Bridging: Fix bouncing native bridge

This fixes a situation in Asterisk 1.8 and 11 where ast_channel_bridge
could cause a bouncing native bridge. In the case of the
dial_LS_options test, this was a remote RTP bridge which caused the
audio path to continually cycle between Asterisk and the remote
endpoints generating a large number of SIP messages and delaying the
test long enough to cause it to fail (checking timing was part of the
test). The root cause was that the code to decide whether to use native
bridging was expecting a time-remaining value of 0 to be the default
instead of the actual default value of -1. A value of 0 or negative
numbers could also be generated by preceding code in some
circumstances. Both issues are addressed in this patch.

ASTERISK-24211 #close
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/3987/
........

Merged revisions 423006 from http://svn.asterisk.org/svn/asterisk/branches/1.8


git-svn-id: http://svn.asterisk.org/svn/asterisk/branches/11@423010 f38db490-d61c-443f-a65b-d21fe96a405b