time matches the given time specification.
---
> 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.

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= Info about application 'Hangup' =- 

[Synopsis]
Hang up the calling channel

[Description]
  Hangup([causecode]): This application will hang up the calling channel.
If a causecode is given the channel's hangup cause will be set to the given
value.

Differenz zum internen Hilfetext in Asterisk 1.2: 

8,10c8
<   Hangup([causecode]): This application will hang up the calling channel.
< If a causecode is given the channel's hangup cause will be set to the given
< value.
---
>   Hangup(): This application will hang up the calling channel.

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= 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.

Differenz zum internen Hilfetext in Asterisk 1.2: 

- keine - 

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= 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.

Differenz zum internen Hilfetext in Asterisk 1.2: 

- keine - 

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= Info about application 'Log' =- 

[Synopsis]
Send arbitrary text to a selected log level

[Description]
Log(<level>|<message>)
  level must be one of ERROR, WARNING, NOTICE, DEBUG, VERBOSE, DTMF

Differenz zum internen Hilfetext in Asterisk 1.2: 

-- in Version 1.2 nicht vorhanden --

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= 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()

Differenz zum internen Hilfetext in Asterisk 1.2: 

- keine - 

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= 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.

Differenz zum internen Hilfetext in Asterisk 1.2: 

- keine - 

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= 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.

Differenz zum internen Hilfetext in Asterisk 1.2: 

22,23c22,23
<          may be possible that stack-intensive applications in deeply nested macros
<          could cause asterisk to crash earlier than this limit.
---
>          may be possible that stack-intensive applications in deeply nested
>          macros could cause asterisk to crash earlier than this limit.

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= Info about application 'MacroExclusive' =- 

[Synopsis]
Exclusive Macro Implementation

[Description]
  MacroExclusive(macroname|arg1|arg2...):
Executes macro defined in the context 'macro-macroname'
Only one call at a time may run the macro.
(we'll wait if another call is busy executing in the Macro)
Arguments and return values as in application Macro()

Differenz zum internen Hilfetext in Asterisk 1.2: 

-- in Version 1.2 nicht vorhanden --

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= 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.

Differenz zum internen Hilfetext in Asterisk 1.2: 

- keine - 

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= 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()

Differenz zum internen Hilfetext in Asterisk 1.2: 

- keine - 

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= 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.

Differenz zum internen Hilfetext in Asterisk 1.2: 

- keine - 

Mit d oder D werden dynamisch Konferenzräume angelegt, in der meetme.conf können sie statisch definiert werden.

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= 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 with review
      'I' -- announce user join/leave without review
      'l' -- set listen only mode (Listen only, no talking)
      'm' -- set initially muted
      'M' -- enable music on hold when the conference has a single caller
      'o' -- set talker optimization - treats talkers who aren't speaking as
             being muted, meaning (a) No encode is done on transmission and
             (b) Received audio that is not registered as talking is omitted
             causing no buildup in background noise
      'p' -- allow user to exit the conference by pressing '#'
      'P' -- always prompt for the pin even if it is specified
      'q' -- quiet mode (don't play enter/leave sounds)
      'r' -- Record conference (records as ${MEETME_RECORDINGFILE}
             using format ${MEETME_RECORDINGFORMAT}). Default filename is
             meetme-conf-rec-${CONFNO}-${UNIQUEID} and the default format is
             wav.
      's' -- Present menu (user or admin) when '*' is received ('send' to menu)
      't' -- set talk only mode. (Talk only, no listening)
      'T' -- set talker detection (sent to manager interface and meetme list)
      'w[(<secs>)]'
          -- wait until the marked user enters the conference
      'x' -- close the conference when last marked user exits
      'X' -- allow user to exit the conference by entering a valid single
             digit extension ${MEETME_EXIT_CONTEXT} or the current context
             if that variable is not defined.
      '1' -- do not play message when first person enters

Differenz zum internen Hilfetext in Asterisk 1.2: 

8,11c8,11
<   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 '#'.
---
>   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 '#'.
20,21c20,21
<              Default: conf-background.agi  (Note: This does not work with
<              non-Zap channels in the same conference)
---
>              Default: conf-background.agi
>              (Note: This does not work with non-Zap channels in the same conference)
27,30c27,28
<       'i' -- announce user join/leave with review
<       'I' -- announce user join/leave without review
<       'l' -- set listen only mode (Listen only, no talking)
<       'm' -- set initially muted
---
>       'i' -- announce user join/leave
>       'm' -- set monitor only mode (Listen only, no talking)
32,35d29
<       'o' -- set talker optimization - treats talkers who aren't speaking as
<              being muted, meaning (a) No encode is done on transmission and
<              (b) Received audio that is not registered as talking is omitted
<              causing no buildup in background noise
41,42c35
<              meetme-conf-rec-${CONFNO}-${UNIQUEID} and the default format is
<              wav.
---
>              meetme-conf-rec-${CONFNO}-${UNIQUEID} and the default format is wav.
52d44
<       '1' -- do not play message when first person enters

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= 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 one user
      'M' -- Mute one user
      'n' -- Unmute all users in the conference
      'N' -- Mute all non-admin users in the conference
      'r' -- Reset one user's volume settings
      'R' -- Reset all users volume settings
      's' -- Lower entire conference speaking volume
      'S' -- Raise entire conference speaking volume
      't' -- Lower one user's talk volume
      'T' -- Raise one user's talk volume
      'u' -- Lower one user's listen volume
      'U' -- Raise one user's listen volume
      'v' -- Lower entire conference listening volume
      'V' -- Raise entire conference listening volume

Differenz zum internen Hilfetext in Asterisk 1.2: 

14,27c14,17
<       'm' -- Unmute one user
<       'M' -- Mute one user
<       'n' -- Unmute all users in the conference
<       'N' -- Mute all non-admin users in the conference
<       'r' -- Reset one user's volume settings
<       'R' -- Reset all users volume settings
<       's' -- Lower entire conference speaking volume
<       'S' -- Raise entire conference speaking volume
<       't' -- Lower one user's talk volume
<       'T' -- Raise one user's talk volume
<       'u' -- Lower one user's listen volume
<       'U' -- Raise one user's listen volume
<       'v' -- Lower entire conference listening volume
<       'V' -- Raise entire conference listening volume
---
>       'm' -- Unmute conference
>       'M' -- Mute conference
>       'n' -- Unmute entire conference (except admin)
>       'N' -- Mute entire conference (except admin)

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= 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.

Differenz zum internen Hilfetext in Asterisk 1.2: 

10,12c10,11
< 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.
---
> 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.

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= 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)

Differenz zum internen Hilfetext in Asterisk 1.2: 

- keine - 

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= 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.

Differenz zum internen Hilfetext in Asterisk 1.2: 

- keine - 

Einige Anwender, die viele (50-500) Gespräche gleichzeitig aufnehmen, berichten von stark verbesserter Performance, wenn man auf eine RAM-Disk aufzeichnet (weniger Such-Operationen) und die Dateien erst nach Gesprächsende auf die Festplatte (lokal oder gemountet) kopiert.

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= 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.

Differenz zum internen Hilfetext in Asterisk 1.2: 

- keine - 

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= Info about application 'Morsecode' =- 

[Synopsis]
Plays morse code

[Description]
Usage: Morsecode(<string>)
Plays the Morse code equivalent of the passed string.  If the variable
MORSEDITLEN is set, it will use that value for the length (in ms) of the dit
(defaults to 80).  Additionally, if MORSETONE is set, it will use that tone
(in Hz).  The tone default is 800.

Differenz zum internen Hilfetext in Asterisk 1.2: 

-- in Version 1.2 nicht vorhanden --

Das häufig als Alternative installierte und auf mpg123 verlinkte Programm mpg321 funktioniert nicht einwandfrei.

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= 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.

Differenz zum internen Hilfetext in Asterisk 1.2: 

- keine - 

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= 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.

Differenz zum internen Hilfetext in Asterisk 1.2: 

- keine - 

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= 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
.

Differenz zum internen Hilfetext in Asterisk 1.2: 

- keine - 

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= 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.

Differenz zum internen Hilfetext in Asterisk 1.2: 

- keine - 

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= 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.

Differenz zum internen Hilfetext in Asterisk 1.2: 

- keine - 

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= 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
        r - record the page into a file (see 'r' for app_meetme)

Differenz zum internen Hilfetext in Asterisk 1.2: 

14,15c14
<         q - quiet, do not play beep to caller
<         r - record the page into a file (see 'r' for app_meetme)
---
>    q - quiet, do not play beep to caller

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= 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 (or the context specified in features.conf).

If you set the PARKINGEXTEN variable to an extension in your
parking context, park() will park the call on that extension, unless
it already exists. In that case, execution will continue at next
priority.

Differenz zum internen Hilfetext in Asterisk 1.2: 

12,17c12
< context (or the context specified in features.conf).
< 
< If you set the PARKINGEXTEN variable to an extension in your
< parking context, park() will park the call on that extension, unless
< it already exists. In that case, execution will continue at next
< priority.
---
> context.

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= 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 to another channel.

announce template: Colon-separated list of files to announce.  The word PARKED
                   will be replaced by a say_digits of the extension in which
                   the call is parked.
timeout:           Time in seconds before the call returns into the return
                   context.
dial:              The app_dial style resource to call to make the
                   announcement.  Console/dsp calls the console.
return_context:    The goto-style label to jump the call back into after
                   timeout.  Default <priority+1>.

The variable ${PARKEDAT} will contain the parking extension into which the
call was placed.  Use with the Local channel to allow the dialplan to make
use of this information.

Differenz zum internen Hilfetext in Asterisk 1.2: 

9,23c9,14
< Park a call into the parkinglot and announce the call to another channel.
< 
< announce template: Colon-separated list of files to announce.  The word PARKED
<                    will be replaced by a say_digits of the extension in which
<                    the call is parked.
< timeout:           Time in seconds before the call returns into the return
<                    context.
< dial:              The app_dial style resource to call to make the
<                    announcement.  Console/dsp calls the console.
< return_context:    The goto-style label to jump the call back into after
<                    timeout.  Default <priority+1>.
< 
< The variable ${PARKEDAT} will contain the parking extension into which the
< call was placed.  Use with the Local channel to allow the dialplan to make
< use of this information.
---
> 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

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= 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.

Differenz zum internen Hilfetext in Asterisk 1.2: 

- keine - 

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= Info about application 'PauseMonitor' =- 

[Synopsis]
Pause monitoring of a channel

[Description]
PauseMonitor
Pauses monitoring of a channel until it is re-enabled by a call to UnpauseMonitor.

Differenz zum internen Hilfetext in Asterisk 1.2: 

-- in Version 1.2 nicht vorhanden --

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= 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)

Differenz zum internen Hilfetext in Asterisk 1.2: 

- keine - 

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= Info about application 'Pickup' =- 

[Synopsis]
Directed Call Pickup

[Description]
  Pickup(extension[@context][&extension2@context...]): This application can pickup any ringing channel
that is calling the specified extension. If no context is specified, the current
context will be used. If you use the special string "PICKUPMARK" for the context parameter, for example
10@PICKUPMARK, this application tries to find a channel which has defined a channel variable with the same content
as "extension".

Differenz zum internen Hilfetext in Asterisk 1.2: 

8c8
<   Pickup(extension[@context][&extension2@context...]): This application can pickup any ringing channel
---
>   Pickup(extension[@context]): This application can pickup any ringing channel
10,12c10,11
< context will be used. If you use the special string "PICKUPMARK" for the context parameter, for example
< 10@PICKUPMARK, this application tries to find a channel which has defined a channel variable with the same content
< as "extension".
---
> context will be used.
> 

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= 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

Differenz zum internen Hilfetext in Asterisk 1.2: 

- keine - 

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= 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.

Differenz zum internen Hilfetext in Asterisk 1.2: 

- keine - 

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= 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

Differenz zum internen Hilfetext in Asterisk 1.2: 

- keine - 

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= Info about application 'Progress' =- 

[Synopsis]
Indicate progress

[Description]
  Progress(): This application will request that in-band progress information
be provided to the calling channel.

Differenz zum internen Hilfetext in Asterisk 1.2: 

- keine - 

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= Info about application 'Queue' =- 

[Synopsis]
Queue a call for a call queue

[Description]
  Queue(queuename[|options[|URL][|announceoverride][|timeout][|AGI]):
Queues an incoming call in a particular call queue as defined in queues.conf.
This application will return to the dialplan if the queue does not exist, or
any of the join options cause the caller to not enter the queue.
The option string may contain zero or more of the following characters:
      'd' -- data-quality (modem) call (minimum delay).
      'h' -- allow callee to hang up by hitting *.
      'H' -- allow caller to hang up by hitting *.
      'n' -- no retries on the timeout; will exit this application and 
             go to the next step.
      'i' -- ignore call forward requests from queue members and do nothing
             when they are requested.
      'r' -- ring instead of playing MOH
      't' -- allow the called user transfer the calling user
      'T' -- to allow the calling user to transfer the call.
      'w' -- allow the called user to write the conversation to disk via Monitor
      'W' -- allow the calling user to write the conversation to disk via Monitor
  In addition to transferring the call, a call may be parked and then picked
up by another user.
  The optional URL will be sent to the called party if the channel supports
it.
  The optional AGI parameter will setup an AGI script to be executed on the 
calling party's channel once they are connected to a queue member.
  The timeout will cause the queue to fail out after a specified number of
seconds, checked between each queues.conf 'timeout' and 'retry' cycle.
  This application sets the following channel variable upon completion:
      QUEUESTATUS    The status of the call as a text string, one of
             TIMEOUT | FULL | JOINEMPTY | LEAVEEMPTY | JOINUNAVAIL | LEAVEUNAVAIL

Differenz zum internen Hilfetext in Asterisk 1.2: 

8c8
<   Queue(queuename[|options[|URL][|announceoverride][|timeout][|AGI]):
---
>   Queue(queuename[|options[|URL][|announceoverride][|timeout]]):
17,19c17
<              go to the next step.
<       'i' -- ignore call forward requests from queue members and do nothing
<              when they are requested.
---
>        go to the next step.
29,30d26
<   The optional AGI parameter will setup an AGI script to be executed on the 
< calling party's channel once they are connected to a queue member.

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= Info about application 'QueueLog' =- 

[Synopsis]
Writes to the queue_log

[Description]
   QueueLog(queuename|uniqueid|agent|event[|additionalinfo]):
Allows you to write your own events into the queue log
Example: QueueLog(101|${UNIQUEID}|${AGENT}|WENTONBREAK|600)

Differenz zum internen Hilfetext in Asterisk 1.2: 

-- in Version 1.2 nicht vorhanden --

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= 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
DEPRECATED: Use GotoIf($[${RAND(1,100)} > <number>]?<label>)

Differenz zum internen Hilfetext in Asterisk 1.2: 

10d9
< DEPRECATED: Use GotoIf($[${RAND(1,100)} > <number>]?<label>)

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= 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 or tone with option i
  maxdigits  -- maximum acceptable number of digits. Stops reading after
                maxdigits have been entered (without requiring the user to
                press the '#' key).
                Defaults to 0 - no limit - wait for the user press the '#' key.
                Any value below 0 means the same. Max accepted value is 255.
  option     -- options are 's' , 'i', 'n'
                's' to return immediately if the line is not up,
                'i' to play  filename as an indication tone from your indications.conf
                'n' to read digits even if the line is not up.
  attempts   -- if greater than 1, that many attempts will be made in the 
                event no data is entered.
  timeout    -- An integer number of seconds to wait for a digit response. If greater
                than 0, that value will override the default timeout.

Read should disconnect if the function fails or errors out.

Differenz zum internen Hilfetext in Asterisk 1.2: 

12c12
<   filename   -- file to play before reading digits or tone with option i
---
>   filename   -- file to play before reading digits.
18,21c18,19
<   option     -- options are 's' , 'i', 'n'
<                 's' to return immediately if the line is not up,
<                 'i' to play  filename as an indication tone from your indications.conf
<                 'n' to read digits even if the line is not up.
---
>   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.
24,25c22
<   timeout    -- An integer number of seconds to wait for a digit response. If greater
<                 than 0, that value will override the default timeout.
---
>   timeout    -- if greater than 0, that value will override the default timeout.

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= 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.

Differenz zum internen Hilfetext in Asterisk 1.2: 

- keine - 

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= 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.  For example, a prefix of 'var_' would make the column 'name'
become the variable ${var_name}.  REALTIMECOUNT will be set with the number
of values read.

Differenz zum internen Hilfetext in Asterisk 1.2: 

11,14c11,13
< All unique column names will be set as channel variables with optional prefix
< to the name.  For example, a prefix of 'var_' would make the column 'name'
< become the variable ${var_name}.  REALTIMECOUNT will be set with the number
< of values read.
---
> 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}
> 

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= 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>.  REALTIMECOUNT will be set with the number of rows
updated or -1 if an error occurs.

Differenz zum internen Hilfetext in Asterisk 1.2: 

11,13c11
< The column <newcol> in 'family' matching column <colmatch>=<value> will be
< updated to <newval>.  REALTIMECOUNT will be set with the number of rows
< updated or -1 if an error occurs.
---
> The column <newcol> in 'family' matching column <colmatch>=<value> will be updated to <newval>

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= 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 (DTMF) instead of default '#'
     'x' : ignore all terminator keys (DTMF) and keep recording until hangup

If filename contains '%d', these characters will be replaced with a number
incremented by one each time the file is recorded. 

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.

Differenz zum internen Hilfetext in Asterisk 1.2: 

21,22c21
<      't' : use alternate '*' terminator key (DTMF) instead of default '#'
<      'x' : ignore all terminator keys (DTMF) and keep recording until hangup
---
>      't' : use alternate '*' terminator key instead of default '#'

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= 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)

Differenz zum internen Hilfetext in Asterisk 1.2: 

- keine - 

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= 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.

Differenz zum internen Hilfetext in Asterisk 1.2: 

- keine - 

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= 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.

Differenz zum internen Hilfetext in Asterisk 1.2: 

- keine - 

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= Info about application 'Return' =- 

[Synopsis]
Return from gosub routine

[Description]
Return()
  Jumps to the last label on the stack, removing it.

Differenz zum internen Hilfetext in Asterisk 1.2: 

- keine - 

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= Info about application 'Ringing' =- 

[Synopsis]
Indicate ringing tone

[Description]
  Ringing(): This application will request that the channel indicate a ringing
tone to the user.

Differenz zum internen Hilfetext in Asterisk 1.2: 

- keine - 

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= 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.

Differenz zum internen Hilfetext in Asterisk 1.2: 

- keine - 

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= 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.

Differenz zum internen Hilfetext in Asterisk 1.2: 

- keine - 

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= 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.

Differenz zum internen Hilfetext in Asterisk 1.2: 

- keine - 

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= 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.

Differenz zum internen Hilfetext in Asterisk 1.2: 

- keine - 

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= 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"

Differenz zum internen Hilfetext in Asterisk 1.2: 

- keine - 

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= 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.

Differenz zum internen Hilfetext in Asterisk 1.2: 

- keine - 

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= 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

Differenz zum internen Hilfetext in Asterisk 1.2: 

- keine - 

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= 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

Differenz zum internen Hilfetext in Asterisk 1.2: 

- keine - 

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= 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        Client failed to load URL (wait enabled)
    UNSUPPORTED   Channel does not support URL transport

If the option 'wait' is specified, execution will wait for an
acknowledgement that the URL has been loaded before continuing

If jumping is specified as an option (the 'j' flag), the client does not
support Asterisk "html" transport, and there exists a step with priority
n + 101, then execution will continue at that step.

SendURL continues normally if the URL was sent correctly or if the channel
does not support HTML transport.  Otherwise, the channel is hung up.

Differenz zum internen Hilfetext in Asterisk 1.2: 

13c13
<     NOLOAD        Client failed to load URL (wait enabled)
---
>     NOLOAD        Clien failed to load URL (wait enabled)
17a18
> and will return -1 if the peer is unable to load the URL
19,24c20,26
< If jumping is specified as an option (the 'j' flag), the client does not
< support Asterisk "html" transport, and there exists a step with priority
< n + 101, then execution will continue at that step.
< 
< SendURL continues normally if the URL was sent correctly or if the channel
< does not support HTML transport.  Otherwise, the channel is hung up.
---
> 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.

Ob globale Variablen auch nach einem Reload von Asterisk noch gültig sind, wird durch die Einstellung clearglobalvars in der extensions.conf bestimmt.

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= 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)

Differenz zum internen Hilfetext in Asterisk 1.2: 

- keine - 

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= Info about application 'SetAMAFlags' =- 

[Synopsis]
Set the AMA Flags

[Description]
  SetAMAFlags([flag]): This application will set the channel's AMA Flags for
  billing purposes.

Differenz zum internen Hilfetext in Asterisk 1.2: 

8,9c8,9
<   SetAMAFlags([flag]): This application will set the channel's AMA Flags for
<   billing purposes.
---
>   SetAMAFlags([flag]): This channel will set the channel's AMA Flags for billing
> purposes.

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= 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

Differenz zum internen Hilfetext in Asterisk 1.2: 

- keine - 

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= 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().

Differenz zum internen Hilfetext in Asterisk 1.2: 

- keine - 

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= 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.

Differenz zum internen Hilfetext in Asterisk 1.2: 

- keine - 

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= 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.

Differenz zum internen Hilfetext in Asterisk 1.2: 

- keine - 

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= 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

Differenz zum internen Hilfetext in Asterisk 1.2: 

17c17
<   VIDEO              : 0x18 - Video
---
>   VIDEO              : 0x18 - Video:

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= 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

Differenz zum internen Hilfetext in Asterisk 1.2: 

- keine - 

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= Info about application 'SIPDtmfMode' =- 

[Synopsis]
Change the dtmfmode for a SIP call

[Description]
SIPDtmfMode(inband|info|rfc2833): Changes the dtmfmode for a SIP call

Differenz zum internen Hilfetext in Asterisk 1.2: 

- keine - 

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= 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.

Differenz zum internen Hilfetext in Asterisk 1.2: 

- keine - 

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= 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

Differenz zum internen Hilfetext in Asterisk 1.2: 

- keine - 

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= Info about application 'StopMonitor' =- 

[Synopsis]
Stop monitoring a channel

[Description]
StopMonitor
Stops monitoring a channel. Has no effect if the channel is not monitored

Differenz zum internen Hilfetext in Asterisk 1.2: 

- keine - 

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= Info about application 'StopPlayTones' =- 

[Synopsis]
Stop playing a tone list

[Description]
Stop playing a tone list

Differenz zum internen Hilfetext in Asterisk 1.2: 

- keine - 

Als Alternative kann die Applikation Backticks() oder Funktion BACKTICKS() aus dem Modul app_backticks^[112] verwenden, mit denen man auch die Ausgabe des Befehls erhält.

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= 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.

Differenz zum internen Hilfetext in Asterisk 1.2: 

- keine - 

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= 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

Differenz zum internen Hilfetext in Asterisk 1.2: 

- keine - 

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= 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.

Differenz zum internen Hilfetext in Asterisk 1.2: 

- keine - 

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= Info about application 'UnpauseMonitor' =- 

[Synopsis]
Unpause monitoring of a channel

[Description]
UnpauseMonitor
Unpauses monitoring of a channel on which monitoring had
previously been paused with PauseMonitor.

Differenz zum internen Hilfetext in Asterisk 1.2: 

-- in Version 1.2 nicht vorhanden --

Interner Hilfetext zu dieser Applikation in Asterisk 1.4: 

  -= 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)

Differenz zum internen Hilfetext in Asterisk 1.2: 

- keine -