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20 authors
  1. Jan 26, 2011
  3. Jan 25, 2011
    • Jeff Peeler's avatar
      Merged revisions 303678 via svnmerge from · d3c7a689
      Jeff Peeler authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r303678 | jpeeler | 2011-01-25 11:02:38 -0600 (Tue, 25 Jan 2011) | 33 lines
  
  Merged revisions 303677 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ................
    r303677 | jpeeler | 2011-01-25 10:59:28 -0600 (Tue, 25 Jan 2011) | 26 lines
    
    Merged revisions 303676 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
    ........
      r303676 | jpeeler | 2011-01-25 10:58:29 -0600 (Tue, 25 Jan 2011) | 20 lines
      
      Fix voicemail sequencing for file based storage.
      
      A previous change was made to account for when the number of voicemail messages
      exceeds the max limit to be handled properly, but it caused gaps in the messages
      to not be properly handled. This has now been resolved.
      
      In later non 1.4 branches, it appears that resequencing wasn't even occurring
      due from what appears and accidental code removal.
      
      (closes issue #18498)
      Reported by: JJCinAZ
      Patches: 
            bug18498v2.patch uploaded by jpeeler (license 325)
      
      (closes issue #18486)
      Reported by: bluefox
      Patches: 
            bug18486.patch uploaded by jpeeler (license 325)
    ........
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      d3c7a689
  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/
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      09213439
  7. Jan 20, 2011
  9. Jan 19, 2011
  11. Jan 08, 2011
  13. Jan 07, 2011
  15. Jan 03, 2011
  17. Dec 29, 2010
  19. Dec 28, 2010
  21. Dec 16, 2010
  23. Dec 07, 2010
  25. Dec 02, 2010
  27. Dec 01, 2010
  29. Nov 30, 2010
  31. Nov 27, 2010
  33. Nov 24, 2010
    • Andrew Parisio's avatar
      Meetme use voicemail greet for join/leave announce · 935930d8
      Andrew Parisio authored 
      Added option v(mailbox@[context]) which tells MeetMe where to look for a users greet file.  If one does not exist it clears the v option and defers to the functionality of i/I as/if set by the MeetMe() command.

Review: https://reviewboard.asterisk.org/r/1009/
(closes issue #18297)
Reported by: parisioa
Patches:
	meetme_final_patch_v.diff uploaded by parisioa (license 1153)



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      935930d8
    • Russell Bryant's avatar
      Merged revisions 296002 via svnmerge from · 712ba231
      Russell Bryant authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r296002 | russell | 2010-11-24 11:13:08 -0600 (Wed, 24 Nov 2010) | 52 lines
  
  Merged revisions 296001 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ................
    r296001 | russell | 2010-11-24 11:03:16 -0600 (Wed, 24 Nov 2010) | 45 lines
    
    Merged revisions 296000 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
    ........
      r296000 | russell | 2010-11-24 10:48:39 -0600 (Wed, 24 Nov 2010) | 38 lines
      
      Handle failures building translation paths more effectively.
      
      The problem scenario occurred on a heavily loaded system that was using the
      codec_dahdi module and exceeded the hardware transcoding capacity.  The failure
      mode at that point was not good.  The report came in to us as an Asterisk
      lock-up.  The "core show locks" shows a ton of threads locked up (but no
      obvious deadlock).  Upon deeper investigation, when the system is in this
      state, the CPU was maxed out.  The CPU was being consumed by the Asterisk
      logger spewing messages on every audio frame for calls set up after transcoder
      capacity was reached.
      
      The purpose of this patch is to make Asterisk handle failures to create a
      translation path in a more graceful manner.  If we can't translate, then the
      call just needs to be dropped, as it's not going to work.  These are the
      changes:
      
      1) In set_format() of channel.c (which is called by set_read_format() and
      set_write_format()), it was ignoring if ast_translator_build_path() failed and
      returned NULL.  It now pays attention to that case and returns a result
      reflecting failure.  With this change in place, the bridging code will
      immediately detect a failure and end the bridge instead of proceeding to try to
      bridge frames that can't be translated and making channel drivers freak out by
      sending them frames in a format they weren't expecting.
      
      2) In ast_indicate_data() of channel.c, failure of ast_playtones_start() was
      ignored.  It is now reflected in the return value of the function.  This didn't
      turn out to have any affect on the bug, but seemed like a good change to leave
      in.
      
      3) In app_dial(), when only sending a call to a single endpoint, it will
      attempt to do some bridging of its own of early audio.  It uses
      make_compatible() when it's going to do this.  However, it ignored failure from
      make compatible.  So, even with the fix from #1, if there was early audio going
      through app_dial, there would still be a period of invalid frames passing
      through.  After detecting failure here, Dial() exits.
      
      ABE-2658
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      712ba231
  35. 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/
    ........
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      7c7486ad
  37. Nov 19, 2010
  39. Nov 12, 2010
  41. Oct 26, 2010
    • Jeff Peeler's avatar
      Merged revisions 293119 via svnmerge from · 4d76ac7a
      Jeff Peeler authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r293119 | jpeeler | 2010-10-26 13:49:08 -0500 (Tue, 26 Oct 2010) | 43 lines
  
  Merged revisions 293118 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ................
    r293118 | jpeeler | 2010-10-26 13:33:24 -0500 (Tue, 26 Oct 2010) | 36 lines
    
    Merged revisions 293004 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
    ........
      r293004 | jpeeler | 2010-10-25 17:55:28 -0500 (Mon, 25 Oct 2010) | 29 lines
      
      Fix inprocess_container in voicemail to correctly restrict max messages.
      
      The comparison function logic was off, so the number of sessions for a given
      mailbox were not being incremented properly. This problem caused the maximum
      number of messages per folder to not be respected when simultaneously leaving
      multiple voicemails just below the threshold. 
      
      These problems should be fixed by the above, but just in case:
      Fixed resequence_mailbox to rely on the actual number of detected number of
      files in a directory rather than just assuming only 10 messages more than the
      maximum had been left. Also if more messages than the maximum are deleted they
      are actually removed now.
      
      
      The second purpose of this commit should have been separated out probably, but
      is related to the above. Again, if the number of messages in a given voicemail
      folder exceeds the maximum set limit make sure to allocate enough space for the
      deleted and heard index tracking array.
      
      A few random fixes:
      There was a forgotten decrement of the inprocess count in imap_store_file.
      
      When using IMAP storage, do not look in the directory where file based storage
      messages may still reside and influence the message count.
      
      Ensure to use only the first format in sendmail.
      
      ABE-2516
    ........
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      4d76ac7a
  43. Oct 21, 2010
  45. Oct 18, 2010
  47. Oct 13, 2010
  49. Oct 06, 2010
  51. Oct 05, 2010
  53. Oct 02, 2010
  55. Sep 30, 2010
  57. Sep 28, 2010
  59. Sep 21, 2010
    • Richard Mudgett's avatar
      Merged revisions 288079-288080 via svnmerge from · 851141c1
      Richard Mudgett authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r288079 | rmudgett | 2010-09-21 15:29:51 -0500 (Tue, 21 Sep 2010) | 2 lines
  
  Protect channel access in CONNECTED_LINE and REDIRECTING interception macro launch code.
........
  r288080 | rmudgett | 2010-09-21 15:29:59 -0500 (Tue, 21 Sep 2010) | 8 lines
  
  Simplify locking code for REDIRECTING interception macro when forwarding a call.
  
  Simplified the locking code by using a local copy of the redirecting party
  information in app_dial.c:do_forward() and app_queue.c:wait_for_answer()
  for launching the REDIRECTING interception macro when a call is forwarded.
  
  Reduced the lock time of the 'o->chan' and 'in' channels.
........


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      851141c1
    • Brett Bryant's avatar
      Merged revisions 287760 via svnmerge from · e8de16e9
      Brett Bryant authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r287760 | bbryant | 2010-09-20 20:00:23 -0400 (Mon, 20 Sep 2010) | 30 lines
  
  Merged revisions 287759 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ................
    r287759 | bbryant | 2010-09-20 19:58:26 -0400 (Mon, 20 Sep 2010) | 23 lines
    
    Merged revisions 287758 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
    ........
      r287758 | bbryant | 2010-09-20 19:57:08 -0400 (Mon, 20 Sep 2010) | 16 lines
      
      Fix misvalidation of meetme pins in conjunction with the 'a' MeetMe flag.
      
      When using the 'a' MeetMe flag and having a user and admin pin setup for your
      conference, using the user pin would gain you admin priviledges. Also, when no
      user pin was set, an admin pin was, the 'a' MeetMe flag wasn't used, and the
      user tried to enter a conference then they were still prompted for a pin and
      forced to hit #.
      
      (closes issue #17908)
      Reported by: kuj
      Patches:
            pins_2.patch uploaded by kuj (license 1111)
            Tested by: kuj
      
            Review: [full review board URL with trailing slash]
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      e8de16e9
  61. Sep 17, 2010