asterisk/include/asterisk/app.h

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/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 1999 - 2005, Digium, Inc.
*
* Mark Spencer <markster@digium.com>
* See http://www.asterisk.org for more information about
* the Asterisk project. Please do not directly contact
* any of the maintainers of this project for assistance;
* the project provides a web site, mailing lists and IRC
* channels for your use.
*
* This program is free software, distributed under the terms of
* the GNU General Public License Version 2. See the LICENSE file
* at the top of the source tree.
*/
/*! \file
* \brief Application convenience functions, designed to give consistent
* look and feel to Asterisk apps.
*/
#ifndef _ASTERISK_APP_H
#define _ASTERISK_APP_H
Merge changes dealing with support for Digium phones. Presence support has been added. This is accomplished by allowing for presence hints in addition to device state hints. A dialplan function called PRESENCE_STATE has been added to allow for setting and reading presence. Presence can be transmitted to Digium phones using custom XML elements in a PIDF presence document. Voicemail has new APIs that allow for moving, removing, forwarding, and playing messages. Messages have had a new unique message ID added to them so that the APIs will work reliably. The state of a voicemail mailbox can be obtained using an API that allows one to get a snapshot of the mailbox. A voicemail Dialplan App called VoiceMailPlayMsg has been added to be able to play back a specific message. Configuration hooks have been added. Configuration hooks allow for a piece of code to be executed when a specific configuration file is loaded by a specific module. This is useful for modules that are dependent on the configuration of other modules. chan_sip now has a public method that allows for a custom SIP INFO request to be sent mid-dialog. Digium phones use this in order to display progress bars when files are played. Messaging support has been expanded a bit. The main visible difference is the addition of an AMI action MessageSend. Finally, a ParkingLots manager action has been added in order to get a list of parking lots. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-04 20:26:12 +00:00
#include "asterisk/stringfields.h"
#include "asterisk/strings.h"
#include "asterisk/threadstorage.h"
Merge changes dealing with support for Digium phones. Presence support has been added. This is accomplished by allowing for presence hints in addition to device state hints. A dialplan function called PRESENCE_STATE has been added to allow for setting and reading presence. Presence can be transmitted to Digium phones using custom XML elements in a PIDF presence document. Voicemail has new APIs that allow for moving, removing, forwarding, and playing messages. Messages have had a new unique message ID added to them so that the APIs will work reliably. The state of a voicemail mailbox can be obtained using an API that allows one to get a snapshot of the mailbox. A voicemail Dialplan App called VoiceMailPlayMsg has been added to be able to play back a specific message. Configuration hooks have been added. Configuration hooks allow for a piece of code to be executed when a specific configuration file is loaded by a specific module. This is useful for modules that are dependent on the configuration of other modules. chan_sip now has a public method that allows for a custom SIP INFO request to be sent mid-dialog. Digium phones use this in order to display progress bars when files are played. Messaging support has been expanded a bit. The main visible difference is the addition of an AMI action MessageSend. Finally, a ParkingLots manager action has been added in order to get a list of parking lots. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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#include "asterisk/file.h"
struct ast_flags64;
#if defined(__cplusplus) || defined(c_plusplus)
extern "C" {
#endif
AST_THREADSTORAGE_EXTERNAL(ast_str_thread_global_buf);
/* IVR stuff */
/*! \brief Callback function for IVR
\return returns 0 on completion, -1 on hangup or digit if interrupted
*/
typedef int (*ast_ivr_callback)(struct ast_channel *chan, char *option, void *cbdata);
typedef enum {
AST_ACTION_UPONE, /*!< adata is unused */
AST_ACTION_EXIT, /*!< adata is the return value for ast_ivr_menu_run if channel was not hungup */
AST_ACTION_CALLBACK, /*!< adata is an ast_ivr_callback */
AST_ACTION_PLAYBACK, /*!< adata is file to play */
AST_ACTION_BACKGROUND, /*!< adata is file to play */
AST_ACTION_PLAYLIST, /*!< adata is list of files, separated by ; to play */
AST_ACTION_MENU, /*!< adata is a pointer to an ast_ivr_menu */
AST_ACTION_REPEAT, /*!< adata is max # of repeats, cast to a pointer */
AST_ACTION_RESTART, /*!< adata is like repeat, but resets repeats to 0 */
AST_ACTION_TRANSFER, /*!< adata is a string with exten\verbatim[@context]\endverbatim */
AST_ACTION_WAITOPTION, /*!< adata is a timeout, or 0 for defaults */
AST_ACTION_NOOP, /*!< adata is unused */
AST_ACTION_BACKLIST, /*!< adata is list of files separated by ; allows interruption */
} ast_ivr_action;
/*!
Special "options" are:
\arg "s" - "start here (one time greeting)"
\arg "g" - "greeting/instructions"
\arg "t" - "timeout"
\arg "h" - "hangup"
\arg "i" - "invalid selection"
*/
struct ast_ivr_option {
char *option;
ast_ivr_action action;
void *adata;
};
struct ast_ivr_menu {
char *title; /*!< Title of menu */
unsigned int flags; /*!< Flags */
struct ast_ivr_option *options; /*!< All options */
};
Merge changes dealing with support for Digium phones. Presence support has been added. This is accomplished by allowing for presence hints in addition to device state hints. A dialplan function called PRESENCE_STATE has been added to allow for setting and reading presence. Presence can be transmitted to Digium phones using custom XML elements in a PIDF presence document. Voicemail has new APIs that allow for moving, removing, forwarding, and playing messages. Messages have had a new unique message ID added to them so that the APIs will work reliably. The state of a voicemail mailbox can be obtained using an API that allows one to get a snapshot of the mailbox. A voicemail Dialplan App called VoiceMailPlayMsg has been added to be able to play back a specific message. Configuration hooks have been added. Configuration hooks allow for a piece of code to be executed when a specific configuration file is loaded by a specific module. This is useful for modules that are dependent on the configuration of other modules. chan_sip now has a public method that allows for a custom SIP INFO request to be sent mid-dialog. Digium phones use this in order to display progress bars when files are played. Messaging support has been expanded a bit. The main visible difference is the addition of an AMI action MessageSend. Finally, a ParkingLots manager action has been added in order to get a list of parking lots. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-04 20:26:12 +00:00
/*!
* \brief Structure used for ast_copy_recording_to_vm in order to cleanly supply
* data needed for making the recording from the recorded file.
*/
struct ast_vm_recording_data {
AST_DECLARE_STRING_FIELDS(
AST_STRING_FIELD(context);
AST_STRING_FIELD(mailbox);
AST_STRING_FIELD(folder);
AST_STRING_FIELD(recording_file);
AST_STRING_FIELD(recording_ext);
AST_STRING_FIELD(call_context);
AST_STRING_FIELD(call_macrocontext);
AST_STRING_FIELD(call_extension);
AST_STRING_FIELD(call_callerchan);
AST_STRING_FIELD(call_callerid);
);
int call_priority;
};
#define AST_IVR_FLAG_AUTORESTART (1 << 0)
#define AST_IVR_DECLARE_MENU(holder, title, flags, foo...) \
static struct ast_ivr_option __options_##holder[] = foo;\
static struct ast_ivr_menu holder = { title, flags, __options_##holder }
enum ast_timelen {
TIMELEN_HOURS,
TIMELEN_MINUTES,
TIMELEN_SECONDS,
TIMELEN_MILLISECONDS,
};
/*! \brief Runs an IVR menu
\return returns 0 on successful completion, -1 on hangup, or -2 on user error in menu */
int ast_ivr_menu_run(struct ast_channel *c, struct ast_ivr_menu *menu, void *cbdata);
/*! \brief Plays a stream and gets DTMF data from a channel
* \param c Which channel one is interacting with
* \param prompt File to pass to ast_streamfile (the one that you wish to play).
* It is also valid for this to be multiple files concatenated by "&".
* For example, "file1&file2&file3".
* \param s The location where the DTMF data will be stored
* \param maxlen Max Length of the data
* \param timeout Timeout length waiting for data(in milliseconds). Set to 0 for standard timeout(six seconds), or -1 for no time out.
*
* This function was designed for application programmers for situations where they need
* to play a message and then get some DTMF data in response to the message. If a digit
* is pressed during playback, it will immediately break out of the message and continue
* execution of your code.
*/
int ast_app_getdata(struct ast_channel *c, const char *prompt, char *s, int maxlen, int timeout);
/*! \brief Full version with audiofd and controlfd. NOTE: returns '2' on ctrlfd available, not '1' like other full functions */
int ast_app_getdata_full(struct ast_channel *c, const char *prompt, char *s, int maxlen, int timeout, int audiofd, int ctrlfd);
/*!
* \brief Run a macro on a channel, placing an optional second channel into autoservice.
* \since 11.0
*
* \details
* This is a shorthand method that makes it very easy to run a
* macro on any given channel. It is perfectly reasonable to
* supply a NULL autoservice_chan here in case there is no
* channel to place into autoservice.
*
* \note It is very important that the autoservice_chan is not
* locked prior to calling. Otherwise, a deadlock could result.
*
* \param autoservice_chan A channel to place into autoservice while the macro is run
* \param macro_chan Channel to execute macro on.
* \param macro_args Macro application argument string.
*
* \retval 0 success
* \retval -1 on error
*/
int ast_app_exec_macro(struct ast_channel *autoservice_chan, struct ast_channel *macro_chan, const char *macro_args);
/*!
* \since 1.8
* \brief Run a macro on a channel, placing an optional second channel into autoservice.
*
* \details
* This is a shorthand method that makes it very easy to run a
* macro on any given channel. It is perfectly reasonable to
* supply a NULL autoservice_chan here in case there is no
* channel to place into autoservice.
*
* \note It is very important that the autoservice_chan is not
* locked prior to calling. Otherwise, a deadlock could result.
*
* \param autoservice_chan A channel to place into autoservice while the macro is run
* \param macro_chan Channel to execute macro on.
* \param macro_name The name of the macro to run.
* \param macro_args The arguments to pass to the macro.
*
* \retval 0 success
* \retval -1 on error
*/
int ast_app_run_macro(struct ast_channel *autoservice_chan,
struct ast_channel *macro_chan, const char *macro_name, const char *macro_args);
/*!
* \since 11
* \brief Run a subroutine on a channel, placing an optional second channel into autoservice.
*
* \details
* This is a shorthand method that makes it very easy to run a
* subroutine on any given channel. It is perfectly reasonable
* to supply a NULL autoservice_chan here in case there is no
* channel to place into autoservice.
*
* \note It is very important that the autoservice_chan is not
* locked prior to calling. Otherwise, a deadlock could result.
*
* \param autoservice_chan A channel to place into autoservice while the subroutine is run
* \param sub_chan Channel to execute subroutine on.
* \param sub_args Gosub application argument string.
*
* \retval 0 success
* \retval -1 on error
*/
int ast_app_exec_sub(struct ast_channel *autoservice_chan, struct ast_channel *sub_chan, const char *sub_args);
/*!
* \since 11
* \brief Run a subroutine on a channel, placing an optional second channel into autoservice.
*
* \details
* This is a shorthand method that makes it very easy to run a
* subroutine on any given channel. It is perfectly reasonable
* to supply a NULL autoservice_chan here in case there is no
* channel to place into autoservice.
*
* \note It is very important that the autoservice_chan is not
* locked prior to calling. Otherwise, a deadlock could result.
*
* \param autoservice_chan A channel to place into autoservice while the subroutine is run
* \param sub_chan Channel to execute subroutine on.
* \param sub_location The location of the subroutine to run.
* \param sub_args The arguments to pass to the subroutine.
*
* \retval 0 success
* \retval -1 on error
*/
int ast_app_run_sub(struct ast_channel *autoservice_chan,
struct ast_channel *sub_chan, const char *sub_location, const char *sub_args);
/*!
* \brief Set voicemail function callbacks
* \param[in] has_voicemail_func set function pointer
* \param[in] inboxcount2_func set function pointer
* \param[in] sayname_func set function pointer
* \param[in] inboxcount_func set function pointer
* \param[in] messagecount_func set function pointer
* \version 1.6.1 Added inboxcount2_func, sayname_func
*/
void ast_install_vm_functions(int (*has_voicemail_func)(const char *mailbox, const char *folder),
int (*inboxcount_func)(const char *mailbox, int *newmsgs, int *oldmsgs),
int (*inboxcount2_func)(const char *mailbox, int *urgentmsgs, int *newmsgs, int *oldmsgs),
int (*messagecount_func)(const char *context, const char *mailbox, const char *folder),
Merge changes dealing with support for Digium phones. Presence support has been added. This is accomplished by allowing for presence hints in addition to device state hints. A dialplan function called PRESENCE_STATE has been added to allow for setting and reading presence. Presence can be transmitted to Digium phones using custom XML elements in a PIDF presence document. Voicemail has new APIs that allow for moving, removing, forwarding, and playing messages. Messages have had a new unique message ID added to them so that the APIs will work reliably. The state of a voicemail mailbox can be obtained using an API that allows one to get a snapshot of the mailbox. A voicemail Dialplan App called VoiceMailPlayMsg has been added to be able to play back a specific message. Configuration hooks have been added. Configuration hooks allow for a piece of code to be executed when a specific configuration file is loaded by a specific module. This is useful for modules that are dependent on the configuration of other modules. chan_sip now has a public method that allows for a custom SIP INFO request to be sent mid-dialog. Digium phones use this in order to display progress bars when files are played. Messaging support has been expanded a bit. The main visible difference is the addition of an AMI action MessageSend. Finally, a ParkingLots manager action has been added in order to get a list of parking lots. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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int (*sayname_func)(struct ast_channel *chan, const char *mailbox, const char *context),
int (*copy_recording_to_vm_func)(struct ast_vm_recording_data *vm_rec_data));
void ast_uninstall_vm_functions(void);
Merge changes dealing with support for Digium phones. Presence support has been added. This is accomplished by allowing for presence hints in addition to device state hints. A dialplan function called PRESENCE_STATE has been added to allow for setting and reading presence. Presence can be transmitted to Digium phones using custom XML elements in a PIDF presence document. Voicemail has new APIs that allow for moving, removing, forwarding, and playing messages. Messages have had a new unique message ID added to them so that the APIs will work reliably. The state of a voicemail mailbox can be obtained using an API that allows one to get a snapshot of the mailbox. A voicemail Dialplan App called VoiceMailPlayMsg has been added to be able to play back a specific message. Configuration hooks have been added. Configuration hooks allow for a piece of code to be executed when a specific configuration file is loaded by a specific module. This is useful for modules that are dependent on the configuration of other modules. chan_sip now has a public method that allows for a custom SIP INFO request to be sent mid-dialog. Digium phones use this in order to display progress bars when files are played. Messaging support has been expanded a bit. The main visible difference is the addition of an AMI action MessageSend. Finally, a ParkingLots manager action has been added in order to get a list of parking lots. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-04 20:26:12 +00:00
/*!
* \brief
* param[in] vm_rec_data Contains data needed to make the recording.
* retval 0 voicemail successfully created from recording.
* retval -1 Failure
*/
int ast_app_copy_recording_to_vm(struct ast_vm_recording_data *vm_rec_data);
/*!
* \brief Determine if a given mailbox has any voicemail
* If folder is NULL, defaults to "INBOX". If folder is "INBOX", includes the
* number of messages in the "Urgent" folder.
* \retval 1 Mailbox has voicemail
* \retval 0 No new voicemail in specified mailbox
* \retval -1 Failure
* \since 1.0
*/
int ast_app_has_voicemail(const char *mailbox, const char *folder);
/*!
* \brief Determine number of new/old messages in a mailbox
* \since 1.0
* \param[in] mailbox Mailbox specification in the format mbox[@context][&mbox2[@context2]][...]
* \param[out] newmsgs Number of messages in the "INBOX" folder. Includes number of messages in the "Urgent" folder, if any.
* \param[out] oldmsgs Number of messages in the "Old" folder.
* \retval 0 Success
* \retval -1 Failure
*/
int ast_app_inboxcount(const char *mailbox, int *newmsgs, int *oldmsgs);
/*!
* \brief Determine number of urgent/new/old messages in a mailbox
* \param[in] mailbox the mailbox context to use
* \param[out] urgentmsgs the urgent message count
* \param[out] newmsgs the new message count
* \param[out] oldmsgs the old message count
* \return Returns 0 for success, negative upon error
* \since 1.6.1
*/
int ast_app_inboxcount2(const char *mailbox, int *urgentmsgs, int *newmsgs, int *oldmsgs);
/*!
* \brief Given a mailbox and context, play that mailbox owner's name to the channel specified
* \param[in] chan Channel on which to play the name
* \param[in] mailbox Mailbox number from which to retrieve the recording
* \param[in] context Mailbox context from which to locate the mailbox number
* \retval 0 Name played without interruption
* \retval dtmf ASCII value of the DTMF which interrupted playback.
* \retval -1 Unable to locate mailbox or hangup occurred.
* \since 1.6.1
*/
int ast_app_sayname(struct ast_channel *chan, const char *mailbox, const char *context);
/*!
* \brief Check number of messages in a given context, mailbox, and folder
* \since 1.4
* \param[in] context Mailbox context
* \param[in] mailbox Mailbox number
* \param[in] folder Mailbox folder
* \return Number of messages in the given context, mailbox, and folder. If folder is NULL, folder "INBOX" is assumed. If folder is "INBOX", includes number of messages in the "Urgent" folder.
*/
int ast_app_messagecount(const char *context, const char *mailbox, const char *folder);
/*! \brief Safely spawn an external program while closing file descriptors
\note This replaces the \b system call in all Asterisk modules
*/
int ast_safe_system(const char *s);
/*!
* \brief Replace the SIGCHLD handler
*
* Normally, Asterisk has a SIGCHLD handler that is cleaning up all zombie
* processes from forking elsewhere in Asterisk. However, if you want to
* wait*() on the process to retrieve information about it's exit status,
* then this signal handler needs to be temporarily replaced.
*
* Code that executes this function *must* call ast_unreplace_sigchld()
* after it is finished doing the wait*().
*/
void ast_replace_sigchld(void);
/*!
* \brief Restore the SIGCHLD handler
*
* This function is called after a call to ast_replace_sigchld. It restores
* the SIGCHLD handler that cleans up any zombie processes.
*/
void ast_unreplace_sigchld(void);
/*!
\brief Send DTMF to a channel
\param chan The channel that will receive the DTMF frames
\param peer (optional) Peer channel that will be autoserviced while the
primary channel is receiving DTMF
\param digits This is a string of characters representing the DTMF digits
to be sent to the channel. Valid characters are
"0123456789*#abcdABCD". Note: You can pass arguments 'f' or
'F', if you want to Flash the channel (if supported by the
channel), or 'w' to add a 500 millisecond pause to the DTMF
sequence.
\param between This is the number of milliseconds to wait in between each
DTMF digit. If zero milliseconds is specified, then the
default value of 100 will be used.
\param duration This is the duration that each DTMF digit should have.
*/
int ast_dtmf_stream(struct ast_channel *chan, struct ast_channel *peer, const char *digits, int between, unsigned int duration);
/*! \brief Stream a filename (or file descriptor) as a generator. */
int ast_linear_stream(struct ast_channel *chan, const char *filename, int fd, int allowoverride);
/*!
* \brief Stream a file with fast forward, pause, reverse, restart.
* \param chan
* \param file filename
* \param fwd, rev, stop, pause, restart, skipms, offsetms
*
* Before calling this function, set this to be the number
* of ms to start from the beginning of the file. When the function
* returns, it will be the number of ms from the beginning where the
* playback stopped. Pass NULL if you don't care.
*/
int ast_control_streamfile(struct ast_channel *chan, const char *file, const char *fwd, const char *rev, const char *stop, const char *pause, const char *restart, int skipms, long *offsetms);
Merge changes dealing with support for Digium phones. Presence support has been added. This is accomplished by allowing for presence hints in addition to device state hints. A dialplan function called PRESENCE_STATE has been added to allow for setting and reading presence. Presence can be transmitted to Digium phones using custom XML elements in a PIDF presence document. Voicemail has new APIs that allow for moving, removing, forwarding, and playing messages. Messages have had a new unique message ID added to them so that the APIs will work reliably. The state of a voicemail mailbox can be obtained using an API that allows one to get a snapshot of the mailbox. A voicemail Dialplan App called VoiceMailPlayMsg has been added to be able to play back a specific message. Configuration hooks have been added. Configuration hooks allow for a piece of code to be executed when a specific configuration file is loaded by a specific module. This is useful for modules that are dependent on the configuration of other modules. chan_sip now has a public method that allows for a custom SIP INFO request to be sent mid-dialog. Digium phones use this in order to display progress bars when files are played. Messaging support has been expanded a bit. The main visible difference is the addition of an AMI action MessageSend. Finally, a ParkingLots manager action has been added in order to get a list of parking lots. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-04 20:26:12 +00:00
/*!
* \brief Stream a file with fast forward, pause, reverse, restart.
* \param chan
* \param file filename
* \param fwd, rev, stop, pause, restart, skipms, offsetms
* \param waitstream callback to invoke when fastforward or rewind occurrs.
*
* Before calling this function, set this to be the number
* of ms to start from the beginning of the file. When the function
* returns, it will be the number of ms from the beginning where the
* playback stopped. Pass NULL if you don't care.
*/
int ast_control_streamfile_w_cb(struct ast_channel *chan,
const char *file,
const char *fwd,
const char *rev,
const char *stop,
const char *pause,
const char *restart,
int skipms,
long *offsetms,
ast_waitstream_fr_cb cb);
/*! \brief Play a stream and wait for a digit, returning the digit that was pressed */
int ast_play_and_wait(struct ast_channel *chan, const char *fn);
Merged revisions 329528 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r329528 | jrose | 2011-07-26 08:52:34 -0500 (Tue, 26 Jul 2011) | 24 lines Merged revisions 329527 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r329527 | jrose | 2011-07-26 08:25:35 -0500 (Tue, 26 Jul 2011) | 17 lines Fixes some voicemail forwarding behavior based around prepend mode. Formerly, prepend forwarding would have the user record a message with no useful prompt and an expectation for the user to push a button on the phone when finished recording. If a length of silence was detected instead, the recording would be canceled and the user would re-enter the voicemail forwarding menu. Subsequent time-outs in prepend recording would also bug out in the sense that they would write over the original message and get sent to the recipient regardless of whether they timed out or were accepted. This patch fixes this issue and adds a prompt which will be played after a timeout informing the user that they needed to press a button. Currently, the sound files that we have are somewhat inadquate for this, so after the call we simply have Allison say "Please try again. Then press pound." which actually relies on two separate sound files. Just one would be more appropriate. reporter: Vlad Povorozniuc Review: https://reviewboard.asterisk.org/r/1327/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@329530 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-26 14:17:13 +00:00
/*!
* \brief Record a file based on input from a channel
* This function will play "auth-thankyou" upon successful recording.
*
* \param chan the channel being recorded
* \param playfile Filename of sound to play before recording begins
* \param recordfile Filename to save the recording
* \param maxtime_sec Longest possible message length in seconds
* \param fmt string containing all formats to be recorded delimited by '|'
* \param duration pointer to integer for storing length of the recording
Merged revisions 337120 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r337120 | mjordan | 2011-09-20 17:49:36 -0500 (Tue, 20 Sep 2011) | 28 lines Merged revisions 337118 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337118 | mjordan | 2011-09-20 17:38:54 -0500 (Tue, 20 Sep 2011) | 21 lines Fix for incorrect voicemail duration in external notifications This patch fixes an issue where the voicemail duration was being reported with a duration significantly less than the actual sound file duration. Voicemails that contained mostly silence were reporting the duration of only the sound in the file, as opposed to the duration of the file with the silence. This patch fixes this by having two durations reported in the __ast_play_and_record family of functions - the sound_duration and the actual duration of the file. The sound_duration, which is optional, now reports the duration of the sound in the file, while the actual full duration of the file is reported in the duration parameter. This allows the voicemail applications to use the sound_duration for minimum duration checking, while reporting the full duration to external parties if the voicemail is kept. (issue ASTERISK-2234) (closes issue ASTERISK-16981) Reported by: Mary Ciuciu, Byron Clark, Brad House, Karsten Wemheuer, KevinH Tested by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1443 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@337124 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-20 23:02:25 +00:00
* \param sound_duration pointer to integer for storing length of the recording minus all silence
Merged revisions 329528 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r329528 | jrose | 2011-07-26 08:52:34 -0500 (Tue, 26 Jul 2011) | 24 lines Merged revisions 329527 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r329527 | jrose | 2011-07-26 08:25:35 -0500 (Tue, 26 Jul 2011) | 17 lines Fixes some voicemail forwarding behavior based around prepend mode. Formerly, prepend forwarding would have the user record a message with no useful prompt and an expectation for the user to push a button on the phone when finished recording. If a length of silence was detected instead, the recording would be canceled and the user would re-enter the voicemail forwarding menu. Subsequent time-outs in prepend recording would also bug out in the sense that they would write over the original message and get sent to the recipient regardless of whether they timed out or were accepted. This patch fixes this issue and adds a prompt which will be played after a timeout informing the user that they needed to press a button. Currently, the sound files that we have are somewhat inadquate for this, so after the call we simply have Allison say "Please try again. Then press pound." which actually relies on two separate sound files. Just one would be more appropriate. reporter: Vlad Povorozniuc Review: https://reviewboard.asterisk.org/r/1327/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@329530 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-26 14:17:13 +00:00
* \param silencethreshold tolerance of noise levels that can be considered silence for the purpose of silence timeout, -1 for default
* \param maxsilence_ms Length of time in milliseconds which will trigger a timeout from silence, -1 for default
* \param path Optional filesystem path to unlock
* \param acceptdtmf Character of DTMF to end and accept the recording
* \param canceldtmf Character of DTMF to end and cancel the recording
*
* \retval -1 failure or hangup
* \retval 'S' Recording ended from silence timeout
* \retval 't' Recording ended from the message exceeding the maximum duration
* \retval dtmfchar Recording ended via the return value's DTMF character for either cancel or accept.
*/
Merged revisions 337120 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r337120 | mjordan | 2011-09-20 17:49:36 -0500 (Tue, 20 Sep 2011) | 28 lines Merged revisions 337118 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337118 | mjordan | 2011-09-20 17:38:54 -0500 (Tue, 20 Sep 2011) | 21 lines Fix for incorrect voicemail duration in external notifications This patch fixes an issue where the voicemail duration was being reported with a duration significantly less than the actual sound file duration. Voicemails that contained mostly silence were reporting the duration of only the sound in the file, as opposed to the duration of the file with the silence. This patch fixes this by having two durations reported in the __ast_play_and_record family of functions - the sound_duration and the actual duration of the file. The sound_duration, which is optional, now reports the duration of the sound in the file, while the actual full duration of the file is reported in the duration parameter. This allows the voicemail applications to use the sound_duration for minimum duration checking, while reporting the full duration to external parties if the voicemail is kept. (issue ASTERISK-2234) (closes issue ASTERISK-16981) Reported by: Mary Ciuciu, Byron Clark, Brad House, Karsten Wemheuer, KevinH Tested by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1443 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@337124 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-20 23:02:25 +00:00
int ast_play_and_record_full(struct ast_channel *chan, const char *playfile, const char *recordfile, int maxtime_sec, const char *fmt, int *duration, int *sound_duration, int silencethreshold, int maxsilence_ms, const char *path, const char *acceptdtmf, const char *canceldtmf);
Merged revisions 329528 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r329528 | jrose | 2011-07-26 08:52:34 -0500 (Tue, 26 Jul 2011) | 24 lines Merged revisions 329527 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r329527 | jrose | 2011-07-26 08:25:35 -0500 (Tue, 26 Jul 2011) | 17 lines Fixes some voicemail forwarding behavior based around prepend mode. Formerly, prepend forwarding would have the user record a message with no useful prompt and an expectation for the user to push a button on the phone when finished recording. If a length of silence was detected instead, the recording would be canceled and the user would re-enter the voicemail forwarding menu. Subsequent time-outs in prepend recording would also bug out in the sense that they would write over the original message and get sent to the recipient regardless of whether they timed out or were accepted. This patch fixes this issue and adds a prompt which will be played after a timeout informing the user that they needed to press a button. Currently, the sound files that we have are somewhat inadquate for this, so after the call we simply have Allison say "Please try again. Then press pound." which actually relies on two separate sound files. Just one would be more appropriate. reporter: Vlad Povorozniuc Review: https://reviewboard.asterisk.org/r/1327/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@329530 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-26 14:17:13 +00:00
/*!
* \brief Record a file based on input from a channel. Use default accept and cancel DTMF.
* This function will play "auth-thankyou" upon successful recording.
*
* \param chan the channel being recorded
* \param playfile Filename of sound to play before recording begins
* \param recordfile Filename to save the recording
* \param maxtime_sec Longest possible message length in seconds
* \param fmt string containing all formats to be recorded delimited by '|'
* \param duration pointer to integer for storing length of the recording
Merged revisions 337120 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r337120 | mjordan | 2011-09-20 17:49:36 -0500 (Tue, 20 Sep 2011) | 28 lines Merged revisions 337118 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337118 | mjordan | 2011-09-20 17:38:54 -0500 (Tue, 20 Sep 2011) | 21 lines Fix for incorrect voicemail duration in external notifications This patch fixes an issue where the voicemail duration was being reported with a duration significantly less than the actual sound file duration. Voicemails that contained mostly silence were reporting the duration of only the sound in the file, as opposed to the duration of the file with the silence. This patch fixes this by having two durations reported in the __ast_play_and_record family of functions - the sound_duration and the actual duration of the file. The sound_duration, which is optional, now reports the duration of the sound in the file, while the actual full duration of the file is reported in the duration parameter. This allows the voicemail applications to use the sound_duration for minimum duration checking, while reporting the full duration to external parties if the voicemail is kept. (issue ASTERISK-2234) (closes issue ASTERISK-16981) Reported by: Mary Ciuciu, Byron Clark, Brad House, Karsten Wemheuer, KevinH Tested by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1443 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@337124 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-20 23:02:25 +00:00
* \param sound_duration pointer to integer for storing length of the recording minus all silence
Merged revisions 329528 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r329528 | jrose | 2011-07-26 08:52:34 -0500 (Tue, 26 Jul 2011) | 24 lines Merged revisions 329527 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r329527 | jrose | 2011-07-26 08:25:35 -0500 (Tue, 26 Jul 2011) | 17 lines Fixes some voicemail forwarding behavior based around prepend mode. Formerly, prepend forwarding would have the user record a message with no useful prompt and an expectation for the user to push a button on the phone when finished recording. If a length of silence was detected instead, the recording would be canceled and the user would re-enter the voicemail forwarding menu. Subsequent time-outs in prepend recording would also bug out in the sense that they would write over the original message and get sent to the recipient regardless of whether they timed out or were accepted. This patch fixes this issue and adds a prompt which will be played after a timeout informing the user that they needed to press a button. Currently, the sound files that we have are somewhat inadquate for this, so after the call we simply have Allison say "Please try again. Then press pound." which actually relies on two separate sound files. Just one would be more appropriate. reporter: Vlad Povorozniuc Review: https://reviewboard.asterisk.org/r/1327/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@329530 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-26 14:17:13 +00:00
* \param silencethreshold tolerance of noise levels that can be considered silence for the purpose of silence timeout, -1 for default
* \param maxsilence_ms length of time in milliseconds which will trigger a timeout from silence, -1 for default
* \param path Optional filesystem path to unlock
*
* \retval -1 failure or hangup
* \retval 'S' Recording ended from silence timeout
* \retval 't' Recording ended from the message exceeding the maximum duration
* \retval dtmfchar Recording ended via the return value's DTMF character for either cancel or accept.
*/
Merged revisions 337120 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r337120 | mjordan | 2011-09-20 17:49:36 -0500 (Tue, 20 Sep 2011) | 28 lines Merged revisions 337118 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337118 | mjordan | 2011-09-20 17:38:54 -0500 (Tue, 20 Sep 2011) | 21 lines Fix for incorrect voicemail duration in external notifications This patch fixes an issue where the voicemail duration was being reported with a duration significantly less than the actual sound file duration. Voicemails that contained mostly silence were reporting the duration of only the sound in the file, as opposed to the duration of the file with the silence. This patch fixes this by having two durations reported in the __ast_play_and_record family of functions - the sound_duration and the actual duration of the file. The sound_duration, which is optional, now reports the duration of the sound in the file, while the actual full duration of the file is reported in the duration parameter. This allows the voicemail applications to use the sound_duration for minimum duration checking, while reporting the full duration to external parties if the voicemail is kept. (issue ASTERISK-2234) (closes issue ASTERISK-16981) Reported by: Mary Ciuciu, Byron Clark, Brad House, Karsten Wemheuer, KevinH Tested by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1443 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@337124 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-20 23:02:25 +00:00
int ast_play_and_record(struct ast_channel *chan, const char *playfile, const char *recordfile, int maxtime_sec, const char *fmt, int *duration, int *sound_duration, int silencethreshold, int maxsilence_ms, const char *path);
Merged revisions 329528 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r329528 | jrose | 2011-07-26 08:52:34 -0500 (Tue, 26 Jul 2011) | 24 lines Merged revisions 329527 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r329527 | jrose | 2011-07-26 08:25:35 -0500 (Tue, 26 Jul 2011) | 17 lines Fixes some voicemail forwarding behavior based around prepend mode. Formerly, prepend forwarding would have the user record a message with no useful prompt and an expectation for the user to push a button on the phone when finished recording. If a length of silence was detected instead, the recording would be canceled and the user would re-enter the voicemail forwarding menu. Subsequent time-outs in prepend recording would also bug out in the sense that they would write over the original message and get sent to the recipient regardless of whether they timed out or were accepted. This patch fixes this issue and adds a prompt which will be played after a timeout informing the user that they needed to press a button. Currently, the sound files that we have are somewhat inadquate for this, so after the call we simply have Allison say "Please try again. Then press pound." which actually relies on two separate sound files. Just one would be more appropriate. reporter: Vlad Povorozniuc Review: https://reviewboard.asterisk.org/r/1327/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@329530 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-26 14:17:13 +00:00
/*!
* \brief Record a file based on input frm a channel. Recording is performed in 'prepend' mode which works a little differently from normal recordings
* This function will not play a success message due to post-recording control in the application this was added for.
*
* \param chan the channel being recorded
* \param playfile Filename of sound to play before recording begins
* \param recordfile Filename to save the recording
* \param maxtime_sec Longest possible message length in seconds
* \param fmt string containing all formats to be recorded delimited by '|'
* \param duration pointer to integer for storing length of the recording
Merged revisions 337120 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r337120 | mjordan | 2011-09-20 17:49:36 -0500 (Tue, 20 Sep 2011) | 28 lines Merged revisions 337118 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337118 | mjordan | 2011-09-20 17:38:54 -0500 (Tue, 20 Sep 2011) | 21 lines Fix for incorrect voicemail duration in external notifications This patch fixes an issue where the voicemail duration was being reported with a duration significantly less than the actual sound file duration. Voicemails that contained mostly silence were reporting the duration of only the sound in the file, as opposed to the duration of the file with the silence. This patch fixes this by having two durations reported in the __ast_play_and_record family of functions - the sound_duration and the actual duration of the file. The sound_duration, which is optional, now reports the duration of the sound in the file, while the actual full duration of the file is reported in the duration parameter. This allows the voicemail applications to use the sound_duration for minimum duration checking, while reporting the full duration to external parties if the voicemail is kept. (issue ASTERISK-2234) (closes issue ASTERISK-16981) Reported by: Mary Ciuciu, Byron Clark, Brad House, Karsten Wemheuer, KevinH Tested by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1443 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@337124 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-20 23:02:25 +00:00
* \param sound_duration pointer to integer for storing length of the recording minus all silence
Merged revisions 329528 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r329528 | jrose | 2011-07-26 08:52:34 -0500 (Tue, 26 Jul 2011) | 24 lines Merged revisions 329527 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r329527 | jrose | 2011-07-26 08:25:35 -0500 (Tue, 26 Jul 2011) | 17 lines Fixes some voicemail forwarding behavior based around prepend mode. Formerly, prepend forwarding would have the user record a message with no useful prompt and an expectation for the user to push a button on the phone when finished recording. If a length of silence was detected instead, the recording would be canceled and the user would re-enter the voicemail forwarding menu. Subsequent time-outs in prepend recording would also bug out in the sense that they would write over the original message and get sent to the recipient regardless of whether they timed out or were accepted. This patch fixes this issue and adds a prompt which will be played after a timeout informing the user that they needed to press a button. Currently, the sound files that we have are somewhat inadquate for this, so after the call we simply have Allison say "Please try again. Then press pound." which actually relies on two separate sound files. Just one would be more appropriate. reporter: Vlad Povorozniuc Review: https://reviewboard.asterisk.org/r/1327/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@329530 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-26 14:17:13 +00:00
* \param beep whether to play a beep to prompt the recording
* \param silencethreshold tolerance of noise levels that can be considered silence for the purpose of silence timeout, -1 for default
* \param maxsilence_ms length of time in milliseconds which will trigger a timeout from silence, -1 for default.
*
* \retval -1 failure or hangup
* \retval 'S' Recording ended from silence timeout
* \retval 't' Recording either exceeded maximum duration or the call was ended via DTMF
*/
Merged revisions 337120 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r337120 | mjordan | 2011-09-20 17:49:36 -0500 (Tue, 20 Sep 2011) | 28 lines Merged revisions 337118 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337118 | mjordan | 2011-09-20 17:38:54 -0500 (Tue, 20 Sep 2011) | 21 lines Fix for incorrect voicemail duration in external notifications This patch fixes an issue where the voicemail duration was being reported with a duration significantly less than the actual sound file duration. Voicemails that contained mostly silence were reporting the duration of only the sound in the file, as opposed to the duration of the file with the silence. This patch fixes this by having two durations reported in the __ast_play_and_record family of functions - the sound_duration and the actual duration of the file. The sound_duration, which is optional, now reports the duration of the sound in the file, while the actual full duration of the file is reported in the duration parameter. This allows the voicemail applications to use the sound_duration for minimum duration checking, while reporting the full duration to external parties if the voicemail is kept. (issue ASTERISK-2234) (closes issue ASTERISK-16981) Reported by: Mary Ciuciu, Byron Clark, Brad House, Karsten Wemheuer, KevinH Tested by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1443 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@337124 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-20 23:02:25 +00:00
int ast_play_and_prepend(struct ast_channel *chan, char *playfile, char *recordfile, int maxtime_sec, char *fmt, int *duration, int *sound_duration, int beep, int silencethreshold, int maxsilence_ms);
enum ast_getdata_result {
AST_GETDATA_FAILED = -1,
AST_GETDATA_COMPLETE = 0,
AST_GETDATA_TIMEOUT = 1,
AST_GETDATA_INTERRUPTED = 2,
/*! indicates a user terminated empty string rather than an empty string resulting
* from a timeout or other factors */
AST_GETDATA_EMPTY_END_TERMINATED = 3,
};
enum AST_LOCK_RESULT {
AST_LOCK_SUCCESS = 0,
AST_LOCK_TIMEOUT = -1,
AST_LOCK_PATH_NOT_FOUND = -2,
AST_LOCK_FAILURE = -3,
};
/*! \brief Type of locking to use in ast_lock_path / ast_unlock_path */
enum AST_LOCK_TYPE {
AST_LOCK_TYPE_LOCKFILE = 0,
AST_LOCK_TYPE_FLOCK = 1,
};
/*!
* \brief Set the type of locks used by ast_lock_path()
* \param type the locking type to use
*/
void ast_set_lock_type(enum AST_LOCK_TYPE type);
/*!
* \brief Lock a filesystem path.
* \param path the path to be locked
* \return one of \ref AST_LOCK_RESULT values
*/
enum AST_LOCK_RESULT ast_lock_path(const char *path);
/*! \brief Unlock a path */
int ast_unlock_path(const char *path);
/*! \brief Read a file into asterisk*/
char *ast_read_textfile(const char *file);
struct ast_group_info;
/*! \brief Split a group string into group and category, returning a default category if none is provided. */
int ast_app_group_split_group(const char *data, char *group, int group_max, char *category, int category_max);
/*! \brief Set the group for a channel, splitting the provided data into group and category, if specified. */
int ast_app_group_set_channel(struct ast_channel *chan, const char *data);
/*! \brief Get the current channel count of the specified group and category. */
int ast_app_group_get_count(const char *group, const char *category);
/*! \brief Get the current channel count of all groups that match the specified pattern and category. */
int ast_app_group_match_get_count(const char *groupmatch, const char *category);
/*! \brief Discard all group counting for a channel */
int ast_app_group_discard(struct ast_channel *chan);
/*! \brief Update all group counting for a channel to a new one */
int ast_app_group_update(struct ast_channel *oldchan, struct ast_channel *newchan);
/*! \brief Write Lock the group count list */
int ast_app_group_list_wrlock(void);
/*! \brief Read Lock the group count list */
int ast_app_group_list_rdlock(void);
/*! \brief Get the head of the group count list */
struct ast_group_info *ast_app_group_list_head(void);
/*! \brief Unlock the group count list */
int ast_app_group_list_unlock(void);
/*!
\brief Define an application argument
\param name The name of the argument
*/
#define AST_APP_ARG(name) char *name
/*!
\brief Declare a structure to hold an application's arguments.
\param name The name of the structure
\param arglist The list of arguments, defined using AST_APP_ARG
This macro declares a structure intended to be used in a call
to ast_app_separate_args(). The structure includes all the
arguments specified, plus an argv array that overlays them and an
argc argument counter. The arguments must be declared using AST_APP_ARG,
and they will all be character pointers (strings).
\note The structure is <b>not</b> initialized, as the call to
ast_app_separate_args() will perform that function before parsing
the arguments.
*/
#define AST_DECLARE_APP_ARGS(name, arglist) AST_DEFINE_APP_ARGS_TYPE(, arglist) name = { 0, }
/*!
\brief Define a structure type to hold an application's arguments.
\param type The name of the structure type
\param arglist The list of arguments, defined using AST_APP_ARG
This macro defines a structure type intended to be used in a call
to ast_app_separate_args(). The structure includes all the
arguments specified, plus an argv array that overlays them and an
argc argument counter. The arguments must be declared using AST_APP_ARG,
and they will all be character pointers (strings).
\note This defines a structure type, but does not declare an instance
of the structure. That must be done separately.
*/
#define AST_DEFINE_APP_ARGS_TYPE(type, arglist) \
struct type { \
unsigned int argc; \
char *argv[0]; \
arglist \
}
/*!
\brief Performs the 'standard' argument separation process for an application.
\param args An argument structure defined using AST_DECLARE_APP_ARGS
\param parse A modifiable buffer containing the input to be parsed
This function will separate the input string using the standard argument
separator character ',' and fill in the provided structure, including
the argc argument counter field.
*/
#define AST_STANDARD_APP_ARGS(args, parse) \
args.argc = __ast_app_separate_args(parse, ',', 1, args.argv, ((sizeof(args) - offsetof(typeof(args), argv)) / sizeof(args.argv[0])))
#define AST_STANDARD_RAW_ARGS(args, parse) \
args.argc = __ast_app_separate_args(parse, ',', 0, args.argv, ((sizeof(args) - offsetof(typeof(args), argv)) / sizeof(args.argv[0])))
/*!
\brief Performs the 'nonstandard' argument separation process for an application.
\param args An argument structure defined using AST_DECLARE_APP_ARGS
\param parse A modifiable buffer containing the input to be parsed
\param sep A nonstandard separator character
This function will separate the input string using the nonstandard argument
separator character and fill in the provided structure, including
the argc argument counter field.
*/
#define AST_NONSTANDARD_APP_ARGS(args, parse, sep) \
args.argc = __ast_app_separate_args(parse, sep, 1, args.argv, ((sizeof(args) - offsetof(typeof(args), argv)) / sizeof(args.argv[0])))
#define AST_NONSTANDARD_RAW_ARGS(args, parse, sep) \
args.argc = __ast_app_separate_args(parse, sep, 0, args.argv, ((sizeof(args) - offsetof(typeof(args), argv)) / sizeof(args.argv[0])))
/*!
\brief Separate a string into arguments in an array
\param buf The string to be parsed (this must be a writable copy, as it will be modified)
\param delim The character to be used to delimit arguments
\param remove_chars Remove backslashes and quote characters, while parsing
\param array An array of 'char *' to be filled in with pointers to the found arguments
\param arraylen The number of elements in the array (i.e. the number of arguments you will accept)
Note: if there are more arguments in the string than the array will hold, the last element of
the array will contain the remaining arguments, not separated.
The array will be completely zeroed by this function before it populates any entries.
\return The number of arguments found, or zero if the function arguments are not valid.
*/
unsigned int __ast_app_separate_args(char *buf, char delim, int remove_chars, char **array, int arraylen);
#define ast_app_separate_args(a,b,c,d) __ast_app_separate_args(a,b,1,c,d)
/*!
\brief A structure to hold the description of an application 'option'.
Application 'options' are single-character flags that can be supplied
to the application to affect its behavior; they can also optionally
accept arguments enclosed in parenthesis.
These structures are used by the ast_app_parse_options function, uses
this data to fill in a flags structure (to indicate which options were
supplied) and array of argument pointers (for those options that had
arguments supplied).
*/
struct ast_app_option {
/*! \brief The flag bit that represents this option. */
uint64_t flag;
/*! \brief The index of the entry in the arguments array
that should be used for this option's argument. */
unsigned int arg_index;
};
#define BEGIN_OPTIONS {
#define END_OPTIONS }
/*!
\brief Declares an array of options for an application.
\param holder The name of the array to be created
\param options The actual options to be placed into the array
\sa ast_app_parse_options
This macro declares a 'static const' array of \c struct \c ast_option
elements to hold the list of available options for an application.
Each option must be declared using either the AST_APP_OPTION()
or AST_APP_OPTION_ARG() macros.
Example usage:
\code
enum my_app_option_flags {
OPT_JUMP = (1 << 0),
OPT_BLAH = (1 << 1),
OPT_BLORT = (1 << 2),
};
enum my_app_option_args {
OPT_ARG_BLAH = 0,
OPT_ARG_BLORT,
!! this entry tells how many possible arguments there are,
and must be the last entry in the list
OPT_ARG_ARRAY_SIZE,
};
AST_APP_OPTIONS(my_app_options, {
AST_APP_OPTION('j', OPT_JUMP),
AST_APP_OPTION_ARG('b', OPT_BLAH, OPT_ARG_BLAH),
AST_APP_OPTION_BLORT('B', OPT_BLORT, OPT_ARG_BLORT),
});
static int my_app_exec(struct ast_channel *chan, void *data)
{
char *options;
struct ast_flags opts = { 0, };
char *opt_args[OPT_ARG_ARRAY_SIZE];
... do any argument parsing here ...
if (ast_app_parse_options(my_app_options, &opts, opt_args, options)) {
return -1;
}
}
\endcode
*/
#define AST_APP_OPTIONS(holder, options...) \
static const struct ast_app_option holder[128] = options
/*!
\brief Declares an application option that does not accept an argument.
\param option The single character representing the option
\param flagno The flag index to be set if this option is present
\sa AST_APP_OPTIONS, ast_app_parse_options
*/
#define AST_APP_OPTION(option, flagno) \
[option] = { .flag = flagno }
/*!
\brief Declares an application option that accepts an argument.
\param option The single character representing the option
\param flagno The flag index to be set if this option is present
\param argno The index into the argument array where the argument should
be placed
\sa AST_APP_OPTIONS, ast_app_parse_options
*/
#define AST_APP_OPTION_ARG(option, flagno, argno) \
[option] = { .flag = flagno, .arg_index = argno + 1 }
/*!
\brief Parses a string containing application options and sets flags/arguments.
\param options The array of possible options declared with AST_APP_OPTIONS
\param flags The flag structure to have option flags set
\param args The array of argument pointers to hold arguments found
\param optstr The string containing the options to be parsed
\return zero for success, non-zero if an error occurs
\sa AST_APP_OPTIONS
*/
int ast_app_parse_options(const struct ast_app_option *options, struct ast_flags *flags, char **args, char *optstr);
/*!
\brief Parses a string containing application options and sets flags/arguments.
\param options The array of possible options declared with AST_APP_OPTIONS
\param flags The 64-bit flag structure to have option flags set
\param args The array of argument pointers to hold arguments found
\param optstr The string containing the options to be parsed
\return zero for success, non-zero if an error occurs
\sa AST_APP_OPTIONS
*/
int ast_app_parse_options64(const struct ast_app_option *options, struct ast_flags64 *flags, char **args, char *optstr);
/*! \brief Given a list of options array, return an option string based on passed flags
\param options The array of possible options declared with AST_APP_OPTIONS
\param flags The flags of the options that you wish to populate the buffer with
\param buf The buffer to fill with the string of options
\param len The maximum length of buf
*/
void ast_app_options2str64(const struct ast_app_option *options, struct ast_flags64 *flags, char *buf, size_t len);
/*! \brief Present a dialtone and collect a certain length extension.
\return Returns 1 on valid extension entered, -1 on hangup, or 0 on invalid extension.
\note Note that if 'collect' holds digits already, new digits will be appended, so be sure it's initialized properly */
int ast_app_dtget(struct ast_channel *chan, const char *context, char *collect, size_t size, int maxlen, int timeout);
/*! \brief Allow to record message and have a review option */
int ast_record_review(struct ast_channel *chan, const char *playfile, const char *recordfile, int maxtime, const char *fmt, int *duration, const char *path);
/*!\brief Decode an encoded control or extended ASCII character
* \param[in] stream String to decode
* \param[out] result Decoded character
* \param[out] consumed Number of characters used in stream to encode the character
* \retval -1 Stream is of zero length
* \retval 0 Success
*/
int ast_get_encoded_char(const char *stream, char *result, size_t *consumed);
/*!\brief Decode a stream of encoded control or extended ASCII characters
* \param[in] stream Encoded string
* \param[out] result Decoded string
* \param[in] result_len Maximum size of the result buffer
* \return A pointer to the result string
*/
char *ast_get_encoded_str(const char *stream, char *result, size_t result_len);
/*! \brief Decode a stream of encoded control or extended ASCII characters */
int ast_str_get_encoded_str(struct ast_str **str, int maxlen, const char *stream);
/*!
* \brief Common routine for child processes, to close all fds prior to exec(2)
* \param[in] n starting file descriptor number for closing all higher file descriptors
* \since 1.6.1
*/
void ast_close_fds_above_n(int n);
/*!
* \brief Common routine to safely fork without a chance of a signal handler firing badly in the child
* \param[in] stop_reaper flag to determine if sigchld handler is replaced or not
* \since 1.6.1
*/
int ast_safe_fork(int stop_reaper);
/*!
* \brief Common routine to cleanup after fork'ed process is complete (if reaping was stopped)
* \since 1.6.1
*/
void ast_safe_fork_cleanup(void);
/*!
* \brief Common routine to parse time lengths, with optional time unit specifier
* \param[in] timestr String to parse
* \param[in] defunit Default unit type
* \param[out] result Resulting value, specified in milliseconds
* \retval 0 Success
* \retval -1 Failure
* \since 1.8
*/
int ast_app_parse_timelen(const char *timestr, int *result, enum ast_timelen defunit);
#if defined(__cplusplus) || defined(c_plusplus)
}
#endif
#endif /* _ASTERISK_APP_H */