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Move routines from voicemail to app for general use (part of bug #752)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@3801 65c4cc65-6c06-0410-ace0-fbb531ad65f3
1.0
Mark Spencer 21 years ago
parent 877f1f64c2
commit 9d8c71ee86
3 changed files with 678 additions and 650 deletions
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466
app.c
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@ -1,11 +1,11 @@
/*
* Asterisk -- A telephony toolkit for Linux.
*
* Channel Management
* Convenient Application Routines
*
* Copyright (C) 1999, Mark Spencer
* Copyright (C) 1999-2004, Digium, Inc.
*
* Mark Spencer <markster@linux-support.net>
* Mark Spencer <markster@digium.com>
*
* This program is free software, distributed under the terms of
* the GNU General Public License
@ -31,6 +31,8 @@
#include "asterisk.h"
#include "astconf.h"
#define MAX_OTHER_FORMATS 10
/* set timeout to 0 for "standard" timeouts. Set timeout to -1 for
"ludicrous time" (essentially never times out) */
int ast_app_getdata(struct ast_channel *c, char *prompt, char *s, int maxlen, int timeout)
@ -485,3 +487,461 @@ int ast_control_streamfile(struct ast_channel *chan, char *file, char *fwd, char
return res;
}
int ast_play_and_wait(struct ast_channel *chan, char *fn)
{
int d;
d = ast_streamfile(chan, fn, chan->language);
if (d)
return d;
d = ast_waitstream(chan, AST_DIGIT_ANY);
ast_stopstream(chan);
return d;
}
static int global_silence_threshold = 128;
static int global_maxsilence = 0;
int ast_play_and_record(struct ast_channel *chan, char *playfile, char *recordfile, int maxtime, char *fmt, int *duration, int silencethreshold, int maxsilence)
{
char d, *fmts;
char comment[256];
int x, fmtcnt=1, res=-1,outmsg=0;
struct ast_frame *f;
struct ast_filestream *others[MAX_OTHER_FORMATS];
char *sfmt[MAX_OTHER_FORMATS];
char *stringp=NULL;
time_t start, end;
struct ast_dsp *sildet; /* silence detector dsp */
int totalsilence = 0;
int dspsilence = 0;
int gotsilence = 0; /* did we timeout for silence? */
int rfmt=0;
if (silencethreshold < 0)
silencethreshold = global_silence_threshold;
if (maxsilence < 0)
maxsilence = global_maxsilence;
/* barf if no pointer passed to store duration in */
if (duration == NULL) {
ast_log(LOG_WARNING, "Error play_and_record called without duration pointer\n");
return -1;
}
ast_log(LOG_DEBUG,"play_and_record: %s, %s, '%s'\n", playfile ? playfile : "<None>", recordfile, fmt);
snprintf(comment,sizeof(comment),"Playing %s, Recording to: %s on %s\n", playfile ? playfile : "<None>", recordfile, chan->name);
if (playfile) {
d = ast_play_and_wait(chan, playfile);
if (d > -1)
d = ast_streamfile(chan, "beep",chan->language);
if (!d)
d = ast_waitstream(chan,"");
if (d < 0)
return -1;
}
fmts = ast_strdupa(fmt);
stringp=fmts;
strsep(&stringp, "|");
ast_log(LOG_DEBUG,"Recording Formats: sfmts=%s\n", fmts);
sfmt[0] = ast_strdupa(fmts);
while((fmt = strsep(&stringp, "|"))) {
if (fmtcnt > MAX_OTHER_FORMATS - 1) {
ast_log(LOG_WARNING, "Please increase MAX_OTHER_FORMATS in app_voicemail.c\n");
break;
}
sfmt[fmtcnt++] = ast_strdupa(fmt);
}
time(&start);
end=start; /* pre-initialize end to be same as start in case we never get into loop */
for (x=0;x<fmtcnt;x++) {
others[x] = ast_writefile(recordfile, sfmt[x], comment, O_TRUNC, 0, 0700);
ast_verbose( VERBOSE_PREFIX_3 "x=%i, open writing: %s format: %s, %p\n", x, recordfile, sfmt[x], others[x]);
if (!others[x]) {
break;
}
}
sildet = ast_dsp_new(); /* Create the silence detector */
if (!sildet) {
ast_log(LOG_WARNING, "Unable to create silence detector :(\n");
return -1;
}
ast_dsp_set_threshold(sildet, silencethreshold);
if (maxsilence > 0) {
rfmt = chan->readformat;
res = ast_set_read_format(chan, AST_FORMAT_SLINEAR);
if (res < 0) {
ast_log(LOG_WARNING, "Unable to set to linear mode, giving up\n");
return -1;
}
}
if (x == fmtcnt) {
/* Loop forever, writing the packets we read to the writer(s), until
we read a # or get a hangup */
f = NULL;
for(;;) {
res = ast_waitfor(chan, 2000);
if (!res) {
ast_log(LOG_DEBUG, "One waitfor failed, trying another\n");
/* Try one more time in case of masq */
res = ast_waitfor(chan, 2000);
if (!res) {
ast_log(LOG_WARNING, "No audio available on %s??\n", chan->name);
res = -1;
}
}
if (res < 0) {
f = NULL;
break;
}
f = ast_read(chan);
if (!f)
break;
if (f->frametype == AST_FRAME_VOICE) {
/* write each format */
for (x=0;x<fmtcnt;x++) {
res = ast_writestream(others[x], f);
}
/* Silence Detection */
if (maxsilence > 0) {
dspsilence = 0;
ast_dsp_silence(sildet, f, &dspsilence);
if (dspsilence)
totalsilence = dspsilence;
else
totalsilence = 0;
if (totalsilence > maxsilence) {
/* Ended happily with silence */
if (option_verbose > 2)
ast_verbose( VERBOSE_PREFIX_3 "Recording automatically stopped after a silence of %d seconds\n", totalsilence/1000);
ast_frfree(f);
gotsilence = 1;
outmsg=2;
break;
}
}
/* Exit on any error */
if (res) {
ast_log(LOG_WARNING, "Error writing frame\n");
ast_frfree(f);
break;
}
} else if (f->frametype == AST_FRAME_VIDEO) {
/* Write only once */
ast_writestream(others[0], f);
} else if (f->frametype == AST_FRAME_DTMF) {
if (f->subclass == '#') {
if (option_verbose > 2)
ast_verbose( VERBOSE_PREFIX_3 "User ended message by pressing %c\n", f->subclass);
res = '#';
outmsg = 2;
ast_frfree(f);
break;
}
}
if (f->subclass == '0') {
/* Check for a '0' during message recording also, in case caller wants operator */
if (option_verbose > 2)
ast_verbose(VERBOSE_PREFIX_3 "User cancelled by pressing %c\n", f->subclass);
res = '0';
outmsg = 0;
ast_frfree(f);
break;
}
if (maxtime) {
time(&end);
if (maxtime < (end - start)) {
if (option_verbose > 2)
ast_verbose( VERBOSE_PREFIX_3 "Took too long, cutting it short...\n");
outmsg = 2;
res = 't';
ast_frfree(f);
break;
}
}
ast_frfree(f);
}
if (end == start) time(&end);
if (!f) {
if (option_verbose > 2)
ast_verbose( VERBOSE_PREFIX_3 "User hung up\n");
res = -1;
outmsg=1;
}
} else {
ast_log(LOG_WARNING, "Error creating writestream '%s', format '%s'\n", recordfile, sfmt[x]);
}
*duration = end - start;
for (x=0;x<fmtcnt;x++) {
if (!others[x])
break;
if (totalsilence)
ast_stream_rewind(others[x], totalsilence-200);
else
ast_stream_rewind(others[x], 200);
ast_truncstream(others[x]);
ast_closestream(others[x]);
}
if (rfmt) {
if (ast_set_read_format(chan, rfmt)) {
ast_log(LOG_WARNING, "Unable to restore format %s to channel '%s'\n", ast_getformatname(rfmt), chan->name);
}
}
if (outmsg) {
if (outmsg > 1) {
/* Let them know recording is stopped */
ast_streamfile(chan, "auth-thankyou", chan->language);
ast_waitstream(chan, "");
}
}
return res;
}
int ast_play_and_prepend(struct ast_channel *chan, char *playfile, char *recordfile, int maxtime, char *fmt, int *duration, int beep, int silencethreshold, int maxsilence)
{
char d = 0, *fmts;
char comment[256];
int x, fmtcnt=1, res=-1,outmsg=0;
struct ast_frame *f;
struct ast_filestream *others[MAX_OTHER_FORMATS];
struct ast_filestream *realfiles[MAX_OTHER_FORMATS];
char *sfmt[MAX_OTHER_FORMATS];
char *stringp=NULL;
time_t start, end;
struct ast_dsp *sildet; /* silence detector dsp */
int totalsilence = 0;
int dspsilence = 0;
int gotsilence = 0; /* did we timeout for silence? */
int rfmt=0;
char prependfile[80];
if (silencethreshold < 0)
silencethreshold = global_silence_threshold;
if (maxsilence < 0)
maxsilence = global_maxsilence;
/* barf if no pointer passed to store duration in */
if (duration == NULL) {
ast_log(LOG_WARNING, "Error play_and_prepend called without duration pointer\n");
return -1;
}
ast_log(LOG_DEBUG,"play_and_prepend: %s, %s, '%s'\n", playfile ? playfile : "<None>", recordfile, fmt);
snprintf(comment,sizeof(comment),"Playing %s, Recording to: %s on %s\n", playfile ? playfile : "<None>", recordfile, chan->name);
if (playfile || beep) {
if (!beep)
d = ast_play_and_wait(chan, playfile);
if (d > -1)
d = ast_streamfile(chan, "beep",chan->language);
if (!d)
d = ast_waitstream(chan,"");
if (d < 0)
return -1;
}
strncpy(prependfile, recordfile, sizeof(prependfile) -1);
strncat(prependfile, "-prepend", sizeof(prependfile) - strlen(prependfile) - 1);
fmts = ast_strdupa(fmt);
stringp=fmts;
strsep(&stringp, "|");
ast_log(LOG_DEBUG,"Recording Formats: sfmts=%s\n", fmts);
sfmt[0] = ast_strdupa(fmts);
while((fmt = strsep(&stringp, "|"))) {
if (fmtcnt > MAX_OTHER_FORMATS - 1) {
ast_log(LOG_WARNING, "Please increase MAX_OTHER_FORMATS in app_voicemail.c\n");
break;
}
sfmt[fmtcnt++] = ast_strdupa(fmt);
}
time(&start);
end=start; /* pre-initialize end to be same as start in case we never get into loop */
for (x=0;x<fmtcnt;x++) {
others[x] = ast_writefile(prependfile, sfmt[x], comment, O_TRUNC, 0, 0700);
ast_verbose( VERBOSE_PREFIX_3 "x=%i, open writing: %s format: %s, %p\n", x, prependfile, sfmt[x], others[x]);
if (!others[x]) {
break;
}
}
sildet = ast_dsp_new(); /* Create the silence detector */
if (!sildet) {
ast_log(LOG_WARNING, "Unable to create silence detector :(\n");
return -1;
}
ast_dsp_set_threshold(sildet, silencethreshold);
if (maxsilence > 0) {
rfmt = chan->readformat;
res = ast_set_read_format(chan, AST_FORMAT_SLINEAR);
if (res < 0) {
ast_log(LOG_WARNING, "Unable to set to linear mode, giving up\n");
return -1;
}
}
if (x == fmtcnt) {
/* Loop forever, writing the packets we read to the writer(s), until
we read a # or get a hangup */
f = NULL;
for(;;) {
res = ast_waitfor(chan, 2000);
if (!res) {
ast_log(LOG_DEBUG, "One waitfor failed, trying another\n");
/* Try one more time in case of masq */
res = ast_waitfor(chan, 2000);
if (!res) {
ast_log(LOG_WARNING, "No audio available on %s??\n", chan->name);
res = -1;
}
}
if (res < 0) {
f = NULL;
break;
}
f = ast_read(chan);
if (!f)
break;
if (f->frametype == AST_FRAME_VOICE) {
/* write each format */
for (x=0;x<fmtcnt;x++) {
if (!others[x])
break;
res = ast_writestream(others[x], f);
}
/* Silence Detection */
if (maxsilence > 0) {
dspsilence = 0;
ast_dsp_silence(sildet, f, &dspsilence);
if (dspsilence)
totalsilence = dspsilence;
else
totalsilence = 0;
if (totalsilence > maxsilence) {
/* Ended happily with silence */
if (option_verbose > 2)
ast_verbose( VERBOSE_PREFIX_3 "Recording automatically stopped after a silence of %d seconds\n", totalsilence/1000);
ast_frfree(f);
gotsilence = 1;
outmsg=2;
break;
}
}
/* Exit on any error */
if (res) {
ast_log(LOG_WARNING, "Error writing frame\n");
ast_frfree(f);
break;
}
} else if (f->frametype == AST_FRAME_VIDEO) {
/* Write only once */
ast_writestream(others[0], f);
} else if (f->frametype == AST_FRAME_DTMF) {
/* stop recording with any digit */
if (option_verbose > 2)
ast_verbose( VERBOSE_PREFIX_3 "User ended message by pressing %c\n", f->subclass);
res = 't';
outmsg = 2;
ast_frfree(f);
break;
}
if (maxtime) {
time(&end);
if (maxtime < (end - start)) {
if (option_verbose > 2)
ast_verbose( VERBOSE_PREFIX_3 "Took too long, cutting it short...\n");
res = 't';
outmsg=2;
ast_frfree(f);
break;
}
}
ast_frfree(f);
}
if (end == start) time(&end);
if (!f) {
if (option_verbose > 2)
ast_verbose( VERBOSE_PREFIX_3 "User hung up\n");
res = -1;
outmsg=1;
#if 0
/* delete all the prepend files */
for (x=0;x<fmtcnt;x++) {
if (!others[x])
break;
ast_closestream(others[x]);
ast_filedelete(prependfile, sfmt[x]);
}
#endif
}
} else {
ast_log(LOG_WARNING, "Error creating writestream '%s', format '%s'\n", prependfile, sfmt[x]);
}
*duration = end - start;
#if 0
if (outmsg > 1) {
#else
if (outmsg) {
#endif
struct ast_frame *fr;
for (x=0;x<fmtcnt;x++) {
snprintf(comment, sizeof(comment), "Opening the real file %s.%s\n", recordfile, sfmt[x]);
realfiles[x] = ast_readfile(recordfile, sfmt[x], comment, O_RDONLY, 0, 0);
if (!others[x] || !realfiles[x])
break;
if (totalsilence)
ast_stream_rewind(others[x], totalsilence-200);
else
ast_stream_rewind(others[x], 200);
ast_truncstream(others[x]);
/* add the original file too */
while ((fr = ast_readframe(realfiles[x]))) {
ast_writestream(others[x],fr);
}
ast_closestream(others[x]);
ast_closestream(realfiles[x]);
ast_filerename(prependfile, recordfile, sfmt[x]);
#if 0
ast_verbose("Recording Format: sfmts=%s, prependfile %s, recordfile %s\n", sfmt[x],prependfile,recordfile);
#endif
ast_filedelete(prependfile, sfmt[x]);
}
}
if (rfmt) {
if (ast_set_read_format(chan, rfmt)) {
ast_log(LOG_WARNING, "Unable to restore format %s to channel '%s'\n", ast_getformatname(rfmt), chan->name);
}
}
if (outmsg) {
if (outmsg > 1) {
/* Let them know it worked */
ast_streamfile(chan, "auth-thankyou", chan->language);
ast_waitstream(chan, "");
}
}
return res;
}

847
apps/app_voicemail.c
View File

File diff suppressed because it is too large Load Diff

15
include/asterisk/app.h
Unescape View File

@ -4,9 +4,9 @@
* Application convenience functions, designed to give consistent
* look and feel to asterisk apps.
*
* Copyright (C) 1999, Mark Spencer
* Copyright (C) 1999-2004, Digium, Inc.
*
* Mark Spencer <markster@linux-support.net>
* Mark Spencer <markster@digium.com>
*
* This program is free software, distributed under the terms of
* the GNU General Public License
@ -57,6 +57,17 @@ int ast_linear_stream(struct ast_channel *chan, const char *filename, int fd, in
//! Stream a file with fast forward, pause, reverse.
int ast_control_streamfile(struct ast_channel *chan, char *file, char *fwd, char *rev, char *stop, char *pause, int skipms);
//! Play a stream and wait for a digit, returning the digit that was pressed
int ast_play_and_wait(struct ast_channel *chan, char *fn);
//! Record a file for a max amount of time, in a given list of formats separated by '|', outputting the duration of the recording, and with a maximum
// permitted silence time of 'maxsilence' under 'silencethreshold' or use '-1' for either or both parameters for defaults.
int ast_play_and_record(struct ast_channel *chan, char *playfile, char *recordfile, int maxtime, char *fmt, int *duration, int silencethreshold, int maxsilence);
//! Record a message and prepend the message to the given record file after playing the optional playfile (or a beep), storing the duration in 'duration' and with a maximum
// permitted silence time of 'maxsilence' under 'silencethreshold' or use '-1' for either or both parameters for defaults.
int ast_play_and_prepend(struct ast_channel *chan, char *playfile, char *recordfile, int maxtime, char *fmt, int *duration, int beep, int silencethreshold, int maxsilence);
#if defined(__cplusplus) || defined(c_plusplus)
}
#endif

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