[asterisk-commits] oej: trunk r56665 - in /trunk: apps/ channels/
include/asterisk/ main/ pbx/ res/
asterisk-commits at lists.digium.com
asterisk-commits at lists.digium.com
Sat Feb 24 13:29:42 MST 2007
Author: oej
Date: Sat Feb 24 14:29:41 2007
New Revision: 56665
URL: http://svn.digium.com/view/asterisk?view=rev&rev=56665
Log:
Doxygen additions, corrections
Modified:
trunk/apps/app_amd.c
trunk/apps/app_meetme.c
trunk/channels/chan_h323.c
trunk/include/asterisk/ael_structs.h
trunk/include/asterisk/config.h
trunk/include/asterisk/doxyref.h
trunk/include/asterisk/http.h
trunk/include/asterisk/jingle.h
trunk/include/asterisk/utils.h
trunk/include/asterisk/zapata.h
trunk/main/channel.c
trunk/main/config.c
trunk/main/rtp.c
trunk/pbx/pbx_ael.c
trunk/res/res_limit.c
Modified: trunk/apps/app_amd.c
URL: http://svn.digium.com/view/asterisk/trunk/apps/app_amd.c?view=diff&rev=56665&r1=56664&r2=56665
==============================================================================
--- trunk/apps/app_amd.c (original)
+++ trunk/apps/app_amd.c Sat Feb 24 14:29:41 2007
@@ -18,6 +18,14 @@
* A license has been granted to Digium (via disclaimer) for the use of
* this code.
*/
+
+/*! \file
+ *
+ * \brief Answering machine detection
+ *
+ * \author Claude Klimos (claude.klimos at aheeva.com)
+ */
+
#include "asterisk.h"
Modified: trunk/apps/app_meetme.c
URL: http://svn.digium.com/view/asterisk/trunk/apps/app_meetme.c?view=diff&rev=56665&r1=56664&r2=56665
==============================================================================
--- trunk/apps/app_meetme.c (original)
+++ trunk/apps/app_meetme.c Sat Feb 24 14:29:41 2007
@@ -334,6 +334,7 @@
int actual; /*!< Actual volume adjustment (for channels that can't adjust) */
};
+/*! \brief The MeetMe User object */
struct ast_conf_user {
int user_no; /*!< User Number */
int userflags; /*!< Flags as set in the conference */
Modified: trunk/channels/chan_h323.c
URL: http://svn.digium.com/view/asterisk/trunk/channels/chan_h323.c?view=diff&rev=56665&r1=56664&r2=56665
==============================================================================
--- trunk/channels/chan_h323.c (original)
+++ trunk/channels/chan_h323.c Sat Feb 24 14:29:41 2007
@@ -120,10 +120,9 @@
setpeercapabilities_cb on_setpeercapabilities;
onhold_cb on_hold;
-/* global debug flag */
-int h323debug;
-
-/*! Global jitterbuffer configuration - by default, jb is disabled */
+int h323debug; /*!< global debug flag */
+
+/*! \brief Global jitterbuffer configuration - by default, jb is disabled */
static struct ast_jb_conf default_jbconf =
{
.flags = 0,
@@ -156,79 +155,81 @@
static call_options_t global_options;
-/** Private structure of a OpenH323 channel */
+/*! \brief Private structure of a OpenH323 channel */
struct oh323_pvt {
- ast_mutex_t lock; /* Channel private lock */
- call_options_t options; /* Options to be used during call setup */
- int alreadygone; /* Whether or not we've already been destroyed by our peer */
- int needdestroy; /* if we need to be destroyed */
- call_details_t cd; /* Call details */
- struct ast_channel *owner; /* Who owns us */
- struct sockaddr_in sa; /* Our peer */
- struct sockaddr_in redirip; /* Where our RTP should be going if not to us */
- int nonCodecCapability; /* non-audio capability */
- int outgoing; /* Outgoing or incoming call? */
- char exten[AST_MAX_EXTENSION]; /* Requested extension */
- char context[AST_MAX_CONTEXT]; /* Context where to start */
- char accountcode[256]; /* Account code */
- char rdnis[80]; /* Referring DNIS, if available */
- int amaflags; /* AMA Flags */
- struct ast_rtp *rtp; /* RTP Session */
- struct ast_dsp *vad; /* Used for in-band DTMF detection */
- int nativeformats; /* Codec formats supported by a channel */
- int needhangup; /* Send hangup when Asterisk is ready */
- int hangupcause; /* Hangup cause from OpenH323 layer */
- int newstate; /* Pending state change */
- int newcontrol; /* Pending control to send */
- int newdigit; /* Pending DTMF digit to send */
- int newduration; /* Pending DTMF digit duration to send */
- int pref_codec; /* Preferred codec */
- int peercapability; /* Capabilities learned from peer */
- int jointcapability; /* Common capabilities for local and remote side */
- struct ast_codec_pref peer_prefs; /* Preferenced list of codecs which remote side supports */
- int dtmf_pt[2]; /* Payload code used for RFC2833/CISCO messages */
- int curDTMF; /* DTMF tone being generated to Asterisk side */
- int DTMFsched; /* Scheduler descriptor for DTMF */
- int update_rtp_info; /* Configuration of fd's array is pending */
- int recvonly; /* Peer isn't wish to receive our voice stream */
- int txDtmfDigit; /* DTMF digit being to send to H.323 side */
- int noInbandDtmf; /* Inband DTMF processing by DSP isn't available */
- int connection_established; /* Call got CONNECT message */
- int got_progress; /* Call got PROGRESS message, pass inband audio */
- struct oh323_pvt *next; /* Next channel in list */
+ ast_mutex_t lock; /*!< Channel private lock */
+ call_options_t options; /*!<!< Options to be used during call setup */
+ int alreadygone; /*!< Whether or not we've already been destroyed by our peer */
+ int needdestroy; /*!< if we need to be destroyed */
+ call_details_t cd; /*!< Call details */
+ struct ast_channel *owner; /*!< Who owns us */
+ struct sockaddr_in sa; /*!< Our peer */
+ struct sockaddr_in redirip; /*!< Where our RTP should be going if not to us */
+ int nonCodecCapability; /*!< non-audio capability */
+ int outgoing; /*!< Outgoing or incoming call? */
+ char exten[AST_MAX_EXTENSION]; /*!< Requested extension */
+ char context[AST_MAX_CONTEXT]; /*!< Context where to start */
+ char accountcode[256]; /*!< Account code */
+ char rdnis[80]; /*!< Referring DNIS, if available */
+ int amaflags; /*!< AMA Flags */
+ struct ast_rtp *rtp; /*!< RTP Session */
+ struct ast_dsp *vad; /*!< Used for in-band DTMF detection */
+ int nativeformats; /*!< Codec formats supported by a channel */
+ int needhangup; /*!< Send hangup when Asterisk is ready */
+ int hangupcause; /*!< Hangup cause from OpenH323 layer */
+ int newstate; /*!< Pending state change */
+ int newcontrol; /*!< Pending control to send */
+ int newdigit; /*!< Pending DTMF digit to send */
+ int newduration; /*!< Pending DTMF digit duration to send */
+ int pref_codec; /*!< Preferred codec */
+ int peercapability; /*!< Capabilities learned from peer */
+ int jointcapability; /*!< Common capabilities for local and remote side */
+ struct ast_codec_pref peer_prefs; /*!< Preferenced list of codecs which remote side supports */
+ int dtmf_pt[2]; /*!< Payload code used for RFC2833/CISCO messages */
+ int curDTMF; /*!< DTMF tone being generated to Asterisk side */
+ int DTMFsched; /*!< Scheduler descriptor for DTMF */
+ int update_rtp_info; /*!< Configuration of fd's array is pending */
+ int recvonly; /*!< Peer isn't wish to receive our voice stream */
+ int txDtmfDigit; /*!< DTMF digit being to send to H.323 side */
+ int noInbandDtmf; /*!< Inband DTMF processing by DSP isn't available */
+ int connection_established; /*!< Call got CONNECT message */
+ int got_progress; /*!< Call got PROGRESS message, pass inband audio */
+ struct oh323_pvt *next; /*!< Next channel in list */
} *iflist = NULL;
-static struct ast_user_list {
+/*! \brief H323 User list */
+static struct h323_user_list {
ASTOBJ_CONTAINER_COMPONENTS(struct oh323_user);
} userl;
-static struct ast_peer_list {
+/*! \brief H323 peer list */
+static struct h323_peer_list {
ASTOBJ_CONTAINER_COMPONENTS(struct oh323_peer);
} peerl;
-static struct ast_alias_list {
+/*! \brief H323 alias list */
+static struct h323_alias_list {
ASTOBJ_CONTAINER_COMPONENTS(struct oh323_alias);
} aliasl;
-/** Asterisk RTP stuff */
+/* Asterisk RTP stuff */
static struct sched_context *sched;
static struct io_context *io;
-/** Protect the interface list (oh323_pvt) */
-AST_MUTEX_DEFINE_STATIC(iflock);
-
-/* Protect the monitoring thread, so only one process can kill or start it, and not
+AST_MUTEX_DEFINE_STATIC(iflock); /*!< Protect the interface list (oh323_pvt) */
+
+/*! \brief Protect the H.323 monitoring thread, so only one process can kill or start it, and not
when it's doing something critical. */
AST_MUTEX_DEFINE_STATIC(monlock);
-/* Protect the H.323 capabilities list, to avoid more than one channel to set the capabilities simultaneaously in the h323 stack. */
+/*! \brief Protect the H.323 capabilities list, to avoid more than one channel to set the capabilities simultaneaously in the h323 stack. */
AST_MUTEX_DEFINE_STATIC(caplock);
-/* Protect the reload process */
+/*! \brief Protect the reload process */
AST_MUTEX_DEFINE_STATIC(h323_reload_lock);
static int h323_reloading = 0;
-/* This is the thread for the monitor which checks for input on the channels
+/*! \brief This is the thread for the monitor which checks for input on the channels
which are not currently in use. */
static pthread_t monitor_thread = AST_PTHREADT_NULL;
static int restart_monitor(void);
@@ -336,7 +337,7 @@
return 0;
}
-/* Channel and private structures should be already locked */
+/*! \brief Channel and private structures should be already locked */
static void __oh323_update_info(struct ast_channel *c, struct oh323_pvt *pvt)
{
if (c->nativeformats != pvt->nativeformats) {
@@ -402,7 +403,7 @@
}
}
-/* Only channel structure should be locked */
+/*! \brief Only channel structure should be locked */
static void oh323_update_info(struct ast_channel *c)
{
struct oh323_pvt *pvt = c->tech_pvt;
@@ -546,7 +547,7 @@
return 0;
}
-/**
+/*! \brief
* Send (play) the specified digit to the channel.
*
*/
@@ -584,7 +585,7 @@
return 0;
}
-/**
+/*! \brief
* Make a call over the specified channel to the specified
* destination.
* Returns -1 on error, 0 on success.
@@ -757,9 +758,9 @@
return 0;
}
+/*! \brief Retrieve audio/etc from channel. Assumes pvt->lock is already held. */
static struct ast_frame *oh323_rtp_read(struct oh323_pvt *pvt)
{
- /* Retrieve audio/etc from channel. Assumes pvt->lock is already held. */
struct ast_frame *f;
/* Only apply it for the first packet, we just need the correct ip/port */
@@ -1004,7 +1005,7 @@
return 0;
}
-/* Private structure should be locked on a call */
+/*! \brief Private structure should be locked on a call */
static struct ast_channel *__oh323_new(struct oh323_pvt *pvt, int state, const char *host)
{
struct ast_channel *ch;
@@ -1811,7 +1812,7 @@
return tmpc;
}
-/** Find a call by alias */
+/*! \brief Find a call by alias */
static struct oh323_alias *find_alias(const char *source_aliases, int realtime)
{
struct oh323_alias *a;
@@ -1824,7 +1825,7 @@
return a;
}
-/**
+/*! \brief
* Callback for sending digits from H.323 up to asterisk
*
*/
@@ -1895,10 +1896,10 @@
return res;
}
-/**
+/*! \brief
* Callback function used to inform the H.323 stack of the local rtp ip/port details
*
- * Returns the local RTP information
+ * \return Returns the local RTP information
*/
static struct rtp_info *external_rtp_create(unsigned call_reference, const char * token)
{
@@ -1936,7 +1937,7 @@
return info;
}
-/**
+/*! \brief
* Definition taken from rtp.c for rtpPayloadType because we need it here.
*/
struct rtpPayloadType {
@@ -1944,7 +1945,7 @@
int code;
};
-/**
+/*! \brief
* Call-back function passing remote ip/port information from H.323 to asterisk
*
* Returns nothing
@@ -2054,7 +2055,7 @@
return;
}
-/**
+/*! \brief
* Call-back function to signal asterisk that the channel has been answered
* Returns nothing
*/
@@ -2108,7 +2109,7 @@
return 0;
}
-/**
+/*! \brief
* Call-back function for incoming calls
*
* Returns 1 on success
@@ -2228,7 +2229,7 @@
return &pvt->options;
}
-/**
+/*! \brief
* Call-back function to start PBX when OpenH323 ready to serve incoming call
*
* Returns 1 on success
@@ -2307,7 +2308,7 @@
return 1;
}
-/**
+/*! \brief
* Call-back function to establish an outgoing H.323 call
*
* Returns 1 on success
@@ -2320,7 +2321,7 @@
return 1;
}
-/**
+/*! \brief
* Call-back function to signal asterisk that the channel is ringing
* Returns nothing
*/
@@ -2346,7 +2347,7 @@
return;
}
-/**
+/*! \brief
* Call-back function to cleanup communication
* Returns nothing,
*/
Modified: trunk/include/asterisk/ael_structs.h
URL: http://svn.digium.com/view/asterisk/trunk/include/asterisk/ael_structs.h?view=diff&rev=56665&r1=56664&r2=56665
==============================================================================
--- trunk/include/asterisk/ael_structs.h (original)
+++ trunk/include/asterisk/ael_structs.h Sat Feb 24 14:29:41 2007
@@ -1,3 +1,27 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 1999 - 2007, Digium, Inc.
+ *
+ * Mark Spencer <markster at 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 Structures for AEL - the Asterisk extension language
+ *
+ * \ref pbx_ael.c
+ */
+
#ifndef _ASTERISK_AEL_STRUCTS_H
#define _ASTERISK_AEL_STRUCTS_H
@@ -22,8 +46,7 @@
# endif
-typedef enum
-{
+typedef enum {
PV_WORD, /* an ident, string, name, label, etc. A user-supplied string. */ /* 0 */
PV_MACRO, /* 1 */
PV_CONTEXT, /* 2 */
Modified: trunk/include/asterisk/config.h
URL: http://svn.digium.com/view/asterisk/trunk/include/asterisk/config.h?view=diff&rev=56665&r1=56664&r2=56665
==============================================================================
--- trunk/include/asterisk/config.h (original)
+++ trunk/include/asterisk/config.h Sat Feb 24 14:29:41 2007
@@ -33,6 +33,8 @@
struct ast_category;
+/*! \brief Structure for variables, used for configurations and for channel variables
+*/
struct ast_variable {
char *name;
char *value;
@@ -50,6 +52,7 @@
typedef struct ast_config *realtime_multi_get(const char *database, const char *table, va_list ap);
typedef int realtime_update(const char *database, const char *table, const char *keyfield, const char *entity, va_list ap);
+/*! \brief Configuration engine structure, used to define realtime drivers */
struct ast_config_engine {
char *name;
config_load_func *load_func;
Modified: trunk/include/asterisk/doxyref.h
URL: http://svn.digium.com/view/asterisk/trunk/include/asterisk/doxyref.h?view=diff&rev=56665&r1=56664&r2=56665
==============================================================================
--- trunk/include/asterisk/doxyref.h (original)
+++ trunk/include/asterisk/doxyref.h Sat Feb 24 14:29:41 2007
@@ -16,7 +16,7 @@
* at the top of the source tree.
*/
-/* \file This file generates Doxygen pages from files in the /doc
+/*! \file This file generates Doxygen pages from files in the /doc
directory of the Asterisk source code tree
*/
Modified: trunk/include/asterisk/http.h
URL: http://svn.digium.com/view/asterisk/trunk/include/asterisk/http.h?view=diff&rev=56665&r1=56664&r2=56665
==============================================================================
--- trunk/include/asterisk/http.h (original)
+++ trunk/include/asterisk/http.h Sat Feb 24 14:29:41 2007
@@ -147,6 +147,7 @@
*/
typedef struct ast_str *(*ast_http_callback)(struct sockaddr_in *requestor, const char *uri, struct ast_variable *params, int *status, char **title, int *contentlength);
+/*! \brief Definition of a URI reachable in the embedded HTTP server */
struct ast_http_uri {
AST_LIST_ENTRY(ast_http_uri) entry;
const char *description;
Modified: trunk/include/asterisk/jingle.h
URL: http://svn.digium.com/view/asterisk/trunk/include/asterisk/jingle.h?view=diff&rev=56665&r1=56664&r2=56665
==============================================================================
--- trunk/include/asterisk/jingle.h (original)
+++ trunk/include/asterisk/jingle.h Sat Feb 24 14:29:41 2007
@@ -15,6 +15,15 @@
* the GNU General Public License Version 2. See the LICENSE file
* at the top of the source tree.
*/
+
+/*! \file
+ * \brief Jingle definitions for chan_jingle
+ *
+ * \ref chan_jingle.c
+ *
+ * \author Matt O'Gorman <mogorman at digium.com>
+ */
+
#ifndef _ASTERISK_JINGLE_H
#define _ASTERISK_JINGLE_H
Modified: trunk/include/asterisk/utils.h
URL: http://svn.digium.com/view/asterisk/trunk/include/asterisk/utils.h?view=diff&rev=56665&r1=56664&r2=56665
==============================================================================
--- trunk/include/asterisk/utils.h (original)
+++ trunk/include/asterisk/utils.h Sat Feb 24 14:29:41 2007
@@ -39,8 +39,8 @@
#include "asterisk/logger.h"
#include "asterisk/compiler.h"
-/*! \note
- \verbatim
+/*!
+\note \verbatim
Note:
It is very important to use only unsigned variables to hold
bit flags, as otherwise you can fall prey to the compiler's
@@ -141,6 +141,8 @@
#define AST_FLAGS_ALL UINT_MAX
+/*! \brief Structure used to handle boolean flags
+*/
struct ast_flags {
unsigned int flags;
};
@@ -150,22 +152,21 @@
char buf[1024];
};
+/*! \brief Thread-safe gethostbyname function to use in Asterisk */
struct hostent *ast_gethostbyname(const char *host, struct ast_hostent *hp);
-/* ast_md5_hash
- \brief Produces MD5 hash based on input string */
+/*! \brief Produces MD5 hash based on input string */
void ast_md5_hash(char *output, char *input);
-/* ast_sha1_hash
- \brief Produces SHA1 hash based on input string */
+/*! \brief Produces SHA1 hash based on input string */
void ast_sha1_hash(char *output, char *input);
int ast_base64encode_full(char *dst, const unsigned char *src, int srclen, int max, int linebreaks);
int ast_base64encode(char *dst, const unsigned char *src, int srclen, int max);
int ast_base64decode(unsigned char *dst, const char *src, int max);
-/*! ast_uri_encode
- \brief Turn text string to URI-encoded %XX version
- At this point, we're converting from ISO-8859-x (8-bit), not UTF8
+/*! \brief Turn text string to URI-encoded %XX version
+
+\note At this point, we're converting from ISO-8859-x (8-bit), not UTF8
as in the SIP protocol spec
If doreserved == 1 we will convert reserved characters also.
RFC 2396, section 2.4
@@ -238,7 +239,7 @@
int ast_utils_init(void);
int ast_wait_for_input(int fd, int ms);
-/*! ast_carefulwrite
+/*!
\brief Try to write string, but wait no more than ms milliseconds
before timing out.
@@ -249,7 +250,7 @@
*/
int ast_carefulwrite(int fd, char *s, int len, int timeoutms);
-/*! Compares the source address and port of two sockaddr_in */
+/*! \brief Compares the source address and port of two sockaddr_in */
static force_inline int inaddrcmp(const struct sockaddr_in *sin1, const struct sockaddr_in *sin2)
{
return ((sin1->sin_addr.s_addr != sin2->sin_addr.s_addr)
Modified: trunk/include/asterisk/zapata.h
URL: http://svn.digium.com/view/asterisk/trunk/include/asterisk/zapata.h?view=diff&rev=56665&r1=56664&r2=56665
==============================================================================
--- trunk/include/asterisk/zapata.h (original)
+++ trunk/include/asterisk/zapata.h Sat Feb 24 14:29:41 2007
@@ -16,6 +16,11 @@
* Sometimes one really wonders why we need a copyright
* for less than ten lines of preprocessor directives...
*/
+
+/*! \file
+ * \brief Stub to find zaptel headers
+ */
+
/*
* Stub to find the zaptel headers. The configure script will
Modified: trunk/main/channel.c
URL: http://svn.digium.com/view/asterisk/trunk/main/channel.c?view=diff&rev=56665&r1=56664&r2=56665
==============================================================================
--- trunk/main/channel.c (original)
+++ trunk/main/channel.c Sat Feb 24 14:29:41 2007
@@ -72,12 +72,15 @@
struct ast_trans_pvt *path;
};
+/*! \brief List of SPY structures
+*/
struct ast_channel_spy_list {
struct channel_spy_trans read_translator;
struct channel_spy_trans write_translator;
AST_LIST_HEAD_NOLOCK(, ast_channel_spy) list;
};
+/*! \brief Definition of the Whisper buffer */
struct ast_channel_whisper_buffer {
ast_mutex_t lock;
struct ast_slinfactory sf;
@@ -88,10 +91,10 @@
/* uncomment if you have problems with 'monitoring' synchronized files */
#if 0
#define MONITOR_CONSTANT_DELAY
-#define MONITOR_DELAY 150 * 8 /* 150 ms of MONITORING DELAY */
+#define MONITOR_DELAY 150 * 8 /*!< 150 ms of MONITORING DELAY */
#endif
-/*! Prevent new channel allocation if shutting down. */
+/*! \brief Prevent new channel allocation if shutting down. */
static int shutting_down;
static int uniqueint;
@@ -101,22 +104,25 @@
AST_THREADSTORAGE(state2str_threadbuf);
#define STATE2STR_BUFSIZE 32
-/*! 100ms */
-#define AST_DEFAULT_EMULATE_DTMF_DURATION 100
-
+#define AST_DEFAULT_EMULATE_DTMF_DURATION 100 /*!< 100ms */
+
+/*! \brief List of channel drivers */
struct chanlist {
const struct ast_channel_tech *tech;
AST_LIST_ENTRY(chanlist) list;
};
-/*! the list of registered channel types */
+/*! \brief the list of registered channel types */
static AST_LIST_HEAD_NOLOCK_STATIC(backends, chanlist);
-/*! the list of channels we have. Note that the lock for this list is used for
+/*! \brief the list of channels we have. Note that the lock for this list is used for
both the channels list and the backends list. */
static AST_LIST_HEAD_STATIC(channels, ast_channel);
-/*! map AST_CAUSE's to readable string representations */
+/*! \brief map AST_CAUSE's to readable string representations
+ *
+ * \ref causes.h
+*/
const struct ast_cause {
int cause;
const char *name;
@@ -184,6 +190,7 @@
return var;
}
+/*! \brief Show channel types - CLI command */
static int show_channeltypes(int fd, int argc, char *argv[])
{
#define FORMAT "%-10.10s %-40.40s %-12.12s %-12.12s %-12.12s\n"
@@ -211,6 +218,7 @@
}
+/*! \brief Show details about a channel driver - CLI command */
static int show_channeltype(int fd, int argc, char *argv[])
{
struct chanlist *cl = NULL;
@@ -428,6 +436,7 @@
return 0;
}
+/*! \brief Unregister channel driver */
void ast_channel_unregister(const struct ast_channel_tech *tech)
{
struct chanlist *chan;
@@ -451,6 +460,7 @@
AST_LIST_UNLOCK(&channels);
}
+/*! \brief Get handle to channel driver based on name */
const struct ast_channel_tech *ast_get_channel_tech(const char *name)
{
struct chanlist *chanls;
Modified: trunk/main/config.c
URL: http://svn.digium.com/view/asterisk/trunk/main/config.c?view=diff&rev=56665&r1=56664&r2=56665
==============================================================================
--- trunk/main/config.c (original)
+++ trunk/main/config.c Sat Feb 24 14:29:41 2007
@@ -70,6 +70,7 @@
static int lline_buffer_size;
+/*! \brief Structure to keep comments for rewriting configuration files */
struct ast_comment {
struct ast_comment *next;
char cmt[0];
Modified: trunk/main/rtp.c
URL: http://svn.digium.com/view/asterisk/trunk/main/rtp.c?view=diff&rev=56665&r1=56664&r2=56665
==============================================================================
--- trunk/main/rtp.c (original)
+++ trunk/main/rtp.c Sat Feb 24 14:29:41 2007
@@ -3277,10 +3277,6 @@
return res;
}
-/*! \brief Bridge calls. If possible and allowed, initiate
- re-invite so the peers exchange media directly outside
- of Asterisk.
-*/
/*! \page AstRTPbridge The Asterisk RTP bridge
The RTP bridge is called from the channel drivers that are using the RTP
subsystem in Asterisk - like SIP, H.323 and Jingle/Google Talk.
@@ -3306,6 +3302,12 @@
- ast_rtp_bridge()
- ast_channel_early_bridge()
- ast_channel_bridge()
+ - rtp.c
+ - rtp.h
+*/
+/*! \brief Bridge calls. If possible and allowed, initiate
+ re-invite so the peers exchange media directly outside
+ of Asterisk.
*/
enum ast_bridge_result ast_rtp_bridge(struct ast_channel *c0, struct ast_channel *c1, int flags, struct ast_frame **fo, struct ast_channel **rc, int timeoutms)
{
Modified: trunk/pbx/pbx_ael.c
URL: http://svn.digium.com/view/asterisk/trunk/pbx/pbx_ael.c?view=diff&rev=56665&r1=56664&r2=56665
==============================================================================
--- trunk/pbx/pbx_ael.c (original)
+++ trunk/pbx/pbx_ael.c Sat Feb 24 14:29:41 2007
@@ -70,7 +70,7 @@
development, this code can be properly re-instated
*/
-/* null definitions for structs passed down the infrastructure */
+/*! \brief null definitions for structs passed down the infrastructure */
struct argapp
{
struct argapp *next;
@@ -151,7 +151,7 @@
static void remove_spaces_before_equals(char *str);
static void substitute_commas(char *str);
-/* I am adding this code to substitute commas with vertbars in the args to apps */
+/*! \brief I am adding this code to substitute commas with vertbars in the args to apps */
static void substitute_commas(char *str)
{
char *p = str;
Modified: trunk/res/res_limit.c
URL: http://svn.digium.com/view/asterisk/trunk/res/res_limit.c?view=diff&rev=56665&r1=56664&r2=56665
==============================================================================
--- trunk/res/res_limit.c (original)
+++ trunk/res/res_limit.c Sat Feb 24 14:29:41 2007
@@ -10,6 +10,14 @@
* This code is released by the author with no restrictions on usage.
*
*/
+
+/*! \file
+ *
+ * \brief Resource limits
+ *
+ * \author Tilghman Lesher <res_limit_200607 at the-tilghman.com>
+ */
+
#include "asterisk.h"
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