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Merged revisions 168191 via svnmerge from

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https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r168191 | rmudgett | 2009-01-09 15:28:42 -0600 (Fri, 09 Jan 2009) | 3 lines
  
  *  Fix for JIRA AST-175/ABE-1757
  *  Miscellaneous doxygen comments added.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168192 65c4cc65-6c06-0410-ace0-fbb531ad65f3
1.6.2
Richard Mudgett 17 years ago
parent 87318da8ea
commit 4d1500edbd
1 changed files with 221 additions and 21 deletions
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242
channels/chan_misdn.c
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@ -142,83 +142,272 @@ enum misdn_chan_state {
#define ORG_MISDN 2
struct hold_info {
/*!
* \brief Logical port the channel call record is HOLDED on
* because the B channel is no longer associated.
*/
int port;
/*!
* \brief Original B channel number the HOLDED call was using.
* \note Used only for debug display messages.
*/
int channel;
};
/*!
* \brief Channel call record structure
*/
struct chan_list {
/*!
* \brief The "allowed_bearers" string read in from /etc/asterisk/misdn.conf
*/
char allowed_bearers[BUFFERSIZE + 1];
/*!
* \brief State of the channel
*/
enum misdn_chan_state state;
/*!
* \brief TRUE if a hangup needs to be queued
* \note This is a debug flag only used to catch calls to hangup_chan() that are already hungup.
*/
int need_queue_hangup;
/*!
* \brief TRUE if a channel can be hung up by calling asterisk directly when done.
*/
int need_hangup;
/*!
* \brief TRUE if we could send an AST_CONTROL_BUSY if needed.
*/
int need_busy;
/*!
* \brief Who originally created this channel. ORG_AST or ORG_MISDN
*/
int originator;
/*!
* \brief TRUE of we are not to respond immediately to a SETUP message. Check the dialplan first.
* \note The "noautorespond_on_setup" boolean read in from /etc/asterisk/misdn.conf
*/
int noautorespond_on_setup;
int norxtone;
int norxtone; /* Boolean assigned values but the value is not used. */
/*!
* \brief TRUE if we are not to generate tones (Playtones)
*/
int notxtone;
/*!
* \brief TRUE if echo canceller is enabled. Value is toggled.
*/
int toggle_ec;
/*!
* \brief TRUE if you want to send Tone Indications to an incoming
* ISDN channel on a TE Port.
* \note The "incoming_early_audio" boolean read in from /etc/asterisk/misdn.conf
*/
int incoming_early_audio;
/*!
* \brief TRUE if DTMF digits are to be passed inband only.
* \note It is settable by the misdn_set_opt() application.
*/
int ignore_dtmf;
/*!
* \brief Pipe file descriptor handles array.
* Read from pipe[0], write to pipe[1]
*/
int pipe[2];
/*!
* \brief Read buffer for inbound audio from pipe[0]
*/
char ast_rd_buf[4096];
/*!
* \brief Inbound audio frame returned by misdn_read().
*/
struct ast_frame frame;
int faxdetect; /*!< 0:no 1:yes 2:yes+nojump */
/*!
* \brief Fax detection option. (0:no 1:yes 2:yes+nojump)
* \note The "faxdetect" option string read in from /etc/asterisk/misdn.conf
* \note It is settable by the misdn_set_opt() application.
*/
int faxdetect;
/*!
* \brief Number of seconds to detect a Fax machine when detection enabled.
* \note 0 disables the timeout.
* \note The "faxdetect_timeout" value read in from /etc/asterisk/misdn.conf
*/
int faxdetect_timeout;
/*!
* \brief Starting time of fax detection with timeout when nonzero.
*/
struct timeval faxdetect_tv;
/*!
* \brief TRUE if a fax has been detected.
*/
int faxhandled;
/*!
* \brief TRUE if we will use the Asterisk DSP to detect DTMF/Fax
* \note The "astdtmf" boolean read in from /etc/asterisk/misdn.conf
*/
int ast_dsp;
/*!
* \brief Jitterbuffer length
* \note The "jitterbuffer" value read in from /etc/asterisk/misdn.conf
*/
int jb_len;
/*!
* \brief Jitterbuffer upper threshold
* \note The "jitterbuffer_upper_threshold" value read in from /etc/asterisk/misdn.conf
*/
int jb_upper_threshold;
/*!
* \brief Allocated jitterbuffer controller
* \note misdn_jb_init() creates the jitterbuffer.
* \note Must use misdn_jb_destroy() to clean up.
*/
struct misdn_jb *jb;
/*!
* \brief Allocated DSP controller
* \note ast_dsp_new() creates the DSP controller.
* \note Must use ast_dsp_free() to clean up.
*/
struct ast_dsp *dsp;
/*!
* \brief Allocated audio frame sample translator
* \note ast_translator_build_path() creates the translator path.
* \note Must use ast_translator_free_path() to clean up.
*/
struct ast_trans_pvt *trans;
/*!
* \brief Associated Asterisk channel structure.
*/
struct ast_channel * ast;
int dummy;
//int dummy; /* Not used */
/*!
* \brief Associated B channel structure.
*/
struct misdn_bchannel *bc;
/*!
* \brief HOLDED channel information
*/
struct hold_info hold_info;
/*!
* \brief From associated B channel: Layer 3 process ID
* \note Used to find the HOLDED channel call record when retrieving a call.
*/
unsigned int l3id;
/*!
* \brief From associated B channel: B Channel mISDN driver layer ID from mISDN_get_layerid()
* \note Used only for debug display messages.
*/
int addr;
char context[BUFFERSIZE];
/*!
* \brief Incoming call dialplan context identifier.
* \note The "context" string read in from /etc/asterisk/misdn.conf
*/
char context[AST_MAX_CONTEXT];
/*!
* \brief The configured music-on-hold class to use for this call.
* \note The "musicclass" string read in from /etc/asterisk/misdn.conf
*/
char mohinterpret[MAX_MUSICCLASS];
int zero_read_cnt;
//int zero_read_cnt; /* Not used */
/*!
* \brief Number of outgoing audio frames dropped since last debug gripe message.
*/
int dropped_frame_cnt;
/*!
* \brief TRUE if we must do the ringback tones.
* \note The "far_alerting" boolean read in from /etc/asterisk/misdn.conf
*/
int far_alerting;
/*!
* \brief TRUE if NT should disconnect an overlap dialing call when a timeout occurs.
* \note The "nttimeout" boolean read in from /etc/asterisk/misdn.conf
*/
int nttimeout;
/*!
* \brief Other channel call record PID
* \note Value imported from Asterisk environment variable MISDN_PID
*/
int other_pid;
/*!
* \brief Bridged other channel call record
* \note Pointer set when other_pid imported from Asterisk environment
* variable MISDN_PID by either side.
*/
struct chan_list *other_ch;
/*!
* \brief Tone zone sound used for dialtone generation.
* \note Used as a boolean. Non-NULL to prod generation if enabled.
*/
const struct ind_tone_zone_sound *ts;
/*!
* \brief Enables overlap dialing for the set amount of seconds. (0 = Disabled)
* \note The "overlapdial" value read in from /etc/asterisk/misdn.conf
*/
int overlap_dial;
/*!
* \brief Overlap dialing timeout Task ID. -1 if not running.
*/
int overlap_dial_task;
/*!
* \brief overlap_tv access lock.
*/
ast_mutex_t overlap_tv_lock;
/*!
* \brief Overlap timer start time. Timer restarted for every digit received.
*/
struct timeval overlap_tv;
struct chan_list *peer;
//struct chan_list *peer; /* Not used */
/*!
* \brief Next channel call record in the list.
*/
struct chan_list *next;
struct chan_list *prev;
struct chan_list *first;
//struct chan_list *prev; /* Not used */
//struct chan_list *first; /* Not used */
};
@ -317,6 +506,9 @@ static int *misdn_out_calls;
struct chan_list dummy_cl;
/*!
* \brief Global channel call record list head.
*/
struct chan_list *cl_te=NULL;
ast_mutex_t cl_te_lock;
@ -436,7 +628,7 @@ static void print_facility(struct FacParm *fac, struct misdn_bchannel *bc)
break;
#endif
case Fac_CD:
chan_misdn_log(1,bc->port," --> calldeflect to: %s, screened: %s\n", fac->u.CDeflection.DeflectedToNumber,
chan_misdn_log(1,bc->port," --> calldeflect to: %s, presentable: %s\n", fac->u.CDeflection.DeflectedToNumber,
fac->u.CDeflection.PresentationAllowed ? "yes" : "no");
break;
case Fac_AOCDCurrency:
@ -1768,6 +1960,7 @@ static struct ast_cli_entry chan_misdn_clis[] = {
AST_CLI_DEFINE(handle_cli_misdn_toggle_echocancel, "Toggle EchoCancel on mISDN Channel"),
};
/*! \brief Updates caller ID information from config */
static int update_config(struct chan_list *ch, int orig)
{
struct ast_channel *ast;
@ -1956,7 +2149,6 @@ static int read_config(struct chan_list *ch, int orig)
int port;
int hdlc = 0;
char lang[BUFFERSIZE + 1];
char localmusicclass[BUFFERSIZE + 1];
char faxdetect[BUFFERSIZE + 1];
char buf[256];
char buf2[256];
@ -1981,8 +2173,7 @@ static int read_config(struct chan_list *ch, int orig)
misdn_cfg_get(port, MISDN_CFG_LANGUAGE, lang, sizeof(lang));
ast_string_field_set(ast, language, lang);
misdn_cfg_get(port, MISDN_CFG_MUSICCLASS, localmusicclass, sizeof(localmusicclass));
ast_string_field_set(ast, musicclass, localmusicclass);
misdn_cfg_get(port, MISDN_CFG_MUSICCLASS, ch->mohinterpret, sizeof(ch->mohinterpret));
misdn_cfg_get(port, MISDN_CFG_TXGAIN, &bc->txgain, sizeof(bc->txgain));
misdn_cfg_get(port, MISDN_CFG_RXGAIN, &bc->rxgain, sizeof(bc->rxgain));
@ -2048,6 +2239,8 @@ static int read_config(struct chan_list *ch, int orig)
if (orig == ORG_AST) {
char callerid[BUFFERSIZE + 1];
/* ORIGINATOR Asterisk (outgoing call) */
misdn_cfg_get(port, MISDN_CFG_TE_CHOOSE_CHANNEL, &(bc->te_choose_channel), sizeof(bc->te_choose_channel));
if (strstr(faxdetect, "outgoing") || strstr(faxdetect, "both")) {
@ -2071,7 +2264,8 @@ static int read_config(struct chan_list *ch, int orig)
debug_numplan(port, bc->cpnnumplan, "CTON");
ch->overlap_dial = 0;
} else { /** ORIGINATOR MISDN **/
} else {
/* ORIGINATOR MISDN (incoming call) */
char prefix[BUFFERSIZE + 1] = "";
if (strstr(faxdetect, "incoming") || strstr(faxdetect, "both")) {
@ -2317,7 +2511,7 @@ static int misdn_answer(struct ast_channel *ast)
}
if (!p->bc) {
chan_misdn_log(1, 0, " --> Got Answer, but theres no bc obj ??\n");
chan_misdn_log(1, 0, " --> Got Answer, but there is no bc obj ??\n");
ast_queue_hangup_with_cause(ast, AST_CAUSE_PROTOCOL_ERROR);
}
@ -2525,7 +2719,7 @@ static int misdn_indication(struct ast_channel *ast, int cond, const void *data,
start_bc_tones(p);
break;
case AST_CONTROL_HOLD:
ast_moh_start(ast,data,ast->musicclass);
ast_moh_start(ast, data, p->mohinterpret);
chan_misdn_log(1, p->bc->port, " --> *\tHOLD pid:%d\n", p->bc ? p->bc->pid : -1);
break;
case AST_CONTROL_UNHOLD:
@ -3228,6 +3422,8 @@ static struct ast_channel *misdn_request(const char *type, int format, void *dat
char cfg_group[BUFFERSIZE + 1];
struct robin_list *rr = NULL;
/* Group dial */
if (misdn_cfg_is_group_method(group, METHOD_ROUND_ROBIN)) {
chan_misdn_log(4, port, " --> STARTING ROUND ROBIN...\n");
rr = get_robin_position(group);
@ -3319,7 +3515,8 @@ static struct ast_channel *misdn_request(const char *type, int format, void *dat
, group);
return NULL;
}
} else { /* 'Normal' Port dial * Port dial */
} else {
/* 'Normal' Port dial * Port dial */
if (channel)
chan_misdn_log(1, port, " --> preselected_channel: %d\n", channel);
newbc = misdn_lib_get_free_bc(port, channel, 0, dec);
@ -3858,6 +4055,7 @@ static void send_cause2ast(struct ast_channel *ast, struct misdn_bchannel *bc, s
}
/*! \brief Import parameters from the dialplan environment variables */
void import_ch(struct ast_channel *chan, struct misdn_bchannel *bc, struct chan_list *ch)
{
const char *tmp;
@ -3891,6 +4089,7 @@ void import_ch(struct ast_channel *chan, struct misdn_bchannel *bc, struct chan_
}
}
/*! \brief Export parameters to the dialplan environment variables */
void export_ch(struct ast_channel *chan, struct misdn_bchannel *bc, struct chan_list *ch)
{
char tmp[32];
@ -4364,9 +4563,9 @@ cb_events(enum event_e event, struct misdn_bchannel *bc, void *user_data)
}
/*
added support for s extension hope it will help those poor cretains
which haven't overlap dial.
*/
* added support for s extension hope it will help those poor cretains
* which haven't overlap dial.
*/
misdn_cfg_get(bc->port, MISDN_CFG_ALWAYS_IMMEDIATE, &ai, sizeof(ai));
if (ai) {
do_immediate_setup(bc, ch, chan);
@ -5116,7 +5315,8 @@ static int load_module(void)
" jb - Set jitter buffer length, optarg is length\n"
" jt - Set jitter buffer upper threshold, optarg is threshold\n"
" jn - Disable jitter buffer\n"
" n - disable DSP on channel, disables: Echocancel, DTMF Detection and Volume Control.\n"
" n - Disable mISDN DSP on channel.\n"
" Disables: echo cancel, DTMF detection, and volume control.\n"
" p - Caller ID presentation,\n"
" optarg is either 'allowed' or 'restricted'\n"
" s - Send Non-inband DTMF as inband\n"

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