diff options
| author | Mark Michelson <mmichelson@digium.com> | 2008-11-20 17:39:06 +0000 |
|---|---|---|
| committer | Mark Michelson <mmichelson@digium.com> | 2008-11-20 17:39:06 +0000 |
| commit | 7a554a7386478fe93db31dd2ba5a421063f6cdbb (patch) | |
| tree | 1025fd910d77eff81d9815d9107328a46b4ec689 /apps | |
| parent | d12263a16a2eb4e3877a12a63b5531395d34ad4b (diff) | |
Merged revisions 158053 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r158053 | mmichelson | 2008-11-20 11:33:06 -0600 (Thu, 20 Nov 2008) | 12 lines
Make sure to set the hangup cause on the calling channel in the case
that ast_call() fails. For incoming SIP channels, this was causing
us to send a 603 instead of a 486 when the call-limit was reached on
the destination channel.
(closes issue #13867)
Reported by: still_nsk
Patches:
13867.diff uploaded by putnopvut (license 60)
Tested by: blitzrage
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@158066 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Diffstat (limited to 'apps')
| -rw-r--r-- | apps/app_dial.c | 3 |
1 files changed, 3 insertions, 0 deletions
diff --git a/apps/app_dial.c b/apps/app_dial.c index 38d2bbd28..4f53eeba8 100644 --- a/apps/app_dial.c +++ b/apps/app_dial.c @@ -1818,6 +1818,9 @@ static int dial_exec_full(struct ast_channel *chan, void *data, struct ast_flags /* Again, keep going even if there's an error */ ast_debug(1, "ast call on peer returned %d\n", res); ast_verb(3, "Couldn't call %s\n", numsubst); + if (tc->hangupcause) { + chan->hangupcause = tc->hangupcause; + } ast_hangup(tc); tc = NULL; ast_free(tmp); |