Merged revisions 297075 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r297075 | jpeeler | 2010-12-01 11:53:13 -0600 (Wed, 01 Dec 2010) | 37 lines Merged revisions 297073 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r297073 | jpeeler | 2010-12-01 11:52:46 -0600 (Wed, 01 Dec 2010) | 30 lines Merged revisions 297072 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r297072 | jpeeler | 2010-12-01 11:50:09 -0600 (Wed, 01 Dec 2010) | 23 lines Fix not stopping MOH when transfered local channel queue member is answered. The problem here is only present when local channels are used with the MOH passthru option as well as no optimization (/nm). I will describe the slightly bizarre scenario that was used to test, where phones B and C are queue members: Phone A dials into a queue with two members using local channels and the above options. Phone B answers. Phone A blind transfers phone B into the same queue. Phone A hangs up. Phone C answers, but phone B didn't stop playing MOH. In this scenario, the unhold frame that should have gotten to phone B never arrived due to the masquerade from the blind transfer. This is usually fine since app_queue manages the starting and stopping of MOH. However, with the passthrough option enabled when app_queue attempts to stop MOH it tries to do so on the local channel rather than the real channel. The easiest solution was to just make sure to send an unhold frame during the transfer since it wouldn't make sense to have MOH playing after a transfer anyway. This only modifies SIP transfers, but the other transfers did not seem to be a problem. If DTMF based transfers were a problem it might be okay to add ast_moh_stop to finishup, but I didn't want to have to add that unless required. ABE-2624 ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@297076 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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@ -22424,6 +22424,9 @@ static int handle_request_refer(struct sip_pvt *p, struct sip_request *req, int
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/* For blind transfers, move the call to the new extensions. For attended transfers on multiple
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servers - generate an INVITE with Replaces. Either way, let the dial plan decided */
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/* indicate before masquerade so the indication actually makes it to the real channel
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when using local channels with MOH passthru */
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ast_indicate(current.chan2, AST_CONTROL_UNHOLD);
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res = ast_async_goto(current.chan2, p->refer->refer_to_context, p->refer->refer_to, 1);
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if (!res) {
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