From 6e30ccd242e9316d2de20086724a6f5d340115cb Mon Sep 17 00:00:00 2001 From: Joshua Colp <jcolp@digium.com> Date: Mon, 1 Dec 2014 13:41:08 +0000 Subject: [PATCH] app_record: Fix bug where using the 'k' option and hanging up would trim 1/4 of a second of the recording. The Record dialplan function trims 1/4 of a second from the end of recordings in case they are terminated because of DTMF. When hanging up, however, you don't want this to happen. This change makes it so on hangup this does not occur. ASTERISK-24530 #close Reported by: Ben Smithurst patches: app_record_v2.diff submitted by Ben Smithurst (license 6529) Review: https://reviewboard.asterisk.org/r/4201/ ........ Merged revisions 428653 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 428654 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@428655 65c4cc65-6c06-0410-ace0-fbb531ad65f3 --- apps/app_record.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) diff --git a/apps/app_record.c b/apps/app_record.c index 4008fc0123..43a1e490f3 100644 --- a/apps/app_record.c +++ b/apps/app_record.c @@ -447,8 +447,13 @@ static int record_exec(struct ast_channel *chan, const char *data) if (gotsilence) { ast_stream_rewind(s, silence - 1000); ast_truncstream(s); - } else if (!gottimeout) { - /* Strip off the last 1/4 second of it */ + } else if (!gottimeout && f) { + /* + * Strip off the last 1/4 second of it, if we didn't end because of a timeout, + * or a hangup. This must mean we ended because of a DTMF tone and while this + * 1/4 second stripping is very old code the most likely explanation is that it + * relates to stripping a partial DTMF tone. + */ ast_stream_rewind(s, 250); ast_truncstream(s); } -- GitLab