From: Mark Michelson <mmichelson@digium.com>
Date: Thu, 11 Sep 2014 22:17:39 +0000 (+0000)
Subject: Remove undocumented default behavior of ast_play_and_record_full acceptdtmf.
X-Git-Tag: 14.0.0-beta1~1667
X-Git-Url: http://git.ipfire.org/cgi-bin/gitweb.cgi?a=commitdiff_plain;h=c212a71f0bd464e93eed947a03ba9414d1d42207;p=thirdparty%2Fasterisk.git

Remove undocumented default behavior of ast_play_and_record_full acceptdtmf.

ast_play_and_record_full() has a parameter called "acceptdtmf" that is a
string of acceptable DTMF digits that may be pressed by a caller to end
and accept the recording.

ARI uses this function in order to perform recording, and it provides
options for what is passed as acceptdtmf to ast_play_and_record_full().
By default, ARI passes an empty string, with the intention that no DTMF
can be used to end the recording.

The problem is that ast_play_and_record_full() attempts to be "helpful"
by setting "#" as the acceptdtmf if an empty string or NULL pointer
has been passed in. With ARI, this results in unexpected behavior
occurring if you have attempted to intercept "#" yourself in order
to perform some other manipulation of the live recording.

This change removes the "helpful" behavior by no longer accepting
"#" as a default acceptdtmf if none is specified by the caller of
ast_play_and_record_full(). This makes the ARI scenario work as
expected.

The other callers of ast_play_and_record_full() are app_voicemail
and app_minivm, and in both cases, they pass an explicit "#" to
ast_play_and_record_full() as acceptdtmf, so they are unaffected
by this change.
........

Merged revisions 422964 from http://svn.asterisk.org/svn/asterisk/branches/12
........

Merged revisions 422965 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422967 65c4cc65-6c06-0410-ace0-fbb531ad65f3
---

diff --git a/main/app.c b/main/app.c
index 82237479cb..a6c7514cde 100644
--- a/main/app.c
+++ b/main/app.c
@@ -1829,7 +1829,7 @@ static const char default_canceldtmf[] = "";
 
 int ast_play_and_record_full(struct ast_channel *chan, const char *playfile, const char *recordfile, int maxtime, const char *fmt, int *duration, int *sound_duration, int beep, int silencethreshold, int maxsilence, const char *path, const char *acceptdtmf, const char *canceldtmf, int skip_confirmation_sound, enum ast_record_if_exists if_exists)
 {
-	return __ast_play_and_record(chan, playfile, recordfile, maxtime, fmt, duration, sound_duration, beep, silencethreshold, maxsilence, path, 0, S_OR(acceptdtmf, default_acceptdtmf), S_OR(canceldtmf, default_canceldtmf), skip_confirmation_sound, if_exists);
+	return __ast_play_and_record(chan, playfile, recordfile, maxtime, fmt, duration, sound_duration, beep, silencethreshold, maxsilence, path, 0, S_OR(acceptdtmf, ""), S_OR(canceldtmf, default_canceldtmf), skip_confirmation_sound, if_exists);
 }
 
 int ast_play_and_record(struct ast_channel *chan, const char *playfile, const char *recordfile, int maxtime, const char *fmt, int *duration, int *sound_duration, int silencethreshold, int maxsilence, const char *path)