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

  (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

diff --git a/CHANGES b/CHANGES
index 39bd923..74f20fb 100644 (file)
--- a/CHANGES
+++ b/CHANGES
@@ -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

diff --git a/apps/app_dial.c b/apps/app_dial.c
index 0ef64b3..c24a566 100644 (file)
--- a/apps/app_dial.c
+++ b/apps/app_dial.c
@@ -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