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

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 9c82c6f..c2afeb5 100644 (file)
--- a/channels/chan_sip.c
+++ b/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);