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20 authors
  1. Jan 20, 2011
  3. Jan 19, 2011
  5. 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
    • Matthew Nicholson's avatar
      Merged revisions 302314 via svnmerge from · 785e3a14
      Matthew Nicholson authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r302314 | mnicholson | 2011-01-18 15:43:21 -0600 (Tue, 18 Jan 2011) | 18 lines
  
  Merged revisions 302313 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ................
    r302313 | mnicholson | 2011-01-18 15:40:03 -0600 (Tue, 18 Jan 2011) | 11 lines
    
    Merged revisions 302311 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
    ........
      r302311 | mnicholson | 2011-01-18 15:35:03 -0600 (Tue, 18 Jan 2011) | 4 lines
      
      URI encode the user part of the contact header.
      
      ABE-2705
    ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@302315 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      785e3a14
    • 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/
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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.
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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

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  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/
    ........
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      a05aeff3
  7. Jan 17, 2011
  9. Jan 14, 2011
  11. Jan 13, 2011
  13. Jan 12, 2011