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Merged revisions 158066 via svnmerge from

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https://origsvn.digium.com/svn/asterisk/trunk

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r158066 | mmichelson | 2008-11-20 11:39:06 -0600 (Thu, 20 Nov 2008) | 20 lines

Merged revisions 158053 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@158067 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Mark Michelson
2008-11-20 17:40:20 +00:00
parent 1c1ed7b9ed
commit 7122ee6adb
2 changed files with 7 additions and 1 deletions
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3
apps/app_dial.c
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@@ -1610,6 +1610,9 @@ 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);