From b0d83789106a52ff1b75b6b14c2d16711ac10d77 Mon Sep 17 00:00:00 2001 From: Tilghman Lesher Date: Sat, 24 Nov 2007 06:24:46 +0000 Subject: [PATCH] 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 --- apps/app_voicemail.c | 5 ++++- main/app.c | 14 +++++++++++--- 2 files changed, 15 insertions(+), 4 deletions(-) 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++) { -- 2.47.2