From c34f40e710b704381105f7eedf7c6a5509f728ad Mon Sep 17 00:00:00 2001
From: David Vossel <dvossel@digium.com>
Date: Thu, 4 Nov 2010 21:39:51 +0000
Subject: [PATCH] Fixes ringback tone on sip semi-attended transfer.

ABE-2168


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@293924 65c4cc65-6c06-0410-ace0-fbb531ad65f3
---
 channels/chan_sip.c | 4 ++++
 1 file changed, 4 insertions(+)

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 81bc410ead..6708a63ba0 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -21833,6 +21833,10 @@ 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);