diff --git a/apps/app_voicemail.c b/apps/app_voicemail.c index 29d9f09f418518d8ba73bbff0efc0e6e0542a4ed..fd3512920389e514eeab2718c75e7c9eceabc588 100644 --- a/apps/app_voicemail.c +++ b/apps/app_voicemail.c @@ -936,6 +936,7 @@ static int make_file(char *dest, int len, char *dir, int num) /*! \brief basically mkdir -p $dest/$context/$ext/$folder * \param dest String. base directory. + * \param len Length of dest. * \param context String. Ignored if is null or empty string. * \param ext String. Ignored if is null or empty string. * \param folder String. Ignored if is null or empty string. diff --git a/channels/chan_sip.c b/channels/chan_sip.c index 2308df01861786e114d57bda7a507e033fab1113..881e707a67588574853b1432ad68af2cfe9e87b7 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -678,7 +678,7 @@ struct sip_history { AST_LIST_HEAD_NOLOCK(sip_history_head, sip_history); /*!< history list, entry in sip_pvt */ -/*! \brief sip_auth: Creadentials for authentication to other SIP services */ +/*! \brief sip_auth: Credentials for authentication to other SIP services */ struct sip_auth { char realm[AST_MAX_EXTENSION]; /*!< Realm in which these credentials are valid */ char username[256]; /*!< Username */ diff --git a/formats/format_ogg_vorbis.c b/formats/format_ogg_vorbis.c index 5fc17c338631caa9e8ef72e35f2c104b1144c00a..9d80b63d44395557c89b53b06ccffb36d2900198 100644 --- a/formats/format_ogg_vorbis.c +++ b/formats/format_ogg_vorbis.c @@ -250,7 +250,8 @@ static int ogg_vorbis_rewrite(struct ast_filestream *s, /*! * \brief Write out any pending encoded data. - * \param s A OGG/Vorbis filestream. + * \param s An OGG/Vorbis filestream. + * \param f The file to write to. */ static void write_stream(struct vorbis_desc *s, FILE *f) { @@ -276,9 +277,9 @@ static void write_stream(struct vorbis_desc *s, FILE *f) /*! * \brief Write audio data from a frame to an OGG/Vorbis filestream. - * \param fs A OGG/Vorbis filestream. - * \param f An frame containing audio to be written to the filestream. - * \return -1 ifthere was an error, 0 on success. + * \param fs An OGG/Vorbis filestream. + * \param f A frame containing audio to be written to the filestream. + * \return -1 if there was an error, 0 on success. */ static int ogg_vorbis_write(struct ast_filestream *fs, struct ast_frame *f) { diff --git a/include/asterisk/channel.h b/include/asterisk/channel.h index bf67fc9d21c219dbda40783ffc1fc50f36ebf30c..26412b3fe2278167b84bd3f321bfc1818353e492 100644 --- a/include/asterisk/channel.h +++ b/include/asterisk/channel.h @@ -801,6 +801,8 @@ struct ast_channel *ast_waitfor_nandfds(struct ast_channel **chan, int n, int *f /*! \brief Waits for input on a group of channels Wait for input on an array of channels for a given # of milliseconds. \return Return channel with activity, or NULL if none has activity. + \param chan an array of pointers to channels + \param n number of channels that are to be waited upon \param ms time "ms" is modified in-place, if applicable */ struct ast_channel *ast_waitfor_n(struct ast_channel **chan, int n, int *ms); @@ -928,9 +930,11 @@ struct ast_channel *ast_walk_channel_by_exten_locked(const struct ast_channel *c int ast_waitfordigit(struct ast_channel *c, int ms); /*! \brief Wait for a digit - Same as ast_waitfordigit() with audio fd for outputing read audio and ctrlfd to monitor for reading. + Same as ast_waitfordigit() with audio fd for outputting read audio and ctrlfd to monitor for reading. * \param c channel to wait for a digit on * \param ms how many milliseconds to wait + * \param audiofd audio file descriptor to write to if audio frames are received + * \param ctrlfd control file descriptor to monitor for reading * \return Returns 1 if ctrlfd becomes available */ int ast_waitfordigit_full(struct ast_channel *c, int ms, int audiofd, int ctrlfd); @@ -1014,7 +1018,7 @@ int ast_str2cause(const char *name) attribute_pure; /*! Gives the string form of a given channel state */ /*! - * \param state state to get the name of + * \param ast_channel_state state to get the name of * Give a name to a state * Returns the text form of the binary state given */ @@ -1344,6 +1348,7 @@ int ast_channel_whisper_start(struct ast_channel *chan); /*! \brief Feed an audio frame into the whisper buffer on a channel \param chan The channel to whisper onto + \param f The frame to to whisper onto chan \return 0 for success, non-zero for failure */ int ast_channel_whisper_feed(struct ast_channel *chan, struct ast_frame *f); diff --git a/include/asterisk/doxyref.h b/include/asterisk/doxyref.h index e9fb631107e8f343b21dafba169577da82d272aa..49e2828160cb114c753cfe614f0747a53976d406 100644 --- a/include/asterisk/doxyref.h +++ b/include/asterisk/doxyref.h @@ -119,7 +119,7 @@ DUNDi is not itself a Voice-over IP signaling or media protocol. Instead, it pub * \arg \ref cdr_drivers * \arg \ref Config_cdr CDR configuration files * - * \verbinclude cdr.txt + * \verbinclude cdrdriver.txt */ /*! \page AstREADME README - the general administrator introduction @@ -427,14 +427,11 @@ DUNDi is not itself a Voice-over IP signaling or media protocol. Instead, it pub /*! \page SoundFiles Sound files * \section SecSound Asterisk Sound files - * Asterisk includes a large amount of sound files. Many of these + * Asterisk includes a large number of sound files. Many of these * are used by applications and demo scripts within asterisk. * * Additional sound files are available in the asterisk-addons - * repository on cvs.digium.com - * - * \section SoundList List of included sound files - * \verbinclude sounds.txt + * repository on svn.digium.com */ /*! \addtogroup cdr_drivers Module: CDR Drivers diff --git a/include/asterisk/file.h b/include/asterisk/file.h index bf7efef8979bfb6f09c61cab67852c5455445a21..029bcf25d96362f5049c014514f90ce251e39aeb 100644 --- a/include/asterisk/file.h +++ b/include/asterisk/file.h @@ -218,7 +218,7 @@ int ast_filecopy(const char *oldname, const char *newname, const char *fmt); /*! Waits for a stream to stop or digit to be pressed */ /*! - * \param c channel to waitstram on + * \param c channel to waitstream on * \param breakon string of DTMF digits to break upon * Begins playback of a stream... * Wait for a stream to stop or for any one of a given digit to arrive, Returns 0 @@ -228,7 +228,7 @@ int ast_waitstream(struct ast_channel *c, const char *breakon); /*! Waits for a stream to stop or digit matching a valid one digit exten to be pressed */ /*! - * \param c channel to waitstram on + * \param c channel to waitstream on * \param context string of context to match digits to break upon * Begins playback of a stream... * Wait for a stream to stop or for any one of a valid extension digit to arrive, Returns 0 @@ -238,7 +238,7 @@ int ast_waitstream_exten(struct ast_channel *c, const char *context); /*! Same as waitstream but allows stream to be forwarded or rewound */ /*! - * \param c channel to waitstram on + * \param c channel to waitstream on * \param breakon string of DTMF digits to break upon * \param forward DTMF digit to fast forward upon * \param rewind DTMF digit to rewind upon diff --git a/include/asterisk/frame.h b/include/asterisk/frame.h index fa1c1289128b926d767419ae1233498576664403..cc3aa5d832a70f2933e864b66eca092e6e79c96c 100644 --- a/include/asterisk/frame.h +++ b/include/asterisk/frame.h @@ -473,7 +473,7 @@ struct ast_frame *ast_smoother_read(struct ast_smoother *s); void ast_frame_dump(const char *name, struct ast_frame *f, char *prefix); -/*! \par AudioCodecPref Audio Codec Preferences +/*! \page AudioCodecPref Audio Codec Preferences In order to negotiate audio codecs in the order they are configured in <channel>.conf for a device, we set up codec preference lists in addition to the codec capabilities setting. The capabilities diff --git a/include/asterisk/lock.h b/include/asterisk/lock.h index 2f63ee183e5964cabf7081cbf3a4eadcb5d65c71..bcd2fdabef93fcd27b58df868e7256927c6ce4b6 100644 --- a/include/asterisk/lock.h +++ b/include/asterisk/lock.h @@ -22,7 +22,7 @@ * - See \ref LockDef */ -/* \page LockDef Asterisk thread locking models +/*! \page LockDef Asterisk thread locking models * * This file provides different implementation of the functions, * depending on the platform, the use of DEBUG_THREADS, and the way diff --git a/include/asterisk/module.h b/include/asterisk/module.h index 4ce6e9460b692e57e59bde1909a1bbc174a60a4d..1d00f45049c213d1dfad1dc0fe62982e9d12d919 100644 --- a/include/asterisk/module.h +++ b/include/asterisk/module.h @@ -77,9 +77,9 @@ enum ast_module_load_result { enum ast_module_load_result ast_load_resource(const char *resource_name); /*! - * \brief Unloads a module. + * \brief Unload a module. * \param resource_name The name of the module to unload. - * \param unload_mode The force flag. This should be set using one of the AST_FORCE flags. + * \param ast_module_unload_mode The force flag. This should be set using one of the AST_FORCE flags. * * This function unloads a module. It will only unload modules that are not in * use (usecount not zero), unless #AST_FORCE_FIRM or #AST_FORCE_HARD is diff --git a/main/app.c b/main/app.c index d3e9f2a100651b39cf04cf14a5fd52663d54e602..60cfe236f77a4be4717faacfca4abf858f6f1f02 100644 --- a/main/app.c +++ b/main/app.c @@ -99,8 +99,12 @@ int ast_app_dtget(struct ast_channel *chan, const char *context, char *collect, return res; } -/*! \param timeout set timeout to 0 for "standard" timeouts. Set timeout to -1 for - "ludicrous time" (essentially never times out) */ +/*! \param c The channel to read from + * \param prompt The file to stream to the channel + * \param s The string to read in to. Must be at least the size of your length + * \param maxlen How many digits to read (maximum) + * \param timeout set timeout to 0 for "standard" timeouts. Set timeout to -1 for + * "ludicrous time" (essentially never times out) */ int ast_app_getdata(struct ast_channel *c, char *prompt, char *s, int maxlen, int timeout) { int res,to,fto; diff --git a/main/ast_expr2f.c b/main/ast_expr2f.c index 7d790924d8e4901732e0db5d064f653f241acd76..3693e4842153b8d1a5195c3087a32c77c1e5f329 100644 --- a/main/ast_expr2f.c +++ b/main/ast_expr2f.c @@ -2722,8 +2722,8 @@ YY_BUFFER_STATE ast_yy_scan_string (yyconst char * yystr , yyscan_t yyscanner) /** Setup the input buffer state to scan the given bytes. The next call to ast_yylex() will * scan from a @e copy of @a bytes. - * @param bytes the byte buffer to scan - * @param len the number of bytes in the buffer pointed to by @a bytes. + * @param yybytes the byte buffer to scan + * @param _yybytes_len the number of bytes in the buffer pointed to by @a bytes. * @param yyscanner The scanner object. * @return the newly allocated buffer state object. */ @@ -2884,7 +2884,7 @@ void ast_yyset_lineno (int line_number , yyscan_t yyscanner) } /** Set the current column. - * @param line_number + * @param column_no * @param yyscanner The scanner object. */ void ast_yyset_column (int column_no , yyscan_t yyscanner) diff --git a/main/asterisk.c b/main/asterisk.c index e92dd055d14a18403023a4d3314e23251811617b..b475e0385ae2bd7156fa187711474f3dd1de9a17 100644 --- a/main/asterisk.c +++ b/main/asterisk.c @@ -144,7 +144,7 @@ int daemon(int, int); /* defined in libresolv of all places */ ast_verbose("certain conditions. Type 'show license' for details.\n"); \ ast_verbose("=========================================================================\n") -/*! \defgroup main_options +/*! \defgroup main_options Main Configuration Options \brief Main configuration options from \ref Config_ast "asterisk.conf" or the operating system command line when starting Asterisk Some of them can be changed in the CLI