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authordvossel <dvossel@f38db490-d61c-443f-a65b-d21fe96a405b>2010-11-04 21:39:51 +0000
committerdvossel <dvossel@f38db490-d61c-443f-a65b-d21fe96a405b>2010-11-04 21:39:51 +0000
commitf3eab8b45e2a0cd1c789f5cc1677bcf0f64efab4 (patch)
treee771c5001b7816e7ad36c1278d865f41ae0525ae
parentf3b4c6871e8b13415bc57c7b098dd5669db7ceeb (diff)
Fixes ringback tone on sip semi-attended transfer.
ABE-2168 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@293924 f38db490-d61c-443f-a65b-d21fe96a405b
-rw-r--r--channels/chan_sip.c4
1 files changed, 4 insertions, 0 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 81bc410ea..6708a63ba 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);