res_pjsip_session: Delay sending BYE if a re-INVITE transaction is in progress.

res_pjsip_session: Delay sending BYE if a re-INVITE transaction is in progress.

Given the scenario where a PJSIP channel is in a native RTP bridge with direct
media and the channel is then hung up the code will currently re-INVITE the channel
back to Asterisk and send a BYE at the same time. Many SIP implementations dislike
this greatly.

This change makes it so that if a re-INVITE transaction is in progress the BYE
is queued to occur after the completion of the transaction (be it through normal
means or a timeout).

Review: https://reviewboard.asterisk.org/r/4248/
........

Merged revisions 429409 from http://svn.asterisk.org/svn/asterisk/branches/13

git-svn-id: https://origsvn.digium.com/svn/asterisk/certified/branches/13.1@429864 65c4cc65-6c06-0410-ace0-fbb531ad65f3