[svn-commits] rmudgett: branch group/issue14292 r200469 - in /team/group/issue14292: ./ app...

SVN commits to the Digium repositories svn-commits at lists.digium.com
Fri Jun 12 16:56:29 CDT 2009


Author: rmudgett
Date: Fri Jun 12 16:56:19 2009
New Revision: 200469

URL: http://svn.asterisk.org/svn-view/asterisk?view=rev&rev=200469
Log:
Merged revisions 200468 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/team/group/issue14068

................
  r200468 | rmudgett | 2009-06-12 16:47:08 -0500 (Fri, 12 Jun 2009) | 812 lines
  
  Bring branch up-to-date after freezing it because of the new signalling module.
  
  Merged revisions 198088,198139,198146,198182-198183,198186,198217,198248,198285,198312,198371,198375,198434,198437-198438,198442,198470,198498,198500,198511,198529-198530,198558,198561,198565,198597,198626,198661,198666,198670,198725,198727,198729,198762,198791,198824,198856,198892,198954,198958,199051,199091,199139,199227,199298,199368,199370,199372,199374,199376,199409,199411,199413,199446,199479,199514,199547,199588,199630,199696,199743,199781,199818,199857,199923,199957-199958,200000,200038-200039,200108,200146,200190,200254,200290,200326,200361,200428,200430 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/trunk
  
  ................
    r198088 | jpeeler | 2009-05-29 14:19:51 -0500 (Fri, 29 May 2009) | 9 lines
    
    New signaling module to handle analog operations in chan_dahdi
    
    This branch splits all the analog signaling logic out of chan_dahdi.c into
    sig_analog.c. Functionality in theory should not change at all. As noted
    in the code, there is still some unused code remaining that will be cleaned
    up in a later commit.
    
    Review: https://reviewboard.asterisk.org/r/253/
  ................
    r198139 | eliel | 2009-05-29 14:46:07 -0500 (Fri, 29 May 2009) | 15 lines
    
    Simplify the Makefile and avoid needing to specify each object file.
    
    Instead of specifying every object file, use make's magic to generate
    it.
    This will generate less conflicts in team branches when a new file is
    added in trunk.
    
    (closes issue #15226)
    Reported by: eliel
    Patches:
          makefile uploaded by eliel (license 64)
    
          Review: http://reviewboard.asterisk.org/r/269/ 
  ................
    r198146 | russell | 2009-05-29 15:06:59 -0500 (Fri, 29 May 2009) | 38 lines
    
    Resolve issues with choppy sound when using res_timing_pthread.
    
    The situation that caused this problem was when continuous mode was being
    turned on and off while a rate was set for a timing interface.  A very easy
    way to replicate this bug was to do a Playback() from behind a Local channel.
    In this scenario, a rate gets set on the channel for doing file playback.
    At the same time, continuous mode gets turned on and off about every 20 ms
    as frames get queued on to the PBX side channel from the other side of the
    Local channel.
    
    Essentially, this module treated continuous mode and a set rate as mutually
    exclusive states for the timer to be in.  When I dug deep enough, I observed
    the following pattern:
    
       1) Set timer to tick every 20 ms.
       2) Wait almost 20 ms ...
       3) Continuous mode gets turned on for a queued up frame
       4) Continuous mode gets turned off
       5) The timer goes back to its tick per 20 ms. state but starts counting
          at 0 ms.
       6) Goto step 2.
    
    Sometimes, res_timing_pthread would make it 20 ms and produce a timer tick,
    but not most of the time.  This is what produced the choppy sound (or sometimes
    no sound at all).
    
    Now, the module treats continuous mode and a set rate as completely independent
    timer modes.  They can be enabled and disabled independently of each other and
    things work as expected.
    
    
    (closes issue #14412)
    Reported by: dome
    Patches:
          issue14412.diff.txt uploaded by russell (license 2)
          issue14412-1.6.1.0.diff.txt uploaded by russell (license 2)
    Tested by: DennisD, russell
  ................
    r198182 | twilson | 2009-05-29 17:21:42 -0500 (Fri, 29 May 2009) | 2 lines
    
    Add a couple of TODO items so I don't forget
  ................
    r198183 | russell | 2009-05-29 17:33:31 -0500 (Fri, 29 May 2009) | 2 lines
    
    Improve handling of trying to ACK too many timer expirations.
  ................
    r198186 | russell | 2009-05-29 18:04:31 -0500 (Fri, 29 May 2009) | 2 lines
    
    Suggesting that only a single timing module be loaded is no longer necessary.
  ................
    r198217 | eliel | 2009-05-29 20:04:57 -0500 (Fri, 29 May 2009) | 10 lines
    
    Remove not used code in the Agent channel.
    
    This code was there because of the AgentCallbackLogin() application.
    ->loginchan[] member was only used by AgentCallbackLogin().
    Agent where dumped to astdb if they where logged in using AgentCallbacklogin()
    so they are not being dumper anymore.
    
    Review: https://reviewboard.asterisk.org/r/267/
  ................
    r198248 | file | 2009-05-29 21:31:48 -0500 (Fri, 29 May 2009) | 2 lines
    
    When removing all packets from a dialog we also need to free the data if present.
  ................
    r198285 | seanbright | 2009-05-29 22:26:06 -0500 (Fri, 29 May 2009) | 15 lines
    
    Merged revisions 198251 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
    ........
      r198251 | seanbright | 2009-05-29 22:46:41 -0400 (Fri, 29 May 2009) | 8 lines
      
      Treat an empty FORWARD_CONTEXT the same way we treat a missing one.
      
      (closes issue #15056)
      Reported by: p_lindheimer
      Patches:
            05292009_bug15056.diff uploaded by seanbright (license 71)
      Tested by: p_lindheimer
    ........
  ................
    r198312 | russell | 2009-05-29 22:43:23 -0500 (Fri, 29 May 2009) | 12 lines
    
    Merged revisions 198311 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
    ........
      r198311 | russell | 2009-05-29 22:42:46 -0500 (Fri, 29 May 2009) | 5 lines
      
      Fix a crash that occurred when MWI SMDI messages expired.
      
      (closes issue #14561)
      Reported by: cmoss28
    ........
  ................
    r198371 | seanbright | 2009-05-30 14:38:58 -0500 (Sat, 30 May 2009) | 19 lines
    
    Merged revisions 198370 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
    ........
      r198370 | seanbright | 2009-05-30 15:36:20 -0400 (Sat, 30 May 2009) | 12 lines
      
      Properly terminate AMI JabberSend response messages.
      
      The response message (either Error or Success) needs an extra trailing \r\n
      after the fields to inform the client that the message is complete.
      
      (closes issue #14876)
      Reported by: srt
      Patches:
            05302009_1.4_res_jabber.c.diff uploaded by seanbright (license 71)
            asterisk_14876.patch uploaded by srt (license 378)
            trunk-14876-2.diff uploaded by phsultan (license 73)
    ........
  ................
    r198375 | seanbright | 2009-05-30 15:11:33 -0500 (Sat, 30 May 2009) | 13 lines
    
    Properly terminate the receive buffer before sending to iksemel.
    
    aji_io_recv takes the maximum number of bytes to read (instead of the total
    buffer size), so we have to subtract 1 from our buffer size.  Without this, when
    we receive packets that are larger than our buffer, iksemel will choke and
    things get wonky.
    
    (closes issue #15232)
    Reported by: lp0
    Patches:
          05302009_res_jabber.c.patch uploaded by seanbright (license 71)
    Tested by: seanbright, lp0
  ................
    r198434 | russell | 2009-05-30 20:19:30 -0500 (Sat, 30 May 2009) | 2 lines
    
    Constify the ast_frame arg to ast_queue_frame().
  ................
    r198437 | eliel | 2009-05-30 20:22:15 -0500 (Sat, 30 May 2009) | 11 lines
    
    Avoid a crash when res_timing_dahdi is unloaded but wasn't properly loaded.
    
    if dahdi_test_timer() fails, timing_funcs_handle remains NULL causing a crash
    when calling ast_unregister_timing_interface() with a NULL pointer.
    
    (closes issue #15234)
    Reported by: eliel
    Patches:
          timing_dahdi1.diff uploaded by eliel (license 64)
  ................
    r198438 | russell | 2009-05-30 20:40:02 -0500 (Sat, 30 May 2009) | 2 lines
    
    Constification and remove some unused code.
  ................
    r198442 | eliel | 2009-05-30 21:09:06 -0500 (Sat, 30 May 2009) | 3 lines
    
    Filter the say.o object, it is being added later.
  ................
    r198470 | tilghman | 2009-05-31 12:52:28 -0500 (Sun, 31 May 2009) | 2 lines
    
    Fix documentation for FIELDQTY.
  ................
    r198498 | file | 2009-06-01 08:31:27 -0500 (Mon, 01 Jun 2009) | 5 lines
    
    Fix a bug where the Event and Content-Type headers were added twice to outgoing SIP NOTIFY messages.
    
    (closes issue #15239)
    Reported by: pj
  ................
    r198500 | mmichelson | 2009-06-01 09:02:05 -0500 (Mon, 01 Jun 2009) | 6 lines
    
    Remove documentation for the 'exten' argument to the AGENT function.
    
    Since AgentCallbackLogin has been removed, this should not be documented
    any more.
  ................
    r198511 | mmichelson | 2009-06-01 09:19:49 -0500 (Mon, 01 Jun 2009) | 3 lines
    
    Add missing unlock of local pvt.
  ................
    r198529 | mmichelson | 2009-06-01 09:42:57 -0500 (Mon, 01 Jun 2009) | 5 lines
    
    Remove extra lock from local_indicate in connected line case.
    
    Oh, and this fixes a deadlock I was seeing.
  ................
    r198530 | mmichelson | 2009-06-01 09:45:43 -0500 (Mon, 01 Jun 2009) | 3 lines
    
    Remove extra lock from app_queue.
  ................
    r198558 | dvossel | 2009-06-01 10:23:21 -0500 (Mon, 01 Jun 2009) | 3 lines
    
    Fixed an issue in the threadstorage cli functions resulting from the constification of struct ast_cli_args in r196072.
  ................
    r198561 | eliel | 2009-06-01 10:38:48 -0500 (Mon, 01 Jun 2009) | 2 lines
    
    Move static documentation of E|Dead|AGI() application and manager action to XML.
  ................
    r198565 | eliel | 2009-06-01 11:09:42 -0500 (Mon, 01 Jun 2009) | 2 lines
    
    Move JabberSend manager action from static docs to the AstXML form.
  ................
    r198597 | eliel | 2009-06-01 12:53:38 -0500 (Mon, 01 Jun 2009) | 3 lines
    
    Do not add say.o in a separate line.
  ................
    r198626 | tilghman | 2009-06-01 13:40:35 -0500 (Mon, 01 Jun 2009) | 2 lines
    
    Add information for new meetme realtime fields
  ................
    r198661 | eliel | 2009-06-01 14:37:30 -0500 (Mon, 01 Jun 2009) | 6 lines
    
    Moved more static documentation to the new AstXML form.
    
    Moved more static docs to XML (pplications and manager actions):
    Monitor, StopMonitor, ChangeMonitor, PauseMonitor, UnpauseMonitor.
  ................
    r198666 | tilghman | 2009-06-01 15:09:56 -0500 (Mon, 01 Jun 2009) | 13 lines
    
    Blocked revisions 198665 via svnmerge
    
    ........
      r198665 | tilghman | 2009-06-01 15:07:04 -0500 (Mon, 01 Jun 2009) | 7 lines
      
      If using the old deprecated format, a reload would cause the class to disappear.
      (closes issue #14759)
       Reported by: lidocaineus
       Patches: 
             20090518__issue14759.diff.txt uploaded by tilghman (license 14)
       Tested by: lmadsen
    ........
  ................
    r198670 | russell | 2009-06-01 15:17:50 -0500 (Mon, 01 Jun 2009) | 2 lines
    
    Minor whitespace fix.
  ................
    r198725 | tilghman | 2009-06-01 15:33:50 -0500 (Mon, 01 Jun 2009) | 8 lines
    
    Add INCrement and DECrement functions
    (closes issue #15025)
     Reported by: greenfieldtech
     Patches: 
           func_math.c.patch_v4 uploaded by greenfieldtech (license 369)
           slightly modified by me
     Tested by: greenfieldtech, lmadsen
  ................
    r198727 | mmichelson | 2009-06-01 15:57:31 -0500 (Mon, 01 Jun 2009) | 13 lines
    
    Add the ability to execute connected line interception macros.
    
    When connected line updates are received or generated in the middle
    of an application call, it is now possible to execute a macro to
    manipulate the connected line data. This way, phone numbers may be
    manipulated to be more presentable to users, names may be changed 
    for...whatever reason, or whatever else needs to be done may be.
    
    Review: https://reviewboard.asterisk.org/r/256
    
    AST-165
  ................
    r198729 | russell | 2009-06-01 16:03:18 -0500 (Mon, 01 Jun 2009) | 2 lines
    
    Tell the IAX2 parser about more control frame types.
  ................
    r198762 | file | 2009-06-02 08:12:59 -0500 (Tue, 02 Jun 2009) | 5 lines
    
    Fix a bug where we were passing in address information that should remain unmodified to a function that may modify it.
    
    (closes issue #15243)
    Reported by: pj
  ................
    r198791 | file | 2009-06-02 08:48:06 -0500 (Tue, 02 Jun 2009) | 5 lines
    
    Correct documentation for the register line, specifically where the domain should be specified.
    
    (closes issue #14367)
    Reported by: Nick_Lewis
  ................
    r198824 | dvossel | 2009-06-02 12:55:35 -0500 (Tue, 02 Jun 2009) | 8 lines
    
    fixes issue with channels not going down after transfer
    
    Iax2 currently does not support native bridging if the timeoutms value is set.  We check for that in iax2_bridge, but then set timeoutms to 0 by default.  If the timeoutms is not provided it is set to -1. By setting timeoutms to 0 it is processed causing a bridging retry loop.
    
    (closes issue #15216)
    Reported by: oxymoron
    Tested by: dvossel
  ................
    r198856 | dvossel | 2009-06-02 16:17:49 -0500 (Tue, 02 Jun 2009) | 10 lines
    
    Generic call forward api, ast_call_forward()
    
    The function ast_call_forward() forwards a call to an extension specified in an ast_channel's call_forward string.  After an ast_channel is called, if the channel's call_forward string is set this function can be used to forward the call to a new channel and terminate the original one.  I have included this api call in both channel.c's ast_request_and_dial() and feature.c's feature_request_and_dial().  App_dial and app_queue already contain call forward logic specific for their application and options.
    
    (closes issue #13630)
    Reported by: festr
    
    Review: https://reviewboard.asterisk.org/r/271/
  ................
    r198892 | dvossel | 2009-06-03 10:51:10 -0500 (Wed, 03 Jun 2009) | 15 lines
    
    Blocked revisions 198891 via svnmerge
    
    ........
      r198891 | dvossel | 2009-06-03 10:49:46 -0500 (Wed, 03 Jun 2009) | 10 lines
      
      Generic call forward api, ast_call_forward()
      
      The function ast_call_forward() forwards a call to an extension specified in an ast_channel's call_forward string.  After an ast_channel is called, if the channel's call_forward string is set this function can be used to forward the call to a new channel and terminate the original one.  I have included this api call in both channel.c's ast_request_and_dial() and res_feature.c's feature_request_and_dial().  App_dial and app_queue already contain call forward logic specific for their application and options.
      
      (closes issue #13630)
      Reported by: festr
      
      Review: https://reviewboard.asterisk.org/r/271/
    ........
  ................
    r198954 | dvossel | 2009-06-03 15:30:10 -0500 (Wed, 03 Jun 2009) | 3 lines
    
    ast_call_forward() todo notes and originate flag copy.
  ................
    r198958 | seanbright | 2009-06-03 15:49:11 -0500 (Wed, 03 Jun 2009) | 17 lines
    
    Blocked revisions 198957 via svnmerge
    
    ........
      r198957 | seanbright | 2009-06-03 16:39:10 -0400 (Wed, 03 Jun 2009) | 11 lines
      
      Fix a possible crash in pbx_spool.
      
      We were trying to reference members of a struct that had previously been freed.
      This patch makes sure that we free the struct after it has been removed from
      the spooler queue.
      
      (closes issue #15072)
      Reported by: garlew
      Patches:
            spool.diff uploaded by garlew (license 376)
    ........
  ................
    r199051 | seanbright | 2009-06-04 09:31:24 -0500 (Thu, 04 Jun 2009) | 47 lines
    
    Merged revisions 199022 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
    ........
      r199022 | seanbright | 2009-06-04 10:14:57 -0400 (Thu, 04 Jun 2009) | 40 lines
      
      Safely handle AMI connections/reload requests that occur during startup.
      
      During asterisk startup, a lock on the list of modules is obtained by the
      primary thread while each module is initialized.  Issue 13778 pointed out a
      problem with this approach, however.  Because the AMI is loaded before other
      modules, it is possible for a module reload to be issued by a connected client
      (via Action: Command), causing a deadlock.
      
      The resolution for 13778 was to move initialization of the manager to happen
      after the other modules had already been lodaded.  While this fixed this
      particular issue, it caused a problem for users (like FreePBX) who call AMI
      scripts via an #exec in a configuration file (See issue 15189).
      
      The solution I have come up with is to defer any reload requests that come in
      until after the server is fully booted.  When a call comes in to
      ast_module_reload (from wherever) before we are fully booted, the request is
      added to a queue of pending requests.  Once we are done booting up, we then
      execute these deferred requests in turn.
      
      Note that I have tried to make this a bit more intelligent in that it will not
      queue up more than 1 request for the same module to be reloaded, and if a
      general reload request comes in ('module reload') the queue is flushed and we
      only issue a single deferred reload for the entire system.
      
      As for how this will impact existing installations - Before 13778, a reload
      issued before module initialization was completed would result in a deadlock.
      After 13778, you simply couldn't connect to the manager during startup (which
      causes problems with #exec-that-calls-AMI configuration files).  I believe this
      is a good general purpose solution that won't negatively impact existing
      installations.
      
      (closes issue #15189)
      (closes issue #13778)
      Reported by: p_lindheimer
      Patches:
            06032009_15189_deferred_reloads.diff uploaded by seanbright (license 71)
      Tested by: p_lindheimer, seanbright
      
      Review: https://reviewboard.asterisk.org/r/272/
    ........
  ................
    r199091 | eliel | 2009-06-04 11:29:50 -0500 (Thu, 04 Jun 2009) | 11 lines
    
    Move static docs to the new AstXML form.
    
    Move SMDI_MSG and SMDI_MSG_RETRIEVE functions statis documentation
    to XML.
    
    (issue #15245)
    Reported by: eliel
    Patches:
          res_smdi_static_conversion.txt uploaded by lmadsen (license 10)
  ................
    r199139 | dvossel | 2009-06-04 14:10:16 -0500 (Thu, 04 Jun 2009) | 9 lines
    
    Merged revisions 199138 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
    ........
      r199138 | dvossel | 2009-06-04 14:00:15 -0500 (Thu, 04 Jun 2009) | 3 lines
      
      Additional updates to AST-2009-001
    ........
  ................
    r199227 | mmichelson | 2009-06-05 08:51:08 -0500 (Fri, 05 Jun 2009) | 14 lines
    
    Correct "dahdi show channels" output when specifying a group.
    
    Since a DAHDI channel may belong to multiple groups, we need to use
    a bitwise and instead of equivalence to determine whether to display
    the channel information.
    
    
    (closes issue #15248)
    Reported by: gentian
    Patches:
          15248.patch uploaded by mmichelson (license 60)
    Tested by: gentian
  ................
    r199298 | dvossel | 2009-06-05 16:21:22 -0500 (Fri, 05 Jun 2009) | 21 lines
    
    Merged revisions 199297 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
    ........
      r199297 | dvossel | 2009-06-05 16:19:56 -0500 (Fri, 05 Jun 2009) | 14 lines
      
      Fixes issue with hints giving unexpected results.
      
      Hints with two or more devices that include ONHOLD gave unexpected results.
      
      (closes issue #15057)
      Reported by: p_lindheimer
      Patches:
            onhold_trunk.diff uploaded by dvossel (license 671)
            pbx.c.1.4.patch uploaded by p (license 558)
            devicestate.c.trunk.patch uploaded by p (license 671)
      Tested by: p_lindheimer, dvossel
      
      Review: https://reviewboard.asterisk.org/r/254/
    ........
  ................
    r199368 | russell | 2009-06-06 16:38:54 -0500 (Sat, 06 Jun 2009) | 2 lines
    
    Switch from "echo -n" to printf.  On my mac, the -n was just getting printed out.
  ................
    r199370 | russell | 2009-06-06 16:40:56 -0500 (Sat, 06 Jun 2009) | 2 lines
    
    Constify a string and strip trailing whitespace.
  ................
    r199372 | russell | 2009-06-06 16:42:31 -0500 (Sat, 06 Jun 2009) | 1 line
    
    minor tweak
  ................
    r199374 | eliel | 2009-06-06 16:56:58 -0500 (Sat, 06 Jun 2009) | 10 lines
    
    Move function SYSINFO documentation to XML.
    
    Move function SYSINFO static documentation to the new AstXML form.
    
    (issue #15245)
    Reported by: eliel
    Patches:
          func_sysinfo_static_conversion.txt uploaded by lmadsen (license 10)
  ................
    r199376 | eliel | 2009-06-06 17:16:47 -0500 (Sat, 06 Jun 2009) | 12 lines
    
    Move function MINIVMACCOUNT and MINIVMCOUNTER static documentation to XML.
    
    Move function MINIVMACCOUNT and MINIVMCOUNTER statis documentation to the new
    AstXML form.
    
    (issue #15245)
    Reported by: eliel
    Patches:
          app_minivm_static_conversion.txt uploaded by lmadsen (license 10)
          (with minor changes by me)
  ................
    r199409 | eliel | 2009-06-06 17:27:48 -0500 (Sat, 06 Jun 2009) | 10 lines
    
    Move function MEETME_INFO documentation to XML.
    
    Move function MEETME_INFO static documentation to the new AstXML form.
    
    (issue #15245)
    Reported by: eliel
    Patches:
          app_meetme_static_conversion.txt uploaded by lmadsen (license 10)
  ................
    r199411 | eliel | 2009-06-06 17:45:42 -0500 (Sat, 06 Jun 2009) | 12 lines
    
    Move function PP_EACH_USER and PP_EACH_EXTENSION documentation to XML.
    
    Move function PP_EACH_USER and PP_EACH_EXTENSION documentation to the new
    AstXML form.
    
    (issue #15245)
    Reported by: eliel
    Patches:
          res_phoneprov_static_conversion.txt uploaded by lmadsen (license 10)
    	(with PP_EACH_USER add by me)
  ................
    r199413 | eliel | 2009-06-06 18:03:15 -0500 (Sat, 06 Jun 2009) | 11 lines
    
    Move music on hold related applications documentation to XML.
    
    Move MusicOnHold, SetMusicOnHold, StartMusicOnHold, StopMusicOnHold static
    documentation to the new AstXML form.
    
    (issue #15245)
    Reported by: eliel
    Patches:
          res_musiconhold_static_conversion.txt uploaded by lmadsen (license 10)
          (with some fixes and formatting by me)
  ................
    r199446 | eliel | 2009-06-06 18:28:38 -0500 (Sat, 06 Jun 2009) | 11 lines
    
    Move AGI command 'gosub' static documentation to XML.
    
    Move AGI command 'gosub' statis documentation to the new AstXML form.
    
    (issue #15245)
    Reported by: eliel
    Patches:
          app_stack_static_conversion.txt uploaded by lmadsen (license 10)
          (with minor changes by me)
  ................
    r199479 | russell | 2009-06-07 09:55:51 -0500 (Sun, 07 Jun 2009) | 2 lines
    
    Global var cleanup - constification and removing unused vars.
  ................
    r199514 | eliel | 2009-06-07 12:29:44 -0500 (Sun, 07 Jun 2009) | 10 lines
    
    Move application ExternalIVR static documentation to XML.
    
    Move application ExternalIVR static documentation to the new AstXML form.
    
    (issue #15245)
    Reported by: eliel
    Patches:
          app_externalivr.diff uploaded by eliel (license 64)
  ................
    r199547 | eliel | 2009-06-07 14:15:41 -0500 (Sun, 07 Jun 2009) | 10 lines
    
    Move OSP* applications static documentation to XML.
    
    Move OSP* applications static documentation to the new AstXML form.
    
    (closes issue #15245)
    Reported by: eliel
    Patches:
          app_osplookup_static_conversion.txt uploaded by lmadsen (license 10)
  ................
    r199588 | mmichelson | 2009-06-08 12:32:04 -0500 (Mon, 08 Jun 2009) | 9 lines
    
    Fix a deadlock that could occur when setting rtp stats on SIP calls.
    
    (closes issue #15143)
    Reported by: cristiandimache
    Patches:
          15143.patch uploaded by mmichelson (license 60)
    Tested by: cristiandimache
  ................
    r199630 | seanbright | 2009-06-08 14:33:09 -0500 (Mon, 08 Jun 2009) | 32 lines
    
    Merged revisions 199626,199628 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
    ........
      r199626 | seanbright | 2009-06-08 15:24:32 -0400 (Mon, 08 Jun 2009) | 21 lines
      
      Increase the size of our thread stack on 64 bit processors.
      
      We were setting the stack size for each thread to 240KB regardless of
      architecture, which meant that in some scenarios we actually had less available
      stack space on 64 bit processors (pointers use 8 bytes instead of 4).  So now we
      calculate the stack size we reserve based on the platform's __WORDSIZE, which
      gives us:
      
           32 bit -> 240KB
           64 bit -> 496KB
          128 bit -> 1008KB (that's right, we're ready for 128 bit processors)
      
      Patch typed by me but written by several members of #asterisk-dev, including
      Kevin, Tilghman, and Qwell.
      
      (closes issue #14932)
      Reported by: jpiszcz
      Patches:
            06052009_issue14932.patch uploaded by seanbright (license 71)
      Tested by: seanbright
    ........
      r199628 | seanbright | 2009-06-08 15:28:33 -0400 (Mon, 08 Jun 2009) | 2 lines
      
      Fix a typo in the stack size calculation just introduced.
    ........
  ................
    r199696 | tilghman | 2009-06-08 17:08:44 -0500 (Mon, 08 Jun 2009) | 1 line
    
    Add sigaction janitor
  ................
    r199743 | dvossel | 2009-06-09 11:22:04 -0500 (Tue, 09 Jun 2009) | 11 lines
    
    module load priority
    
    This patch adds the option to give a module a load priority. The value represents the order in which a module's load() function is initialized.  The lower the value, the higher the priority.  The value is only checked if the AST_MODFLAG_LOAD_ORDER flag is set.  If the AST_MODFLAG_LOAD_ORDER flag is not set, the value will never be read and the module will be given the lowest possible priority
    on load.  Since some modules are reliant on a timing interface, the timing modules have been given a high load priorty.
    
    (closes issue #15191)
    Reported by: alecdavis
    Tested by: dvossel
    
    Review: https://reviewboard.asterisk.org/r/262/
  ................
    r199781 | seanbright | 2009-06-09 13:08:53 -0500 (Tue, 09 Jun 2009) | 2 lines
    
    Fix all of the parallel build warnings issued when running make -j#.
  ................
    r199818 | dvossel | 2009-06-09 15:47:57 -0500 (Tue, 09 Jun 2009) | 11 lines
    
    CLI NOTIFY sending wrong transport type.
    
    SIP's cli NOTIFY command only used UDP rather than copying the transport type from the peer.
    
    (closes issue #15283)
    Reported by: jthurman
    Patches:
          sip-notify-tcp-svn199728.patch uploaded by jthurman (license 614)
    Tested by: jthurman, dvossel
  ................
    r199857 | seanbright | 2009-06-10 11:10:23 -0500 (Wed, 10 Jun 2009) | 9 lines
    
    Merged revisions 199856 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
    ........
      r199856 | seanbright | 2009-06-10 12:08:35 -0400 (Wed, 10 Jun 2009) | 2 lines
      
      __WORDSIZE is not available on all platforms, so use sizeof(void *) instead.
    ........
  ................
    r199923 | mmichelson | 2009-06-10 13:58:12 -0500 (Wed, 10 Jun 2009) | 8 lines
    
    Use ast_channel_unref to instead of ast_free on a newly created channel.
    
    Also I removed an unnecessary free of a cid_name. This will be freed properly
    in the channel destructor.
    
    Reported by mnicholson in #asterisk-dev.
  ................
    r199957 | dbrooks | 2009-06-10 15:00:45 -0500 (Wed, 10 Jun 2009) | 10 lines
    
    Fixes the argument order in definition of new_find_extension().
    
    In the definition of new_find_extension(), the arguments 'callerid' and
    'label' were swapped. The prototype declaration and all calls to the
    function are ordered 'callerid' then 'label', but the function itself
    was ordered 'label' then 'callerid'.
    
    (closes issue #15303)
    Reported by: JimDickenson
  ................
    r199958 | mmichelson | 2009-06-10 15:15:48 -0500 (Wed, 10 Jun 2009) | 6 lines
    
    Only try to use the invite_branch on outgoing INVITEs with auth credentials.
    
    I have added a comment to the code to help ease understanding of the logic here
    as well.
  ................
    r200000 | seanbright | 2009-06-10 15:40:41 -0500 (Wed, 10 Jun 2009) | 1 line
    
    Remove some trailing whitespace and steal revision 200000.
  ................
    r200038 | lmadsen | 2009-06-11 07:13:49 -0500 (Thu, 11 Jun 2009) | 14 lines
    
    Blocked revisions 200037 via svnmerge
    
    ........
      r200037 | lmadsen | 2009-06-11 08:12:06 -0400 (Thu, 11 Jun 2009) | 8 lines
      
      Fix path for .flavor and .version.
      
      (issue #14737)
      Reported by: davidw
      Patches:
            flavor.patch uploaded by davidw (license 780)
      Tested by: davidw
    ........
  ................
    r200039 | lmadsen | 2009-06-11 07:15:09 -0500 (Thu, 11 Jun 2009) | 8 lines
    
    Fix path for .flavor and .version
    
    (issue #14737)
    Reported by: davidw
    Patches:
          flavor.patch uploaded by davidw (license 780)
    Tested by: davidw
  ................
    r200108 | eliel | 2009-06-11 10:40:03 -0500 (Thu, 11 Jun 2009) | 9 lines
    
    Release the allocated channel decreasing the reference counter.
    
    When allocating the channel use ao2_ref(-1) to release it, instead of calling
    ast_free().
    Also avoid freeing structures inside that channel (on error) if they will be
    released by the channel destructor being called if the reference counter reachs
    0.
  ................
    r200146 | mmichelson | 2009-06-11 16:17:14 -0500 (Thu, 11 Jun 2009) | 5 lines
    
    Fix a crash due to a potentially NULL p->options.
    
    Thanks to mnicholson for pointing it out.
  ................
    r200190 | seanbright | 2009-06-11 17:21:32 -0500 (Thu, 11 Jun 2009) | 8 lines
    
    Blocked revisions 200185 via svnmerge
    
    ........
      r200185 | seanbright | 2009-06-11 18:20:31 -0400 (Thu, 11 Jun 2009) | 2 lines
      
      Backport fix for parallel build warnings from trunk r199781.
    ........
  ................
    r200254 | seanbright | 2009-06-11 21:20:19 -0500 (Thu, 11 Jun 2009) | 5 lines
    
    Call chgrp instead of chown when setting run directory group ownership.
    
    (issue #13153)
    Reported by: pabelanger
  ................
    r200290 | mmichelson | 2009-06-12 09:55:07 -0500 (Fri, 12 Jun 2009) | 3 lines
    
    Fix a potential crash from trying to access a NULL channel pointer.
  ................
    r200326 | mmichelson | 2009-06-12 10:37:30 -0500 (Fri, 12 Jun 2009) | 4 lines
    
    Fix some bad locking stemming from trying to forward a call to a non-existent
    extension from a queue.
  ................
    r200361 | mmichelson | 2009-06-12 14:07:51 -0500 (Fri, 12 Jun 2009) | 16 lines
    
    Merged revisions 200360 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
    ........
      r200360 | mmichelson | 2009-06-12 14:06:41 -0500 (Fri, 12 Jun 2009) | 10 lines
      
      Suppress a warning message and give a better return code when generating
      inband ringing after a call is answered.
      
      (closes issue #15158)
      Reported by: madkins
      Patches:
            15158.patch uploaded by mmichelson (license 60)
      Tested by: madkins
    ........
  ................
    r200428 | seanbright | 2009-06-12 14:42:26 -0500 (Fri, 12 Jun 2009) | 6 lines
    
    First shot at an upstart script for asterisk on Ubuntu.
    
    This works relatively well (assuming you are using /var/run/asterisk) as your
    run directory and upstart 0.3.9.  Needs to be generalized and eventually added
    to the 'make install' target for Ubuntu.
  ................
    r200430 | seanbright | 2009-06-12 14:46:25 -0500 (Fri, 12 Jun 2009) | 1 line
    
    Include basic installation and usage instructions for upstart script.
  ................
................

Added:
    team/group/issue14292/channels/sig_analog.c
      - copied unchanged from r200468, team/group/issue14068/channels/sig_analog.c
    team/group/issue14292/channels/sig_analog.h
      - copied unchanged from r200468, team/group/issue14068/channels/sig_analog.h
    team/group/issue14292/contrib/upstart/
      - copied from r200468, team/group/issue14068/contrib/upstart/
    team/group/issue14292/contrib/upstart/asterisk.upstart-0.3.9
      - copied unchanged from r200468, team/group/issue14068/contrib/upstart/asterisk.upstart-0.3.9
Modified:
    team/group/issue14292/   (props changed)
    team/group/issue14292/Makefile
    team/group/issue14292/apps/app_adsiprog.c
    team/group/issue14292/apps/app_alarmreceiver.c
    team/group/issue14292/apps/app_amd.c
    team/group/issue14292/apps/app_authenticate.c
    team/group/issue14292/apps/app_chanisavail.c
    team/group/issue14292/apps/app_channelredirect.c
    team/group/issue14292/apps/app_chanspy.c
    team/group/issue14292/apps/app_confbridge.c
    team/group/issue14292/apps/app_controlplayback.c
    team/group/issue14292/apps/app_dahdibarge.c
    team/group/issue14292/apps/app_dahdiras.c
    team/group/issue14292/apps/app_db.c
    team/group/issue14292/apps/app_dial.c
    team/group/issue14292/apps/app_dictate.c
    team/group/issue14292/apps/app_directed_pickup.c
    team/group/issue14292/apps/app_directory.c
    team/group/issue14292/apps/app_disa.c
    team/group/issue14292/apps/app_dumpchan.c
    team/group/issue14292/apps/app_echo.c
    team/group/issue14292/apps/app_exec.c
    team/group/issue14292/apps/app_externalivr.c
    team/group/issue14292/apps/app_fax.c
    team/group/issue14292/apps/app_jack.c
    team/group/issue14292/apps/app_macro.c
    team/group/issue14292/apps/app_meetme.c
    team/group/issue14292/apps/app_minivm.c
    team/group/issue14292/apps/app_osplookup.c
    team/group/issue14292/apps/app_queue.c
    team/group/issue14292/apps/app_stack.c
    team/group/issue14292/build_tools/make_version_c
    team/group/issue14292/build_tools/make_version_h
    team/group/issue14292/channels/Makefile
    team/group/issue14292/channels/chan_agent.c
    team/group/issue14292/channels/chan_dahdi.c
    team/group/issue14292/channels/chan_iax2.c
    team/group/issue14292/channels/chan_local.c
    team/group/issue14292/channels/chan_sip.c
    team/group/issue14292/channels/iax2-parser.c
    team/group/issue14292/configs/agents.conf.sample
    team/group/issue14292/configs/modules.conf.sample
    team/group/issue14292/configs/sip.conf.sample
    team/group/issue14292/contrib/init.d/rc.debian.asterisk
    team/group/issue14292/contrib/scripts/meetme.sql
    team/group/issue14292/doc/janitor-projects.txt
    team/group/issue14292/doc/tex/channelvariables.tex
    team/group/issue14292/funcs/func_math.c
    team/group/issue14292/funcs/func_strings.c
    team/group/issue14292/funcs/func_sysinfo.c
    team/group/issue14292/include/asterisk/_private.h
    team/group/issue14292/include/asterisk/app.h
    team/group/issue14292/include/asterisk/channel.h
    team/group/issue14292/include/asterisk/devicestate.h
    team/group/issue14292/include/asterisk/frame.h
    team/group/issue14292/include/asterisk/module.h
    team/group/issue14292/include/asterisk/utils.h
    team/group/issue14292/main/Makefile
    team/group/issue14292/main/app.c
    team/group/issue14292/main/asterisk.c
    team/group/issue14292/main/channel.c
    team/group/issue14292/main/devicestate.c
    team/group/issue14292/main/dial.c
    team/group/issue14292/main/features.c
    team/group/issue14292/main/loader.c
    team/group/issue14292/main/manager.c
    team/group/issue14292/main/pbx.c
    team/group/issue14292/main/rtp_engine.c
    team/group/issue14292/main/threadstorage.c
    team/group/issue14292/res/res_agi.c
    team/group/issue14292/res/res_calendar.c
    team/group/issue14292/res/res_jabber.c
    team/group/issue14292/res/res_monitor.c
    team/group/issue14292/res/res_musiconhold.c
    team/group/issue14292/res/res_phoneprov.c
    team/group/issue14292/res/res_smdi.c
    team/group/issue14292/res/res_timing_dahdi.c
    team/group/issue14292/res/res_timing_pthread.c
    team/group/issue14292/res/res_timing_timerfd.c
    team/group/issue14292/sample.call

Propchange: team/group/issue14292/
------------------------------------------------------------------------------
Binary property 'branch-1.4-blocked' - no diff available.

Propchange: team/group/issue14292/
------------------------------------------------------------------------------
Binary property 'branch-1.4-merged' - no diff available.

Propchange: team/group/issue14292/
------------------------------------------------------------------------------
--- issue14292-integrated (original)
+++ issue14292-integrated Fri Jun 12 16:56:19 2009
@@ -1,1 +1,1 @@
-/team/group/issue14068:1-199816
+/team/group/issue14068:1-200468

Propchange: team/group/issue14292/
------------------------------------------------------------------------------
--- svnmerge-integrated (original)
+++ svnmerge-integrated Fri Jun 12 16:56:19 2009
@@ -1,1 +1,1 @@
-/trunk:1-198083
+/trunk:1-200463

Modified: team/group/issue14292/Makefile
URL: http://svn.asterisk.org/svn-view/asterisk/team/group/issue14292/Makefile?view=diff&rev=200469&r1=200468&r2=200469
==============================================================================
--- team/group/issue14292/Makefile (original)
+++ team/group/issue14292/Makefile Fri Jun 12 16:56:19 2009
@@ -371,24 +371,24 @@
 	menuselect/menuselect --check-deps menuselect.makeopts $(GLOBAL_MAKEOPTS) $(USER_MAKEOPTS)
 
 $(MOD_SUBDIRS_EMBED_LDSCRIPT):
-	@echo "EMBED_LDSCRIPTS+="`$(SILENTMAKE) -C $(@:-embed-ldscript=) SUBDIR=$(@:-embed-ldscript=) __embed_ldscript` >> makeopts.embed_rules
+	+ at echo "EMBED_LDSCRIPTS+="`$(SILENTMAKE) -C $(@:-embed-ldscript=) SUBDIR=$(@:-embed-ldscript=) __embed_ldscript` >> makeopts.embed_rules
 
 $(MOD_SUBDIRS_EMBED_LDFLAGS):
-	@echo "EMBED_LDFLAGS+="`$(SILENTMAKE) -C $(@:-embed-ldflags=) SUBDIR=$(@:-embed-ldflags=) __embed_ldflags` >> makeopts.embed_rules
+	+ at echo "EMBED_LDFLAGS+="`$(SILENTMAKE) -C $(@:-embed-ldflags=) SUBDIR=$(@:-embed-ldflags=) __embed_ldflags` >> makeopts.embed_rules
 
 $(MOD_SUBDIRS_EMBED_LIBS):
-	@echo "EMBED_LIBS+="`$(SILENTMAKE) -C $(@:-embed-libs=) SUBDIR=$(@:-embed-libs=) __embed_libs` >> makeopts.embed_rules
+	+ at echo "EMBED_LIBS+="`$(SILENTMAKE) -C $(@:-embed-libs=) SUBDIR=$(@:-embed-libs=) __embed_libs` >> makeopts.embed_rules
 
 $(MOD_SUBDIRS_MENUSELECT_TREE):
-	@$(SUBMAKE) -C $(@:-menuselect-tree=) SUBDIR=$(@:-menuselect-tree=) moduleinfo

[... 9025 lines stripped ...]



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