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asterisk/apps/app_amd.c

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/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 2003 - 2006, Aheeva Technology.
*
* Claude Klimos (claude.klimos@aheeva.com)
*
* 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.
*
* A license has been granted to Digium (via disclaimer) for the use of
* this code.
*/
/*! \file
*
* \brief Answering machine detection
*
* \author Claude Klimos (claude.klimos@aheeva.com)
*/
/*** MODULEINFO
<support_level>extended</support_level>
***/
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/module.h"
#include "asterisk/lock.h"
#include "asterisk/channel.h"
#include "asterisk/dsp.h"
#include "asterisk/pbx.h"
#include "asterisk/config.h"
#include "asterisk/app.h"
/*** DOCUMENTATION
<application name="AMD" language="en_US">
<synopsis>
Attempt to detect answering machines.
</synopsis>
<syntax>
<parameter name="initialSilence" required="false">
<para>Is maximum initial silence duration before greeting.</para>
<para>If this is exceeded set as MACHINE</para>
</parameter>
<parameter name="greeting" required="false">
<para>is the maximum length of a greeting.</para>
<para>If this is exceeded set as MACHINE</para>
</parameter>
<parameter name="afterGreetingSilence" required="false">
<para>Is the silence after detecting a greeting.</para>
<para>If this is exceeded set as HUMAN</para>
</parameter>
<parameter name="totalAnalysis Time" required="false">
<para>Is the maximum time allowed for the algorithm</para>
<para>to decide HUMAN or MACHINE</para>
</parameter>
<parameter name="miniumWordLength" required="false">
<para>Is the minimum duration of Voice considered to be a word</para>
</parameter>
<parameter name="betweenWordSilence" required="false">
<para>Is the minimum duration of silence after a word to
consider the audio that follows to be a new word</para>
</parameter>
<parameter name="maximumNumberOfWords" required="false">
<para>Is the maximum number of words in a greeting</para>
<para>If this is exceeded set as MACHINE</para>
</parameter>
<parameter name="silenceThreshold" required="false">
<para>How long do we consider silence</para>
</parameter>
<parameter name="maximumWordLength" required="false">
<para>Is the maximum duration of a word to accept.</para>
<para>If exceeded set as MACHINE</para>
</parameter>
</syntax>
<description>
<para>This application attempts to detect answering machines at the beginning
of outbound calls. Simply call this application after the call
has been answered (outbound only, of course).</para>
<para>When loaded, AMD reads amd.conf and uses the parameters specified as
default values. Those default values get overwritten when the calling AMD
with parameters.</para>
<para>This application sets the following channel variables:</para>
<variablelist>
<variable name="AMDSTATUS">
<para>This is the status of the answering machine detection</para>
<value name="MACHINE" />
<value name="HUMAN" />
<value name="NOTSURE" />
<value name="HANGUP" />
</variable>
<variable name="AMDCAUSE">
<para>Indicates the cause that led to the conclusion</para>
<value name="TOOLONG">
Total Time.
</value>
<value name="INITIALSILENCE">
Silence Duration - Initial Silence.
</value>
<value name="HUMAN">
Silence Duration - afterGreetingSilence.
</value>
<value name="LONGGREETING">
Voice Duration - Greeting.
</value>
<value name="MAXWORDLENGTH">
Word Count - maximum number of words.
</value>
</variable>
</variablelist>
</description>
<see-also>
<ref type="application">WaitForSilence</ref>
<ref type="application">WaitForNoise</ref>
</see-also>
</application>
***/
static const char app[] = "AMD";
#define STATE_IN_WORD 1
#define STATE_IN_SILENCE 2
/* Some default values for the algorithm parameters. These defaults will be overwritten from amd.conf */
static int dfltInitialSilence = 2500;
static int dfltGreeting = 1500;
static int dfltAfterGreetingSilence = 800;
static int dfltTotalAnalysisTime = 5000;
static int dfltMinimumWordLength = 100;
static int dfltBetweenWordsSilence = 50;
static int dfltMaximumNumberOfWords = 3;
static int dfltSilenceThreshold = 256;
static int dfltMaximumWordLength = 5000; /* Setting this to a large default so it is not used unless specify it in the configs or command line */
/* Set to the lowest ms value provided in amd.conf or application parameters */
static int dfltMaxWaitTimeForFrame = 50;
static void isAnsweringMachine(struct ast_channel *chan, const char *data)
{
int res = 0;
struct ast_frame *f = NULL;
struct ast_dsp *silenceDetector = NULL;
int dspsilence = 0, framelength = 0;
struct ast_format readFormat;
int inInitialSilence = 1;
int inGreeting = 0;
int voiceDuration = 0;
int silenceDuration = 0;
int iTotalTime = 0;
int iWordsCount = 0;
int currentState = STATE_IN_WORD;
int consecutiveVoiceDuration = 0;
char amdCause[256] = "", amdStatus[256] = "";
char *parse = ast_strdupa(data);
/* Lets set the initial values of the variables that will control the algorithm.
The initial values are the default ones. If they are passed as arguments
when invoking the application, then the default values will be overwritten
by the ones passed as parameters. */
int initialSilence = dfltInitialSilence;
int greeting = dfltGreeting;
int afterGreetingSilence = dfltAfterGreetingSilence;
int totalAnalysisTime = dfltTotalAnalysisTime;
int minimumWordLength = dfltMinimumWordLength;
int betweenWordsSilence = dfltBetweenWordsSilence;
int maximumNumberOfWords = dfltMaximumNumberOfWords;
int silenceThreshold = dfltSilenceThreshold;
int maximumWordLength = dfltMaximumWordLength;
int maxWaitTimeForFrame = dfltMaxWaitTimeForFrame;
AST_DECLARE_APP_ARGS(args,
AST_APP_ARG(argInitialSilence);
AST_APP_ARG(argGreeting);
AST_APP_ARG(argAfterGreetingSilence);
AST_APP_ARG(argTotalAnalysisTime);
AST_APP_ARG(argMinimumWordLength);
AST_APP_ARG(argBetweenWordsSilence);
AST_APP_ARG(argMaximumNumberOfWords);
AST_APP_ARG(argSilenceThreshold);
AST_APP_ARG(argMaximumWordLength);
);
ast_format_clear(&readFormat);
ast_callerid restructuring 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: http://svn.digium.com/svn/asterisk/trunk@276347 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-14 15:48:36 +00:00
ast_verb(3, "AMD: %s %s %s (Fmt: %s)\n", chan->name,
S_COR(chan->caller.ani.number.valid, chan->caller.ani.number.str, "(N/A)"),
ast_callerid restructuring 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: http://svn.digium.com/svn/asterisk/trunk@276347 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-14 15:48:36 +00:00
S_COR(chan->redirecting.from.number.valid, chan->redirecting.from.number.str, "(N/A)"),
ast_getformatname(&chan->readformat));
/* Lets parse the arguments. */
if (!ast_strlen_zero(parse)) {
/* Some arguments have been passed. Lets parse them and overwrite the defaults. */
AST_STANDARD_APP_ARGS(args, parse);
if (!ast_strlen_zero(args.argInitialSilence))
initialSilence = atoi(args.argInitialSilence);
if (!ast_strlen_zero(args.argGreeting))
greeting = atoi(args.argGreeting);
if (!ast_strlen_zero(args.argAfterGreetingSilence))
afterGreetingSilence = atoi(args.argAfterGreetingSilence);
if (!ast_strlen_zero(args.argTotalAnalysisTime))
totalAnalysisTime = atoi(args.argTotalAnalysisTime);
if (!ast_strlen_zero(args.argMinimumWordLength))
minimumWordLength = atoi(args.argMinimumWordLength);
if (!ast_strlen_zero(args.argBetweenWordsSilence))
betweenWordsSilence = atoi(args.argBetweenWordsSilence);
if (!ast_strlen_zero(args.argMaximumNumberOfWords))
maximumNumberOfWords = atoi(args.argMaximumNumberOfWords);
if (!ast_strlen_zero(args.argSilenceThreshold))
silenceThreshold = atoi(args.argSilenceThreshold);
if (!ast_strlen_zero(args.argMaximumWordLength))
maximumWordLength = atoi(args.argMaximumWordLength);
} else {
ast_debug(1, "AMD using the default parameters.\n");
}
/* Find lowest ms value, that will be max wait time for a frame */
if (maxWaitTimeForFrame > initialSilence)
maxWaitTimeForFrame = initialSilence;
if (maxWaitTimeForFrame > greeting)
maxWaitTimeForFrame = greeting;
if (maxWaitTimeForFrame > afterGreetingSilence)
maxWaitTimeForFrame = afterGreetingSilence;
if (maxWaitTimeForFrame > totalAnalysisTime)
maxWaitTimeForFrame = totalAnalysisTime;
if (maxWaitTimeForFrame > minimumWordLength)
maxWaitTimeForFrame = minimumWordLength;
if (maxWaitTimeForFrame > betweenWordsSilence)
maxWaitTimeForFrame = betweenWordsSilence;
/* Now we're ready to roll! */
ast_verb(3, "AMD: initialSilence [%d] greeting [%d] afterGreetingSilence [%d] "
"totalAnalysisTime [%d] minimumWordLength [%d] betweenWordsSilence [%d] maximumNumberOfWords [%d] silenceThreshold [%d] maximumWordLength [%d] \n",
initialSilence, greeting, afterGreetingSilence, totalAnalysisTime,
minimumWordLength, betweenWordsSilence, maximumNumberOfWords, silenceThreshold, maximumWordLength);
/* Set read format to signed linear so we get signed linear frames in */
ast_format_copy(&readFormat, &chan->readformat);
if (ast_set_read_format_by_id(chan, AST_FORMAT_SLINEAR) < 0 ) {
ast_log(LOG_WARNING, "AMD: Channel [%s]. Unable to set to linear mode, giving up\n", chan->name );
pbx_builtin_setvar_helper(chan , "AMDSTATUS", "");
pbx_builtin_setvar_helper(chan , "AMDCAUSE", "");
return;
}
/* Create a new DSP that will detect the silence */
if (!(silenceDetector = ast_dsp_new())) {
ast_log(LOG_WARNING, "AMD: Channel [%s]. Unable to create silence detector :(\n", chan->name );
pbx_builtin_setvar_helper(chan , "AMDSTATUS", "");
pbx_builtin_setvar_helper(chan , "AMDCAUSE", "");
return;
}
/* Set silence threshold to specified value */
ast_dsp_set_threshold(silenceDetector, silenceThreshold);
/* Now we go into a loop waiting for frames from the channel */
while ((res = ast_waitfor(chan, 2 * maxWaitTimeForFrame)) > -1) {
/* If we fail to read in a frame, that means they hung up */
if (!(f = ast_read(chan))) {
ast_verb(3, "AMD: Channel [%s]. HANGUP\n", chan->name);
ast_debug(1, "Got hangup\n");
strcpy(amdStatus, "HANGUP");
res = 1;
break;
}
if (f->frametype == AST_FRAME_VOICE || f->frametype == AST_FRAME_NULL || f->frametype == AST_FRAME_CNG) {
/* If the total time exceeds the analysis time then give up as we are not too sure */
if (f->frametype == AST_FRAME_VOICE) {
framelength = (ast_codec_get_samples(f) / DEFAULT_SAMPLES_PER_MS);
} else {
framelength = 2 * maxWaitTimeForFrame;
}
iTotalTime += framelength;
if (iTotalTime >= totalAnalysisTime) {
ast_verb(3, "AMD: Channel [%s]. Too long...\n", chan->name );
ast_frfree(f);
strcpy(amdStatus , "NOTSURE");
sprintf(amdCause , "TOOLONG-%d", iTotalTime);
break;
}
/* Feed the frame of audio into the silence detector and see if we get a result */
if (f->frametype != AST_FRAME_VOICE)
dspsilence += 2 * maxWaitTimeForFrame;
else {
dspsilence = 0;
ast_dsp_silence(silenceDetector, f, &dspsilence);
}
if (dspsilence > 0) {
silenceDuration = dspsilence;
if (silenceDuration >= betweenWordsSilence) {
if (currentState != STATE_IN_SILENCE ) {
ast_verb(3, "AMD: Channel [%s]. Changed state to STATE_IN_SILENCE\n", chan->name);
}
/* Find words less than word duration */
if (consecutiveVoiceDuration < minimumWordLength && consecutiveVoiceDuration > 0){
ast_verb(3, "AMD: Channel [%s]. Short Word Duration: %d\n", chan->name, consecutiveVoiceDuration);
}
currentState = STATE_IN_SILENCE;
consecutiveVoiceDuration = 0;
}
if (inInitialSilence == 1 && silenceDuration >= initialSilence) {
ast_verb(3, "AMD: Channel [%s]. ANSWERING MACHINE: silenceDuration:%d initialSilence:%d\n",
chan->name, silenceDuration, initialSilence);
ast_frfree(f);
strcpy(amdStatus , "MACHINE");
sprintf(amdCause , "INITIALSILENCE-%d-%d", silenceDuration, initialSilence);
res = 1;
break;
}
if (silenceDuration >= afterGreetingSilence && inGreeting == 1) {
ast_verb(3, "AMD: Channel [%s]. HUMAN: silenceDuration:%d afterGreetingSilence:%d\n",
chan->name, silenceDuration, afterGreetingSilence);
ast_frfree(f);
strcpy(amdStatus , "HUMAN");
sprintf(amdCause , "HUMAN-%d-%d", silenceDuration, afterGreetingSilence);
res = 1;
break;
}
} else {
consecutiveVoiceDuration += framelength;
voiceDuration += framelength;
/* If I have enough consecutive voice to say that I am in a Word, I can only increment the
number of words if my previous state was Silence, which means that I moved into a word. */
if (consecutiveVoiceDuration >= minimumWordLength && currentState == STATE_IN_SILENCE) {
iWordsCount++;
ast_verb(3, "AMD: Channel [%s]. Word detected. iWordsCount:%d\n", chan->name, iWordsCount);
currentState = STATE_IN_WORD;
}
if (consecutiveVoiceDuration >= maximumWordLength){
ast_verb(3, "AMD: Channel [%s]. Maximum Word Length detected. [%d]\n", chan->name, consecutiveVoiceDuration);
ast_frfree(f);
strcpy(amdStatus , "MACHINE");
sprintf(amdCause , "MAXWORDLENGTH-%d", consecutiveVoiceDuration);
break;
}
if (iWordsCount >= maximumNumberOfWords) {
ast_verb(3, "AMD: Channel [%s]. ANSWERING MACHINE: iWordsCount:%d\n", chan->name, iWordsCount);
ast_frfree(f);
strcpy(amdStatus , "MACHINE");
sprintf(amdCause , "MAXWORDS-%d-%d", iWordsCount, maximumNumberOfWords);
res = 1;
break;
}
if (inGreeting == 1 && voiceDuration >= greeting) {
ast_verb(3, "AMD: Channel [%s]. ANSWERING MACHINE: voiceDuration:%d greeting:%d\n", chan->name, voiceDuration, greeting);
ast_frfree(f);
strcpy(amdStatus , "MACHINE");
sprintf(amdCause , "LONGGREETING-%d-%d", voiceDuration, greeting);
res = 1;
break;
}
if (voiceDuration >= minimumWordLength ) {
if (silenceDuration > 0)
ast_verb(3, "AMD: Channel [%s]. Detected Talk, previous silence duration: %d\n", chan->name, silenceDuration);
silenceDuration = 0;
}
if (consecutiveVoiceDuration >= minimumWordLength && inGreeting == 0) {
/* Only go in here once to change the greeting flag when we detect the 1st word */
if (silenceDuration > 0)
ast_verb(3, "AMD: Channel [%s]. Before Greeting Time: silenceDuration: %d voiceDuration: %d\n", chan->name, silenceDuration, voiceDuration);
inInitialSilence = 0;
inGreeting = 1;
}
}
}
ast_frfree(f);
}
if (!res) {
/* It took too long to get a frame back. Giving up. */
ast_verb(3, "AMD: Channel [%s]. Too long...\n", chan->name);
strcpy(amdStatus , "NOTSURE");
sprintf(amdCause , "TOOLONG-%d", iTotalTime);
}
/* Set the status and cause on the channel */
pbx_builtin_setvar_helper(chan , "AMDSTATUS" , amdStatus);
pbx_builtin_setvar_helper(chan , "AMDCAUSE" , amdCause);
/* Restore channel read format */
if (readFormat.id && ast_set_read_format(chan, &readFormat))
ast_log(LOG_WARNING, "AMD: Unable to restore read format on '%s'\n", chan->name);
/* Free the DSP used to detect silence */
ast_dsp_free(silenceDetector);
return;
}
static int amd_exec(struct ast_channel *chan, const char *data)
{
isAnsweringMachine(chan, data);
return 0;
}
static int load_config(int reload)
{
struct ast_config *cfg = NULL;
char *cat = NULL;
struct ast_variable *var = NULL;
struct ast_flags config_flags = { reload ? CONFIG_FLAG_FILEUNCHANGED : 0 };
dfltSilenceThreshold = ast_dsp_get_threshold_from_settings(THRESHOLD_SILENCE);
if (!(cfg = ast_config_load("amd.conf", config_flags))) {
ast_log(LOG_ERROR, "Configuration file amd.conf missing.\n");
return -1;
} else if (cfg == CONFIG_STATUS_FILEUNCHANGED) {
return 0;
} else if (cfg == CONFIG_STATUS_FILEINVALID) {
ast_log(LOG_ERROR, "Config file amd.conf is in an invalid format. Aborting.\n");
return -1;
}
cat = ast_category_browse(cfg, NULL);
while (cat) {
if (!strcasecmp(cat, "general") ) {
var = ast_variable_browse(cfg, cat);
while (var) {
if (!strcasecmp(var->name, "initial_silence")) {
dfltInitialSilence = atoi(var->value);
} else if (!strcasecmp(var->name, "greeting")) {
dfltGreeting = atoi(var->value);
} else if (!strcasecmp(var->name, "after_greeting_silence")) {
dfltAfterGreetingSilence = atoi(var->value);
} else if (!strcasecmp(var->name, "silence_threshold")) {
dfltSilenceThreshold = atoi(var->value);
} else if (!strcasecmp(var->name, "total_analysis_time")) {
dfltTotalAnalysisTime = atoi(var->value);
} else if (!strcasecmp(var->name, "min_word_length")) {
dfltMinimumWordLength = atoi(var->value);
} else if (!strcasecmp(var->name, "between_words_silence")) {
dfltBetweenWordsSilence = atoi(var->value);
} else if (!strcasecmp(var->name, "maximum_number_of_words")) {
dfltMaximumNumberOfWords = atoi(var->value);
} else if (!strcasecmp(var->name, "maximum_word_length")) {
dfltMaximumWordLength = atoi(var->value);
} else {
ast_log(LOG_WARNING, "%s: Cat:%s. Unknown keyword %s at line %d of amd.conf\n",
app, cat, var->name, var->lineno);
}
var = var->next;
}
}
cat = ast_category_browse(cfg, cat);
}
ast_config_destroy(cfg);
ast_verb(3, "AMD defaults: initialSilence [%d] greeting [%d] afterGreetingSilence [%d] "
"totalAnalysisTime [%d] minimumWordLength [%d] betweenWordsSilence [%d] maximumNumberOfWords [%d] silenceThreshold [%d] maximumWordLength [%d]\n",
dfltInitialSilence, dfltGreeting, dfltAfterGreetingSilence, dfltTotalAnalysisTime,
dfltMinimumWordLength, dfltBetweenWordsSilence, dfltMaximumNumberOfWords, dfltSilenceThreshold, dfltMaximumWordLength);
return 0;
}
static int unload_module(void)
{
return ast_unregister_application(app);
}
static int load_module(void)
{
if (load_config(0))
return AST_MODULE_LOAD_DECLINE;
if (ast_register_application_xml(app, amd_exec))
return AST_MODULE_LOAD_FAILURE;
return AST_MODULE_LOAD_SUCCESS;
}
static int reload(void)
{
if (load_config(1))
return AST_MODULE_LOAD_DECLINE;
return AST_MODULE_LOAD_SUCCESS;
}
AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_DEFAULT, "Answering Machine Detection Application",
.load = load_module,
.unload = unload_module,
.reload = reload,
);