[asterisk-commits] bebuild: tag 1.8.12.0-rc3 r364763 - in /tags/1.8.12.0-rc3: ./ channels/

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Tue May 1 12:31:55 CDT 2012


Author: bebuild
Date: Tue May  1 12:31:50 2012
New Revision: 364763

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=364763
Log:
Merge 364706 for 1.8.12.0-rc3

Removed:
    tags/1.8.12.0-rc3/asterisk-1.8.12.0-rc2-summary.html
    tags/1.8.12.0-rc3/asterisk-1.8.12.0-rc2-summary.txt
Modified:
    tags/1.8.12.0-rc3/   (props changed)
    tags/1.8.12.0-rc3/.version
    tags/1.8.12.0-rc3/ChangeLog
    tags/1.8.12.0-rc3/channels/chan_sip.c

Propchange: tags/1.8.12.0-rc3/
------------------------------------------------------------------------------
--- svn:mergeinfo (original)
+++ svn:mergeinfo Tue May  1 12:31:50 2012
@@ -1,1 +1,1 @@
-/branches/1.8:363102,363106,363141
+/branches/1.8:363102,363106,363141,364706

Modified: tags/1.8.12.0-rc3/.version
URL: http://svnview.digium.com/svn/asterisk/tags/1.8.12.0-rc3/.version?view=diff&rev=364763&r1=364762&r2=364763
==============================================================================
--- tags/1.8.12.0-rc3/.version (original)
+++ tags/1.8.12.0-rc3/.version Tue May  1 12:31:50 2012
@@ -1,1 +1,1 @@
-1.8.12.0-rc2
+1.8.12.0-rc3

Modified: tags/1.8.12.0-rc3/ChangeLog
URL: http://svnview.digium.com/svn/asterisk/tags/1.8.12.0-rc3/ChangeLog?view=diff&rev=364763&r1=364762&r2=364763
==============================================================================
--- tags/1.8.12.0-rc3/ChangeLog (original)
+++ tags/1.8.12.0-rc3/ChangeLog Tue May  1 12:31:50 2012
@@ -1,3 +1,21 @@
+2012-05-01  Asterisk Development Team <asteriskteam at digium.com>
+
+	* Asterisk 1.8.12.0-rc3 Released.
+
+        * channels/chan_sip.c: Revert revision 360862
+
+        Revision 360862 was intended to improve identities sent in
+	dialog-info NOTIFY requests. Some users reported that hint became
+	broken once this was done. It's not clear exactly what part of
+	the patch has caused this regression, but broken hints are bad.
+
+        For now, this revision is being reverted so that the next releases of
+        Asterisk do not have bad behavior in them.  The original reported
+ 	issue will have to be fixed differently in the next version of
+	Asterisk.
+
+	(issue ASTERISK-16735)
+
 2012-04-24  Asterisk Development Team <asteriskteam at digium.com>
 
 	* Asterisk 1.8.12.0-rc2 Released.

Modified: tags/1.8.12.0-rc3/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/tags/1.8.12.0-rc3/channels/chan_sip.c?view=diff&rev=364763&r1=364762&r2=364763
==============================================================================
--- tags/1.8.12.0-rc3/channels/chan_sip.c (original)
+++ tags/1.8.12.0-rc3/channels/chan_sip.c Tue May  1 12:31:50 2012
@@ -12617,8 +12617,6 @@
 		if ((state & AST_EXTENSION_RINGING) && sip_cfg.notifyringing) {
 			const char *local_display = exten;
 			char *local_target = ast_strdupa(mto);
-			const char *remote_display = exten;
-			char *remote_target = ast_strdupa(mfrom);
 
 			/* There are some limitations to how this works.  The primary one is that the
 			   callee must be dialing the same extension that is being monitored.  Simply dialing
@@ -12628,28 +12626,16 @@
 
 				if ((caller = ast_channel_callback(find_calling_channel, NULL, p, 0))) {
 					char *cid_num;
-					char *connected_num;
 					int need;
 
 					ast_channel_lock(caller);
 					cid_num = S_COR(caller->caller.id.number.valid,
 						caller->caller.id.number.str, "");
 					need = strlen(cid_num) + strlen(p->fromdomain) + sizeof("sip:@");
-					remote_target = alloca(need);
-					snprintf(remote_target, need, "sip:%s@%s", cid_num, p->fromdomain);
-
-					remote_display = ast_strdupa(S_COR(caller->caller.id.name.valid,
+					local_target = alloca(need);
+					snprintf(local_target, need, "sip:%s@%s", cid_num, p->fromdomain);
+					local_display = ast_strdupa(S_COR(caller->caller.id.name.valid,
 						caller->caller.id.name.str, ""));
-
-					connected_num = S_COR(caller->connected.id.number.valid,
-						caller->connected.id.number.str, "");
-					need = strlen(connected_num) + strlen(p->fromdomain) + sizeof("sip:@");
-					local_target = alloca(need);
-					snprintf(local_target, need, "sip:%s@%s", connected_num, p->fromdomain);
-
-					local_display = ast_strdupa(S_COR(caller->connected.id.name.valid,
-						caller->connected.id.name.str, ""));
-
 					ast_channel_unlock(caller);
 					caller = ast_channel_unref(caller);
 				}
@@ -12671,10 +12657,10 @@
 						"<target uri=\"%s\"/>\n"
 						"</remote>\n"
 						"<local>\n"
-						"<identity display=\"%s\">%s</identity>\n"
+						"<identity>%s</identity>\n"
 						"<target uri=\"%s\"/>\n"
 						"</local>\n",
-						remote_display, remote_target, remote_target, local_display, local_target, local_target);
+						local_display, local_target, local_target, mto, mto);
 			} else {
 				ast_str_append(tmp, 0, "<dialog id=\"%s\" direction=\"recipient\">\n", exten);
 			}




More information about the asterisk-commits mailing list