https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r313615 | rmudgett | 2011-04-13 12:18:49 -0500 (Wed, 13 Apr 2011) | 5 lines
  
  * Add missing channel lock to handle_cli_agi_add_cmd().
  
  * Flush any Async AGI commands left over from earlier Async AGI control of
  the call.
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  r313588 | rmudgett | 2011-04-13 11:31:50 -0500 (Wed, 13 Apr 2011) | 55 lines
  
  Merged revisions 313579 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ................
    r313579 | rmudgett | 2011-04-13 11:29:49 -0500 (Wed, 13 Apr 2011) | 48 lines
    
    Merged revisions 313545 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
    ........
      r313545 | rmudgett | 2011-04-13 11:21:24 -0500 (Wed, 13 Apr 2011) | 41 lines
      
      Asterisk does not hangup a channel after endpoint hangs up.
      
      If the call that the dialplan started an AGI script for is hungup while
      the AGI script is in the middle of a command then the AGI script is not
      notified of the hangup.  There are many AGI Exec commands that this can
      happen with.  The reported applications have been: Background, Wait, Read,
      and Dial.  Also the AGI Get Data command.
      
      * Don't wait on the Asterisk channel after it has hung up.  The channel is
      likely to never need servicing again.
      
      * Restored the AGI script's ability to return the AGI_RESULT_HANGUP value
      in run_agi().  It previously only could return AGI_RESULT_SUCCESS or
      AGI_RESULT_FAILURE after the DeadAGI and AGI applications were merged.
      
      (closes issue #17954)
      Reported by: mn3250
      Patches:
            issue17954_v1.8.patch uploaded by rmudgett (license 664)
            issue17954_v1.6.2.patch uploaded by rmudgett (license 664)
            issue17954_v1.4.patch uploaded by rmudgett (license 664)
      Tested by: rmudgett
      JIRA SWP-2171
      
      (closes issue #18492)
      Reported by: devmod
      Tested by: rmudgett
      JIRA SWP-2761
      
      (closes issue #18935)
      Reported by: nvitaly
      Tested by: astmiv, rmudgett
      JIRA SWP-3216
      
      (closes issue #17393)
      Reported by: siby
      Tested by: rmudgett
      JIRA SWP-2727
      
      Review: https://reviewboard.asterisk.org/r/1165/
    ........
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(closes issue #19076)
Reported by: lmadsen
Patches: 
      __20110408-channel-description.txt uploaded by lmadsen (license 10)
Tested by: lmadsen

Review: https://reviewboard.asterisk.org/r/1163/

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  r313517 | rmudgett | 2011-04-12 17:35:53 -0500 (Tue, 12 Apr 2011) | 12 lines
  
  Bring the dumpchan application inline with "core show channel".
  
  * Added fields that are in "core show channel" to dumpchan output.
  
  * Fixed reuse of formatbuf before the previous string stored there was
  used by snprintf.  All output strings now have their own buffer.
  
  * Adjusted the buffer sizes to not be so abusive of the stack now that
  there are more buffers.
  
  Change requested by oej.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@313527 65c4cc65-6c06-0410-ace0-fbb531ad65f3

IPv6 support for ooh323,
bindaddr, peers and users ip can be IPv4 or IPv6 addr
correction for multi-homed mode (0.0.0.0 or :: bindaddr)
can work in dual 6/4 mode with :: bindaddr
gatekeeper mode isn't supported in v6 mode while

(issue #18278)
Reported by: may213
Patches: 
      ipv6-ooh323.patch uploaded by may213 (license 454)

Review: https://reviewboard.asterisk.org/r/1004/



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also went ahead and fixed the problem it introduces before committing.

........
  r313435 | jrose | 2011-04-12 13:44:44 -0500 (Tue, 12 Apr 2011) | 1 line

  fixing stupid mistake with putting code before variable declaration
  ........

    Merged revisions 313433 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	
    ........

      r313432 | jrose | 2011-04-12 13:12:29 -0500 (Tue, 12 Apr 2011) | 14 lines

      reload Chan_dahdi memory leak caused by variables

      chan_dahdi reloading with variables set via setvar in chan_dahdi.conf would
      stay in the dahdi_pvt structs for individual channels (causing them to just
      continue adding the new ones to the list) and also there was a memory leak
      causes by the conf objects. This patch resolves both of these by using 
      ast_variables_destroy during the loading process.

      (closes issue #17450)
      Reported by: nahuelgreco
      Patches:
          patch.diff uploaded by jrose (license 1225)
          Tested by: tilghman, jrose
      Review: https://reviewboard.asterisk.org/r/1170/
    
    ........
																	  
  ........																							
  
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  r313368 | rmudgett | 2011-04-11 18:03:02 -0500 (Mon, 11 Apr 2011) | 2 lines
  
  Backport a restructuring change from trunk to make the next change stand out.
........
  r313369 | rmudgett | 2011-04-11 18:08:02 -0500 (Mon, 11 Apr 2011) | 13 lines
  
  Frames from the inbound channel should go to all outbound channels in app_dial.c.
  
  In app_dial.c:wait_for_answer() frames from the inbound channel should be
  sent to all outbound channels instead of only if there is just one
  outbound channel.
  
  Control frames like AST_CONTROL_CONNECTED_LINE need to be passed to all of
  the the outbound channels.  This can happen if a blond transfer is done by
  a remote switch on the inbound channel.
  
  JIRA AST-443
  JIRA SWP-2730
........


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  r313366 | rmudgett | 2011-04-11 17:27:25 -0500 (Mon, 11 Apr 2011) | 2 lines
  
  Added "Connected Line ID" and "Connected Line ID Name" to "core show channel" output.
........


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  r313279 | lmadsen | 2011-04-11 14:36:40 -0500 (Mon, 11 Apr 2011) | 21 lines
  
  Merged revisions 313278 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ................
    r313278 | lmadsen | 2011-04-11 14:33:03 -0500 (Mon, 11 Apr 2011) | 14 lines
    
    Merged revisions 313277 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
    ........
      r313277 | lmadsen | 2011-04-11 14:30:20 -0500 (Mon, 11 Apr 2011) | 6 lines
      
      Fix detection of OpenSSL 1.0
      
      (closes issue #19093)
      Reported by: tzafrir
      Patches: 
            detect_openssl_10.diff uploaded by tzafrir (license 46)
    ........
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  r313190 | rmudgett | 2011-04-11 10:40:30 -0500 (Mon, 11 Apr 2011) | 39 lines
  
  Merged revisions 313189 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ................
    r313189 | rmudgett | 2011-04-11 10:32:53 -0500 (Mon, 11 Apr 2011) | 32 lines
    
    Merged revisions 313188 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
    ........
      r313188 | rmudgett | 2011-04-11 10:27:52 -0500 (Mon, 11 Apr 2011) | 25 lines
      
      Stuck channel using FEATD_MF if caller hangs up at the right time.
      
      The cause was actually a caller hanging up just at the end of the Feature
      Group D DTMF tones that setup the call.  The reason for this is a "guard
      timer" that's implemented using ast_safe_sleep(100).  If the caller
      happens to hang up AFTER the final tone of the DTMF string but BEFORE the
      end of that ast_safe_sleep(), then ast_safe_sleep() will return non-zero.
      This causes the code to bounce to the end of ss_thread(), but it does NOT
      tear down the call properly.
      
      This should be a rare occurrence because the caller has to hang up at
      EXACTLY the right time.  Nonetheless, it was happening quite regularly on
      the reporter's system.  It's not easily reproducible, unless you purposely
      increase the guard-time to 2000 or more.  Once you do that, you can
      reproduce it every time by watching the DTMF debug and hanging up just as
      it ends.
      
      Simply add an ast_hangup() before goto quit.
      
      (closes issue #15671)
      Reported by: jcromes
      Patches:
            issue15671.patch uploaded by pabelanger (license 224)
      Tested by: jcromes
    ........
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  r313142 | may | 2011-04-10 00:56:17 +0400 (Sun, 10 Apr 2011) | 3 lines
  
  fix trivial bug in ooh323_indicate on AST_CONTROL_SRC...
  check p->rtp is not null
........


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Factor out the equivalent function for analog.


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  r313048 | jrose | 2011-04-07 08:35:33 -0500 (Thu, 07 Apr 2011) | 16 lines
  
  Merged revisions 313047 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ........
    r313047 | jrose | 2011-04-07 08:23:01 -0500 (Thu, 07 Apr 2011) | 9 lines
    
    Makes parking lots clear and rebuild properly when features reload is invoked from CLI
    
    Before, default parkinglot in context parkedcalls with ext 700 would always be present and when reload was invoked, the previous parkinglots would not be cleared.
    
    (closes issue #18801)
    Reported by: mickecarlsson
    
    Review: https://reviewboard.asterisk.org/r/1161/
  ........
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  r313001 | alecdavis | 2011-04-07 22:19:31 +1200 (Thu, 07 Apr 2011) | 13 lines
  
  Fix ISDN calling subaddr User Specified Odd/Even Flag
  
  Calculation of the Odd/Even flag was wrong.
  Implement correct algo, and set odd/even=0 if data would be truncated.
  Only allow automatic calculation of the O/E flag, don't let dialplan influence.
  
  (closes issue #19062)
  Reported by: festr
  Patches: 
        bug19062.diff2.txt uploaded by alecdavis (license 585)
  Tested by: festr, alecdavis, rmudgett
........


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  r312949 | rmudgett | 2011-04-05 13:45:24 -0500 (Tue, 05 Apr 2011) | 6 lines
  
  Crash if ISDN span layer 1 is down on initial load.
  
  Regression from -r312575 B channel shifting during negotiation.
  
  * Also combine updating the alarm flag with clearing the resetting flag.
........


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  r312889 | rmudgett | 2011-04-05 11:19:35 -0500 (Tue, 05 Apr 2011) | 5 lines
  
  Add 416 response to OPTIONS packet.
  
  RFC3261 Section 11.2 says the response code to an OPTIONS packet needs to
  be the same as if it were an INVITE.
........


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  r312866 | rmudgett | 2011-04-05 10:38:14 -0500 (Tue, 05 Apr 2011) | 15 lines
  
  Responding to OPTIONS packet with 404 because Asterisk not looking for "s" extension.
  
  The get_destination() function was not using the "s" extension when the
  request URI did not specify an extension.  This is a regression caused
  when the URI parsing code was extracted into parse_uri().
  
  Made get_destination() substitute the "s" extension when the parsed URI
  results in an empty string.
  
  (closes issue #18348)
  Reported by: shmaize
  Patches:
        issue18348_v1.8.patch uploaded by rmudgett (license 664)
  Tested by: shmaize
........


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  r312766 | mnicholson | 2011-04-05 09:14:50 -0500 (Tue, 05 Apr 2011) | 22 lines
  
  Merged revisions 312764 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ................
    r312764 | mnicholson | 2011-04-05 09:13:07 -0500 (Tue, 05 Apr 2011) | 15 lines
    
    Merged revisions 312761 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
    ........
      r312761 | mnicholson | 2011-04-05 09:10:34 -0500 (Tue, 05 Apr 2011) | 8 lines
      
      Limit the number of unauthenticated manager sessions and also limit the time they have to authenticate.
      
      AST-2011-005
      
      (closes issue #18996)
      Reported by: tzafrir
      Tested by: mnicholson
    ........
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It was only used in a debug message and may not be correct anyway.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@312716 65c4cc65-6c06-0410-ace0-fbb531ad65f3

In handle_cli_dialplan_add_extension, const char pointer *into_context is used instead of a->argv[5] to improve readability.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@312680 65c4cc65-6c06-0410-ace0-fbb531ad65f3

If the user invokes 'dialplan add extension' into a non-existing context, the context will be created
and a message informing the user of the context being created will be issued in cli.

(closes issue #17431)
Reported by: leearcher
Patches:
      context_auto_create.diff uploaded by kobaz (license 834)
Tested by: leearcher, kobaz, jrose


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  r312575 | rmudgett | 2011-04-04 11:10:50 -0500 (Mon, 04 Apr 2011) | 52 lines
  
  Merged revisions 312574 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ................
    r312574 | rmudgett | 2011-04-04 11:00:02 -0500 (Mon, 04 Apr 2011) | 45 lines
    
    Merged revisions 312573 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
    ........
      r312573 | rmudgett | 2011-04-04 10:49:30 -0500 (Mon, 04 Apr 2011) | 38 lines
      
      Issues with ISDN calls changing B channels during call negotiations.
      
      The handling of the PROCEEDING message was not using the correct call
      structure if the B channel was changed.  (The same for PROGRESS.) The call
      was also not hungup if the new B channel is not provisioned or is busy.
      
      * Made all call connection messages (SETUP_ACKNOWLEDGE, PROCEEDING,
      PROGRESS, ALERTING, CONNECT, CONNECT_ACKNOWLEDGE) ensure that they are
      using the correct structure and B channel.  If there is any problem with
      the operations then the call is now hungup with an appropriate cause code.
      
      * Made miscellaneous messages (INFORMATION, FACILITY, NOTIFY) find the
      correct structure by looking for the call and not using the channel ID.
      NOTIFY is an exception with versions of libpri before v1.4.11 because a
      call pointer is not available for Asterisk to use.
      
      * Made all hangup messages (DISCONNECT, RELEASE, RELEASE_COMPLETE) find
      the correct structure by looking for the call and not using the channel
      ID.
      
      (closes issue #18313)
      Reported by: destiny6628
      Tested by: rmudgett
      JIRA SWP-2620
      
      (closes issue #18231)
      Reported by: destiny6628
      Tested by: rmudgett
      JIRA SWP-2924
      
      (closes issue #18488)
      Reported by: jpokorny
      JIRA SWP-2929
      
      JIRA AST-437 (The issues fixed here are most likely causing this JIRA issue.)
      JIRA DAHDI-406
      JIRA LIBPRI-33 (Stuck resetting flag likely fixed)
    ........
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  r312509 | rmudgett | 2011-04-01 18:15:42 -0500 (Fri, 01 Apr 2011) | 22 lines
  
  When a call going out an NT-PTMP port gets rejected, Asterisk crashes.
  
  If a call is sent to an ISDN phone that rejects the call with
  RELEASE_COMPLETE(cause: call reject(21), or busy(17)) Asterisk crashes.
  
  I could not get my setup to crash.  However, I could see the possibility
  from a race condition between queuing an AST_CONTROL_BUSY to the core and
  then queueing an AST_CONTROL_HANGUP.  If the AST_CONTROL_BUSY is processed
  before the AST_CONTROL_HANGUP is queued, the ast_channel could be
  destroyed out from under chan_misdn.
  
  Avoid this particular crash scenario by not queueing the
  AST_CONTROL_HANGUP if the AST_CONTROL_BUSY was queued.
  
  (closes issue #18408)
  Reported by: wimpy
  Patches:
        issue18408_v1.8.patch uploaded by rmudgett (license 664)
  Tested by: rmudgett, wimpy
  
  JIRA SWP-2679
........


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  r312461 | rmudgett | 2011-04-01 16:31:39 -0500 (Fri, 01 Apr 2011) | 25 lines
  
  CallCompletionRequest()/CallCompletionCancel() exit non-zero if fail.
  
  The CallCompletionRequest()/CallCompletionCancel() dialplan applications
  exit nonzero on normal failure conditions.  The nonzero exit causes the
  dialplan to hangup immediately.  The dialplan author has no opportunity to
  report success/failure to the user.
  
  * Made always return zero so the dialplan can continue.
  
  * Made set CC_REQUEST_RESULT/CC_REQUEST_REASON and
  CC_CANCEL_RESULT/CC_CANCEL_REASON channel variables respectively.  Also
  documented the values set.
  
  * Reduced the warning about no core instance in CallCompletionCancel() to
  a debug message.  It is a normal event and should not be output at the
  WARNING level.
  
  (closes issue #18763)
  Reported by: p_lindheimer
  Patches:
        ccss.patch uploaded by p lindheimer (license 558) Modified
  Tested by: p_lindheimer, rmudgett
  
  JIRA SWP-3042
........


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In chan_dahdi.conf, the user can now use length 4 patterns in addition to the usual length 2 patterns.  The s
ntax remains the same and the method used to track the pattern history will only change when using the length
 4 patterns.

(closes issue SWP-3250)
Code:
        jrose
        rmudgett


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................
  r312286 | tilghman | 2011-04-01 05:44:33 -0500 (Fri, 01 Apr 2011) | 2 lines
  
  Reload must react correctly against a possibly changed table, so dropping the conditional reload flag.
................
  r312288 | tilghman | 2011-04-01 05:58:45 -0500 (Fri, 01 Apr 2011) | 21 lines
  
  Merged revisions 312287 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ................
    r312287 | tilghman | 2011-04-01 05:51:24 -0500 (Fri, 01 Apr 2011) | 14 lines
    
    Merged revisions 312285 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
    ........
      r312285 | tilghman | 2011-04-01 05:36:42 -0500 (Fri, 01 Apr 2011) | 7 lines
      
      Found some leaking file descriptors while looking at ast_FD_SETSIZE dead code.
      
      (issue #18969)
       Reported by: oej
       Patches: 
             20110315__issue18969__14.diff.txt uploaded by tilghman (license 14)
    ........
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  r312211 | alecdavis | 2011-04-01 22:03:11 +1300 (Fri, 01 Apr 2011) | 36 lines
  
  Merged revisions 312210 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ................
    r312210 | alecdavis | 2011-04-01 21:47:29 +1300 (Fri, 01 Apr 2011) | 29 lines
    
    Merged revisions 312174 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
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      r312174 | alecdavis | 2011-04-01 21:29:49 +1300 (Fri, 01 Apr 2011) | 23 lines
      
      voicemail: get real last_message_index and count_messages, ODBC resequence
      
      change last_message_index to read the max msgnum stored in the database
      change count_messages to actually count the number of messages.
      
      last_message_index change:
        This fixed overwriting of the last message if msgnum=0 was missing.
        Previously every incoming message would overwrite msgnum=1.
      count_messages change:
        allows us to detect when requencing is required in opneA_mailbox.
      resequence enabled for ODBC storage:
        Assists with fixing up corrupt databases with gaps, but only when
        a user actively opens there mailboxes.
      
      (closes issue #18692,#18582,#19032)
      Reported by: elguero
      Patches: 
            based on odbc_resequence_mailbox2.1.diff uploaded by elguero (license 37)
      Tested by: elguero, nivek, alecdavis
      
      Review: https://reviewboard.asterisk.org/r/1153/
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  r312117 | alecdavis | 2011-04-01 20:32:12 +1300 (Fri, 01 Apr 2011) | 29 lines
  
  Merged revisions 312103 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
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    r312103 | alecdavis | 2011-04-01 20:25:54 +1300 (Fri, 01 Apr 2011) | 22 lines
    
    Merged revisions 312070 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
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      r312070 | alecdavis | 2011-04-01 19:46:56 +1300 (Fri, 01 Apr 2011) | 16 lines
      
      app_voicemail: close_mailbox needs to respect additional messages while mailbox is open.
      
      close_mailbox leave gaps in message sequence if messages are deleted and new messages
      arrive during this time, this is because the shuffle down to slot 0, only shuffles
      the number of pre-existing messages when mailbox is opened, ignoring new arrivals.
      
      Fix: in close_mailbox re-evaluate number of messages before the shuffle, this then includes new arrivals.
      
      Happens on filebased or ODBC storage.
      
      (issues #19032,#18582,#18692,#18998)
      Reported by: alecdavis,tootai,afosorio
      
      Review: https://reviewboard.asterisk.org/r/1153/
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  r312022 | rmudgett | 2011-03-31 15:11:40 -0500 (Thu, 31 Mar 2011) | 14 lines
  
  chan_misdn segfaults when DEBUG_THREADS is enabled.
  
  The segfault happens because jb->mutexjb is uninitialized from the
  ast_malloc().  The internals of ast_mutex_init() were assuming a nonzero
  value meant mutex tracking initialization had already happened.  Recent
  changes to mutex tracking code to reduce excessive memory consumption
  exposed this uninitialized value.
  
  Converted misdn_jb_init() to use ast_calloc() instead of ast_malloc().
  Also eliminated redundant zero initialization code in the routine.
  
  (closes issue #18975)
  Reported by: irroot
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  r311930 | tilghman | 2011-03-31 01:43:18 -0500 (Thu, 31 Mar 2011) | 6 lines
  
  Incorrect default example; the field is actually internally named "clid", not "callerid".
  
  (closes issue #19040)
  Reported by: wcselby
  Tested by: tilghman
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  r311874 | rmudgett | 2011-03-29 20:56:05 -0500 (Tue, 29 Mar 2011) | 1 line
  
  Update some setup_dahdi_int() comments.
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  r311799 | tilghman | 2011-03-29 02:08:39 -0500 (Tue, 29 Mar 2011) | 7 lines
  
  Remove extraneous check from integer-type fields.
  
  (closes issue #19027)
   Reported by: mlehner
   
  Review: https://reviewboard.asterisk.org/r/1149/
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  r311751 | russell | 2011-03-28 17:00:01 -0500 (Mon, 28 Mar 2011) | 2 lines
  
  Cross-reference VoiceMail() and VoiceMailMain() in the xml docs.
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  r311687 | may | 2011-03-28 01:47:13 +0400 (Mon, 28 Mar 2011) | 2 lines
  
  correct return values in ooh323_indicate for AST_CONTROL_T38_PARAMETERS
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  r311615 | bbryant | 2011-03-23 17:54:11 -0400 (Wed, 23 Mar 2011) | 8 lines
  
  This patch fixes a bug with MeetMe behavior where the 'P' option for always
  prompting for a pin is ignored for the first caller.
  
  (closes issue #18070)
  Reported by: mav3rick
  
  Review: https://reviewboard.asterisk.org/r/1132/
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