[Asterisk-code-review] app_dial: Expanded A option to add caller announcement (asterisk[master])
George Joseph
asteriskteam at digium.com
Wed Jun 23 13:28:34 CDT 2021
George Joseph has submitted this change. ( https://gerrit.asterisk.org/c/asterisk/+/15941 )
Change subject: app_dial: Expanded A option to add caller announcement
......................................................................
app_dial: Expanded A option to add caller announcement
Hitherto, the A option has made it possible to play
audio upon answer to the called party only. This option
is expanded to allow for playback of an audio file to
the caller instead of or in addition to the audio
played to the answerer.
ASTERISK-29442
Change-Id: If6eed3ff5c341dc8c588c8210987f2571e891e5e
---
M apps/app_dial.c
A doc/CHANGES-staging/app_dial_announcement.txt
2 files changed, 68 insertions(+), 17 deletions(-)
Approvals:
Joshua Colp: Looks good to me, but someone else must approve
Benjamin Keith Ford: Looks good to me, but someone else must approve
George Joseph: Looks good to me, approved; Approved for Submit
diff --git a/apps/app_dial.c b/apps/app_dial.c
index d611083..51f5f45 100644
--- a/apps/app_dial.c
+++ b/apps/app_dial.c
@@ -93,11 +93,17 @@
</parameter>
<parameter name="options" required="false">
<optionlist>
- <option name="A">
- <argument name="x" required="true">
+ <option name="A" argsep=":">
+ <argument name="x">
<para>The file to play to the called party</para>
</argument>
- <para>Play an announcement to the called party, where <replaceable>x</replaceable> is the prompt to be played</para>
+ <argument name="y">
+ <para>The file to play to the calling party</para>
+ </argument>
+ <para>Play an announcement to the called and/or calling parties, where <replaceable>x</replaceable>
+ is the prompt to be played to the called party and <replaceable>y</replaceable> is the prompt
+ to be played to the caller. The files may be different and will be played to each party
+ simultaneously.</para>
</option>
<option name="a">
<para>Immediately answer the calling channel when the called channel answers in
@@ -2941,33 +2947,71 @@
int digit = 0;
struct ast_channel *chans[2];
struct ast_channel *active_chan;
+ char *calledfile = NULL, *callerfile = NULL;
+ int calledstream = 0, callerstream = 0;
chans[0] = chan;
chans[1] = peer;
- /* we need to stream the announcement to the called party when the OPT_ARG_ANNOUNCE (-A) is setted */
+ /* we need to stream the announcement(s) when the OPT_ARG_ANNOUNCE (-A) is set */
+ callerfile = opt_args[OPT_ARG_ANNOUNCE];
+ calledfile = strsep(&callerfile, ":");
- /* stream the file */
- res = ast_streamfile(peer, opt_args[OPT_ARG_ANNOUNCE], ast_channel_language(peer));
- if (res) {
- res = 0;
- ast_log(LOG_ERROR, "error streaming file '%s' to callee\n", opt_args[OPT_ARG_ANNOUNCE]);
+ /* stream the file(s) */
+ if (!ast_strlen_zero(calledfile)) {
+ res = ast_streamfile(peer, calledfile, ast_channel_language(peer));
+ if (res) {
+ res = 0;
+ ast_log(LOG_ERROR, "error streaming file '%s' to callee\n", calledfile);
+ } else {
+ calledstream = 1;
+ }
+ }
+ if (!ast_strlen_zero(callerfile)) {
+ res = ast_streamfile(chan, callerfile, ast_channel_language(chan));
+ if (res) {
+ res = 0;
+ ast_log(LOG_ERROR, "error streaming file '%s' to caller\n", callerfile);
+ } else {
+ callerstream = 1;
+ }
}
+ /* can't use ast_waitstream, because we're streaming two files at once, and can't block
+ We'll need to handle both channels at once. */
+
ast_channel_set_flag(peer, AST_FLAG_END_DTMF_ONLY);
- while (ast_channel_stream(peer)) {
- int ms;
+ while (ast_channel_stream(peer) || ast_channel_stream(chan)) {
+ int mspeer, mschan;
- ms = ast_sched_wait(ast_channel_sched(peer));
+ mspeer = ast_sched_wait(ast_channel_sched(peer));
+ mschan = ast_sched_wait(ast_channel_sched(chan));
- if (ms < 0 && !ast_channel_timingfunc(peer)) {
- ast_stopstream(peer);
+ if (calledstream) {
+ if (mspeer < 0 && !ast_channel_timingfunc(peer)) {
+ ast_stopstream(peer);
+ calledstream = 0;
+ }
+ }
+ if (callerstream) {
+ if (mschan < 0 && !ast_channel_timingfunc(chan)) {
+ ast_stopstream(chan);
+ callerstream = 0;
+ }
+ }
+
+ if (!calledstream && !callerstream) {
break;
}
- if (ms < 0)
- ms = 1000;
- active_chan = ast_waitfor_n(chans, 2, &ms);
+ if (mspeer < 0)
+ mspeer = 1000;
+
+ if (mschan < 0)
+ mschan = 1000;
+
+ /* wait for the lowest maximum of the two */
+ active_chan = ast_waitfor_n(chans, 2, (mspeer > mschan ? &mschan : &mspeer));
if (active_chan) {
struct ast_channel *other_chan;
struct ast_frame *fr = ast_read(active_chan);
@@ -3017,6 +3061,7 @@
ast_frfree(fr);
}
ast_sched_runq(ast_channel_sched(peer));
+ ast_sched_runq(ast_channel_sched(chan));
}
ast_channel_clear_flag(peer, AST_FLAG_END_DTMF_ONLY);
}
diff --git a/doc/CHANGES-staging/app_dial_announcement.txt b/doc/CHANGES-staging/app_dial_announcement.txt
new file mode 100644
index 0000000..3947b0e
--- /dev/null
+++ b/doc/CHANGES-staging/app_dial_announcement.txt
@@ -0,0 +1,6 @@
+Subject: app_dial announcement option
+
+The A option for Dial now supports
+playing audio to the caller as well
+as the called party.
+
--
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Gerrit-Project: asterisk
Gerrit-Branch: master
Gerrit-Change-Id: If6eed3ff5c341dc8c588c8210987f2571e891e5e
Gerrit-Change-Number: 15941
Gerrit-PatchSet: 6
Gerrit-Owner: N A <mail at interlinked.x10host.com>
Gerrit-Reviewer: Benjamin Keith Ford <bford at digium.com>
Gerrit-Reviewer: Friendly Automation
Gerrit-Reviewer: George Joseph <gjoseph at digium.com>
Gerrit-Reviewer: Joshua Colp <jcolp at sangoma.com>
Gerrit-Reviewer: Mark Murawski <markm at intellasoft.net>
Gerrit-MessageType: merged
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