[asterisk-commits] mmichelson: trunk r158066 - in /trunk: ./ apps/app_dial.c channels/chan_sip.c

[SVN commits to the Asterisk project]

Author: mmichelson
Date: Thu Nov 20 11:39:06 2008
New Revision: 158066

URL: http://svn.digium.com/view/asterisk?view=rev&rev=158066
Log:
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


........

Modified:
    trunk/   (props changed)
    trunk/apps/app_dial.c
    trunk/channels/chan_sip.c

Propchange: trunk/
------------------------------------------------------------------------------
Binary property 'branch-1.4-merged' - no diff available.

Modified: trunk/apps/app_dial.c
URL: http://svn.digium.com/view/asterisk/trunk/apps/app_dial.c?view=diff&rev=158066&r1=158065&r2=158066
==============================================================================
--- trunk/apps/app_dial.c (original)
+++ trunk/apps/app_dial.c Thu Nov 20 11:39:06 2008
@@ -1818,6 +1818,9 @@
 			/* 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);

Modified: trunk/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/trunk/channels/chan_sip.c?view=diff&rev=158066&r1=158065&r2=158066
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Thu Nov 20 11:39:06 2008
@@ -4870,8 +4870,11 @@
 
 	res = update_call_counter(p, INC_CALL_RINGING);
 
-	if (res == -1)
+	if (res == -1) {
 		return res;
+	} else {
+		ast->hangupcause = AST_CAUSE_USER_BUSY;
+	}
 
 	p->callingpres = ast->cid.cid_pres;
 	p->jointcapability = ast_translate_available_formats(p->capability, p->prefcodec);