Multiple revisions 396884,400075,400093,401446,401960

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  r396884 | jbigelow | 2013-08-16 17:45:10 -0500 (Fri, 16 Aug 2013) | 8 lines
  
  Add test suite events to indicate when a feature is detected or not
  
  These are needed by the bridge test suite tests for them to be able to run
  against Asterisk 11.
  
  Review: https://reviewboard.asterisk.org/r/2751/
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  r400075 | mjordan | 2013-09-28 16:59:12 -0500 (Sat, 28 Sep 2013) | 16 lines
  
  Add check for openSUSE when detecting bfd library
  
  In ASTERISK-17842, some additional library checks were added to the configure
  script so that the bfd library could be found on CentOS and Fedora systems.
  
  As it turns out, openSUSE requires an additional library. This patch adds
  another check to the configure script for openSUSE that will add that library.
  
  Review: https://reviewboard.asterisk.org/r/2885/
  
  (closes issue AST-1169)
  Reported by: Guenther Kelleter
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  Merged revisions 400073 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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  r400093 | mjordan | 2013-09-28 17:21:37 -0500 (Sat, 28 Sep 2013) | 23 lines
  
  res_rtp_asterisk: Correct erroneous lost packet information in RTCP reports
  
  RTCP's calculation of the number of lost packets in an RTP stream is based on
  that stream's sequence number count, the number of received packets, and how
  many packets we expect to receive. When the SSRC for an RTP stream changes,
  there can - and almost always will be - a large jump in the next packet's
  timestamp and sequence number. If we don't reset the number of received
  packets, sequence number count, and other metrics used by RTCP, the next RR/SR
  report will use the previous SSRC's values to calculate the lost packet count
  for the new SSRC - resulting in a very large number of lost packets.
  
  This patch modifies res_rtp_asterisk such that, if it detects a SSRC change, it
  will reset the various values used by the RTCP calculations. From the
  perspective of RTCP, this appears as a new media stream - which is what it is.
  
  Review: https://reviewboard.asterisk.org/r/2886/
  
  (closes issue AST-1174)
  Reported by: Thomas Arimont
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  Merged revisions 400089 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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  r401446 | mjordan | 2013-10-22 17:42:24 -0500 (Tue, 22 Oct 2013) | 15 lines
  
  res_rtp_asterisk: Fix crash when RTCP is not available during SSRC change
  
  In r400089, a patch was put in to correct erroneous RTCP statistic resets.
  Unfortunately, ast_rtp_read can be called on an RTP instance that does not
  have RTCP information. This patch prevents that crash by only resetting
  the statistics if we do actually have an RTCP instance.
  
  (issue AST-1174)
  
  (closes issue ASTERISK-22667)
  Reported by: John Bigelow
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  Merged revisions 401445 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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  r401960 | sgriepentrog | 2013-10-25 15:44:40 -0500 (Fri, 25 Oct 2013) | 15 lines
  
  pbx.c: fix confused match caller id that deleted exten still in hash
  
  This fixes a bug where a zero length callerid match adjacent to a no
  match callerid extension entry would be deleted together, which then
  resulted in hashtable references to free'd memory.  A third state of
  the matchcid value has been added to indicate match to any extension
  which allows enforcing comparison of matchcid on/off without errors.
  
  (closes issue AST-1235)
  Reported by: Guenther Kelleter
  Review: https://reviewboard.asterisk.org/r/2930/
  ........
  
  Merged revisions 401959 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 396884,400075,400093,401446,401960 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/certified/branches/11.6@402382 65c4cc65-6c06-0410-ace0-fbb531ad65f3