[Asterisk-code-review] res_tonedetect: Add call progress tone detection (asterisk[18])
Friendly Automation
asteriskteam at digium.com
Fri Nov 19 08:10:55 CST 2021
Friendly Automation has submitted this change. ( https://gerrit.asterisk.org/c/asterisk/+/17548 )
Change subject: res_tonedetect: Add call progress tone detection
......................................................................
res_tonedetect: Add call progress tone detection
Makes basic call progress tone detection available
in a tech-agnostic manner with the addition of the
ToneScan application. This can determine if the channel
has encountered a busy signal, SIT tones, dial tone,
modem, fax machine, etc. A few basic async progress
tone detect options are also added to the TONE_DETECT
function.
ASTERISK-29720 #close
Change-Id: Ia02437e0450473031e294798b8cb421fb8f24e90
---
A doc/CHANGES-staging/tonescan.txt
M res/res_tonedetect.c
2 files changed, 364 insertions(+), 7 deletions(-)
Approvals:
George Joseph: Looks good to me, approved
Friendly Automation: Approved for Submit
diff --git a/doc/CHANGES-staging/tonescan.txt b/doc/CHANGES-staging/tonescan.txt
new file mode 100644
index 0000000..cbed34f
--- /dev/null
+++ b/doc/CHANGES-staging/tonescan.txt
@@ -0,0 +1,6 @@
+Subject: ToneScan application
+
+A new application, ToneScan, allows for
+synchronous detection of call progress
+signals such as dial tone, busy tone,
+Special Information Tones, and modems.
diff --git a/res/res_tonedetect.c b/res/res_tonedetect.c
index 1d5db83..b65a69b 100644
--- a/res/res_tonedetect.c
+++ b/res/res_tonedetect.c
@@ -93,13 +93,90 @@
<ref type="application">PlayTones</ref>
</see-also>
</application>
+ <application name="ToneScan" language="en_US">
+ <synopsis>
+ Wait for period of time while scanning for call progress tones
+ </synopsis>
+ <syntax>
+ <parameter name="zone" required="false">
+ <para>Call progress zone. Default is the system default.</para>
+ </parameter>
+ <parameter name="timeout" required="false">
+ <para>Maximum amount of time, in seconds, to wait for call progress
+ or signal tones. Default is forever.</para>
+ </parameter>
+ <parameter name="threshold" required="false">
+ <para>DSP threshold required for a match. A higher number will
+ require a longer match and may reduce false positives, at the
+ expense of false negatives. Default is 1.</para>
+ </parameter>
+ <parameter name="options" required="false">
+ <optionlist>
+ <option name="f">
+ <para>Enable fax machine detection. By default, this is disabled.</para>
+ </option>
+ <option name="v">
+ <para>Enable voice detection. By default, this is disabled.</para>
+ </option>
+ </optionlist>
+ </parameter>
+ </syntax>
+ <description>
+ <para>Waits for a a distinguishable call progress tone and then exits.
+ Unlike a conventional scanner, this is not currently capable of
+ scanning for modem carriers.</para>
+ <variablelist>
+ <variable name="TONESCANSTATUS">
+ This indicates the result of the scan.
+ <value name="RINGING">
+ Audible ringback tone
+ </value>
+ <value name="BUSY">
+ Busy tone
+ </value>
+ <value name="SIT">
+ Special Information Tones
+ </value>
+ <value name="VOICE">
+ Human voice detected
+ </value>
+ <value name="DTMF">
+ DTMF digit
+ </value>
+ <value name="FAX">
+ Fax (answering)
+ </value>
+ <value name="MODEM">
+ Modem (answering)
+ </value>
+ <value name="DIALTONE">
+ Dial tone
+ </value>
+ <value name="NUT">
+ UK Number Unobtainable tone
+ </value>
+ <value name="TIMEOUT">
+ Timeout reached before any positive detection
+ </value>
+ <value name="HANGUP">
+ Caller hung up before any positive detection
+ </value>
+ </variable>
+ </variablelist>
+ </description>
+ <see-also>
+ <ref type="application">WaitForTone</ref>
+ </see-also>
+ </application>
<function name="TONE_DETECT" language="en_US">
<synopsis>
Asynchronously detects a tone
</synopsis>
<syntax>
<parameter name="freq" required="true">
- <para>Frequency of the tone to detect.</para>
+ <para>Frequency of the tone to detect. To disable frequency
+ detection completely (e.g. for signal detection only),
+ specify 0 for the frequency.</para>
</parameter>
<parameter name="duration_ms" required="false">
<para>Minimum duration of tone, in ms. Default is 500ms.
@@ -108,6 +185,18 @@
</parameter>
<parameter name="options">
<optionlist>
+ <option name="a">
+ <para>Match immediately on Special Information Tones, instead of or in addition
+ to a particular frequency.</para>
+ </option>
+ <option name="b">
+ <para>Match immediately on a busy signal, instead of or in addition to
+ a particular frequency.</para>
+ </option>
+ <option name="c">
+ <para>Match immediately on a dial tone, instead of or in addition to
+ a particular frequency.</para>
+ </option>
<option name="d">
<para>Custom decibel threshold to use. Default is 16.</para>
</option>
@@ -147,10 +236,18 @@
<literal>rx</literal> to get the number of times a tone has been detected in the
RX direction.</para>
<example title="intercept2600">
- same => n,Set(TONE_DETECT(2600,1000,g(got-2600,s,1))=)
+ same => n,Set(TONE_DETECT(2600,1000,g(got-2600,s,1))=) ; detect 2600 Hz
same => n,Wait(15)
same => n,NoOp(${TONE_DETECT(rx)})
</example>
+ <example title="dropondialtone">
+ same => n,Set(TONE_DETECT(0,,bg(my-hangup,s,1))=) ; disconnect a call if we hear a busy signal
+ same => n,Goto(somewhere-else)
+ same => n(myhangup),Hangup()
+ </example>
+ <example title="removedetector">
+ same => n,Set(TONE_DETECT(0,,x)=) ; remove the detector from the channel
+ </example>
</description>
</function>
***/
@@ -170,6 +267,7 @@
int txcount;
int rxcount;
int hitsrequired;
+ int signalfeatures;
};
enum td_opts {
@@ -181,6 +279,9 @@
OPT_DECIBEL = (1 << 6),
OPT_SQUELCH = (1 << 7),
OPT_HITS_REQ = (1 << 8),
+ OPT_SIT = (1 << 9),
+ OPT_BUSY = (1 << 10),
+ OPT_DIALTONE = (1 << 11),
};
enum {
@@ -193,6 +294,9 @@
};
AST_APP_OPTIONS(td_opts, {
+ AST_APP_OPTION('a', OPT_SIT),
+ AST_APP_OPTION('b', OPT_BUSY),
+ AST_APP_OPTION('c', OPT_DIALTONE),
AST_APP_OPTION_ARG('d', OPT_DECIBEL, OPT_ARG_DECIBEL),
AST_APP_OPTION_ARG('g', OPT_GOTO_RX, OPT_ARG_GOTO_RX),
AST_APP_OPTION_ARG('h', OPT_GOTO_TX, OPT_ARG_GOTO_TX),
@@ -230,6 +334,7 @@
{
struct ast_datastore *datastore = NULL;
struct detect_information *di = NULL;
+ int match = 0;
/* If the audiohook is stopping it means the channel is shutting down.... but we let the datastore destroy take care of it */
if (audiohook->status == AST_AUDIOHOOK_STATUS_DONE) {
@@ -258,6 +363,7 @@
char result = frame->subclass.integer;
if (result == 'q') {
int now;
+ match = 1;
if (direction == AST_AUDIOHOOK_DIRECTION_READ) {
di->rxcount = di->rxcount + 1;
now = di->rxcount;
@@ -275,6 +381,42 @@
}
}
}
+ if (di->signalfeatures && !match) { /* skip unless there are call progress/signal options */
+ int tstate, tcount;
+ tcount = ast_dsp_get_tcount(di->dsp);
+ tstate = ast_dsp_get_tstate(di->dsp);
+ if (tstate > 0) {
+ ast_debug(3, "tcount: %d, tstate: %d\n", tcount, tstate);
+ switch (tstate) {
+ case DSP_TONE_STATE_DIALTONE:
+ if (di->signalfeatures & DSP_FEATURE_WAITDIALTONE) {
+ match = 1;
+ }
+ break;
+ case DSP_TONE_STATE_BUSY:
+ if (di->signalfeatures & DSP_PROGRESS_BUSY) {
+ match = 1;
+ }
+ break;
+ case DSP_TONE_STATE_SPECIAL3:
+ if (di->signalfeatures & DSP_PROGRESS_CONGESTION) {
+ match = 1;
+ }
+ break;
+ default: /* ignore */
+ break;
+ }
+ if (match) {
+ if (direction == AST_AUDIOHOOK_DIRECTION_READ && di->gotorx) {
+ ast_async_parseable_goto(chan, di->gotorx);
+ } else if (di->gototx) {
+ ast_async_parseable_goto(chan, di->gototx);
+ } else {
+ ast_debug(3, "Detected call progress signal, but don't know where to go\n");
+ }
+ }
+ }
+ }
/* this could be the duplicated frame or a new one, doesn't matter */
ast_frfree(frame);
return 0;
@@ -326,8 +468,8 @@
ast_log(LOG_WARNING, "Frequency must be an integer: %s\n", f1);
return -1;
}
- if (*freq1 < 1) {
- ast_log(LOG_WARNING, "Sorry, positive frequencies only: %d\n", *freq1);
+ if (*freq1 < 0) {
+ ast_log(LOG_WARNING, "Sorry, no negative frequencies: %d\n", *freq1);
return -1;
}
if (!ast_strlen_zero(f2)) {
@@ -413,6 +555,23 @@
return 0;
}
+static int parse_signal_features(struct ast_flags *flags)
+{
+ int features = 0;
+
+ if (ast_test_flag(flags, OPT_SIT)) {
+ features |= DSP_PROGRESS_CONGESTION;
+ }
+ if (ast_test_flag(flags, OPT_BUSY)) {
+ features |= DSP_PROGRESS_BUSY;
+ }
+ if (ast_test_flag(flags, OPT_DIALTONE)) {
+ features |= DSP_FEATURE_WAITDIALTONE;
+ }
+
+ return features;
+}
+
static int detect_write(struct ast_channel *chan, const char *cmd, char *data, const char *value)
{
char *parse;
@@ -422,6 +581,7 @@
char *opt_args[OPT_ARG_ARRAY_SIZE];
struct ast_dsp *dsp;
int freq1 = 0, freq2 = 0, duration = 500, db = 16, squelch = 0, hitsrequired = 1;
+ int signalfeatures = 0;
AST_DECLARE_APP_ARGS(args,
AST_APP_ARG(freqs);
@@ -461,6 +621,7 @@
return -1;
}
}
+ signalfeatures = parse_signal_features(&flags);
ast_channel_lock(chan);
if (!(datastore = ast_channel_datastore_find(chan, &detect_datastore, NULL))) {
@@ -481,8 +642,12 @@
ast_log(LOG_WARNING, "Unable to allocate DSP!\n");
return -1;
}
- ast_dsp_set_features(dsp, DSP_FEATURE_FREQ_DETECT);
- ast_dsp_set_freqmode(dsp, freq1, duration, db, squelch);
+ di->signalfeatures = signalfeatures; /* we're not including freq detect */
+ if (freq1 > 0) {
+ signalfeatures |= DSP_FEATURE_FREQ_DETECT;
+ ast_dsp_set_freqmode(dsp, freq1, duration, db, squelch);
+ }
+ ast_dsp_set_features(dsp, signalfeatures);
di->dsp = dsp;
di->txcount = 0;
di->rxcount = 0;
@@ -493,7 +658,12 @@
} else {
di = datastore->data;
dsp = di->dsp;
- ast_dsp_set_freqmode(dsp, freq1, duration, db, squelch);
+ di->signalfeatures = signalfeatures; /* we're not including freq detect */
+ if (freq1 > 0) {
+ signalfeatures |= DSP_FEATURE_FREQ_DETECT;
+ ast_dsp_set_freqmode(dsp, freq1, duration, db, squelch);
+ }
+ ast_dsp_set_features(dsp, signalfeatures);
}
di->duration = duration;
di->gotorx = NULL;
@@ -641,6 +811,185 @@
}
static char *waitapp = "WaitForTone";
+static char *scanapp = "ToneScan";
+
+static int scan_exec(struct ast_channel *chan, const char *data)
+{
+ char *appdata;
+ double timeoutf = 0;
+ int timeout = 0;
+ struct ast_frame *frame = NULL, *frame2 = NULL;
+ struct ast_dsp *dsp = NULL, *dsp2 = NULL;
+ struct timeval start;
+ int remaining_time = 0;
+ int features, match = 0, fax = 0, voice = 0, threshold = 1;
+ AST_DECLARE_APP_ARGS(args,
+ AST_APP_ARG(zone);
+ AST_APP_ARG(timeout);
+ AST_APP_ARG(threshold);
+ AST_APP_ARG(options);
+ );
+
+ appdata = ast_strdupa(data);
+ AST_STANDARD_APP_ARGS(args, appdata);
+
+ if (!ast_strlen_zero(args.timeout) && (sscanf(args.timeout, "%30lf", &timeoutf) != 1 || timeout < 0)) {
+ ast_log(LOG_WARNING, "Invalid timeout: %s\n", args.timeout);
+ pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "ERROR");
+ return -1;
+ }
+ if (!ast_strlen_zero(args.threshold) && (ast_str_to_int(args.threshold, &threshold) || threshold < 1)) {
+ ast_log(LOG_WARNING, "Invalid threshold: %s\n", args.threshold);
+ pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "ERROR");
+ return -1;
+ }
+ timeout = 1000 * timeoutf;
+
+ if (!ast_strlen_zero(args.options) && strchr(args.options, 'f')) {
+ fax = 1;
+ }
+ if (!ast_strlen_zero(args.options) && strchr(args.options, 'v')) {
+ voice = 1;
+ }
+
+ if (!(dsp = ast_dsp_new())) {
+ ast_log(LOG_WARNING, "Unable to allocate DSP!\n");
+ pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "ERROR");
+ return -1;
+ }
+
+ if (!ast_strlen_zero(args.zone)) {
+ if (ast_dsp_set_call_progress_zone(dsp, args.zone)) {
+ ast_log(LOG_WARNING, "Invalid call progress zone: %s\n", args.zone);
+ pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "ERROR");
+ ast_dsp_free(dsp);
+ return -1;
+ }
+ }
+
+ if (fax) {
+ if (!(dsp2 = ast_dsp_new())) {
+ ast_dsp_free(dsp);
+ ast_log(LOG_WARNING, "Unable to allocate DSP!\n");
+ pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "ERROR");
+ return -1;
+ }
+ }
+
+ features = DSP_PROGRESS_RINGING; /* audible ringback tone */
+ features |= DSP_PROGRESS_BUSY; /* busy signal */
+ features |= DSP_PROGRESS_CONGESTION; /* SIT tones (not reorder!) */
+ features |= DSP_PROGRESS_TALK; /* voice. */
+ features |= DSP_FEATURE_WAITDIALTONE; /* dial tone */
+ features |= DSP_FEATURE_FREQ_DETECT; /* modem answer */
+ if (voice) {
+ features |= DSP_TONE_STATE_TALKING; /* voice */
+ }
+ ast_dsp_set_features(dsp, features);
+ /* all modems begin negotiating with Bell 103. An answering modem just sends mark tone, or 2225 Hz */
+ ast_dsp_set_freqmode(dsp, 2225, 400, 16, 0); /* this needs to be pretty short, or the progress tones code will thing this is voice */
+
+ if (fax) { /* fax detect uses same tone detect internals as modem and causes things to not work as intended, so use a separate DSP if needed. */
+ ast_dsp_set_features(dsp2, DSP_FEATURE_FAX_DETECT); /* fax tone */
+ ast_dsp_set_faxmode(dsp2, DSP_FAXMODE_DETECT_CED); /* we only care about the answering side (CED), not originating (CNG) */
+ }
+
+ ast_debug(1, "Starting tone scan, timeout: %d ms, threshold: %d\n", timeout, threshold);
+ start = ast_tvnow();
+ do {
+ if (timeout > 0) {
+ remaining_time = ast_remaining_ms(start, timeout);
+ if (remaining_time <= 0) {
+ pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "TIMEOUT");
+ break;
+ }
+ }
+ if (ast_waitfor(chan, 1000) > 0) {
+ if (!(frame = ast_read(chan))) {
+ ast_debug(1, "Channel '%s' did not return a frame; probably hung up.\n", ast_channel_name(chan));
+ pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "HANGUP");
+ break;
+ } else if (frame->frametype == AST_FRAME_VOICE) {
+ if (fax) {
+ frame2 = ast_frdup(frame);
+ }
+ frame = ast_dsp_process(chan, dsp, frame);
+ if (frame->frametype == AST_FRAME_DTMF) {
+ char result = frame->subclass.integer;
+ match = 1;
+ if (result == 'q') {
+ pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "MODEM");
+ } else {
+ pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "DTMF");
+ }
+ } else if (fax) {
+ char result;
+ frame2 = ast_dsp_process(chan, dsp2, frame2);
+ result = frame->subclass.integer;
+ if (result == AST_FRAME_DTMF) {
+ if (result == 'e') {
+ pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "FAX");
+ match = 1;
+ } else {
+ ast_debug(1, "Ignoring inactionable event\n"); /* shouldn't happen */
+ }
+ }
+ ast_frfree(frame2);
+ }
+ if (!match) {
+ int tstate, tcount;
+ tcount = ast_dsp_get_tcount(dsp);
+ tstate = ast_dsp_get_tstate(dsp);
+ if (tstate > 0) {
+ ast_debug(3, "tcount: %d, tstate: %d\n", tcount, tstate);
+ if (tcount >= threshold) {
+ match = 1;
+ switch (tstate) {
+ case DSP_TONE_STATE_RINGING:
+ pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "RINGING");
+ break;
+ case DSP_TONE_STATE_DIALTONE:
+ pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "DIALTONE");
+ break;
+ case DSP_TONE_STATE_TALKING:
+ /* even if we don't specify this feature, it's still checked, so we always need to handle it.
+ Even if we are looking for it, we need to wait a while or tones will be interpreted
+ as voice, because this will match first (and this should match last). */
+ if (voice && tcount > 15 && tcount >= threshold) {
+ pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "VOICE");
+ } else {
+ match = 0;
+ }
+ break;
+ case DSP_TONE_STATE_BUSY:
+ pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "BUSY");
+ break;
+ case DSP_TONE_STATE_SPECIAL3:
+ pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "SIT");
+ break;
+ case DSP_TONE_STATE_HUNGUP: /* UK only */
+ pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "NUT");
+ break;
+ default:
+ match = 0;
+ ast_debug(1, "Something else we weren't expecting? tstate: %d, #%d\n", tstate, tcount);
+ }
+ }
+ }
+ }
+ }
+ ast_frfree(frame);
+ } else {
+ pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "HANGUP");
+ }
+ } while (!match && (timeout == 0 || remaining_time > 0));
+ ast_dsp_free(dsp);
+ if (dsp2) {
+ ast_dsp_free(dsp2);
+ }
+
+ return 0;
+}
static struct ast_custom_function detect_function = {
.name = "TONE_DETECT",
@@ -653,6 +1002,7 @@
int res;
res = ast_unregister_application(waitapp);
+ res |= ast_unregister_application(scanapp);
res |= ast_custom_function_unregister(&detect_function);
return res;
@@ -663,6 +1013,7 @@
int res;
res = ast_register_application_xml(waitapp, wait_exec);
+ res |= ast_register_application_xml(scanapp, scan_exec);
res |= ast_custom_function_register(&detect_function);
return res;
--
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Gerrit-Project: asterisk
Gerrit-Branch: 18
Gerrit-Change-Id: Ia02437e0450473031e294798b8cb421fb8f24e90
Gerrit-Change-Number: 17548
Gerrit-PatchSet: 2
Gerrit-Owner: N A <mail at interlinked.x10host.com>
Gerrit-Reviewer: Friendly Automation
Gerrit-Reviewer: George Joseph <gjoseph at digium.com>
Gerrit-MessageType: merged
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