[Asterisk-code-review] app_sf: Add full tech-agnostic SF support (asterisk[master])
N A
asteriskteam at digium.com
Mon Sep 13 14:07:16 CDT 2021
N A has uploaded this change for review. ( https://gerrit.asterisk.org/c/asterisk/+/16484 )
Change subject: app_sf: Add full tech-agnostic SF support
......................................................................
app_sf: Add full tech-agnostic SF support
Adds tech-agnostic support for the SF signaling protocol.
This includes SF sender and receiver applications and
Dial application integration.
ASTERISK-29496-p3 #do-not-close
Change-Id: I311bbdf596eff79d50dffdc9175fb3f25c99c9ac
---
M apps/app_dial.c
A apps/app_sf.c
A doc/CHANGES-staging/app_sf.txt
M include/asterisk/app.h
M main/app.c
5 files changed, 649 insertions(+), 8 deletions(-)
git pull ssh://gerrit.asterisk.org:29418/asterisk refs/changes/84/16484/1
diff --git a/apps/app_dial.c b/apps/app_dial.c
index 7d55ad5..a61cfa0 100644
--- a/apps/app_dial.c
+++ b/apps/app_dial.c
@@ -158,6 +158,8 @@
<argument name="progress" />
<argument name="mfprogress" />
<argument name="mfwink" />
+ <argument name="sfprogress" />
+ <argument name="sfwink" />
<para>Send the specified DTMF strings <emphasis>after</emphasis> the called
party has answered, but before the call gets bridged. The
<replaceable>called</replaceable> DTMF string is sent to the called party, and the
@@ -170,6 +172,11 @@
If <replaceable>mfwink</replaceable> is specified, its MF is sent
to the called party immediately after receiving a <literal>WINK</literal> message.</para>
<para>See <literal>SendMF</literal> for valid digits.</para>
+ <para>If <replaceable>sfprogress</replaceable> is specified, its SF is sent
+ to the called party immediately after receiving a <literal>PROGRESS</literal> message.
+ If <replaceable>sfwink</replaceable> is specified, its SF is sent
+ to the called party immediately after receiving a <literal>WINK</literal> message.</para>
+ <para>See <literal>SendSF</literal> for valid digits.</para>
</option>
<option name="E">
<para>Enable echoing of sent MF or SF digits back to caller (e.g. "hearpulsing").
@@ -1214,7 +1221,7 @@
char *opt_args[],
struct privacy_args *pa,
const struct cause_args *num_in, int *result, char *dtmf_progress,
- char *mf_progress, char *mf_wink,
+ char *mf_progress, char *mf_wink, char *sf_progress, char *sf_wink,
const int hearpulsing,
const int ignore_cc,
struct ast_party_id *forced_clid, struct ast_party_id *stored_clid,
@@ -1581,6 +1588,14 @@
ast_mf_stream(c, (hearpulsing ? NULL : in),
(hearpulsing ? in : NULL), mf_progress, 50, 55, 120, 65, 0);
}
+ if (!ast_strlen_zero(sf_progress)) {
+ ast_verb(3,
+ "Sending SF '%s' to %s as result of "
+ "receiving a PROGRESS message.\n",
+ sf_progress, (hearpulsing ? "parties" : "called party"));
+ ast_sf_stream(c, (hearpulsing ? NULL : in),
+ (hearpulsing ? in : NULL), sf_progress, 0, 0);
+ }
if (!ast_strlen_zero(dtmf_progress)) {
ast_verb(3,
"Sending DTMF '%s' to the called party as result of "
@@ -1603,6 +1618,14 @@
ast_mf_stream(c, (hearpulsing ? NULL : in),
(hearpulsing ? in : NULL), mf_wink, 50, 55, 120, 65, 0);
}
+ if (!ast_strlen_zero(sf_wink)) {
+ ast_verb(3,
+ "Sending SF '%s' to %s as result of "
+ "receiving a WINK message.\n",
+ sf_wink, (hearpulsing ? "parties" : "called party"));
+ ast_sf_stream(c, (hearpulsing ? NULL : in),
+ (hearpulsing ? in : NULL), sf_wink, 0, 0);
+ }
}
break;
case AST_CONTROL_VIDUPDATE:
@@ -2277,7 +2300,7 @@
struct ast_bridge_config config = { { 0, } };
struct timeval calldurationlimit = { 0, };
char *dtmfcalled = NULL, *dtmfcalling = NULL, *dtmf_progress = NULL;
- char *mf_progress = NULL, *mf_wink = NULL;
+ char *mf_progress = NULL, *mf_wink = NULL, *sf_progress = NULL, *sf_wink = NULL;
struct privacy_args pa = {
.sentringing = 0,
.privdb_val = 0,
@@ -2412,11 +2435,13 @@
}
if (ast_test_flag64(&opts, OPT_SENDDTMF) && !ast_strlen_zero(opt_args[OPT_ARG_SENDDTMF])) {
- mf_wink = opt_args[OPT_ARG_SENDDTMF];
- dtmfcalled = strsep(&mf_wink, ":");
- dtmfcalling = strsep(&mf_wink, ":");
- dtmf_progress = strsep(&mf_wink, ":");
- mf_progress = strsep(&mf_wink, ":");
+ sf_wink = opt_args[OPT_ARG_SENDDTMF];
+ dtmfcalled = strsep(&sf_wink, ":");
+ dtmfcalling = strsep(&sf_wink, ":");
+ dtmf_progress = strsep(&sf_wink, ":");
+ mf_progress = strsep(&sf_wink, ":");
+ mf_wink = strsep(&sf_wink, ":");
+ sf_progress = strsep(&sf_wink, ":");
}
if (ast_test_flag64(&opts, OPT_DURATION_LIMIT) && !ast_strlen_zero(opt_args[OPT_ARG_DURATION_LIMIT])) {
@@ -2893,7 +2918,7 @@
}
peer = wait_for_answer(chan, &out_chans, &to, peerflags, opt_args, &pa, &num, &result,
- dtmf_progress, mf_progress, mf_wink, (ast_test_flag64(&opts, OPT_HEARPULSING) ? 1 : 0),
+ dtmf_progress, mf_progress, mf_wink, sf_progress, sf_wink, (ast_test_flag64(&opts, OPT_HEARPULSING) ? 1 : 0),
ignore_cc, &forced_clid, &stored_clid, &config);
if (!peer) {
diff --git a/apps/app_sf.c b/apps/app_sf.c
new file mode 100644
index 0000000..0c0b44f
--- /dev/null
+++ b/apps/app_sf.c
@@ -0,0 +1,428 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2021, Naveen Albert
+ *
+ * Naveen Albert <asterisk at phreaknet.org>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ *
+ * \brief SF sender and receiver applications
+ *
+ * \author Naveen Albert <asterisk at phreaknet.org>
+ *
+ * \ingroup applications
+ */
+
+/*** MODULEINFO
+ <support_level>extended</support_level>
+ ***/
+
+#include "asterisk.h"
+
+#include "asterisk/file.h"
+#include "asterisk/pbx.h"
+#include "asterisk/channel.h"
+#include "asterisk/dsp.h"
+#include "asterisk/app.h"
+#include "asterisk/module.h"
+#include "asterisk/indications.h"
+#include "asterisk/conversions.h"
+
+/*** DOCUMENTATION
+ <application name="ReceiveSF" language="en_US">
+ <synopsis>
+ Detects SF digits on a channel and saves them to a variable.
+ </synopsis>
+ <syntax>
+ <parameter name="variable" required="true">
+ <para>The input digits will be stored in the given
+ <replaceable>variable</replaceable> name.</para>
+ </parameter>
+ <parameter name="digits" required="false">
+ <para>Maximum number of digits to read. Default is unlimited.</para>
+ </parameter>
+ <parameter name="timeout">
+ <para>The number of seconds to wait for all digits, if greater
+ than <literal>0</literal>. Can be floating point. Default
+ is no timeout.</para>
+ </parameter>
+ <parameter name="frequency">
+ <para>The frequency for which to detect pulsed digits.
+ Default is 2600 Hz.</para>
+ </parameter>
+ <parameter name="options">
+ <optionlist>
+ <option name="d">
+ <para>Delay audio by a frame to try to extra quelch.</para>
+ </option>
+ <option name="e">
+ <para>Allow receiving extra pulses 11 through 16.</para>
+ </option>
+ <option name="m">
+ <para>Mute conference.</para>
+ </option>
+ <option name="q">
+ <para>Quelch SF from in-band.</para>
+ </option>
+ <option name="r">
+ <para>"Radio" mode (relaxed SF).</para>
+ </option>
+ </optionlist>
+ </parameter>
+ </syntax>
+ <description>
+ <para>Reads SF digits from the user in to the given
+ <replaceable>variable</replaceable>.</para>
+ <para>This application does not automatically answer the channel and
+ should be preceded with <literal>Answer</literal> or
+ <literal>Progress</literal> as needed.</para>
+ <variablelist>
+ <variable name="RECEIVESFSTATUS">
+ <para>This is the status of the read operation.</para>
+ <value name="START" />
+ <value name="ERROR" />
+ <value name="HANGUP" />
+ <value name="TIMEOUT" />
+ </variable>
+ </variablelist>
+ </description>
+ <see-also>
+ <ref type="application">ReceiveMF</ref>
+ <ref type="application">SendMF</ref>
+ <ref type="application">Read</ref>
+ </see-also>
+ </application>
+ <application name="SendSF" language="en_US">
+ <synopsis>
+ Sends arbitrary SF digits on the current or specified channel.
+ </synopsis>
+ <syntax>
+ <parameter name="digits" required="true">
+ <para>List of digits 0-9 to send; w for a half-second pause,
+ also f or F for a flash-hook if the channel supports flash-hook,
+ h or H for 250 ms of 2600 Hz,
+ and W for a wink if the channel supports wink.</para>
+ </parameter>
+ <parameter name="frequency" required="false">
+ <para>Frequency to use. (defaults to 2600 Hz).</para>
+ </parameter>
+ <parameter name="channel" required="false">
+ <para>Channel where digits will be played</para>
+ </parameter>
+ </syntax>
+ <description>
+ <para>It will send all digits or terminate if it encounters an error.</para>
+ </description>
+ <see-also>
+ <ref type="application">SendDTMF</ref>
+ <ref type="application">SendMF</ref>
+ <ref type="application">ReceiveMF</ref>
+ <ref type="application">ReceiveSF</ref>
+ </see-also>
+ </application>
+ ***/
+
+enum read_option_flags {
+ OPT_DELAY = (1 << 0),
+ OPT_MUTE = (1 << 1),
+ OPT_QUELCH = (1 << 2),
+ OPT_RELAXED = (1 << 3),
+ OPT_EXTRAPULSES = (1 << 4),
+};
+
+AST_APP_OPTIONS(read_app_options, {
+ AST_APP_OPTION('d', OPT_DELAY),
+ AST_APP_OPTION('e', OPT_EXTRAPULSES),
+ AST_APP_OPTION('m', OPT_MUTE),
+ AST_APP_OPTION('q', OPT_QUELCH),
+ AST_APP_OPTION('r', OPT_RELAXED),
+});
+
+static const char *readsf_name = "ReceiveSF";
+static const char sendsf_name[] = "SendSF";
+
+static int read_sf_digits(struct ast_channel *chan, char *buf, int timeout, int maxdigits, int freq, int features, int extrapulses) {
+ /* Bell System Technical Journal 39 (Nov. 1960) */
+ #define SF_MIN_OFF 25
+ #define SF_ON 67
+ #define SF_BETWEEN 600
+ #define SF_MIN_DETECT 50
+
+ struct ast_dsp *dsp = NULL;
+ struct ast_frame *frame = NULL;
+ struct timeval start, pulsetimer, digittimer;
+ int remaining_time = timeout;
+ char *str = buf;
+ int hits = 0, digits_read = 0;
+ unsigned short int sf_on = 0;
+
+ if (!(dsp = ast_dsp_new())) {
+ ast_log(LOG_WARNING, "Unable to allocate DSP!\n");
+ pbx_builtin_setvar_helper(chan, "RECEIVESFSTATUS", "ERROR");
+ return -1;
+ }
+ ast_dsp_set_features(dsp, DSP_FEATURE_FREQ_DETECT);
+ /* tolerance is 46 to 76% make break at 8 to 12 pps */
+ ast_dsp_set_freqmode(dsp, freq, SF_MIN_DETECT, 16, 0);
+
+ start = ast_tvnow();
+ *str = 0; /* start with empty output buffer */
+
+ while (timeout == 0 || remaining_time > 0) {
+ if (timeout > 0) {
+ remaining_time = ast_remaining_ms(start, timeout);
+ if (remaining_time <= 0) {
+ pbx_builtin_setvar_helper(chan, "RECEIVESFSTATUS", "TIMEOUT");
+ break;
+ }
+ }
+ if (ast_waitfor(chan, 1000) > 0) {
+ frame = ast_read(chan);
+ if (!frame) {
+ ast_debug(1, "Channel '%s' did not return a frame; probably hung up.\n", ast_channel_name(chan));
+ pbx_builtin_setvar_helper(chan, "RECEIVESFSTATUS", "HANGUP");
+ break;
+ } else if (frame->frametype == AST_FRAME_VOICE) {
+ frame = ast_dsp_process(chan, dsp, frame);
+ if (frame->frametype == AST_FRAME_DTMF) {
+ char result = frame->subclass.integer;
+ if (result == 'q') {
+ sf_on = 1;
+ pulsetimer = ast_tvnow(); /* reset the pulse timer */
+ /* now, we need at least a 33ms pause to register the pulse */
+ }
+ } else {
+ if (sf_on) {
+ int timeleft = ast_remaining_ms(pulsetimer, SF_MIN_OFF);
+ if (timeleft <= 0) {
+ sf_on = 0;
+ /* The pulse needs to end no more than 30ms after we detected it */
+ if (timeleft > -30) {
+ hits++;
+ digittimer = ast_tvnow(); /* reset the digit timer */
+ ast_debug(5, "Detected SF pulse (pulse #%d)\n", hits);
+ if (!(dsp = ast_dsp_new())) {
+ ast_log(LOG_WARNING, "Unable to allocate DSP!\n");
+ pbx_builtin_setvar_helper(chan, "RECEIVESFSTATUS", "ERROR");
+ return -1;
+ }
+ ast_dsp_set_features(dsp, DSP_FEATURE_FREQ_DETECT);
+ ast_dsp_set_freqmode(dsp, freq, SF_MIN_DETECT, 16, 0);
+ } else {
+ ast_debug(5, "SF noise, ignoring, time elapsed was %d ms\n", timeleft);
+ }
+ }
+ } else if (hits > 0 && ast_remaining_ms(digittimer, SF_BETWEEN) <= 0) {
+ /* has the digit finished? */
+ ast_debug(2, "Received SF digit: %d\n", hits);
+ digits_read++;
+ if (hits > 10) {
+ if (extrapulses) {
+ /* dahdi-base.c translates 11 to * and 12 to # */
+ if (hits == 11) {
+ hits = '*';
+ } else if (hits == 12) {
+ hits = '#';
+ } else if (hits == 13) {
+ hits = 'D';
+ } else if (hits == 14) {
+ hits = 'C';
+ } else if (hits == 15) {
+ hits = 'B';
+ } else if (hits == 16) {
+ hits = 'A';
+ } else {
+ ast_debug(3, "Got %d SF pulses, is someone playing with the phone?\n", hits);
+ hits = 'A';
+ }
+ *str++ = hits;
+ } else {
+ ast_debug(2, "Got more than 10 pulses, truncating to 10\n");
+ hits = 0; /* 10 dial pulses = digit 0 */
+ *str++ = hits + '0';
+ }
+ } else {
+ if (hits == 10) {
+ hits = 0; /* 10 dial pulses = digit 0 */
+ }
+ *str++ = hits + '0';
+ }
+ *str = 0;
+ hits = 0;
+ if (maxdigits > 0 && digits_read >= maxdigits) {
+ pbx_builtin_setvar_helper(chan, "RECEIVESFSTATUS", "START");
+ break;
+ }
+ }
+ }
+ }
+ } else {
+ pbx_builtin_setvar_helper(chan, "RECEIVESFSTATUS", "HANGUP");
+ }
+ }
+ if (dsp) {
+ ast_dsp_free(dsp);
+ }
+ ast_debug(3, "channel '%s' - event loop stopped { timeout: %d, remaining_time: %d }\n", ast_channel_name(chan), timeout, remaining_time);
+ return 0;
+}
+
+static int read_sf_exec(struct ast_channel *chan, const char *data)
+{
+ char tmp[256] = "";
+ double tosec;
+ struct ast_flags flags = {0};
+ char *argcopy = NULL;
+ int features = 0, digits = 0, to = 0, freq = 2600;
+
+ AST_DECLARE_APP_ARGS(arglist,
+ AST_APP_ARG(variable);
+ AST_APP_ARG(digits);
+ AST_APP_ARG(timeout);
+ AST_APP_ARG(freq);
+ AST_APP_ARG(options);
+ );
+
+ if (ast_strlen_zero(data)) {
+ ast_log(LOG_WARNING, "ReceiveSF requires an argument (variable)\n");
+ return -1;
+ }
+
+ argcopy = ast_strdupa(data);
+
+ AST_STANDARD_APP_ARGS(arglist, argcopy);
+
+ if (!ast_strlen_zero(arglist.options)) {
+ ast_app_parse_options(read_app_options, &flags, NULL, arglist.options);
+ }
+
+ if (!ast_strlen_zero(arglist.timeout)) {
+ tosec = atof(arglist.timeout);
+ if (tosec <= 0) {
+ to = 0;
+ } else {
+ to = tosec * 1000.0;
+ }
+ }
+
+ if (!ast_strlen_zero(arglist.digits) && (ast_str_to_int(arglist.digits, &digits) || digits <= 0)) {
+ ast_log(LOG_WARNING, "Invalid number of digits: %s\n", arglist.digits);
+ return -1;
+ }
+
+ if (!ast_strlen_zero(arglist.freq) && (ast_str_to_int(arglist.freq, &freq) || freq <= 0)) {
+ ast_log(LOG_WARNING, "Invalid freq: %s\n", arglist.freq);
+ return -1;
+ }
+
+ if (ast_strlen_zero(arglist.variable)) {
+ ast_log(LOG_WARNING, "Invalid! Usage: ReceiveSF(variable[,timeout][,option])\n");
+ return -1;
+ }
+
+ if (ast_test_flag(&flags, OPT_DELAY)) {
+ features |= DSP_DIGITMODE_MUTEMAX;
+ }
+
+ if (ast_test_flag(&flags, OPT_MUTE)) {
+ features |= DSP_DIGITMODE_MUTECONF;
+ }
+
+ if (!ast_test_flag(&flags, OPT_QUELCH)) {
+ features |= DSP_DIGITMODE_NOQUELCH;
+ }
+
+ if (ast_test_flag(&flags, OPT_RELAXED)) {
+ features |= DSP_DIGITMODE_RELAXDTMF;
+ }
+
+ read_sf_digits(chan, tmp, to, digits, freq, features, ast_test_flag(&flags, OPT_EXTRAPULSES));
+ pbx_builtin_setvar_helper(chan, arglist.variable, tmp);
+ if (!ast_strlen_zero(tmp)) {
+ ast_verb(3, "MF digits received: '%s'\n", tmp);
+ } else {
+ ast_verb(3, "No MF digits received.\n");
+ }
+ return 0;
+}
+
+static int sendsf_exec(struct ast_channel *chan, const char *vdata)
+{
+ int res;
+ char *data;
+ int frequency = 2600;
+ struct ast_channel *chan_found = NULL;
+ struct ast_channel *chan_dest = chan;
+ struct ast_channel *chan_autoservice = NULL;
+ AST_DECLARE_APP_ARGS(args,
+ AST_APP_ARG(digits);
+ AST_APP_ARG(frequency);
+ AST_APP_ARG(channel);
+ );
+
+ if (ast_strlen_zero(vdata)) {
+ ast_log(LOG_WARNING, "SendSF requires an argument\n");
+ return 0;
+ }
+
+ data = ast_strdupa(vdata);
+ AST_STANDARD_APP_ARGS(args, data);
+
+ if (ast_strlen_zero(args.digits)) {
+ ast_log(LOG_WARNING, "The digits argument is required (0-9,wf)\n");
+ return 0;
+ }
+ if (!ast_strlen_zero(args.frequency) && (ast_str_to_int(args.frequency, &frequency) || frequency < 1)) {
+ ast_log(LOG_WARNING, "Invalid duration: %s\n", args.frequency);
+ return -1;
+ }
+ if (!ast_strlen_zero(args.channel)) {
+ chan_found = ast_channel_get_by_name(args.channel);
+ if (!chan_found) {
+ ast_log(LOG_WARNING, "No such channel: %s\n", args.channel);
+ return 0;
+ }
+ chan_dest = chan_found;
+ if (chan_found != chan) {
+ chan_autoservice = chan;
+ }
+ }
+ res = ast_sf_stream(chan_dest, chan_autoservice, NULL, args.digits, frequency, 0);
+ ast_channel_cleanup(chan_found);
+
+ return chan_autoservice ? 0 : res;
+}
+
+static int unload_module(void)
+{
+ int res;
+
+ res = ast_unregister_application(readsf_name);
+ res |= ast_unregister_application(sendsf_name);
+
+ return res;
+}
+
+static int load_module(void)
+{
+ int res;
+
+ res = ast_register_application_xml(readsf_name, read_sf_exec);
+ res |= ast_register_application_xml(sendsf_name, sendsf_exec);
+
+ return res;
+}
+
+AST_MODULE_INFO_STANDARD_EXTENDED(ASTERISK_GPL_KEY, "SF Sender and Receiver Applications");
diff --git a/doc/CHANGES-staging/app_sf.txt b/doc/CHANGES-staging/app_sf.txt
new file mode 100644
index 0000000..347ba5d
--- /dev/null
+++ b/doc/CHANGES-staging/app_sf.txt
@@ -0,0 +1,5 @@
+Subject: app_sf
+
+Adds tech-agnostic support for SF signaling through
+SF sender and receiver applications, along with Dial
+integration.
diff --git a/include/asterisk/app.h b/include/asterisk/app.h
index ec34f28..5a8ca86 100644
--- a/include/asterisk/app.h
+++ b/include/asterisk/app.h
@@ -923,6 +923,34 @@
void ast_unreplace_sigchld(void);
/*!
+ * \brief Send a string of SF digits to a channel
+ *
+ * \param chan The channel that will receive the SF digits
+ * \param peer (optional) Peer channel that will be autoserviced while the
+ * primary channel is receiving SF
+ * \param chan2 A second channel that will simultaneously receive SF digits.
+ * This option may only be used if is_external is 0.
+ * \param digits This is a string of characters representing the SF digits
+ * to be sent to the channel. Valid characters are
+ * "0123456789". Note: You can pass arguments 'f' or
+ * 'F', if you want to Flash the channel (if supported by the
+ * channel), or 'w' or 'W' to add a wink (if supported by the
+ * channel).
+ * \param between This is the number of milliseconds to wait in between each
+ * SF digit. If zero milliseconds is specified, then the
+ * default value of 50 will be used.
+ * \param duration This is the duration that each numeric SF digit should have.
+ * Default value is 55.
+ * \param is_external 1 if called by a thread that is not the channel's media
+ * handler thread, 0 if called by the channel's media handler
+ * thread.
+ *
+ * \retval 0 on success.
+ * \retval -1 on failure or a channel hung up.
+ */
+int ast_sf_stream(struct ast_channel *chan, struct ast_channel *peer, struct ast_channel *chan2, const char *digits, int frequency, int is_external);
+
+/*!
* \brief Send a string of MF digits to a channel
*
* \param chan The channel that will receive the MF digits.
diff --git a/main/app.c b/main/app.c
index 30e83c0..0774ee6 100644
--- a/main/app.c
+++ b/main/app.c
@@ -833,6 +833,145 @@
return 0;
}
+static int sf_stream(struct ast_channel *chan, struct ast_channel *chan2, const char *digits, int frequency, int is_external)
+{
+ /* Bell System Technical Journal 39 (Nov. 1960) */
+ #define SF_ON 67
+ #define SF_OFF 33
+ #define SF_BETWEEN 600
+
+ const char *ptr;
+ int res;
+ struct ast_silence_generator *silgen = NULL, *silgen2 = NULL;
+ char *freq;
+ int (*my_sleep)(struct ast_channel *chan, int ms);
+
+ if (is_external) {
+ my_sleep = external_sleep;
+ } else {
+ my_sleep = ast_safe_sleep;
+ }
+
+ /* Need a quiet time before sending digits. */
+ if (ast_opt_transmit_silence) {
+ silgen = ast_channel_start_silence_generator(chan);
+ if (chan2) {
+ silgen2 = ast_channel_start_silence_generator(chan2);
+ }
+ }
+ if (chan2) {
+ ast_autoservice_start(chan2);
+ }
+ res = my_sleep(chan, 100);
+ if (chan2) {
+ ast_autoservice_stop(chan2);
+ }
+ if (res) {
+ goto sf_stream_cleanup;
+ }
+
+ freq = ast_malloc(32);
+ /* pauses need to send audio, so send 0 Hz */
+ snprintf(freq, 31, "%d/%d,%d/%d", frequency, SF_ON, 0, SF_OFF);
+
+ for (ptr = digits; *ptr; ptr++) {
+ if (*ptr == 'w') {
+ /* 'w' -- wait half a second */
+ res = my_sleep(chan, 500);
+ if (res) {
+ break;
+ }
+ } else if (*ptr == 'h' || *ptr == 'H') {
+ /* 'h' -- 2600 Hz for half a second, but
+ only to far end of trunk, not near end */
+ ast_playtones_start(chan, 0, "2600", 0);
+ if (chan2) {
+ ast_playtones_start(chan2, 0, "0", 0);
+ ast_autoservice_start(chan2);
+ }
+ res = my_sleep(chan, 250);
+ ast_senddigit_mf_end(chan);
+ if (chan2) {
+ ast_autoservice_stop(chan2);
+ ast_senddigit_mf_end(chan2);
+ }
+ if (res) {
+ break;
+ }
+ } else if (strchr("0123456789*#ABCDabcdwWfF", *ptr)) {
+ if (*ptr == 'f' || *ptr == 'F') {
+ /* ignore return values if not supported by channel */
+ ast_indicate(chan, AST_CONTROL_FLASH);
+ } else if (*ptr == 'W') {
+ /* ignore return values if not supported by channel */
+ ast_indicate(chan, AST_CONTROL_WINK);
+ } else {
+ /* Character represents valid SF */
+ int beeps;
+ if (*ptr == '*') {
+ beeps = 11;
+ } else if (*ptr == '#') {
+ beeps = 12;
+ } else if (*ptr == 'D') {
+ beeps = 13;
+ } else if (*ptr == 'C') {
+ beeps = 14;
+ } else if (*ptr == 'B') {
+ beeps = 15;
+ } else if (*ptr == 'A') {
+ beeps = 16;
+ } else {
+ beeps = (*ptr == '0') ? 10 : *ptr - '0';
+ }
+ while (beeps-- > 0) {
+ ast_playtones_start(chan, 0, freq, 0);
+ if (chan2) {
+ ast_playtones_start(chan2, 0, freq, 0);
+ ast_autoservice_start(chan2);
+ }
+ res = my_sleep(chan, SF_ON + SF_OFF);
+ ast_senddigit_mf_end(chan);
+ if (chan2) {
+ ast_autoservice_stop(chan2);
+ ast_senddigit_mf_end(chan2);
+ }
+ if (res) {
+ break;
+ }
+ }
+ }
+ /* pause between digits */
+ ast_playtones_start(chan, 0, "0", 0);
+ if (chan2) {
+ ast_playtones_start(chan2, 0, "0", 0);
+ ast_autoservice_start(chan2);
+ }
+ res = my_sleep(chan, SF_BETWEEN);
+ if (chan2) {
+ ast_autoservice_stop(chan2);
+ ast_senddigit_mf_end(chan2);
+ }
+ ast_senddigit_mf_end(chan);
+ if (res) {
+ break;
+ }
+ } else {
+ ast_log(LOG_WARNING, "Illegal SF character '%c' in string. (0-9A-DwWfFhH allowed)\n", *ptr);
+ }
+ }
+ ast_free(freq);
+
+sf_stream_cleanup:
+ if (silgen) {
+ ast_channel_stop_silence_generator(chan, silgen);
+ }
+ if (silgen2) {
+ ast_channel_stop_silence_generator(chan2, silgen2);
+ }
+
+ return res;
+}
+
static int mf_stream(struct ast_channel *chan, struct ast_channel *chan2, const char *digits, int between, unsigned int duration,
unsigned int durationkp, unsigned int durationst, int is_external)
{
@@ -1010,6 +1149,22 @@
return res;
}
+int ast_sf_stream(struct ast_channel *chan, struct ast_channel *peer, struct ast_channel *chan2, const char *digits, int frequency, int is_external)
+{
+ int res;
+ if (frequency <= 0) {
+ frequency = 2600;
+ }
+ if (!is_external && !chan2 && peer && ast_autoservice_start(peer)) {
+ return -1;
+ }
+ res = sf_stream(chan, chan2, digits, frequency, is_external);
+ if (!is_external && !chan2 && peer && ast_autoservice_stop(peer)) {
+ res = -1;
+ }
+ return res;
+}
+
int ast_mf_stream(struct ast_channel *chan, struct ast_channel *peer, struct ast_channel *chan2, const char *digits,
int between, unsigned int duration, unsigned int durationkp, unsigned int durationst, int is_external)
{
--
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Gerrit-Project: asterisk
Gerrit-Branch: master
Gerrit-Change-Id: I311bbdf596eff79d50dffdc9175fb3f25c99c9ac
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Gerrit-PatchSet: 1
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