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20 authors
  1. Jan 31, 2011
  3. Jan 26, 2011
  5. Jan 24, 2011
    • Russell Bryant's avatar
      Merged revisions 303549 via svnmerge from · 09213439
      Russell Bryant authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r303549 | russell | 2011-01-24 14:51:37 -0600 (Mon, 24 Jan 2011) | 45 lines
  
  Merged revisions 303548 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ................
    r303548 | russell | 2011-01-24 14:49:53 -0600 (Mon, 24 Jan 2011) | 38 lines
    
    Merged revisions 303546 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
    ........
      r303546 | russell | 2011-01-24 14:32:21 -0600 (Mon, 24 Jan 2011) | 31 lines
      
      Fix channel redirect out of MeetMe() and other issues with channel softhangup.
      
      Mantis issue #18585 reports that a channel redirect out of MeetMe() stopped
      working properly.  This issue includes a patch that resolves the issue by
      removing a call to ast_check_hangup() from app_meetme.c.  I left that in my
      patch, as it doesn't need to be there.  However, the rest of the patch fixes
      this problem with or without the change to app_meetme.
      
      The key difference between what happens before and after this patch is the
      effect of the END_OF_Q control frame.  After END_OF_Q is hit in ast_read(),
      ast_read() will return NULL.  With the ast_check_hangup() removed, app_meetme
      sees this which causes it to exit as intended.  Checking ast_check_hangup()
      caused app_meetme to exit earlier in the process, and the target of the
      redirect saw the condition where ast_read() returned NULL.
      
      Removing ast_check_hangup() works around the issue in app_meetme, but doesn't
      solve the issue if another application did the same thing.  There are also
      other edge cases where if an application finishes at the same time that a
      redirect happens, the target of the redirect will think that the channel hung
      up.  So, I made some changes in pbx.c to resolve it at a deeper level.  There
      are already places that unset the SOFTHANGUP_ASYNCGOTO flag in an attempt to
      abort the hangup process.  My patch extends this to remove the END_OF_Q frame
      from the channel's read queue, making the "abort hangup" more complete.  This
      same technique was used in every place where a softhangup flag was cleared.
      
      (closes issue #18585)
      Reported by: oej
      Tested by: oej, wedhorn, russell
      
      Review: https://reviewboard.asterisk.org/r/1082/
    ........
  ................
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@303551 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      09213439
    • Matthew Nicholson's avatar
      According to section 19.1.2 of RFC 3261: · e706b570
      Matthew Nicholson authored 
        For each component, the set of valid BNF expansions defines exactly
  which characters may appear unescaped.  All other characters MUST be
  escaped.

This patch modifies ast_uri_encode() to encode strings in line with this recommendation.  This patch also adds an ast_escape_quoted() function which escapes '"' and '\' characters in quoted strings in accordance with section 25.1 of RFC 3261.  The ast_uri_encode() function has also been modified to take an ast_flags struct describing the set of rules it should use when escaping characters to allow for it to escape SIP URIs in addition to HTTP URIs and other types of URIs or variations of those two URI types in the future.

The ast_uri_decode() function has also been modified to accept an ast_flags struct describing the set of rules to use when decoding to enable decoding '+' as ' ' in legacy http URLs.

The unit tests for these functions have also been updated.

ABE-2705

Review: https://reviewboard.asterisk.org/r/1081/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@303509 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      e706b570
  7. Jan 19, 2011
  9. Dec 20, 2010
  11. Dec 18, 2010
  13. Dec 12, 2010
  15. Dec 10, 2010
  17. Dec 03, 2010
    • Matthew Nicholson's avatar
      Merged revisions 297157,297486,297495 via svnmerge from · 23d106b8
      Matthew Nicholson authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r297157 | mnicholson | 2010-12-01 13:47:33 -0600 (Wed, 01 Dec 2010) | 2 lines
  
  Changed some NOTICE and WARNING messages to DEBUG messages.
........
  r297486 | mnicholson | 2010-12-02 15:30:47 -0600 (Thu, 02 Dec 2010) | 6 lines
  
  Add support for reserving a fax session before answering the channel.
  
  Note: this change breaks ABI compatibility.
  
  FAX-217
........
  r297495 | mnicholson | 2010-12-03 09:21:52 -0600 (Fri, 03 Dec 2010) | 4 lines
  
  Print a DEBUG message instead of a WARNING message when the selected fax tech does not support reserving sessions.
  
  Answer the channel before quering it for t.38 support.  This is necessary for the query to work properly over local channels.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@297496 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      23d106b8
  19. Dec 01, 2010
  21. Nov 30, 2010
  23. Nov 29, 2010
  25. Nov 27, 2010
  27. Nov 22, 2010
    • Richard Mudgett's avatar
      Merged revisions 295866 via svnmerge from · 7c7486ad
      Richard Mudgett authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r295866 | rmudgett | 2010-11-22 13:36:10 -0600 (Mon, 22 Nov 2010) | 60 lines
  
  Merged revisions 295843 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ................
    r295843 | rmudgett | 2010-11-22 13:28:23 -0600 (Mon, 22 Nov 2010) | 53 lines
    
    Merged revisions 295790 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
    ........
      r295790 | rmudgett | 2010-11-22 12:46:26 -0600 (Mon, 22 Nov 2010) | 46 lines
      
      The channel redirect function (CLI or AMI) hangs up the call instead of redirecting the call.
      
      To recreate the problem:
      1) Party A calls Party B
      2) Invoke CLI "channel redirect" command to redirect channel call leg
      associated with A.
      3) All associated channels are hung up.
      
      Note that if the CLI command were done on the channel call leg associated
      with B it works.
      
      This regression was a result of the fix for issue #16946
      (https://reviewboard.asterisk.org/r/740/).
      
      The regression affects all features that use an async goto to execute the
      dialplan because of an external event: Channel redirect, AMI redirect, SIP
      REFER, and FAX detection.
      
      The struct ast_channel._softhangup code is a mess.  The variable is used
      for several purposes that do not necessarily result in the call being hung
      up.  I have added doxygen comments to describe how the various _softhangup
      bits are used.  I have corrected all the places where the variable was
      tested in a non-bit oriented manner.
      
      The primary fix is the new AST_CONTROL_END_OF_Q frame.  It acts as a weak
      hangup request so the soft hangup requests that do not normally result in
      a hangup do not hangup.
      
      JIRA SWP-2470
      JIRA SWP-2489
      
      (closes issue #18171)
      Reported by: SantaFox
      (closes issue #18185)
      Reported by: kwemheuer
      (closes issue #18211)
      Reported by: zahir_koradia
      (closes issue #18230)
      Reported by: vmarrone
      (closes issue #18299)
      Reported by: mbrevda
      (closes issue #18322)
      Reported by: nerbos
      
      Review:	https://reviewboard.asterisk.org/r/1013/
    ........
  ................
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@295867 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      7c7486ad
  29. Nov 20, 2010
    • Russell Bryant's avatar
      Merged revisions 295711 via svnmerge from · 9fbbdfb2
      Russell Bryant authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r295711 | russell | 2010-11-19 18:50:00 -0600 (Fri, 19 Nov 2010) | 36 lines
  
  Merged revisions 295710 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ........
    r295710 | russell | 2010-11-19 18:45:51 -0600 (Fri, 19 Nov 2010) | 29 lines
    
    Fix cache of device state changes for multiple servers.
    
    This patch addresses a regression where device states across multiple servers
    were not being processing completely correctly.  The code works to determine
    the overall state by looking at the last known state of a device on each
    server.  However, there was a regression due to some invasive rewrites of how
    the cache works that led to the cache only storing the last device state change
    for a device, regardless of which server it was on.
    
    The code is set up to cache device state change events by ensuring that each
    event in the cache has a unique device name + entity ID (server ID).  The code
    that was responsible for comparing raw information elements (which EID is)
    always returned a match due to a memcmp() with a length of 0.
    
    There isn't much code to fix the actual bug.  This patch also introduces a new
    CLI command that was very useful for debugging this problem.  The command
    allows you to dump the contents of the event cache.
    
    (closes issue #18284)
    Reported by: klaus3000
    Patches:
          issue18284.rev1.txt uploaded by russell (license 2)
    Tested by: russell, klaus3000
    
    (closes issue #18280)
    Reported by: klaus3000
    
    Review: https://reviewboard.asterisk.org/r/1012/
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@295712 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      9fbbdfb2
  31. Nov 18, 2010
  33. Nov 09, 2010
  35. Nov 08, 2010
    • Jeff Peeler's avatar
      Merged revisions 294278 via svnmerge from · 12a40275
      Jeff Peeler authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r294278 | jpeeler | 2010-11-08 15:59:45 -0600 (Mon, 08 Nov 2010) | 23 lines
  
  Merged revisions 294277 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ........
    r294277 | jpeeler | 2010-11-08 15:58:13 -0600 (Mon, 08 Nov 2010) | 16 lines
    
    Fix playback failure when using IAX with the timerfd module.
    
    To fix this issue the alert pipe will now be used when the timerfd module is
    in use. There appeared to be a race that was not solved by adding locking in the
    timerfd module, but needed to be there anyway. The race was between the timer
    being put in non-continuous mode in ast_read on the channel thread and the IAX 
    frame scheduler queuing a frame which would enable continuous mode before the
    non-continuous mode event was read. This race for now is simply avoided.
    
    (closes issue #18110)
    Reported by: tpanton
    Tested by: tpanton
    
    I put tested by tpanton because it was tested on his hardware. Thanks for the
    remote access to debug this issue!
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294279 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      12a40275
  37. Nov 03, 2010
    • Terry Wilson's avatar
      Merged revisions 293803 via svnmerge from · abc94089
      Terry Wilson authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r293803 | twilson | 2010-11-03 11:05:14 -0700 (Wed, 03 Nov 2010) | 25 lines
  
  Avoid valgrind warnings for ast_rtp_instance_get_xxx_address
  
  The documentation for ast_rtp_instance_get_(local/remote)_address stated that
  they returned 0 for success and -1 on failure. Instead, they returned 0 if the
  address structure passed in was already equivalent to the address instance
  local/remote address or 1 otherwise. 90% of the calls to these functions
  completely ignored the return address and passed in an uninitialized struct,
  which would make valgrind complain even though the operation was technically
  safe.
  
  This patch fixes the documentation and converts the get_xxx_address functions
  to void since all they really do is copy the address and cannot fail.
  Additionally two new functions
  (ast_rtp_instance_get_and_cmp_(local/remote)_address) are created for the 3
  times where the return value was actually checked. The
  get_and_cmp_local_address function is currently unused, but exists for the sake
  of symmetry.
  
  The only functional change as a result of this change is that we will not do an
  ast_sockaddr_cmp() on (mostly uninitialized) addresses before doing the
  ast_sockaddr_copy() in the get_*_address functions. So, even though it is an
  API change, it shouldn't have a noticeable change in behavior.
  
  Review: https://reviewboard.asterisk.org/r/995/
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@293809 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      abc94089
  39. Oct 26, 2010
  41. Oct 25, 2010
  43. Oct 14, 2010
  45. Oct 11, 2010
    • David Vossel's avatar
      Merged revisions 291192 via svnmerge from · 0736871c
      David Vossel authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r291192 | dvossel | 2010-10-11 16:38:39 -0500 (Mon, 11 Oct 2010) | 19 lines
  
  Gtalk enhancements and general code cleanup.
  
  This patch includes several chan_gtalk enhancements.
  Two new gtalk.conf options have been added, externip
  and stunadd.  Setting externip allows us to
  manually specify what the external IP address is
  outside of a NAT environment.  Setting the stunaddr
  option to a valid stun server allows for that external
  ip to be retrieved via a STUN server automatically.  This
  external IP is then advertised during call setup as
  a possible candidate.
  
  I have also attempted to clean up chan_gtalk's code
  so it meets our coding guidelines. During this cleanup
  I noticed several things that need to be done in the
  code and made a TODO section at the top of the file.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@291193 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      0736871c
  47. Oct 07, 2010
  49. Oct 05, 2010
  51. Oct 02, 2010
  53. Sep 30, 2010
  55. Sep 28, 2010
  57. Sep 24, 2010
  59. Sep 20, 2010
    • David Vossel's avatar
      Merged revisions 287647 via svnmerge from · 2f3dee23
      David Vossel authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r287647 | dvossel | 2010-09-20 17:09:16 -0500 (Mon, 20 Sep 2010) | 21 lines
  
  Addition of the FrameHook API (AKA AwesomeHooks)
  
  So far all our tools for viewing and manipulating media streams
  within Asterisk have been entirely focused on audio.  That made
  sense then, but is not scalable now.  The FrameHook API lets us
  tap into and manipulate _ANY_ type of media or signaling passed
  on a channel present today or in the future.  This tool is a step
  in the direction of expanding Asterisk's boundaries and will help
  generate some rather interesting applications in the future.
  
  In addition to the FrameHook API, a simple dialplan function
  exercising the api has been included as well.  This function
  is called FRAME_TRACE().  FRAME_TRACE() allows for the internal
  ast_frames read and written to a channel to be output.  Filters
  can be placed on this function to debug only certain types of frames.
  This function could be thought of as an internal way of doing
  ast_frame packet captures.
  
  Review: https://reviewboard.asterisk.org/r/925/
........



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@287648 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      2f3dee23
  61. Sep 15, 2010
  63. Sep 10, 2010