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)



git-svn-id: http://svn.asterisk.org/svn/asterisk/branches/11@409761 f38db490-d61c-443f-a65b-d21fe96a405b