[asterisk-commits] dvossel: branch 1.8 r293924 - /branches/1.8/channels/chan_sip.c

[SVN commits to the Asterisk project]

Author: dvossel
Date: Thu Nov  4 16:39:51 2010
New Revision: 293924

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=293924
Log:
Fixes ringback tone on sip semi-attended transfer.

ABE-2168

Modified:
    branches/1.8/channels/chan_sip.c

Modified: branches/1.8/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/branches/1.8/channels/chan_sip.c?view=diff&rev=293924&r1=293923&r2=293924
==============================================================================
--- branches/1.8/channels/chan_sip.c (original)
+++ branches/1.8/channels/chan_sip.c Thu Nov  4 16:39:51 2010
@@ -21833,6 +21833,10 @@
 
 		ast_indicate(target.chan1, AST_CONTROL_UNHOLD);
 
+		if (current->chan2 && current->chan2->_state == AST_STATE_RING) {
+			ast_indicate(target.chan1, AST_CONTROL_RINGING);
+		}
+
 		if (target.chan2) {
 			ast_channel_queue_connected_line_update(target.chan1, &connected_to_transferee, NULL);
 			ast_channel_queue_connected_line_update(target.chan2, &connected_to_target, NULL);