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

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);