Doxygen Updates - janitor work

Doxygen updates including mistakes, misspellings, missing parameters, updates for Doxygen style.  Some missing txt file links are removed but their content or essense will be included in some later updates.  A majority of the txt files were removed in the 1.6 era but never noted. The HR and EXTREF are simple changes that make the documentation more compatable with more versions of Doxygen.

Further updates coming.

(issue ASTERISK-20259)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373330 65c4cc65-6c06-0410-ace0-fbb531ad65f3
changes/78/78/1
Andrew Latham 13 years ago
parent 448098ca9f
commit 6f61cb50c5

@ -621,7 +621,7 @@ static void send_leave_event(struct ast_channel *chan, const char *conf_name)
* \brief Announce number of users in the conference bridge to the caller
*
* \param conference_bridge Conference bridge to peek at
* \param (OPTIONAL) conference_bridge_user Caller
* \param conference_bridge_user Optional Caller
*
* \note if caller is NULL, the announcment will be sent to all participants in the conference.
* \return Returns 0 on success, -1 if the user hung up
@ -1258,7 +1258,7 @@ static int play_sound_file(struct conference_bridge *conference_bridge, const ch
* \brief Play number into the conference bridge
*
* \param conference_bridge The conference bridge to say the number into
* \param number to say
* \param say_number number to say
*
* \retval 0 success
* \retval -1 failure

@ -22,7 +22,7 @@
*
* \author Mark Spencer <markster@digium.com>
*
* \extref ICES - http://www.icecast.org/ices.php
* ICES - http://www.icecast.org/ices.php
*
* \ingroup applications
*/

@ -1217,6 +1217,7 @@ static int user_max_cmp(void *obj, void *arg, int flags)
* \param dynamic Mark the newly created conference as dynamic
* \param refcount How many references to mark on the conference
* \param chan The asterisk channel
* \param test
*
* \return A pointer to the conference struct, or NULL if it wasn't found and
* make or dynamic were not set.

@ -178,7 +178,7 @@ static void *skel_level_alloc(const char *cat);
* internally by the Config Options code to check if an level has already been added to the
* container that will be swapped for the live container on a successul reload.
*
* \param container A non-active container to search for a level
* \param tmp_container A non-active container to search for a level
* \param category The category associated with the level to check for
* \retval non-NULL The level from the container
* \retval NULL The level does not exist in the container
@ -648,6 +648,14 @@ static int unload_module(void)
return ast_unregister_application(app);
}
/*!
* \brief Load the module
*
* \par The configuration file
*
* The application app_skel uses a configuration file.
* \verbinclude app_skel.conf.sample
*/
static int load_module(void)
{
if (aco_info_init(&cfg_info)) {

@ -25,11 +25,11 @@
* \brief PostgreSQL CDR logger
*
* \author Matthew D. Hardeman <mhardemn@papersoft.com>
* \extref PostgreSQL http://www.postgresql.org/
* PostgreSQL http://www.postgresql.org/
*
* See also
* \arg \ref Config_cdr
* \extref PostgreSQL http://www.postgresql.org/
* PostgreSQL http://www.postgresql.org/
* \ingroup cdr_drivers
*/

@ -21,7 +21,8 @@
* \brief RADIUS CDR Support
*
* \author Philippe Sultan
* \extref The Radius Client Library - http://developer.berlios.de/projects/radiusclient-ng/
* The Radius Client Library
* http://developer.berlios.de/projects/radiusclient-ng/
*
* \arg See also \ref AstCDR
* \ingroup cdr_drivers
@ -187,7 +188,8 @@ static int build_radius_record(VALUE_PAIR **tosend, struct ast_cdr *cdr)
}
/* Setting Acct-Session-Id & User-Name attributes for proper generation
of Acct-Unique-Session-Id on server side */
* of Acct-Unique-Session-Id on server side
*/
/* Channel */
if (!rc_avpair_add(rh, tosend, PW_USER_NAME, &cdr->channel, strlen(cdr->channel), 0))
return -1;

@ -22,7 +22,7 @@
* \brief Store CDR records in a SQLite database.
*
* \author Holger Schurig <hs4233@mail.mn-solutions.de>
* \extref SQLite http://www.sqlite.org/
* SQLite http://www.sqlite.org/
*
* See also
* \arg \ref Config_cdr

@ -29,11 +29,11 @@
* \brief PostgreSQL CEL logger
*
* \author Steve Murphy <murf@digium.com>
* \extref PostgreSQL http://www.postgresql.org/
* PostgreSQL http://www.postgresql.org/
*
* See also
* \arg \ref Config_cel
* \extref PostgreSQL http://www.postgresql.org/
* PostgreSQL http://www.postgresql.org/
* \ingroup cel_drivers
*/

@ -20,7 +20,7 @@
*
* \brief RADIUS CEL Support
* \author Philippe Sultan
* \extref The Radius Client Library - http://developer.berlios.de/projects/radiusclient-ng/
* The Radius Client Library - http://developer.berlios.de/projects/radiusclient-ng/
*
* \arg See also \ref AstCEL
* \ingroup cel_drivers

@ -29,7 +29,7 @@
*
* \ingroup channel_drivers
*
* \extref Portaudio http://www.portaudio.com/
* Portaudio http://www.portaudio.com/
*
* To install portaudio v19 from svn, check it out using the following command:
* - svn co https://www.portaudio.com/repos/portaudio/branches/v19-devel

@ -29,7 +29,7 @@
*
* \par See also
* \arg Config_h323
* \extref OpenH323 http://www.voxgratia.org/
* OpenH323 http://www.voxgratia.org/
*
* \ingroup channel_drivers
*/

@ -22,7 +22,7 @@
*
* \brief Jingle Channel Driver
*
* \extref Iksemel http://iksemel.jabberstudio.org/
* Iksemel http://iksemel.jabberstudio.org/
*
* \ingroup channel_drivers
*/

@ -24,7 +24,7 @@
*
* \author Christian Richter <crich@beronet.com>
*
* \extref MISDN http://www.misdn.org/
* MISDN http://www.misdn.org/
*
* \ingroup channel_drivers
*/

@ -22,7 +22,7 @@
*
* \brief Motif Jingle Channel Driver
*
* \extref Iksemel http://iksemel.jabberstudio.org/
* Iksemel http://iksemel.jabberstudio.org/
*
* \ingroup channel_drivers
*/

@ -65,7 +65,8 @@ int sdp_crypto_process(struct sdp_crypto *p, const char *attr, struct ast_rtp_in
* \details
* The offer is stored on the sdp_crypto struct in a_crypto
*
* \param A valid sdp_crypto struct
* \param p A valid sdp_crypto struct
* \param taglen Length
*
* \retval 0 success
* \retval nonzero failure

@ -1523,9 +1523,9 @@ AST_TEST_DEFINE(parse_contact_header_test)
* item is found that is not supported, it is copied to the unsupported
* out buffer.
*
* \param option list
* \param options list
* \param unsupported out buffer (optional)
* \param unsupported out buffer length (optional)
* \param unsupported_len out buffer length (optional)
*/
unsigned int parse_sip_options(const char *options, char *unsupported, size_t unsupported_len)
{

@ -26,7 +26,7 @@
*
* \ingroup codecs
*
* \extref The Speex library - http://www.speex.org
* The Speex library - http://www.speex.org
*
*/
@ -44,7 +44,8 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include <speex/speex.h>
/* We require a post 1.1.8 version of Speex to enable preprocessing
and better type handling */
* and better type handling
*/
#ifdef _SPEEX_TYPES_H
#include <speex/speex_preprocess.h>
#endif

@ -26,7 +26,7 @@
*
* \ingroup functions
*
* \extref The Speex 1.2 library - http://www.speex.org
* The Speex 1.2 library - http://www.speex.org
* \note Requires the 1.2 version of the Speex library (which might not be what you find in Linux packages)
*/

@ -189,7 +189,7 @@ enum ast_acl_sense ast_apply_ha(const struct ast_ha *ha, const struct ast_sockad
* contained in the acl container. It will deny if any of the ast_ha lists
* fail, and it will pass only if all of the rules pass.
*
* \param acl The head of the list of ACLs to evaluate
* \param acl_list The head of the list of ACLs to evaluate
* \param addr An ast_sockaddr whose address is considered when matching rules
* \param purpose Context for which the ACL is being applied - Establishes purpose of a notice when rejected
*

@ -331,6 +331,9 @@ typedef void (ast_vm_msg_play_cb)(struct ast_channel *chan, const char *playfile
/*!
* \brief Set voicemail function callbacks
*
* \param copy_recording_to_vm_func, vm_index_to_foldername, vm_mailbox_snapshot_create
* \param vm_mailbox_snapshot_destroy, vm_msg_move, vm_msg_remove, vm_msg_forward, vm_msg_play
* \param[in] has_voicemail_func set function pointer
* \param[in] inboxcount_func set function pointer
* \param[in] inboxcount2_func set function pointer
@ -411,7 +414,10 @@ 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[in] mailbox Mailbox specification in the format
* /code
* mbox[\@context][&mbox2[\@context2]][...]
* /code
* \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
@ -495,7 +501,7 @@ struct ast_vm_mailbox_snapshot *ast_vm_mailbox_snapshot_destroy(struct ast_vm_ma
* \param context The voicemail context for the mailbox
* \param num_msgs The number of messages to move
* \param oldfolder The folder from where messages should be moved
* \param old_msg_nums The message IDs of the messages to move
* \param old_msg_ids The message IDs of the messages to move
* \param newfolder The folder to which messages should be moved
* new folder. This array must be num_msgs sized.
*
@ -557,10 +563,12 @@ int ast_vm_msg_forward(const char *from_mailbox,
/*!
* \brief Play a voicemail msg back on a channel.
*
* \param chan
* \param mailbox msg is in.
* \param context of mailbox.
* \param voicemail folder to look in.
* \param message number in the voicemailbox to playback to the channel.
* \param folder voicemail folder to look in.
* \param msg_num message number in the voicemailbox to playback to the channel.
* \param cb
*
* \retval 0 success
* \retval -1 failure
@ -643,7 +651,7 @@ int ast_control_streamfile(struct ast_channel *chan, const char *file, const cha
* \param chan
* \param file filename
* \param fwd, rev, stop, pause, restart, skipms, offsetms
* \param waitstream callback to invoke when fastforward or rewind occurrs.
* \param cb 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

@ -197,7 +197,7 @@ int ast_bridge_features_unregister(enum ast_bridge_builtin_feature feature);
* \param dtmf DTMF string to be activated upon
* \param callback Function to execute upon activation
* \param hook_pvt Unique data
* \param Optional destructor callback for hook_pvt data
* \param destructor Optional destructor callback for hook_pvt data
*
* \retval 0 on success
* \retval -1 on failure
@ -226,9 +226,9 @@ int ast_bridge_features_hook(struct ast_bridge_features *features,
/*! \brief Set a callback on the features structure to receive talking notifications on.
*
* \param features Bridge features structure
* \param talker_cb, Callback function to execute when talking events occur in the bridge core.
* \param talker_cb Callback function to execute when talking events occur in the bridge core.
* \param pvt_data Optional unique data that will be passed with the talking events.
* \param Optional destructor callback for pvt data.
* \param talker_destructor Optional destructor callback for pvt data.
*
* \retval 0, success
* \retval -1, failure

@ -26,7 +26,7 @@
\author Russell Bryant <russell@digium.com>
\AsteriskTrunkWarning
<hr/>
<hr>
\section ArchTOC Table of Contents
@ -64,7 +64,7 @@
-# \ref ArchThreadDebugging
-# \ref ArchMemoryDebugging
<hr/>
<hr>
\section ArchIntro Introduction
@ -295,7 +295,7 @@ many events to one call. The CEL modules look very similar to CDR modules.
CEL modules typically live in the <code>cel/</code> directory in the source tree.
For a list of CEL handlers, see \ref cel_drivers.
For a list of CEL handlers, see cel_drivers.
For additional information about the CEL API, see
<code>include/asterisk/cel.h</code>.
@ -407,7 +407,7 @@ When an inbound channel executes an application such as <code>Dial()</code>, an
outbound channel is created and bridged to the inbound channel once it answers.
Dialplan applications always execute in the context of a channel thread. Dialplan
functions \i almost always do, as well. However, it is possible to read and write
functions almost always do, as well. However, it is possible to read and write
dialplan functions from an asynchronous interface such as the Asterisk CLI or the
manager interface (AMI). However, it is still always the channel thread that is
the owner of the ast_channel data structure.
@ -710,7 +710,7 @@ The implementation of this memory debugging system can be found in
<code>main/astmm.c</code>.
<hr/>
<hr>
Return to the \ref ArchTOC
*/

@ -23,7 +23,7 @@
*
* \AsteriskTrunkWarning
*
* <hr/>
* <hr>
*
* \section Introduction Introduction
* This document will instruct you how to setup a local git mirror of the
@ -32,7 +32,7 @@
* Why would you want that? for starters, it's a fast repository browser
* and works well even when offline. More on why and why not at 'Pros and Cons'
* in the end of this document.
* <hr/>
* <hr>
*
* \section Setup Setup
*
@ -120,7 +120,7 @@
echo 'exclude res_snmp' >build_tools/conf
\endverbatim
*
* <hr/>
* <hr>
*
* \section Update Update
* The main Asterisk repository tends to get new commits occasionally. I
@ -133,7 +133,7 @@
\endverbatim
*
* Next, get all updates.
* <hr/>
* <hr>
*
* \section Usage Usage
*
@ -183,7 +183,7 @@
git svn rebase --fetch-all
\endverbatim
*
* <hr/>
* <hr>
*
* \section ProsAndCons Pros and Cons
*
@ -233,5 +233,5 @@
* to test that. Even if it will, it will require an extra step of manual
* editing.
*
* <hr/>
* <hr>
*/

@ -23,23 +23,23 @@
*
* \AsteriskTrunkWarning
*
* <hr/>
* <hr>
*
* \section CommitMsgFormatting Commit Message Formatting
*
* The following illustrates the basic outline for commit messages:
*
\verbatim
<One-liner summary of changes>
<Empty Line>
<Verbose description of the changes>
<Empty Line>
<Special Tags>
\endverbatim
* \verbatim
* <One-liner summary of changes>
*
* <Empty Line>
*
* <Verbose description of the changes>
*
* <Empty Line>
*
* <Special Tags>
* \endverbatim
*
* Some commit history viewers treat the first line of commit messages as the
* summary for the commit. So, an effort should be made to format our commit
@ -52,7 +52,7 @@
* \note For trivial commits, such as "fix the build", or "fix spelling mistake",
* the verbose description may not be necessary.
*
* <hr/>
* <hr>
*
* \section CommitMsgTags Special Tags for Commit Messages
*
@ -77,12 +77,12 @@
*
* Here is an example of what the template will generate for you:
*
\verbatim
(closes issue #1234)
Reported by: SomeGuy
Patches:
fix_bug_1234.diff uploaded by SomeDeveloper (license 5678)
\endverbatim
* \verbatim
* (closes issue #1234)
* Reported by: SomeGuy
* Patches:
* fix_bug_1234.diff uploaded by SomeDeveloper (license 5678)
* \endverbatim
*
* If the patch being committed was written by the person doing the commit,
* and is not available to reference as an upload to the issue, there is no
@ -97,7 +97,7 @@
*
* \note The trailing slash in the review URL is required.
*
* <hr/>
* <hr>
*
* \section CommitMsgSvnmerge Commit Messages with svnmerge
*

@ -23,7 +23,7 @@
*
* \AsteriskTrunkWarning
*
* <hr/>
* <hr>
* \section WorkflowDescription Description of the Issue Tracker Workflow
*
* (This document is most beneficial for Asterisk bug marshals, however it is good

@ -133,7 +133,7 @@
*
* \arg \ref ReleaseStatus
*
* <hr/>
* <hr>
*
* \section commitmonitoring Commit Monitoring
*
@ -141,7 +141,7 @@
* <a href="http://lists.digium.com/">http://lists.digium.com</a>. The Digium
* mailing list server hosts a %number of mailing lists for commits.
*
* <hr/>
* <hr>
*
* \section ast10policy Asterisk 1.0
*
@ -154,7 +154,7 @@
*
* No commits should be made to the Asterisk 1.0 branch.
*
* <hr/>
* <hr>
*
* \section ast12policy Asterisk 1.2
*
@ -181,7 +181,7 @@
* - 1.2.X.Y - a release that contains fixes to the security patches released in
* version 1.2.X
*
* <hr/>
* <hr>
*
* \section ast14policy Asterisk 1.4
*
@ -208,7 +208,7 @@
* - 1.4.X.Y - a release that contains very few changes on top of 1.4.X. This
* may be for a security patch, or for a regression introduced in 1.4.X.
*
* <hr/>
* <hr>
*
* \section ast16policy Asterisk 1.6
*
@ -244,7 +244,7 @@
* - 1.6.X.Y - a release that contains fixes for bugs or security issues identified
* in the 1.6.X release series.
*
* <hr/>
* <hr>
*
* \section asttrunk Asterisk Trunk
*
@ -269,7 +269,7 @@
* tested and reviewed such that there is reasonable belief that the code
* is ready to go.
*
* <hr/>
* <hr>
*
* \section astteam Asterisk Team Branches
*

@ -23,7 +23,7 @@
*
* \AsteriskTrunkWarning
*
* <hr/>
* <hr>
*
* \section ReviewboardGuidelines Usage Guidelines
*
@ -52,7 +52,7 @@
* testing and should not have blatant coding guidelines violations. Lack of
* these things is careless and shows disrespect for those reviewing your code.
*
* <hr/>
* <hr>
*
* \section ReviewboardPosting Posting Code to Reviewboard
*
@ -61,25 +61,25 @@
* The easiest way to post a patch to reviewboard is by using the
* post-review tool. We have post-review in our repotools svn repository.
*
\verbatim
$ svn co http://svn.digium.com/svn/repotools
\endverbatim
* \verbatim
* $ svn co http://svn.digium.com/svn/repotools
* \endverbatim
*
* Essentially, post-review is a script that will take the output of "svn
* diff" and create a review request out of it for you. So, once you have
* a working copy with the changes you expect in the output of "svn diff",
* you just run the following command:
*
\verbatim
$ post-review
\endverbatim
* \verbatim
* $ post-review
* \endverbatim
*
* If it complains about not knowing which reviewboard server to use, add
* the server option:
*
\verbatim
$ post-review --server=https://reviewboard.asterisk.org
\endverbatim
* \verbatim
* $ post-review --server=https://reviewboard.asterisk.org
* \endverbatim
*
* \subsection postreviewnewfiles Dealing with New Files
*
@ -90,15 +90,15 @@
*
* You would start by getting your changes applied to a trunk working copy:
*
\verbatim
$ cd .../trunk
\endverbatim
* \verbatim
* $ cd .../trunk
* \endverbatim
*
* Then, apply the changes from your branch:
*
\verbatim
$ svn merge .../trunk .../team/group/my_new_code
\endverbatim
* \verbatim
* $ svn merge .../trunk .../team/group/my_new_code
* \endverbatim
*
* Now, the code is merged into your working copy. However, for a new
* file, subversion treats it as a copy of existing content and not new
@ -106,10 +106,10 @@
* it to show up in the diff, use the following commands so svn treats it
* as new content and publishes it in the diff:
*
\verbatim
$ svn revert my_new_file.c
$ svn add my_new_file.c
\endverbatim
* \verbatim
* $ svn revert my_new_file.c
* $ svn add my_new_file.c
* \endverbatim
*
* Now, it should work, and you can run "post-review" as usual.
*
@ -121,7 +121,7 @@
* Apply the current version of the diff to a working copy as described above,
* and then run the following command:
*
\verbatim
$ post-review -r <review request number>
\endverbatim
* \verbatim
* $ post-review -r <review request number>
* \endverbatim
*/

@ -68,7 +68,7 @@
* \arg \b Main: Asterisk Developer's website http://www.asterisk.org/developers/
* \arg \b Bugs: The Issue Tracker https://issues.asterisk.org
* \arg \b Lists: List Server http://lists.digium.com
* \arg \b Wiki: The Asterisk Wiki http://www.voip-info.org
* \arg \b Wiki: The Asterisk Wiki http://wiki.asterisk..org
* \arg \b Docs: The Asterisk Documentation Project http://www.asteriskdocs.org
* \arg \b Digium: The Asterisk Company http://www.digium.com
*/
@ -92,8 +92,6 @@
* \arg \ref AstExtState
* \arg \ref AstDataRetrieval
*
* \subsection model_txt Generic Model
* \verbinclude model.txt
* \subsection channel_txt Channels
* \arg See \ref Def_Channel
*/
@ -224,7 +222,6 @@
* - \ref pbx_retrieve_variable()
* - \ref AstChanVar
*
* \verbinclude channelvariables.tex
*/
/*!
@ -281,7 +278,6 @@
* \arg \ref enum.c
* \arg \ref func_enum.c
*
* \verbinclude enum.txt
*/
/*!
@ -334,7 +330,6 @@
/*!
* \page Config_ast Asterisk.conf
* \verbinclude asterisk-conf.txt
*/
/*!
@ -373,11 +368,9 @@
* IAX2 is implemented in \ref chan_iax2.c
* \arg \link Config_iax iax.conf Configuration file example \endlink
* \section iaxreadme IAX readme file
* \verbinclude iax.txt
* \section Config_iax IAX Configuration example
* \verbinclude iax.conf.sample
* \section iaxjitter IAX Jitterbuffer information
* \verbinclude jitterbuffer.txt
*/
/*!
@ -409,7 +402,6 @@
* \page README_misdn MISDN documentation
* \arg See \ref Config_misdn
* \section mISDN configuration
* \verbinclude misdn.txt
*/
/*!
@ -728,7 +720,6 @@
* \arg Configuration file:
* \verbinclude res_config_sqlite.conf
* \arg SQL tables:
* \verbinclude res_config_sqlite.txt
* \arg See also:
* http://www.sqlite.org
*/

@ -21,7 +21,7 @@
* \arg \ref AJI_intro
* \ref res_jabber.c
* \author Matt O'Gorman <mogorman@digium.com>
* \extref IKSEMEL http://iksemel.jabberstudio.org
* IKSEMEL http://iksemel.jabberstudio.org
*
* \page AJI_intro AJI - The Asterisk Jabber Interface
*

@ -75,7 +75,6 @@ const char *ast_setlocale(const char *locale);
* \param len Size of the chunk of memory buf.
* \param format A string specifying the format of time to be placed into buf.
* \param tm Pointer to the broken out time to be used for the format.
* \param locale Text string specifying the locale to be used for language strings.
* \retval An integer value specifying the number of bytes placed into buf or -1 on error.
*/
int ast_strftime(char *buf, size_t len, const char *format, const struct ast_tm *tm);
@ -87,7 +86,6 @@ int ast_strftime_locale(char *buf, size_t len, const char *format, const struct
* \param s A string specifying some portion of a date and time.
* \param format The format in which the string, s, is expected.
* \param tm The broken-out time structure into which the parsed data is expected.
* \param locale Text string specifying the locale to be used for language strings.
* \retval A pointer to the first character within s not used to parse the date and time.
*/
char *ast_strptime(const char *s, const char *format, struct ast_tm *tm);

@ -104,9 +104,11 @@ int ast_odbc_smart_execute(struct odbc_obj *obj, SQLHSTMT stmt) __attribute__((d
* \brief Retrieves a connected ODBC object
* \param name The name of the ODBC class for which a connection is needed.
* \param flags One or more of the following flags:
* \li RES_ODBC_SANITY_CHECK Whether to ensure that a connection is valid before returning the handle. Usually unnecessary.
* \li RES_ODBC_INDEPENDENT_CONNECTION Return a handle which is independent from all others. Usually used when starting a transaction.
* \li RES_ODBC_CONNECTED Only return a connected handle. Intended for use with peers which use idlecheck, which are checked periodically for reachability.
* \li RES_ODBC_SANITY_CHECK Whether to ensure that a connection is valid before returning the handle. Usually unnecessary.
* \li RES_ODBC_INDEPENDENT_CONNECTION Return a handle which is independent from all others. Usually used when starting a transaction.
* \li RES_ODBC_CONNECTED Only return a connected handle. Intended for use with peers which use idlecheck, which are checked periodically for reachability.
* \param file, function, lineno
*
* \return ODBC object
* \retval NULL if there is no connection available with the requested name.
*

@ -29,9 +29,10 @@ extern "C" {
/*!
* \brief Send a customized SIP INFO request
*
* \param chan Channel
* \param headers The headers to add to the INFO request
* \param content_type The content type header to add
* \param conten The body of the INFO request
* \param content The body of the INFO request
* \param useragent_filter If non-NULL, only send the INFO if the
* recipient's User-Agent contains useragent_filter as a substring
*

@ -19,7 +19,7 @@
/*! \file
* \brief XMPP Interface
* \author Joshua Colp <jcolp@digium.com>
* \extref IKSEMEL http://iksemel.jabberstudio.org
* IKSEMEL http://iksemel.jabberstudio.org
*/
#ifndef _ASTERISK_XMPP_H

@ -371,7 +371,7 @@ struct ast_acl_list *ast_duplicate_acl_list(struct ast_acl_list *original)
* \param netmask The netmask configured in the host access rule.
* \param result The resultant address after applying the netmask to the given address
* \retval 0 Successfully applied netmask
* \reval -1 Failed to apply netmask
* \retval -1 Failed to apply netmask
*/
static int apply_netmask(const struct ast_sockaddr *addr, const struct ast_sockaddr *netmask,
struct ast_sockaddr *result)

@ -2142,8 +2142,8 @@ void ast_yyset_lineno (int line_number , yyscan_t yyscanner)
}
/** Set the current column.
* @param line_number
* @param yyscanner The scanner object.
* \param column_no line_number
* \param yyscanner The scanner object.
*/
void ast_yyset_column (int column_no , yyscan_t yyscanner)
{

@ -21,6 +21,15 @@
/*!
* \mainpage Asterisk -- The Open Source Telephony Project
*
* \par Welcome
*
* This documentation created by the Doxygen project clearly explains the
* internals of the Asterisk software. This documentation contains basic
* examples, developer documentation, support information, and information
* for upgrading.
*
*
*
* \par Developer Documentation for Asterisk
*
* This is the main developer documentation for Asterisk. It is
@ -44,19 +53,19 @@
* of <a href="http://www.digium.com">Digium, Inc</a>.
*
* \author Mark Spencer <markster@digium.com>
* Also see \ref AstCREDITS
*
* 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.
*
* \todo Add pages for mailinglists, IRC, etc...
*/
/*! \file
\brief Top level source file for Asterisk - the Open Source PBX. Implementation
of PBX core functions and CLI interface.
* \brief Top level source file for Asterisk - the Open Source PBX.
* Implementation of PBX core functions and CLI interface.
*/
/*** MODULEINFO
@ -1024,7 +1033,8 @@ static struct sigaction ignore_sig_handler = {
AST_MUTEX_DEFINE_STATIC(safe_system_lock);
/*! \brief Keep track of how many threads are currently trying to wait*() on
* a child process */
* a child process
*/
static unsigned int safe_system_level = 0;
static struct sigaction safe_system_prev_handler;
@ -1200,7 +1210,7 @@ static void ast_network_puts(const char *string)
}
/*!
* write the string to the console, and all attached
* \brief write the string to the console, and all attached
* console clients
*/
void ast_console_puts(const char *string)
@ -1537,10 +1547,10 @@ static int ast_tryconnect(void)
}
/*! \brief Urgent handler
Called by soft_hangup to interrupt the poll, read, or other
system call. We don't actually need to do anything though.
Remember: Cannot EVER ast_log from within a signal handler
*
* Called by soft_hangup to interrupt the poll, read, or other
* system call. We don't actually need to do anything though.
* Remember: Cannot EVER ast_log from within a signal handler
*/
static void _urg_handler(int num)
{
@ -1628,8 +1638,10 @@ static void set_icon(char *text)
fprintf(stdout, "\033]1;%s\007", text);
}
/*! \brief We set ourselves to a high priority, that we might pre-empt everything
else. If your PBX has heavy activity on it, this is a good thing. */
/*! \brief We set ourselves to a high priority, that we might pre-empt
* everything else. If your PBX has heavy activity on it, this is a
* good thing.
*/
int ast_set_priority(int pri)
{
struct sched_param sched;
@ -1737,7 +1749,8 @@ static int can_safely_quit(shutdown_nice_t niceness, int restart)
}
/* Re-acquire lock and check if someone changed the niceness, in which
* case someone else has taken over the shutdown. */
* case someone else has taken over the shutdown.
*/
ast_mutex_lock(&safe_system_lock);
if (shuttingdown != niceness) {
if (shuttingdown == NOT_SHUTTING_DOWN && option_verbose && ast_opt_console) {
@ -3478,7 +3491,10 @@ int main(int argc, char *argv[])
if (getenv("HOME"))
snprintf(filename, sizeof(filename), "%s/.asterisk_history", getenv("HOME"));
/* Check for options */
/*! \brief Check for options
*
* \todo Document these options
*/
while ((c = getopt(argc, argv, "BC:cde:FfG:ghIiL:M:mnpqRrs:TtU:VvWXx:")) != -1) {
/*!\note Please keep the ordering here to alphabetical, capital letters
* first. This will make it easier in the future to select unused

@ -568,6 +568,8 @@ static enum ast_device_state cc_state_to_devstate_map[] = {
* \intenral
* \brief lookup the ast_device_state mapped to cc_state
*
* \param state
*
* \return the correponding DEVICE STATE from the cc_state_to_devstate_map
* when passed an internal state.
*/
@ -1638,7 +1640,7 @@ struct extension_child_dialstring {
*
* \details
* This serves mainly as a key when searching for a particular dialstring.
* For instance, let's say that we have called device SIP/400@somepeer. This
* For instance, let's say that we have called device SIP/400\@somepeer. This
* device offers call completion, but then due to some unforeseen circumstance,
* this device backs out and makes CC unavailable. When that happens, we need
* to find the dialstring that corresponds to that device, and we use the

@ -25,7 +25,7 @@
* This program implements a tiny http server
* and was inspired by micro-httpd by Jef Poskanzer
*
* \extref GMime http://spruce.sourceforge.net/gmime/
* GMime http://spruce.sourceforge.net/gmime/
*
* \ref AstHTTP - AMI over the http protocol
*/

@ -22,7 +22,7 @@
*
* \author Mark Spencer <markster@digium.com>
*
* \extref OpenSSL http://www.openssl.org - for AMI/SSL
* OpenSSL http://www.openssl.org - for AMI/SSL
*
* At the moment this file contains a number of functions, namely:
*

@ -336,7 +336,7 @@ static void SHA1ProcessMessageBlock(SHA1Context *context)
/*!
* \brief This helper function finishes off the digest calculations.
* \param context [in/out] The context to pad.
* \param Pad_byte [in] The last byte to add to the message block
* \param Pad_Byte [in] The last byte to add to the message block
* before the 0-padding and length. This will contain the last
* bits of the message followed by another single bit. If the
* message was an exact multiple of 8-bits long, Pad_Byte will
@ -359,7 +359,7 @@ static void SHA1Finalize(SHA1Context * context, uint8_t Pad_Byte)
/*!
* \brief Pad message to be 512 bits.
* \param context [in/out] The context to pad.
* \param Pad_byte [in] Last padding byte.
* \param Pad_Byte [in] Last padding byte.
*
* According to the standard, the message must be padded to the next
* even multiple of 512 bits. The first padding bit must be a '1'.

@ -70,7 +70,7 @@ int __ast_str_helper(struct ast_str **buf, ssize_t max_len,
}
/*
* Ask vsnprintf how much space we need. Remember that vsnprintf
* does not count the final <code>'\0'</code> so we must add 1.
* does not count the final <code>'\\0'</code> so we must add 1.
*/
va_copy(aq, ap);
res = vsnprintf((*buf)->__AST_STR_STR + offset, (*buf)->__AST_STR_LEN - offset, fmt, aq);

@ -281,8 +281,9 @@ static inline float tdd_getcarrier(float *cr, float *ci, int bit)
} while(0);
/*! Generate TDD hold tone
* \param buf Result buffer
* \todo How big should this be??? */
* \param outbuf, buf Result buffer
* \todo How big should this be?
*/
int tdd_gen_holdtone(unsigned char *buf)
{
int bytes = 0;

@ -20,7 +20,7 @@
*
* \author Eliel C. Sardanons (LU1ALY) <eliels@gmail.com>
*
* \extref libxml2 http://www.xmlsoft.org/
* libxml2 http://www.xmlsoft.org/
*/
/*** MODULEINFO

@ -22,7 +22,7 @@
*
* \author Tilghman Lesher <res_config_curl_v1@the-tilghman.com>
*
* \extref Depends on the CURL library - http://curl.haxx.se/
* Depends on the CURL library - http://curl.haxx.se/
*
*/

@ -28,7 +28,7 @@
* \author Carl-Einar Thorner <cthorner@voicerd.com>
* \author Russell Bryant <russell@digium.com>
*
* \extref OpenLDAP http://www.openldap.org
* OpenLDAP http://www.openldap.org
*/
/*** MODULEINFO

@ -19,7 +19,7 @@
* \author Mark Spencer <markster@digium.com>
* \author Manuel Guesdon <mguesdon@oxymium.net> - PostgreSQL RealTime Driver Author/Adaptor
*
* \extref PostgreSQL http://www.postgresql.org
* PostgreSQL http://www.postgresql.org
*/
/*** MODULEINFO

@ -22,7 +22,7 @@
*
* \author Mark Spencer <markster@digium.com>
*
* \extref Uses the OpenSSL library, available at
* Uses the OpenSSL library, available at
* http://www.openssl.org/
*/

@ -22,7 +22,7 @@
*
* \author Tilghman Lesher <res_curl_v1@the-tilghman.com>
*
* \extref Depends on the CURL library - http://curl.haxx.se/
* Depends on the CURL library - http://curl.haxx.se/
*
*/

@ -23,7 +23,7 @@
* References:
* - http://www.xmpp.org - The XMPP standards foundation
*
* \extref Iksemel http://code.google.com/p/iksemel/
* Iksemel http://code.google.com/p/iksemel/
*
* \todo If you unload this module, chan_gtalk/jingle will be dead. How do we handle that?
* \todo Dialplan applications need RETURN variable, like JABBERSENDSTATUS
@ -3304,6 +3304,7 @@ static void aji_init_event_distribution(struct aji_client *client)
/*!
* \brief Callback for handling PubSub events
* \param data void pointer to aji_client structure
* \param pak A pak
* \return IKS_FILTER_EAT
*/
static int aji_handle_pubsub_event(void *data, ikspak *pak)
@ -3476,8 +3477,10 @@ static void aji_publish_device_state(struct aji_client *client, const char *devi
/*!
* \brief Publish MWI to a PubSub node
* \param client the configured XMPP client we use to connect to a XMPP server
* \param device the name of the device whose state to publish
* \param device_state the state to publish
* \param mailbox The mailbox
* \param context The context
* \param oldmsgs Old messages
* \param newmsgs New messages
* \return void
*/
static void aji_publish_mwi(struct aji_client *client, const char *mailbox,
@ -3833,6 +3836,7 @@ static void aji_create_pubsub_collection(struct aji_client *client, const char
/*!
* \brief Create a PubSub leaf node.
* \param client the configured XMPP client we use to connect to a XMPP server
* \param collection_name The name to use for this collection
* \param leaf_name The name to use for this collection
* \return void.
*/
@ -3847,6 +3851,7 @@ const char *leaf_name)
* \param client the configured XMPP client we use to connect to a XMPP server
* \param node_type the type of node to create
* \param name the name of the node to create
* \param collection_name The name to use for this collection
* \return iks*
*/
static iks* aji_create_pubsub_node(struct aji_client *client, const char *node_type, const

@ -24,7 +24,7 @@
* \author Russell Bryant <russell@digium.com>
*
* Here is a useful mailing list post that describes SMDI protocol details:
* \ref http://lists.digium.com/pipermail/asterisk-dev/2003-June/000884.html
* http://lists.digium.com/pipermail/asterisk-dev/2003-June/000884.html
*
* \todo This module currently has its own mailbox monitoring thread. This should
* be converted to MWI subscriptions and just let the optional global voicemail

@ -13,7 +13,7 @@
*
* \author Thorsten Lockert <tholo@voop.as>
*
* \extref Uses the Net-SNMP libraries available at
* Uses the Net-SNMP libraries available at
* http://net-snmp.sourceforge.net/
*/

@ -22,7 +22,7 @@
*
* \author Joshua Colp <jcolp@digium.com>
*
* \extref Iksemel http://code.google.com/p/iksemel/
* Iksemel http://code.google.com/p/iksemel/
*
* A reference module for interfacting Asterisk directly as a client or component with
* an XMPP/Jabber compliant server.
@ -1017,6 +1017,7 @@ static void xmpp_pubsub_create_affiliations(struct ast_xmpp_client *client, cons
* \param client the configured XMPP client we use to connect to a XMPP server
* \param node_type the type of node to create
* \param name the name of the node to create
* \param collection_name
* \return void
*/
static void xmpp_pubsub_create_node(struct ast_xmpp_client *client, const char *node_type, const
@ -1078,6 +1079,7 @@ static void xmpp_pubsub_create_collection(struct ast_xmpp_client *client, const
/*!
* \brief Create a PubSub leaf node.
* \param client the configured XMPP client we use to connect to a XMPP server
* \param collection_name
* \param leaf_name The name to use for this collection
* \return void.
*/
@ -1090,8 +1092,10 @@ static void xmpp_pubsub_create_leaf(struct ast_xmpp_client *client, const char *
/*!
* \brief Publish MWI to a PubSub node
* \param client the configured XMPP client we use to connect to a XMPP server
* \param device the name of the device whose state to publish
* \param device_state the state to publish
* \param mailbox The Mailbox
* \param context The Context
* \param oldmsgs Old messages
* \param newmsgs New Messages
* \return void
*/
static void xmpp_pubsub_publish_mwi(struct ast_xmpp_client *client, const char *mailbox,
@ -1278,6 +1282,7 @@ static void xmpp_pubsub_subscribe(struct ast_xmpp_client *client, const char *no
/*!
* \brief Callback for handling PubSub events
* \param data void pointer to ast_xmpp_client structure
* \param pak A pak
* \return IKS_FILTER_EAT
*/
static int xmpp_pubsub_handle_event(void *data, ikspak *pak)

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