kenmirrors/asterisk
1
0
Fork 0
mirror of https://github.com/asterisk/asterisk.git synced 2025-10-12 15:45:18 +00:00
Code Issues Packages Projects Releases Wiki Activity

Doxygen additions, corrections

Browse Source 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@56665 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Olle Johansson
2007-02-24 20:29:41 +00:00
parent e916cf45da
commit 75d387acbc
15 changed files with 170 additions and 97 deletions
Download Patch File Download Diff File Expand all files Collapse all files

141
channels/chan_h323.c
View File

@@ -120,10 +120,9 @@ setcapabilities_cb on_setcapabilities;
setpeercapabilities_cb on_setpeercapabilities;
onhold_cb on_hold;
/* global debug flag */
int h323debug;
int h323debug; /*!< global debug flag */
/*! Global jitterbuffer configuration - by default, jb is disabled */
/*! \brief Global jitterbuffer configuration - by default, jb is disabled */
static struct ast_jb_conf default_jbconf =
{
.flags = 0,
@@ -156,79 +155,81 @@ static unsigned int unique = 0;
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);
AST_MUTEX_DEFINE_STATIC(iflock); /*!< Protect the interface list (oh323_pvt) */
/* Protect the monitoring thread, so only one process can kill or start it, and not
/*! \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 @@ static int oh323_simulate_dtmf_end(void *data)
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 @@ static void __oh323_update_info(struct ast_channel *c, struct oh323_pvt *pvt)
}
}
/* 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 @@ static int oh323_digit_begin(struct ast_channel *c, char digit)
return 0;
}
/**
/*! \brief
* Send (play) the specified digit to the channel.
*
*/
@@ -584,7 +585,7 @@ static int oh323_digit_end(struct ast_channel *c, char digit, unsigned int durat
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 @@ static int oh323_hangup(struct ast_channel *c)
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 @@ static int __oh323_rtp_create(struct oh323_pvt *pvt)
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 @@ static struct ast_channel *oh323_request(const char *type, int format, void *dat
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 @@ static struct oh323_alias *find_alias(const char *source_aliases, int realtime)
return a;
}
/**
/*! \brief
* Callback for sending digits from H.323 up to asterisk
*
*/
@@ -1895,10 +1896,10 @@ static int receive_digit(unsigned call_reference, char digit, const char *token,
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 @@ static struct rtp_info *external_rtp_create(unsigned call_reference, const char
return info;
}
/**
/*! \brief
* Definition taken from rtp.c for rtpPayloadType because we need it here.
*/
struct rtpPayloadType {
@@ -1944,7 +1945,7 @@ struct rtpPayloadType {
int code;
};
/**
/*! \brief
* Call-back function passing remote ip/port information from H.323 to asterisk
*
* Returns nothing
@@ -2054,7 +2055,7 @@ static void setup_rtp_connection(unsigned call_reference, const char *remoteIp,
return;
}
/**
/*! \brief
* Call-back function to signal asterisk that the channel has been answered
* Returns nothing
*/
@@ -2108,7 +2109,7 @@ static int progress(unsigned call_reference, const char *token, int inband)
return 0;
}
/**
/*! \brief
* Call-back function for incoming calls
*
* Returns 1 on success
@@ -2228,7 +2229,7 @@ static call_options_t *setup_incoming_call(call_details_t *cd)
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 @@ static int answer_call(unsigned call_reference, const char *token)
return 1;
}
/**
/*! \brief
* Call-back function to establish an outgoing H.323 call
*
* Returns 1 on success
@@ -2320,7 +2321,7 @@ static int setup_outgoing_call(call_details_t *cd)
return 1;
}
/**
/*! \brief
* Call-back function to signal asterisk that the channel is ringing
* Returns nothing
*/
@@ -2346,7 +2347,7 @@ static void chan_ringing(unsigned call_reference, const char *token)
return;
}
/**
/*! \brief
* Call-back function to cleanup communication
* Returns nothing,
*/