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20 authors
  1. Jan 31, 2011
  3. Jan 28, 2011
  5. Jan 27, 2011
  7. Jan 26, 2011
  9. Jan 25, 2011
  11. 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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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
  13. Jan 20, 2011
    • Richard Mudgett's avatar
      Merged revisions 303153 via svnmerge from · 9974f89a
      Richard Mudgett authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r303153 | rmudgett | 2011-01-20 14:31:20 -0600 (Thu, 20 Jan 2011) | 22 lines
  
  Merged revision 303098 from
  https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
  
  ..........
    r303098 | rmudgett | 2011-01-20 12:11:45 -0600 (Thu, 20 Jan 2011) | 15 lines
  
    CC_INTERFACES does not get built correctly with local channels.
  
    If local channels are used with CCSS, CC_INTERFACES gets garbage prepended
    to it so the CC recall fails.  Also CC_INTERFACES gets "&(null)" appended
    to it.
  
    * Initialize the buffer to eliminate the prepended garbage.
  
    * Filter out the empty interface strings to eliminate the latter.
  
    * Added a diagnostic message if the CC_INTERFACES is ever empty.
  
    JIRA ABE-2740
    JIRA SWP-2848
  ..........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@303154 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      9974f89a
    • Shaun Ruffell's avatar
      Merged revisions 303107 via svnmerge from · 80f6848c
      Shaun Ruffell authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r303107 | sruffell | 2011-01-20 13:57:31 -0600 (Thu, 20 Jan 2011) | 23 lines
  
  Merged revisions 303106 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ........
    r303106 | sruffell | 2011-01-20 13:56:34 -0600 (Thu, 20 Jan 2011) | 15 lines
    
    main/features: Use POLLPRI when waiting for events on parked channels.
    
    This change resolves a regression in the 1.6.2 when converting from
    select to poll.  The DAHDI timers use POLLPRI to indicate that the timer
    fired, but features was not waiting for that flag.  The result was no
    audio for MOH when a call was parked and res_timing_dahdi was in use.
    
    This patch is slightly modified from the one on the mantis issue.  It does
    not set an exception on the channel if the POLLPRI flag is set.
    
    (closes issue #18262)
    Reported by: francesco_r
    Patches:
          patch_park_moh-trunk-2.txt uploaded by cjacobsen (license 1029)
          Tested by: francesco_r, rfrantik, one47
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@303108 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      80f6848c
  15. Jan 19, 2011
  17. Jan 18, 2011
    • Richard Mudgett's avatar
      Merged revisions 302318 via svnmerge from · 8cd1ac53
      Richard Mudgett authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r302318 | rmudgett | 2011-01-18 16:04:14 -0600 (Tue, 18 Jan 2011) | 1 line
  
  Use the expanded format type instead of plain int.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@302319 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      8cd1ac53
    • Jeff Peeler's avatar
      Merged revisions 302266 via svnmerge from · b1f9f1e7
      Jeff Peeler authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r302266 | jpeeler | 2011-01-18 14:19:57 -0600 (Tue, 18 Jan 2011) | 34 lines
  
  Merged revisions 302265 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ........
    r302265 | jpeeler | 2011-01-18 14:13:52 -0600 (Tue, 18 Jan 2011) | 27 lines
    
    Convert device state callbacks to ao2 objects to fix a deadlock in chan_sip.
    
    Lock scenario presented here:
    Thread 1
     holds ast_rdlock_contexts &conlock
     holds handle_statechange hints
     holds handle_statechange hint
      waiting for cb_extensionstate
       Locked Here: chan_sip.c line 7428 (find_call)
    Thread 2
     holds handle_request_do &netlock
     holds find_call sip_pvt_ptr
      waiting for ast_rdlock_contexts &conlock
       Locked Here: pbx.c line 9911 (ast_rdlock_contexts)
    
    Chan_sip has an established locking order of locking the sip_pvt and then
    getting the context lock. So the as stated by the summary, the operations in
    thread 2 have been modified to no longer require the context lock.
    
    (closes issue #18310)
    Reported by: one47
    Patches: 
          statecbs_ao2.mk2.patch uploaded by one47 (license 23),
          modified by me
    
    Review: https://reviewboard.asterisk.org/r/1072/
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@302270 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      b1f9f1e7
    • Russell Bryant's avatar
      Merged revisions 302267 via svnmerge from · 519b766c
      Russell Bryant authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r302267 | russell | 2011-01-18 14:19:57 -0600 (Tue, 18 Jan 2011) | 5 lines
  
  Don't enable AO2_DEBUG by default if AST_DEVMODE is on.
  
  AO2_DEBUG is not important and is causing a false compiler warning to be
  generated on my Ubuntu Natty dev box.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@302268 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      519b766c
    • Richard Mudgett's avatar
      Merged revisions 302174 via svnmerge from · a05aeff3
      Richard Mudgett authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r302174 | rmudgett | 2011-01-18 12:11:43 -0600 (Tue, 18 Jan 2011) | 102 lines
  
  Merged revisions 302173 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ................
    r302173 | rmudgett | 2011-01-18 12:07:15 -0600 (Tue, 18 Jan 2011) | 95 lines
    
    Merged revisions 302172 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
    ........
      r302172 | rmudgett | 2011-01-18 12:04:36 -0600 (Tue, 18 Jan 2011) | 88 lines
      
      Issues with DTMF triggered attended transfers.
      
      Issue #17999
      1) A calls B. B answers.
      2) B using DTMF dial *2 (code in features.conf for attended transfer).
      3) A hears MOH. B dial number C
      4) C ringing. A hears MOH.
      5) B hangup. A still hears MOH. C ringing.
      6) A hangup. C still ringing until "atxfernoanswertimeout" expires.
      For v1.4 C will ring forever until C answers the dead line. (Issue #17096)
      
      Problem: When A and B hangup, C is still ringing.
      
      Issue #18395
      SIP call limit of B is 1
      1. A call B, B answered
      2. B *2(atxfer) call C
      3. B hangup, C ringing
      4. Timeout waiting for C to answer
      5. Recall to B fails because B has reached its call limit.
      
      Because B reached its call limit, it cannot do anything until the transfer
      it started completes.
      
      Issue #17273
      Same scenario as issue 18395 but party B is an FXS port.  Party B cannot
      do anything until the transfer it started completes.  If B goes back off
      hook before C answers, B hears ringback instead of the expected dialtone.
      
      **********
      Note for the issue #17273 and #18395 fix:
      
      DTMF attended transfer works within the channel bridge.  Unfortunately,
      when either party A or B in the channel bridge hangs up, that channel is
      not completely hung up until the transfer completes.  This is a real
      problem depending upon the channel technology involved.
      
      For chan_dahdi, the channel is crippled until the hangup is complete.
      Either the channel is not useable (analog) or the protocol disconnect
      messages are held up (PRI/BRI/SS7) and the media is not released.
      
      For chan_sip, a call limit of one is going to block that endpoint from any
      further calls until the hangup is complete.
      
      For party A this is a minor problem.  The party A channel will only be in
      this condition while party B is dialing and when party B and C are
      conferring.  The conversation between party B and C is expected to be a
      short one.  Party B is either asking a question of party C or announcing
      party A.  Also party A does not have much incentive to hangup at this
      point.
      
      For party B this can be a major problem during a blonde transfer.  (A
      blonde transfer is our term for an attended transfer that is converted
      into a blind transfer.  :)) Party B could be the operator.  When party B
      hangs up, he assumes that he is out of the original call entirely.  The
      party B channel will be in this condition while party C is ringing, while
      attempting to recall party B, and while waiting between call attempts.
      
      WARNING:
      The ATXFER_NULL_TECH conditional is a hack to fix the problem.  It will
      replace the party B channel technology with a NULL channel driver to
      complete hanging up the party B channel technology.  The consequences of
      this code is that the 'h' extension will not be able to access any channel
      technology specific information like SIP statistics for the call.
      
      ATXFER_NULL_TECH is not defined by default.
      **********
      
      (closes issue #17999)
      Reported by: iskatel
      Tested by: rmudgett
      JIRA SWP-2246
      
      (closes issue #17096)
      Reported by: gelo
      Tested by: rmudgett
      JIRA SWP-1192
      
      (closes issue #18395)
      Reported by: shihchuan
      Tested by: rmudgett
      
      (closes issue #17273)
      Reported by: grecco
      Tested by: rmudgett
      
      Review: https://reviewboard.asterisk.org/r/1047/
    ........
  ................
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@302178 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      a05aeff3
  19. Jan 13, 2011
  21. Jan 12, 2011
  23. Jan 11, 2011
  25. Jan 10, 2011
  27. Jan 04, 2011
  29. Jan 03, 2011
  31. Dec 31, 2010
  33. Dec 29, 2010