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20 authors
  1. Sep 07, 2011
  3. Sep 06, 2011
  5. Sep 02, 2011
    • Richard Mudgett's avatar
      Merged revisions 334357 via svnmerge from · 35e27201
      Richard Mudgett authored 
      https://origsvn.digium.com/svn/asterisk/branches/10

................
  r334357 | rmudgett | 2011-09-02 16:08:16 -0500 (Fri, 02 Sep 2011) | 26 lines
  
  Merged revisions 334355 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
  ........
    r334355 | rmudgett | 2011-09-02 15:59:49 -0500 (Fri, 02 Sep 2011) | 19 lines
    
    MusicOnHold has extra unref which may lead to memory corruption and crash.
    
    The problem happens when a call is disconnected and you had started a MOH 
    class that does not use the files mode.  If you define REF_DEBUG and 
    recreate the problem, it will announce itself with the following warning: 
    Attempt to unref mohclass 0xb70722e0 (default) when only 1 ref remained, 
    and class is still in a container!  
    
    * Fixed moh_alloc() and moh_release() functions not handling the
    state->class reference consistently.
    
    (closes issue ASTERISK-18346)
    Reported by: Mark Murawski
    Patches:
          jira_asterisk_18346_v1.8.patch (license #5621) patch uploaded by rmudgett
    Tested by: rmudgett, Mark Murawski
    
    Review: https://reviewboard.asterisk.org/r/1404/
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@334358 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      35e27201
    • Richard Mudgett's avatar
      Merged revisions 334297 via svnmerge from · 220bf145
      Richard Mudgett authored 
      https://origsvn.digium.com/svn/asterisk/branches/10

................
  r334297 | rmudgett | 2011-09-02 12:15:08 -0500 (Fri, 02 Sep 2011) | 46 lines
  
  Merged revisions 334296 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
  ........
    r334296 | rmudgett | 2011-09-02 12:10:58 -0500 (Fri, 02 Sep 2011) | 39 lines
    
    Fix potential memory allocation failure crashes in config.c.
    
    * Added required checks to the returned memory allocation pointers to
    prevent crashes.
    
    * Made ast_include_rename() create a replacement ast_variable list node if
    the new filename is longer than the available space.  Fixes potential
    crash and memory leak.
    
    * Factored out ast_variable_move() from ast_variable_update() so
    ast_include_rename() can also use it when creating a replacement
    ast_variable list node.
    
    * Made the filename stuffed at the end of the struct a minimum allocated
    size in ast_variable_new() in case ast_include_rename() changes the stored
    filename.
    
    * Constify struct char pointers pointing to strings stuffed at the end of
    the struct for: ast_variable, cache_file_mtime, and ast_config_map.
    
    * Factored out cfmtime_new() to remove inlined code and allow some struct
    pointers to become const.
    
    * Removed the list lock from struct cache_file_mtime that was never used.
    
    * Added doxygen comments to several structure elements and better
    documented what strings are stuffed at the struct end char array.
    
    * Reworked ast_config_text_file_save() and set_fn() to handle allocation
    failure of the include file scratch pad object tracking blank lines.
    
    * Made ast_config_text_file_save() fn[] declared with PATH_MAX to ensure
    it is long enough for any filename with path.  Also reduced the number of
    container fileset buckets from a rediculus 180,000 to 1023.
    
    JIRA AST-618
    
    Review: https://reviewboard.asterisk.org/r/1378/
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@334304 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      220bf145
  7. Sep 01, 2011
  9. Aug 31, 2011
    • Matthew Nicholson's avatar
      Merged revisions 334157 via svnmerge from · 9dd15059
      Matthew Nicholson authored 
      https://origsvn.digium.com/svn/asterisk/branches/10

................
  r334157 | mnicholson | 2011-08-31 13:53:40 -0500 (Wed, 31 Aug 2011) | 11 lines
  
  Merged revisions 334156 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
  ........
    r334156 | mnicholson | 2011-08-31 13:50:33 -0500 (Wed, 31 Aug 2011) | 4 lines
    
    Disable T.38 when we get a invite with image media port set to 0
    
    ASTERISK-17678
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@334158 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      9dd15059
    • Richard Mudgett's avatar
      Optimize chan_sip.c check_rtp_timeout() function. · 89e79698
      Richard Mudgett authored 
      * Make check_rtp_timeout() remember the values returned by
ast_rtp_instance_get_timeout(), ast_rtp_instance_get_hold_timeout(), and
ast_rtp_instance_get_keepalive() instead of repeatedly calling them.

(closes issue ASTERISK-18319)
Reported by: Rob Gagnon
Patches:
      issue-18319-trunk-r333066.diff (License #6159) patch uploaded by Rob Gagnon

Review: https://reviewboard.asterisk.org/r/1377/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@334115 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      89e79698
    • Matthew Nicholson's avatar
      Merged revisions 334064 via svnmerge from · dadc749d
      Matthew Nicholson authored 
      https://origsvn.digium.com/svn/asterisk/branches/10

........
  r334064 | mnicholson | 2011-08-31 11:31:00 -0500 (Wed, 31 Aug 2011) | 4 lines
  
  only alter the gateway_timeout when attching the gateway to a channel
  
  ASTERISK-18219
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@334067 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      dadc749d
    • Richard Mudgett's avatar
      Merged revisions 334013 via svnmerge from · 1961bb61
      Richard Mudgett authored 
      https://origsvn.digium.com/svn/asterisk/branches/10

................
  r334013 | rmudgett | 2011-08-31 11:00:49 -0500 (Wed, 31 Aug 2011) | 30 lines
  
  Merged revisions 334012 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
  ........
    r334012 | rmudgett | 2011-08-31 10:57:12 -0500 (Wed, 31 Aug 2011) | 23 lines
    
    No DAHDI channel available for conference, user introduction disabled.
    
    The following error will consistently occur when trying to dial into a
    MeetMe conference when the server does not have DAHDI hardware installed:
    
    app_meetme.c: No DAHDI channel available for conference, user introduction
    disabled (is chan_dahdi loaded?)
    
    While chan_dahdi is loaded correctly during compilation and install of
    Asterisk/Dahdi, including associated modules, etc., a chan_dahdi.conf
    configuration file in /etc/asterisk is not created by FreePBX if hardware
    does not exist, causing MeetMe to be unable to open a DAHDI pseudo
    channel.
    
    * Allow chan_dahdi to create a pseudo channel when there is no
    chan_dahdi.conf file to load.
    
    (closes issue ASTERISK-17398)
    Reported by: Preston Edwards
    Patches:
          jira_asterisk_17398_v1.8.patch (license #5621) patch uploaded by rmudgett
    Tested by: rmudgett
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@334014 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      1961bb61
    • Richard Mudgett's avatar
      Merged revisions 334010 via svnmerge from · ab17a27f
      Richard Mudgett authored 
      https://origsvn.digium.com/svn/asterisk/branches/10

................
  r334010 | rmudgett | 2011-08-31 10:23:11 -0500 (Wed, 31 Aug 2011) | 50 lines
  
  Merged revisions 334009 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
  ........
    r334009 | rmudgett | 2011-08-31 10:20:31 -0500 (Wed, 31 Aug 2011) | 43 lines
    
    Call pickup race leaves orphaned channels or crashes.
    
    Multiple users attempting to pickup a call that has been forked to
    multiple extensions either crashes or fails a masquerade with a "bad
    things may happen" message.
    
    This is the scenario that is causing all the grief:
    1) Pickup target is selected
    2) target is marked as being picked up in ast_do_pickup()
    3) target is unlocked by ast_do_pickup()
    4) app dial or queue gets a chance to hang up losing calls and calls
    ast_hangup() on target
    5) SINCE A MASQUERADE HAS NOT BEEN SETUP YET BY ast_do_pickup() with
    ast_channel_masquerade(), ast_hangup() completes successfully and the
    channel is no longer in the channels container.
    6) ast_do_pickup() then calls ast_channel_masquerade() to schedule the
    masquerade on the dead channel.
    7) ast_do_pickup() then calls ast_do_masquerade() on the dead channel
    8) bad things happen while doing the masquerade and in the process
    ast_do_masquerade() puts the dead channel back into the channels container
    9) The "orphaned" channel is visible in the channels list if a crash does
    not happen.
    
    This patch does the following:
    
    * Made ast_hangup() set AST_FLAG_ZOMBIE on a successfully hung-up channel
    and not release the channel lock until that has happened.
    
    * Made __ast_channel_masquerade() not setup a masquerade if either channel
    has AST_FLAG_ZOMBIE set.
    
    * Fix chan_agent misuse of AST_FLAG_ZOMBIE since it would no longer work.
    
    (closes issue ASTERISK-18222)
    Reported by: Alec Davis
    Tested by: rmudgett, Alec Davis, irroot, Karsten Wemheuer
    
    (closes issue ASTERISK-18273)
    Reported by: Karsten Wemheuer
    Tested by: rmudgett, Alec Davis, irroot, Karsten Wemheuer
    
    Review: https://reviewboard.asterisk.org/r/1400/
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@334011 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      ab17a27f
    • Kinsey Moore's avatar
      Merged revisions 334007 via svnmerge from · 82229cc6
      Kinsey Moore authored 
      https://origsvn.digium.com/svn/asterisk/branches/10

................
  r334007 | kmoore | 2011-08-31 10:19:30 -0500 (Wed, 31 Aug 2011) | 14 lines
  
  Merged revisions 334006 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
  ........
    r334006 | kmoore | 2011-08-31 10:18:37 -0500 (Wed, 31 Aug 2011) | 7 lines
    
    Correct an AMI protocol violation with SIPshowpeer
    
    The response of SIPshowpeer ends with "\r\n\r\n". Since other commands are
    ended by using \r\n this confuses any interfacing script.
    
    (closes issue ASTERISK-17486)
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@334008 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      82229cc6
  11. Aug 30, 2011
  13. Aug 29, 2011
  15. Aug 28, 2011
  17. Aug 26, 2011
  19. Aug 25, 2011
  21. Aug 24, 2011