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Merged revisions 293924 via svnmerge from 

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

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  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: http://svn.digium.com/svn/asterisk/trunk@294046 f38db490-d61c-443f-a65b-d21fe96a405b
This commit is contained in:
dvossel 2010-11-05 15:26:01 +00:00
parent c4fb7fe108
commit be55442a83
1 changed files with 4 additions and 0 deletions
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4
channels/chan_sip.c
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@ -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);