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author | mmichelson <mmichelson@f38db490-d61c-443f-a65b-d21fe96a405b> | 2008-11-20 17:39:06 +0000 |
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committer | mmichelson <mmichelson@f38db490-d61c-443f-a65b-d21fe96a405b> | 2008-11-20 17:39:06 +0000 |
commit | 056bae58f4c7e630cf704141e1bef3ddf9b9d2e7 (patch) | |
tree | 1025fd910d77eff81d9815d9107328a46b4ec689 /apps/app_dial.c | |
parent | d93047742f3db5b8651cefe74d168be7d5e15d26 (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
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@158066 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'apps/app_dial.c')
-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); |