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authorjpeeler <jpeeler@f38db490-d61c-443f-a65b-d21fe96a405b>2010-12-01 17:50:09 +0000
committerjpeeler <jpeeler@f38db490-d61c-443f-a65b-d21fe96a405b>2010-12-01 17:50:09 +0000
commit677325de6d4cecf2a2ac2cacd516bd5a182519e5 (patch)
tree017611f46637e11e970aeab359e81115a37da92c /channels
parentd73d05741467423b25a7233afde44faa199a3c06 (diff)
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: http://svn.digium.com/svn/asterisk/branches/1.4@297072 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'channels')
-rw-r--r--channels/chan_sip.c3
1 files changed, 3 insertions, 0 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index a06726bd6..3ed71c96f 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -16157,6 +16157,9 @@ static int handle_request_refer(struct sip_pvt *p, struct sip_request *req, int
/* For blind transfers, move the call to the new extensions. For attended transfers on multiple
servers - generate an INVITE with Replaces. Either way, let the dial plan decided */
+ /* indicate before masquerade so the indication actually makes it to the real channel
+ when using local channels with MOH passthru */
+ ast_indicate(current.chan2, AST_CONTROL_UNHOLD);
res = ast_async_goto(current.chan2, p->refer->refer_to_context, p->refer->refer_to, 1);
if (!res) {