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author | Mark Michelson <mmichelson@digium.com> | 2012-01-21 00:10:35 +0000 |
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committer | Mark Michelson <mmichelson@digium.com> | 2012-01-21 00:10:35 +0000 |
commit | ab8ba431b6b5a29c8b0238afa8d763a87ae057e0 (patch) | |
tree | a5477a084def48fe85eb3cd6e6776355e4e6db32 /codecs | |
parent | d0c765497df2fb6a076e98ab5bec1505342431e3 (diff) |
Fix RTP reference leak.
If a blind transfer were initiated using a REFER without a prior
reINVITE to place the call on hold, AND if Asterisk were sending
RTCP reports, then there was a reference for the RTP instance
of the transferer.
This fixes the issue by merging two similar but slightly conflicting
sections of code into a single area. It also adds a stop_media_flows()
call in the case that the transferer's UA never sends a BYE to us
like it is supposed to.
(issue ASTERISK-19192)
Review: https://reviewboard.asterisk.org/r/1681/
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Merged revisions 352014 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 352015 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352018 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Diffstat (limited to 'codecs')
0 files changed, 0 insertions, 0 deletions