From: Tilghman Lesher <tilghman@meg.abyt.es>
Date: Sat, 24 Nov 2007 06:24:46 +0000 (+0000)
Subject: Merged revisions 89540 via svnmerge from
X-Git-Tag: 1.6.0-beta1~3^2~688
X-Git-Url: http://git.ipfire.org/cgi-bin/gitweb.cgi?a=commitdiff_plain;h=b0d83789106a52ff1b75b6b14c2d16711ac10d77;p=thirdparty%2Fasterisk.git

Merged revisions 89540 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r89540 | tilghman | 2007-11-24 00:19:23 -0600 (Sat, 24 Nov 2007) | 9 lines

Currently, zero-length voicemail messages cause a hangup in VoicemailMain.
This change fixes the problem, with a multi-faceted approach.  First, we
do our best to avoid these messages from being created in the first place,
and second, if that fails, we detect when the voicemail message is
zero-length and avoid exiting at that point.
Reported by: dtyoo
Patch by: gkloepfer,tilghman
(Closes issue #11083)

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
---

diff --git a/apps/app_voicemail.c b/apps/app_voicemail.c
index 1ae60eddad..3b1b25b607 100644
--- a/apps/app_voicemail.c
+++ b/apps/app_voicemail.c
@@ -4851,7 +4851,10 @@ static int play_message(struct ast_channel *chan, struct ast_vm_user *vmu, struc
 	if (!res) {
 		make_file(vms->fn, sizeof(vms->fn), vms->curdir, vms->curmsg);
 		vms->heard[vms->curmsg] = 1;
-		res = wait_file(chan, vms, vms->fn);
+		if ((res = wait_file(chan, vms, vms->fn)) < 0) {
+			ast_log(LOG_WARNING, "Playback of message %s failed\n", vms->fn);
+			res = 0;
+		}
 	}
 	DISPOSE(vms->curdir, vms->curmsg);
 	return res;
diff --git a/main/app.c b/main/app.c
index 68310d3592..20064af140 100644
--- a/main/app.c
+++ b/main/app.c
@@ -743,8 +743,6 @@ static int __ast_play_and_record(struct ast_channel *chan, const char *playfile,
 		} else {
 			ast_frfree(f);
 		}
-		if (end == start)
-			end = time(NULL);
 	} else {
 		ast_log(LOG_WARNING, "Error creating writestream '%s', format '%s'\n", recordfile, sfmt[x]);
 	}
@@ -753,7 +751,17 @@ static int __ast_play_and_record(struct ast_channel *chan, const char *playfile,
 		if (silgen)
 			ast_channel_stop_silence_generator(chan, silgen);
 	}
-	*duration = end - start;
+
+	/*!\note
+	 * Instead of asking how much time passed (end - start), calculate the number
+	 * of seconds of audio which actually went into the file.  This fixes a
+	 * problem where audio is stopped up on the network and never gets to us.
+	 *
+	 * Note that we still want to use the number of seconds passed for the max
+	 * message, otherwise we could get a situation where this stream is never
+	 * closed (which would create a resource leak).
+	 */
+	*duration = ast_tellstream(others[0]) / 8000;
 
 	if (!prepend) {
 		for (x = 0; x < fmtcnt; x++) {