kenmirrors/asterisk
1
0
Fork 0
mirror of https://github.com/asterisk/asterisk.git synced 2025-09-05 12:16:00 +00:00
Code Issues Packages Projects Releases Wiki Activity

Make sure to set the hangup cause on the calling channel in the case

Browse Source 
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/branches/1.4@158053 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Mark Michelson
2008-11-20 17:33:06 +00:00
parent 238cd9ea24
commit 3a1a981e2e
2 changed files with 5 additions and 0 deletions
Download Patch File Download Diff File Expand all files Collapse all files

3
apps/app_dial.c
View File

@@ -1383,6 +1383,9 @@ static int dial_exec_full(struct ast_channel *chan, void *data, struct ast_flags
ast_log(LOG_DEBUG, "ast call on peer returned %d\n", res);
if (option_verbose > 2)
ast_verbose(VERBOSE_PREFIX_3 "Couldn't call %s\n", numsubst);
if (tmp->chan->hangupcause) {
chan->hangupcause = tmp->chan->hangupcause;
}
ast_hangup(tmp->chan);
tmp->chan = NULL;
free(tmp);