The purpose of this patch is to eliminate struct ast_callerid since it has turned into a miscellaneous collection of various party information. Eliminate struct ast_callerid and replace it with the following struct organization: struct ast_party_name { char *str; int char_set; int presentation; unsigned char valid; }; struct ast_party_number { char *str; int plan; int presentation; unsigned char valid; }; struct ast_party_subaddress { char *str; int type; unsigned char odd_even_indicator; unsigned char valid; }; struct ast_party_id { struct ast_party_name name; struct ast_party_number number; struct ast_party_subaddress subaddress; char *tag; }; struct ast_party_dialed { struct { char *str; int plan; } number; struct ast_party_subaddress subaddress; int transit_network_select; }; struct ast_party_caller { struct ast_party_id id; char *ani; int ani2; }; The new organization adds some new information as well. * The party name and number now have their own presentation value that can be manipulated independently. ISDN supplies the presentation value for the name and number at different times with the possibility that they could be different. * The party name and number now have a valid flag. Before this change the name or number string could be empty if the presentation were restricted. Most channel drivers assume that the name or number is then simply not available instead of indicating that the name or number was restricted. * The party name now has a character set value. SIP and Q.SIG have the ability to indicate what character set a name string is using so it could be presented properly. * The dialed party now has a numbering plan value that could be useful to have available. The various channel drivers will need to be updated to support the new core features as needed. They have simply been converted to supply current functionality at this time. The following items of note were either corrected or enhanced: * The CONNECTEDLINE() and REDIRECTING() dialplan functions were consolidated into func_callerid.c to share party id handling code. * CALLERPRES() is now deprecated because the name and number have their own presentation values. * Fixed app_alarmreceiver.c write_metadata(). The workstring[] could contain garbage. It also can only contain the caller id number so using ast_callerid_parse() on it is silly. There was also a typo in the CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id number string. ast_callerid_parse() alters the given buffer which in this case is the channel's caller id number string. Then using ast_shrink_phone_number() could alter it even more. * Fixed caller ID name and number memory leak in chan_usbradio.c. * Fixed uninitialized char arrays cid_num[] and cid_name[] in sig_analog.c. * Protected access to a caller channel with lock in chan_sip.c. * Clarified intent of code in app_meetme.c sla_ring_station() and dial_trunk(). Also made save all caller ID data instead of just the name and number strings. * Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge() function. * Corrected some weirdness with app_privacy.c's use of caller presentation. Review: https://reviewboard.asterisk.org/r/702/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276347 65c4cc65-6c06-0410-ace0-fbb531ad65f3
253 lines
7.2 KiB
C
253 lines
7.2 KiB
C
/*
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* Asterisk -- An open source telephony toolkit.
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*
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* Copyright (C) 1999 - 2005, Digium, Inc.
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*
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* Mark Spencer <markster@digium.com>
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*
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* See http://www.asterisk.org for more information about
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* the Asterisk project. Please do not directly contact
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* any of the maintainers of this project for assistance;
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* the project provides a web site, mailing lists and IRC
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* channels for your use.
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*
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* This program is free software, distributed under the terms of
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* the GNU General Public License Version 2. See the LICENSE file
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* at the top of the source tree.
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*/
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/*! \file
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*
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* \brief Playback a file with audio detect
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*
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* \author Mark Spencer <markster@digium.com>
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*
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* \ingroup applications
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*/
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#include "asterisk.h"
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ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
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#include "asterisk/lock.h"
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#include "asterisk/file.h"
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#include "asterisk/channel.h"
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#include "asterisk/pbx.h"
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#include "asterisk/module.h"
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#include "asterisk/translate.h"
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#include "asterisk/utils.h"
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#include "asterisk/dsp.h"
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#include "asterisk/app.h"
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/*** DOCUMENTATION
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<application name="BackgroundDetect" language="en_US">
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<synopsis>
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Background a file with talk detect.
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</synopsis>
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<syntax>
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<parameter name="filename" required="true" />
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<parameter name="sil">
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<para>If not specified, defaults to <literal>1000</literal>.</para>
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</parameter>
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<parameter name="min">
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<para>If not specified, defaults to <literal>100</literal>.</para>
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</parameter>
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<parameter name="max">
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<para>If not specified, defaults to <literal>infinity</literal>.</para>
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</parameter>
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<parameter name="analysistime">
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<para>If not specified, defaults to <literal>infinity</literal>.</para>
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</parameter>
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</syntax>
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<description>
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<para>Plays back <replaceable>filename</replaceable>, waiting for interruption from a given digit (the digit
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must start the beginning of a valid extension, or it will be ignored). During
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the playback of the file, audio is monitored in the receive direction, and if
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a period of non-silence which is greater than <replaceable>min</replaceable> ms yet less than
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<replaceable>max</replaceable> ms is followed by silence for at least <replaceable>sil</replaceable> ms,
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which occurs during the first <replaceable>analysistime</replaceable> ms, then the audio playback is
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aborted and processing jumps to the <replaceable>talk</replaceable> extension, if available.</para>
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</description>
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</application>
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***/
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static char *app = "BackgroundDetect";
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static int background_detect_exec(struct ast_channel *chan, const char *data)
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{
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int res = 0;
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char *tmp;
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struct ast_frame *fr;
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int notsilent = 0;
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struct timeval start = { 0, 0 };
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struct timeval detection_start = { 0, 0 };
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int sil = 1000;
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int min = 100;
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int max = -1;
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int analysistime = -1;
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int continue_analysis = 1;
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int x;
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int origrformat = 0;
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struct ast_dsp *dsp = NULL;
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AST_DECLARE_APP_ARGS(args,
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AST_APP_ARG(filename);
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AST_APP_ARG(silence);
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AST_APP_ARG(min);
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AST_APP_ARG(max);
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AST_APP_ARG(analysistime);
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);
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if (ast_strlen_zero(data)) {
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ast_log(LOG_WARNING, "BackgroundDetect requires an argument (filename)\n");
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return -1;
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}
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tmp = ast_strdupa(data);
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AST_STANDARD_APP_ARGS(args, tmp);
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if (!ast_strlen_zero(args.silence) && (sscanf(args.silence, "%30d", &x) == 1) && (x > 0)) {
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sil = x;
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}
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if (!ast_strlen_zero(args.min) && (sscanf(args.min, "%30d", &x) == 1) && (x > 0)) {
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min = x;
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}
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if (!ast_strlen_zero(args.max) && (sscanf(args.max, "%30d", &x) == 1) && (x > 0)) {
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max = x;
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}
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if (!ast_strlen_zero(args.analysistime) && (sscanf(args.analysistime, "%30d", &x) == 1) && (x > 0)) {
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analysistime = x;
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}
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ast_debug(1, "Preparing detect of '%s', sil=%d, min=%d, max=%d, analysistime=%d\n", args.filename, sil, min, max, analysistime);
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do {
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if (chan->_state != AST_STATE_UP) {
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if ((res = ast_answer(chan))) {
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break;
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}
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}
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origrformat = chan->readformat;
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if ((ast_set_read_format(chan, AST_FORMAT_SLINEAR))) {
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ast_log(LOG_WARNING, "Unable to set read format to linear!\n");
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res = -1;
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break;
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}
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if (!(dsp = ast_dsp_new())) {
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ast_log(LOG_WARNING, "Unable to allocate DSP!\n");
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res = -1;
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break;
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}
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ast_stopstream(chan);
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if (ast_streamfile(chan, tmp, chan->language)) {
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ast_log(LOG_WARNING, "ast_streamfile failed on %s for %s\n", chan->name, (char *)data);
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break;
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}
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detection_start = ast_tvnow();
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while (chan->stream) {
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res = ast_sched_wait(chan->sched);
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if ((res < 0) && !chan->timingfunc) {
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res = 0;
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break;
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}
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if (res < 0) {
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res = 1000;
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}
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res = ast_waitfor(chan, res);
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if (res < 0) {
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ast_log(LOG_WARNING, "Waitfor failed on %s\n", chan->name);
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break;
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} else if (res > 0) {
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fr = ast_read(chan);
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if (continue_analysis && analysistime >= 0) {
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/* If we have a limit for the time to analyze voice
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* frames and the time has not expired */
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if (ast_tvdiff_ms(ast_tvnow(), detection_start) >= analysistime) {
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continue_analysis = 0;
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ast_verb(3, "BackgroundDetect: Talk analysis time complete on %s.\n", chan->name);
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}
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}
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if (!fr) {
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res = -1;
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break;
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} else if (fr->frametype == AST_FRAME_DTMF) {
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char t[2];
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t[0] = fr->subclass.integer;
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t[1] = '\0';
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if (ast_canmatch_extension(chan, chan->context, t, 1,
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S_COR(chan->caller.id.number.valid, chan->caller.id.number.str, NULL))) {
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/* They entered a valid extension, or might be anyhow */
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res = fr->subclass.integer;
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ast_frfree(fr);
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break;
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}
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} else if ((fr->frametype == AST_FRAME_VOICE) && (fr->subclass.codec == AST_FORMAT_SLINEAR) && continue_analysis) {
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int totalsilence;
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int ms;
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res = ast_dsp_silence(dsp, fr, &totalsilence);
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if (res && (totalsilence > sil)) {
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/* We've been quiet a little while */
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if (notsilent) {
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/* We had heard some talking */
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ms = ast_tvdiff_ms(ast_tvnow(), start);
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ms -= sil;
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if (ms < 0)
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ms = 0;
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if ((ms > min) && ((max < 0) || (ms < max))) {
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char ms_str[12];
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ast_debug(1, "Found qualified token of %d ms\n", ms);
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/* Save detected talk time (in milliseconds) */
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snprintf(ms_str, sizeof(ms_str), "%d", ms);
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pbx_builtin_setvar_helper(chan, "TALK_DETECTED", ms_str);
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ast_goto_if_exists(chan, chan->context, "talk", 1);
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res = 0;
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ast_frfree(fr);
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break;
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} else {
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ast_debug(1, "Found unqualified token of %d ms\n", ms);
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}
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notsilent = 0;
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}
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} else {
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if (!notsilent) {
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/* Heard some audio, mark the begining of the token */
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start = ast_tvnow();
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ast_debug(1, "Start of voice token!\n");
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notsilent = 1;
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}
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}
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}
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ast_frfree(fr);
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}
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ast_sched_runq(chan->sched);
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}
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ast_stopstream(chan);
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} while (0);
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if (res > -1) {
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if (origrformat && ast_set_read_format(chan, origrformat)) {
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ast_log(LOG_WARNING, "Failed to restore read format for %s to %s\n",
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chan->name, ast_getformatname(origrformat));
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}
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}
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if (dsp) {
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ast_dsp_free(dsp);
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}
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return res;
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}
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static int unload_module(void)
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{
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return ast_unregister_application(app);
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}
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static int load_module(void)
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{
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return ast_register_application_xml(app, background_detect_exec);
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}
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AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Playback with Talk Detection");
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