Merged revisions 293924 via svnmerge from

https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r293924 | dvossel | 2010-11-04 16:39:51 -0500 (Thu, 04 Nov 2010) | 4 lines
  
  Fixes ringback tone on sip semi-attended transfer.
  
  ABE-2168
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294046 65c4cc65-6c06-0410-ace0-fbb531ad65f3

channels/chan_sip.c

@@ -21939,6 +21939,10 @@ static int local_attended_transfer(struct sip_pvt *transferer, struct sip_dual *
 
 		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);