Blocked revisions 409762
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Correct RTP handling in chan_unistim and fix transfer process broken in previous fix:
- Fixed too early RTP setup with phone, that cause no ringback tone on caller side
- Handle call transfer cancel only in STATE_CALL case (related to ASTERISK-23073)
(Reported by: Németh Tamás, niurkin sil)
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Merged revisions 409761 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409763
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