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app_dial optional parameter to option 'r' to allow play indication from indications.conf

Browse Source 
(closes issue #14504)
  Reported by: alecdavis
  Tested by: alecdavis,jsmith
  Patch
	 app_dial.play_ring_indications.diff7.txt uploaded by alecdavis (license 585)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@235740 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Alec L Davis
2009-12-19 08:59:31 +00:00
parent cf7b67d9d3
commit 7537d3c0cb
2 changed files with 62 additions and 10 deletions
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2
CHANGES
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@@ -82,6 +82,8 @@ Applications
announcements or macros are executed.
* Modified app_dial to set answertime when the called channel answers even if
the called channel hangs up during playback of an announcement.
* Modified app_dial 'r' option to support an additional parameter to play an
indication tone from indications.conf
* Added c() option to app_chanspy. This option allows custom DTMF to be set
to cycle through the next available channel. By default this is still '*'.
* Added x() option to app_chanspy. This option allows DTMF to be set to

70
apps/app_dial.c
View File

@@ -62,6 +62,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/global_datastores.h"
#include "asterisk/dsp.h"
#include "asterisk/cel.h"
#include "asterisk/indications.h"
/*** DOCUMENTATION
<application name="Dial" language="en_US">
@@ -319,8 +320,11 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
it is provided. The current extension is used if a database family/key is not specified.</para>
</option>
<option name="r">
<para>Indicate ringing to the calling party, even if the called party isn't actually ringing. Pass no audio to the calling
<para>Default: Indicate ringing to the calling party, even if the called party isn't actually ringing. Pass no audio to the calling
party until the called channel has answered.</para>
<argument name="tone" required="false">
<para>Indicate progress to calling party. Send audio 'tone' from indications.conf</para>
</argument>
</option>
<option name="S">
<argument name="x" required="true" />
@@ -535,6 +539,7 @@ enum {
OPT_ARG_DURATION_LIMIT,
OPT_ARG_MUSICBACK,
OPT_ARG_CALLEE_MACRO,
OPT_ARG_RINGBACK,
OPT_ARG_CALLEE_GOSUB,
OPT_ARG_CALLEE_GO_ON,
OPT_ARG_PRIVACY,
@@ -572,7 +577,7 @@ AST_APP_OPTIONS(dial_exec_options, BEGIN_OPTIONS
AST_APP_OPTION_ARG('O', OPT_OPERMODE, OPT_ARG_OPERMODE),
AST_APP_OPTION('p', OPT_SCREENING),
AST_APP_OPTION_ARG('P', OPT_PRIVACY, OPT_ARG_PRIVACY),
AST_APP_OPTION('r', OPT_RINGBACK),
AST_APP_OPTION_ARG('r', OPT_RINGBACK, OPT_ARG_RINGBACK),
AST_APP_OPTION_ARG('S', OPT_DURATION_STOP, OPT_ARG_DURATION_STOP),
AST_APP_OPTION('t', OPT_CALLEE_TRANSFER),
AST_APP_OPTION('T', OPT_CALLER_TRANSFER),
@@ -890,6 +895,7 @@ struct privacy_args {
static struct ast_channel *wait_for_answer(struct ast_channel *in,
struct chanlist *outgoing, int *to, struct ast_flags64 *peerflags,
char *opt_args[],
struct privacy_args *pa,
const struct cause_args *num_in, int *result, char *dtmf_progress)
{
@@ -908,11 +914,12 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in,
ast_party_connected_line_init(&connected_caller);
if (single) {
/* Turn off hold music, etc */
if (!ast_test_flag64(outgoing, OPT_MUSICBACK | OPT_RINGBACK))
if (!ast_test_flag64(outgoing, OPT_MUSICBACK | OPT_RINGBACK)) {
ast_deactivate_generator(in);
/* If we are calling a single channel, make them compatible for in-band tone purpose */
ast_channel_make_compatible(outgoing->chan, in);
/* If we are calling a single channel, and not providing ringback or music, */
/* then, make them compatible for in-band tone purpose */
ast_channel_make_compatible(outgoing->chan, in);
}
if (!ast_test_flag64(peerflags, OPT_IGNORE_CONNECTEDLINE) && !ast_test_flag64(outgoing, DIAL_NOCONNECTEDLINE)) {
ast_channel_lock(outgoing->chan);
@@ -1078,7 +1085,7 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in,
/* Setup early media if appropriate */
if (single && CAN_EARLY_BRIDGE(peerflags, in, c))
ast_channel_early_bridge(in, c);
if (!(pa->sentringing) && !ast_test_flag64(outgoing, OPT_MUSICBACK)) {
if (!(pa->sentringing) && !ast_test_flag64(outgoing, OPT_MUSICBACK) && ast_strlen_zero(opt_args[OPT_ARG_RINGBACK])) {
ast_indicate(in, AST_CONTROL_RINGING);
pa->sentringing++;
}
@@ -1576,6 +1583,36 @@ static void end_bridge_callback_data_fixup(struct ast_bridge_config *bconfig, st
bconfig->end_bridge_callback_data = originator;
}
static int dial_handle_playtones(struct ast_channel *chan, const char *data)
{
struct ast_tone_zone_sound *ts = NULL;
int res;
const char *str = data;
if (ast_strlen_zero(str)) {
ast_debug(1,"Nothing to play\n");
return -1;
}
ts = ast_get_indication_tone(chan->zone, str);
if (ts && ts->data[0]) {
res = ast_playtones_start(chan, 0, ts->data, 0);
} else {
res = -1;
}
if (ts) {
ts = ast_tone_zone_sound_unref(ts);
}
if (res) {
ast_log(LOG_WARNING, "Unable to start playtone \'%s\'\n", str);
}
return res;
}
static int dial_exec_full(struct ast_channel *chan, const char *data, struct ast_flags64 *peerflags, int *continue_exec)
{
int res = -1; /* default: error */
@@ -1651,6 +1688,10 @@ static int dial_exec_full(struct ast_channel *chan, const char *data, struct ast
}
}
if (!ast_test_flag64(&opts, OPT_RINGBACK)) {
opt_args[OPT_ARG_RINGBACK] = NULL;
}
if (ast_test_flag64(&opts, OPT_OPERMODE)) {
opermode = ast_strlen_zero(opt_args[OPT_ARG_OPERMODE]) ? 1 : atoi(opt_args[OPT_ARG_OPERMODE]);
ast_verb(3, "Setting operator services mode to %d.\n", opermode);
@@ -1965,12 +2006,21 @@ static int dial_exec_full(struct ast_channel *chan, const char *data, struct ast
}
ast_indicate(chan, AST_CONTROL_PROGRESS);
} else if (ast_test_flag64(outgoing, OPT_RINGBACK)) {
ast_indicate(chan, AST_CONTROL_RINGING);
sentringing++;
if (!ast_strlen_zero(opt_args[OPT_ARG_RINGBACK])) {
if (dial_handle_playtones(chan, opt_args[OPT_ARG_RINGBACK])){
ast_indicate(chan, AST_CONTROL_RINGING);
sentringing++;
} else {
ast_indicate(chan, AST_CONTROL_PROGRESS);
}
} else {
ast_indicate(chan, AST_CONTROL_RINGING);
sentringing++;
}
}
}
peer = wait_for_answer(chan, outgoing, &to, peerflags, &pa, &num, &result, dtmf_progress);
peer = wait_for_answer(chan, outgoing, &to, peerflags, opt_args, &pa, &num, &result, dtmf_progress);
/* The ast_channel_datastore_remove() function could fail here if the
* datastore was moved to another channel during a masquerade. If this is