Add update_peer function to unistim_rtp_glue, improve other unistim_rtp_glue functions conforming to other channel drivers. Do not forget auto-detected and user-selected phone settings on 'unistim reload'
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Merged revisions 409705 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 409745 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@409747 f38db490-d61c-443f-a65b-d21fe96a405b
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: http://svn.asterisk.org/svn/asterisk/trunk@409763 f38db490-d61c-443f-a65b-d21fe96a405b