[asterisk-commits] rmudgett: trunk r168192 - in /trunk: ./ channels/chan_misdn.c
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Fri Jan 9 15:43:40 CST 2009
Author: rmudgett
Date: Fri Jan 9 15:43:30 2009
New Revision: 168192
URL: http://svn.digium.com/view/asterisk?view=rev&rev=168192
Log:
Merged revisions 168191 via svnmerge from
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.
........
Modified:
trunk/ (props changed)
trunk/channels/chan_misdn.c
Propchange: trunk/
------------------------------------------------------------------------------
Binary property 'branch-1.4-merged' - no diff available.
Modified: trunk/channels/chan_misdn.c
URL: http://svn.digium.com/view/asterisk/trunk/channels/chan_misdn.c?view=diff&rev=168192&r1=168191&r2=168192
==============================================================================
--- trunk/channels/chan_misdn.c (original)
+++ trunk/channels/chan_misdn.c Fri Jan 9 15:43:30 2009
@@ -142,83 +142,272 @@
#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];
-
- int zero_read_cnt;
+ /*!
+ * \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; /* 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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
}
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 @@
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 @@
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 @@
, 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 @@
}
+/*! \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 @@
}
}
+/*! \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 @@
}
/*
- 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 @@
" 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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