asterisk/asterisk.git
9 years agoMerged revisions 336792 via svnmerge from
Terry Wilson [Mon, 19 Sep 2011 22:28:17 +0000 (22:28 +0000)]
Merged revisions 336792 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

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  r336792 | twilson | 2011-09-19 17:13:34 -0500 (Mon, 19 Sep 2011) | 9 lines

  Merged revisions 336791 via svnmerge from
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    r336791 | twilson | 2011-09-19 17:07:58 -0500 (Mon, 19 Sep 2011) | 2 lines

    Don't interfere with T.38 reinvites

    This is an update to the fix for ASTERISK-18340 and ASTERISK-17725
  ........
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9 years agoMerged revisions 336789 via svnmerge from
Tilghman Lesher [Mon, 19 Sep 2011 21:42:11 +0000 (21:42 +0000)]
Merged revisions 336789 via svnmerge from
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  r336789 | tilghman | 2011-09-19 16:41:16 -0500 (Mon, 19 Sep 2011) | 2 lines

  Ensure substring will not be found in the previous match.
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9 years agoMerged revisions 336734 via svnmerge from
Tilghman Lesher [Mon, 19 Sep 2011 20:31:09 +0000 (20:31 +0000)]
Merged revisions 336734 via svnmerge from
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  r336734 | tilghman | 2011-09-19 15:29:40 -0500 (Mon, 19 Sep 2011) | 18 lines

  Merged revisions 336733 via svnmerge from
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    r336733 | tilghman | 2011-09-19 15:27:03 -0500 (Mon, 19 Sep 2011) | 11 lines

    Various changes to allow 1.8 to compile on Mac OS X Lion (10.7)

    * Makefile workaround for 10.6 extended to work on 10.7 and later.
    * Now uses the 'weak' symbol for Lion systems, which no longer support
      'weak_import'

    Closes ASTERISK-17612.
    Closes ASTERISK-18213.

    Tested by: tilghman, oej.
  ........
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9 years agoMerged revisions 336717 via svnmerge from
Jonathan Rose [Mon, 19 Sep 2011 20:23:29 +0000 (20:23 +0000)]
Merged revisions 336717 via svnmerge from
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  r336717 | jrose | 2011-09-19 15:16:23 -0500 (Mon, 19 Sep 2011) | 14 lines

  Merged revisions 336716 via svnmerge from
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    r336716 | jrose | 2011-09-19 15:07:36 -0500 (Mon, 19 Sep 2011) | 7 lines

    Document applications that play audio and do not answer unanswered calls.

    This patch is part of an effort to document early media and its usage. If you are
    interested in contributing to this documentation effort, there are probably other
    applications worth documenting as well as an Asterisk wiki article at
    https://wiki.asterisk.org/wiki/display/AST/Early+Media+and+the+Progress+Application
  ........
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9 years agoMerged revisions 336659 via svnmerge from
Richard Mudgett [Mon, 19 Sep 2011 19:03:38 +0000 (19:03 +0000)]
Merged revisions 336659 via svnmerge from
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  r336659 | rmudgett | 2011-09-19 13:51:19 -0500 (Mon, 19 Sep 2011) | 38 lines

  Merged revisions 336658 via svnmerge from
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    r336658 | rmudgett | 2011-09-19 13:46:40 -0500 (Mon, 19 Sep 2011) | 31 lines

    Made Dial d and H options no longer immediately auto-answer the calling leg.

    The Dial d and H options break DTMF attended transfer atxferdropcall
    option.

    1) Party A calls party B.
    2) Party B does a DTMF attended transfer to Party C.

    If the dialplan uses the Dial d or H options to call Party C then the Dial
    application answers the call immediately before initiating the call leg to
    Party C.  The premature answer causes the transfer code to not invoke the
    atxferdropcall=no behavior for a blonde transfer since Party C has
    "answered".  The transfer code thinks that Party B has "consulted" with
    Party C when Party B hangs up and completes the transfer to Party A.
    Party A now hears ringback until Party C actually answers.

    ASTERISK-13294 Dial d option.
    ASTERISK-11067 Dial H option to disconnect before answer.

    The referenced issues made Dial answer with the d and H options because
    many SIP and ISDN phones cannot send DTMF before the call is connected.

    * Made require the dialplan to control when or if the call needs to be
    answered to use the Dial application d and H options.  (The call is no
    longer surprise answered when using the Dial d or H options.)

    Review: https://reviewboard.asterisk.org/r/1381/

    JIRA AST-623
    JIRA AST-666
  ........
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9 years agoUpdate merge 10 branch merge propterty.
Richard Mudgett [Mon, 19 Sep 2011 19:00:16 +0000 (19:00 +0000)]
Update merge 10 branch merge propterty.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@336661 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoRestore 10 branch merge properties.
Richard Mudgett [Mon, 19 Sep 2011 18:57:50 +0000 (18:57 +0000)]
Restore 10 branch merge properties.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@336660 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoRemove weird mergeinfo props that make merges annoying sometimes.
Jason Parker [Mon, 19 Sep 2011 16:22:52 +0000 (16:22 +0000)]
Remove weird mergeinfo props that make merges annoying sometimes.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@336600 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 336572 via svnmerge from
Leif Madsen [Mon, 19 Sep 2011 15:48:53 +0000 (15:48 +0000)]
Merged revisions 336572 via svnmerge from
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  r336572 | lmadsen | 2011-09-19 10:41:16 -0500 (Mon, 19 Sep 2011) | 7 lines

  Update get_ilbc_source.sh script to work again.

  Recently iLBC support in Asterisk has changed after the acquisition of GIPS
  by Google. More information about how this may affect you is available in a
  blog post at:

    http://blogs.asterisk.org/2011/09/19/ilbc-support-in-asterisk-after-googles-acquisition-of-gips/
........

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9 years agoMerged revisions 336570 via svnmerge from
Richard Mudgett [Mon, 19 Sep 2011 15:36:39 +0000 (15:36 +0000)]
Merged revisions 336570 via svnmerge from
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  r336570 | rmudgett | 2011-09-19 10:32:00 -0500 (Mon, 19 Sep 2011) | 11 lines

  Merged revisions 336569 via svnmerge from
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    r336569 | rmudgett | 2011-09-19 10:25:34 -0500 (Mon, 19 Sep 2011) | 4 lines

    Rework sig_pri_hangup() to be simpler and clearer.

    JIRA AST-675
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9 years agoMerged revisions 336502 via svnmerge from
Olle Johansson [Mon, 19 Sep 2011 13:57:26 +0000 (13:57 +0000)]
Merged revisions 336502 via svnmerge from
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  r336502 | oej | 2011-09-19 15:38:53 +0200 (Mån, 19 Sep 2011) | 12 lines

  Merged revisions 336501 via svnmerge from
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    r336501 | oej | 2011-09-19 15:33:50 +0200 (Mån, 19 Sep 2011) | 5 lines

    Add diversion header to a 302 redirect response if we have diversion data

    (closes issue ASTERISK-18143)
     patch by oej
  ........
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9 years agoMerged revisions 336500 via svnmerge from
Gregory Nietsky [Mon, 19 Sep 2011 13:41:52 +0000 (13:41 +0000)]
Merged revisions 336500 via svnmerge from
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  r336500 | irroot | 2011-09-19 15:31:50 +0200 (Mon, 19 Sep 2011) | 19 lines

  Merged revisions 336499 via svnmerge from
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  ........
    r336499 | irroot | 2011-09-19 15:27:52 +0200 (Mon, 19 Sep 2011) | 13 lines

    A long time ago in a galaxy far far away a IPv6 update was made,
    chan_h323 was not updated causeing all to flee to chan_ooh323.

    the brave Jedi [asterisk developers] pondered this miscarrige of justice
    and restored order to the force for the sake of closing out 2 old issues.

    (closes issue ASTERISK-17278)
    (closes issue ASTERISK-17500)
    Reported by: dread, sybasesql
    Tested by: irroot
    Reviewed by: IRC (russellb, kpfleming)
  ........
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9 years agoMerged revisions 336441 via svnmerge from
Olle Johansson [Mon, 19 Sep 2011 12:20:44 +0000 (12:20 +0000)]
Merged revisions 336441 via svnmerge from
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  r336441 | oej | 2011-09-19 14:15:06 +0200 (Mån, 19 Sep 2011) | 9 lines

  Merged revisions 336440 via svnmerge from
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    r336440 | oej | 2011-09-19 14:06:48 +0200 (Mån, 19 Sep 2011) | 2 lines

    Make sure manager_debug option is reset at reload
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9 years agoMerged revisions 336381 via svnmerge from
Olle Johansson [Mon, 19 Sep 2011 10:10:11 +0000 (10:10 +0000)]
Merged revisions 336381 via svnmerge from
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  r336381 | oej | 2011-09-19 12:05:00 +0200 (Mån, 19 Sep 2011) | 16 lines

  Merged revisions 336378 via svnmerge from
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    r336378 | oej | 2011-09-19 11:40:44 +0200 (Mån, 19 Sep 2011) | 9 lines

    Add missing unlock at MWI message sending time

    (closes issue ASTERISK-18573)

    Patches:
       sip_mwi_lock.patch (license #5041) by Gregory Hinton Nietsky

    Thanks to irrot for the reminder, to Gregory for the patch!
  ........
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9 years agoMerged revisions 336316 via svnmerge from
Terry Wilson [Fri, 16 Sep 2011 22:12:24 +0000 (22:12 +0000)]
Merged revisions 336316 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

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  r336316 | twilson | 2011-09-16 17:11:39 -0500 (Fri, 16 Sep 2011) | 9 lines

  Merged revisions 336314 via svnmerge from
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    r336314 | twilson | 2011-09-16 17:10:56 -0500 (Fri, 16 Sep 2011) | 2 lines

    Whitespace fix
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9 years agoMerged revisions 336313 via svnmerge from
Terry Wilson [Fri, 16 Sep 2011 22:11:01 +0000 (22:11 +0000)]
Merged revisions 336313 via svnmerge from
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  r336313 | twilson | 2011-09-16 17:07:00 -0500 (Fri, 16 Sep 2011) | 12 lines

  Merged revisions 336312 via svnmerge from
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    r336312 | twilson | 2011-09-16 17:04:25 -0500 (Fri, 16 Sep 2011) | 5 lines

    Add missing frame types to func_frame_trace

    Also casts control frames to the proper enum so that the compile will catch
    new additions.
  ........
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9 years agoMerged revisions 336307 via svnmerge from
Jonathan Rose [Fri, 16 Sep 2011 21:20:02 +0000 (21:20 +0000)]
Merged revisions 336307 via svnmerge from
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  r336307 | jrose | 2011-09-16 16:09:20 -0500 (Fri, 16 Sep 2011) | 20 lines

  Merged revisions 336294 via svnmerge from
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  ........
    r336294 | jrose | 2011-09-16 14:53:40 -0500 (Fri, 16 Sep 2011) | 13 lines

    Fix bad RTP media bridges in directmedia calls on peers separated by multiple Asterisk nodes.

    In a situation involving devices on separate Asterisk trunks, the remote RTP bridge would
    break when starting a call with directmedia. This patch queues a new type of control frame
    so that our RTP bridge loop can properly detect when these situations occur and check to see
    if peers need to be updated in order to send their media to the proper location.

    (Closes issue ASTERISK-18340)
    Reported by: Thomas Arimont
    (Closes issue ASTERISK-17725)
    Reported by: kwk
    Tested by: twilson, jrose
  ........
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9 years agoMerged revisions 336235 via svnmerge from
Sean Bright [Fri, 16 Sep 2011 19:11:22 +0000 (19:11 +0000)]
Merged revisions 336235 via svnmerge from
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  r336235 | seanbright | 2011-09-16 15:10:39 -0400 (Fri, 16 Sep 2011) | 9 lines

  Merged revisions 336234 via svnmerge from
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    r336234 | seanbright | 2011-09-16 15:06:27 -0400 (Fri, 16 Sep 2011) | 2 lines

    Make a note that inotify won't work with an NFS mounted spooler directory.
  ........
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9 years agoMerged revisions 336167 via svnmerge from
Gregory Nietsky [Fri, 16 Sep 2011 10:16:56 +0000 (10:16 +0000)]
Merged revisions 336167 via svnmerge from
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  r336167 | irroot | 2011-09-16 12:12:03 +0200 (Fri, 16 Sep 2011) | 22 lines

  Merged revisions 336166 via svnmerge from
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    r336166 | irroot | 2011-09-16 12:09:17 +0200 (Fri, 16 Sep 2011) | 16 lines

    The round robin routing routine in chan_misdn.c is broken.

    it rotates between ports but never checks the channels in the ports.

    i have extensivly tested it and verified it works on 1 upto 4 ports.
    before the patch only 1 out of each port was used now all are used as
    expected.

    (closes issue ASTERISK-18413)
    Reported by: irroot
    Tested by: irroot
    Reviewed by: irroot

    Review: https://reviewboard.asterisk.org/r/1410/
  ........
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9 years agoMerged revisions 336094 via svnmerge from
Gregory Nietsky [Thu, 15 Sep 2011 15:59:24 +0000 (15:59 +0000)]
Merged revisions 336094 via svnmerge from
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  r336094 | irroot | 2011-09-15 17:54:46 +0200 (Thu, 15 Sep 2011) | 26 lines

  Merged revisions 336093 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.8

  ........
    r336093 | irroot | 2011-09-15 17:46:21 +0200 (Thu, 15 Sep 2011) | 20 lines

    Locking order in app_queue.c causes deadlocks.

    a channel lock must never be held with the queues container lock held.

    the deadlock occured on masquerade.

    the queues container lock is a relic of the past the old queue module lock.
    with ao2 there is no need to hold this lock when dealing with members this
    patch removes unneeded locks.

    (closes issue ASTERISK-18101)
    (closes issue ASTERISK-18487)
    Reported by: Paul Rolfe, Jason Legault
    Tested by: irroot, Jason Legault, Paul Rolfe
    Reviewed by: Matthew Nicholson

    Review: https://reviewboard.asterisk.org/r/1402/
  ........
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9 years agoMerged revisions 336091 via svnmerge from
David Vossel [Thu, 15 Sep 2011 15:19:51 +0000 (15:19 +0000)]
Merged revisions 336091 via svnmerge from
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  r336091 | dvossel | 2011-09-15 10:19:10 -0500 (Thu, 15 Sep 2011) | 2 lines

  Removes some no-op code found in format_cap.c.
........

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9 years agoMerged revisions 336042 via svnmerge from
Olle Johansson [Thu, 15 Sep 2011 12:50:40 +0000 (12:50 +0000)]
Merged revisions 336042 via svnmerge from
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  r336042 | oej | 2011-09-15 14:46:38 +0200 (Tor, 15 Sep 2011) | 12 lines

  Meetme: Introducing a new option "k" to kill a conference if there's only a single member left.

  When using Meetme as a modular call bridge from third party applications, it's handy to make
  it behave like a normal call bridge. When the second to last person exists, the last person
  will be kicked out of the conference when this option is enabled.

  (closes issue ASTERISK-18234)

  Review: https://reviewboard.asterisk.org/r/1376/

  Patch by oej, sponsored by ClearIT, Solna, Sweden
........

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9 years agoMerged revisions 335991 via svnmerge from
Gregory Nietsky [Thu, 15 Sep 2011 08:40:07 +0000 (08:40 +0000)]
Merged revisions 335991 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

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  r335991 | irroot | 2011-09-15 10:29:12 +0200 (Thu, 15 Sep 2011) | 17 lines

  Merged revisions 335978 via svnmerge from
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    r335978 | irroot | 2011-09-15 10:15:22 +0200 (Thu, 15 Sep 2011) | 11 lines

    lock the channel before calling ast_bridged_channel() to prevent a seg fault.

    AMI agents list called on shutdown causes a segfault, introducing proper locking
    will prevent this.

    (closes issue ASTERISK-18092)

    Reported by: agustina
    Patches: chan_agent.patch (License #5041) patch uploaded by irroot
  ........
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9 years agoMerged revisions 335912 via svnmerge from
Richard Mudgett [Wed, 14 Sep 2011 18:38:43 +0000 (18:38 +0000)]
Merged revisions 335912 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

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  r335912 | rmudgett | 2011-09-14 13:31:15 -0500 (Wed, 14 Sep 2011) | 20 lines

  Merged revisions 335911 via svnmerge from
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  ........
    r335911 | rmudgett | 2011-09-14 13:21:35 -0500 (Wed, 14 Sep 2011) | 13 lines

    Remove unnecessary libpri dependency checks in the configure script.

    Using the --with-pri option with the configure script generated an error
    about not having PRI_L2_PERSISTENCE if you did not have the absolute
    latest libpri SVN checkout installed.

    The AST_EXT_LIB_SETUP_DEPENDENT macro in the configure.ac script seems to
    be for libraries that are dependent upon other libraries and not
    necessarily for optional/added features within a library.

    (closes issue ASTERISK-18535)
    Reported by: Michael Keuter
  ........
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9 years agoMerged revisions 335852 via svnmerge from
Richard Mudgett [Wed, 14 Sep 2011 16:05:38 +0000 (16:05 +0000)]
Merged revisions 335852 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

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  r335852 | rmudgett | 2011-09-14 11:00:37 -0500 (Wed, 14 Sep 2011) | 18 lines

  Merged revisions 335851 via svnmerge from
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  ........
    r335851 | rmudgett | 2011-09-14 10:53:25 -0500 (Wed, 14 Sep 2011) | 11 lines

    Fixed cut-n-paste regression using the wrong variable.

    Fixes the missing DAHDI channels when using the newer chan_dahdi.conf
    sections for channel configuration.

    (closes issue ASTERISK-18496)
    Reported by: Sean Darcy
    Patches:
          jira_asterisk_18496_v1.8.patch (license #5621) patch uploaded by rmudgett
    Tested by: Sean Darcy, rmudgett
  ........
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9 years agoMerged revisions 335791 via svnmerge from
Matthew Nicholson [Wed, 14 Sep 2011 13:29:41 +0000 (13:29 +0000)]
Merged revisions 335791 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

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  r335791 | mnicholson | 2011-09-14 08:28:50 -0500 (Wed, 14 Sep 2011) | 11 lines

  Merged revisions 335790 via svnmerge from
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  ........
    r335790 | mnicholson | 2011-09-14 08:28:16 -0500 (Wed, 14 Sep 2011) | 4 lines

    The tech and data members of fast_originate_helper are not string fields.

    ASTERISK-17709
  ........
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9 years agoMerged revisions 335721 via svnmerge from
Richard Mudgett [Tue, 13 Sep 2011 22:11:20 +0000 (22:11 +0000)]
Merged revisions 335721 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

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  r335721 | rmudgett | 2011-09-13 17:10:44 -0500 (Tue, 13 Sep 2011) | 9 lines

  Merged revisions 335720 via svnmerge from
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  ........
    r335720 | rmudgett | 2011-09-13 17:10:15 -0500 (Tue, 13 Sep 2011) | 1 line

    Remove obsolete todo comment about PICKUPRESULT.
  ........
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9 years agoAdditional updates for parsing dnsmgr.conf
Paul Belanger [Tue, 13 Sep 2011 21:52:59 +0000 (21:52 +0000)]
Additional updates for parsing dnsmgr.conf

Review: https://reviewboard.asterisk.org/r/1432/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@335719 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agodo parse defaultlanguage from asterisk.conf
Tzafrir Cohen [Tue, 13 Sep 2011 21:40:56 +0000 (21:40 +0000)]
do parse defaultlanguage from asterisk.conf

Do parse the option "defaultlanguage" from the [options] section of
asterisk.conf, as in the sample config file. Otherwise the build-time
default language (normally "en") is always the default one.

Review: https://reviewboard.asterisk.org/r/1342/
Signed-off-by: Tzafrir Cohen (License #5035) <tzafrir.cohen@xorcom.com>
Original-Commit: http://svn.digium.com/svn/asterisk/branches/1.8@335716
Original-Commit: http://svn.digium.com/svn/asterisk/branches/10@335717

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@335718 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 335656 via svnmerge from
Tilghman Lesher [Tue, 13 Sep 2011 18:56:45 +0000 (18:56 +0000)]
Merged revisions 335656 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

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  r335656 | tilghman | 2011-09-13 13:55:33 -0500 (Tue, 13 Sep 2011) | 11 lines

  Merged revisions 335655 via svnmerge from
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  ........
    r335655 | tilghman | 2011-09-13 13:52:38 -0500 (Tue, 13 Sep 2011) | 4 lines

    Move mandatory checks closer to the beginning of the file.

    If these are going to fail, they should fail as quickly as possible.
  ........
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9 years agoMerged revisions 335653 via svnmerge from
Matthew Nicholson [Tue, 13 Sep 2011 18:49:26 +0000 (18:49 +0000)]
Merged revisions 335653 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

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  r335653 | mnicholson | 2011-09-13 13:47:57 -0500 (Tue, 13 Sep 2011) | 12 lines

  Merged revisions 335618 via svnmerge from
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  ........
    r335618 | mnicholson | 2011-09-13 13:20:52 -0500 (Tue, 13 Sep 2011) | 5 lines

    Don't limit the size of appdata for manager originate actions.

    ASTERISK-17709
    Patch by: tilghman (with modifications)
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@335654 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoClean up dsp.conf parsing
Paul Belanger [Tue, 13 Sep 2011 18:11:33 +0000 (18:11 +0000)]
Clean up dsp.conf parsing

Review: https://reviewboard.asterisk.org/r/1434/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@335603 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoClean up cdr.conf parsing for [csv] section
Paul Belanger [Tue, 13 Sep 2011 14:25:43 +0000 (14:25 +0000)]
Clean up cdr.conf parsing for [csv] section

Review: https://reviewboard.asterisk.org/r/1427/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@335556 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoClean up dnsmgr.conf parsing
Paul Belanger [Tue, 13 Sep 2011 14:22:58 +0000 (14:22 +0000)]
Clean up dnsmgr.conf parsing

Review: https://reviewboard.asterisk.org/r/1432/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@335555 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 335510 via svnmerge from
Russell Bryant [Tue, 13 Sep 2011 07:35:59 +0000 (07:35 +0000)]
Merged revisions 335510 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

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  r335510 | russell | 2011-09-13 02:24:34 -0500 (Tue, 13 Sep 2011) | 22 lines

  Merged revisions 335497 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.8

  ........
    r335497 | russell | 2011-09-13 02:11:36 -0500 (Tue, 13 Sep 2011) | 15 lines

    Fix a crash in res_ais.

    This patch resolves a crash observed in a load testing environment that
    involved the use of the res_ais module.  I observed some crashes where
    the event delivery callback would get called, but the length parameter
    incidcating how much data there was to read was 0.  The code assumed
    (with good reason I would think) that if this callback got called, there
    was an event available to read.  However, if the rare case that there's
    nothing there, catch it and return instead of blowing up.

    More specifically, the change always ensure that the size of the received
    event in the cluster is always big enough to be a real ast_event.

    Review: https://reviewboard.asterisk.org/r/1423/
  ........
................

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9 years agoMerged revisions 335434 via svnmerge from
Matthew Nicholson [Mon, 12 Sep 2011 15:56:27 +0000 (15:56 +0000)]
Merged revisions 335434 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

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  r335434 | mnicholson | 2011-09-12 10:55:48 -0500 (Mon, 12 Sep 2011) | 13 lines

  Merged revisions 335433 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.8

  ........
    r335433 | mnicholson | 2011-09-12 10:54:41 -0500 (Mon, 12 Sep 2011) | 6 lines

    Properly set caller_warning and callee_warning before we try to use them.

    ASTERISK-18199
    Patch by: elguero
    Testing by: rtang
  ........
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9 years agoDocumentation updates
Olle Johansson [Mon, 12 Sep 2011 14:33:43 +0000 (14:33 +0000)]
Documentation updates

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@335385 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 335346 via svnmerge from
Kinsey Moore [Mon, 12 Sep 2011 14:24:03 +0000 (14:24 +0000)]
Merged revisions 335346 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

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  r335346 | kmoore | 2011-09-12 09:22:15 -0500 (Mon, 12 Sep 2011) | 17 lines

  Merged revisions 335341 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.8

  ........
    r335341 | kmoore | 2011-09-12 09:21:17 -0500 (Mon, 12 Sep 2011) | 10 lines

    Ensure frames are not written to dialed channel if ringback is requested

    When a single channel was dialed and there was media to be forwarded to the
    calling channel, the media was written without regard for ringback causing
    silence to be heard in some circumstances.  This regression was introduced
    when the meaning of "single" changed to mean only the number of channels
    dialed.

    (closes issue ASTERISK-18083)
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@335354 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoSmall documentation updates
Olle Johansson [Mon, 12 Sep 2011 14:22:56 +0000 (14:22 +0000)]
Small documentation updates

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@335349 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoNew sip.conf option for setting default tonezone for channel or individual devices
Olle Johansson [Mon, 12 Sep 2011 13:57:57 +0000 (13:57 +0000)]
New sip.conf option for setting default tonezone for channel or individual devices

Review: https://reviewboard.asterisk.org/r/1429/

(closes issue ASTERISK-18497)

Thanks to russellb for peer review.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@335325 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 335323 via svnmerge from
Olle Johansson [Mon, 12 Sep 2011 13:50:24 +0000 (13:50 +0000)]
Merged revisions 335323 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

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  r335323 | oej | 2011-09-12 15:47:13 +0200 (Mån, 12 Sep 2011) | 19 lines

  Merged revisions 335319 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.8

  ........
    r335319 | oej | 2011-09-12 15:25:30 +0200 (Mån, 12 Sep 2011) | 12 lines

    Lock the peer->mvipvt to avoid crashes with SIP history enabled

    After the launch of 1.6 event-based MWI we have two threads handling the peer->mwipvt,
    which cause issues with SIP history additions in combination with the max limit for
    number of history entries.

    Review: https://reviewboard.asterisk.org/r/1373/

    (closes issue ASTERISK-18288)

    Thanks to irrot for peer review. Work with this bug funded by IPvision AS
  ........
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9 years agoMerged revisions 335321 via svnmerge from
Kinsey Moore [Mon, 12 Sep 2011 13:27:45 +0000 (13:27 +0000)]
Merged revisions 335321 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

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  r335321 | kmoore | 2011-09-12 08:27:04 -0500 (Mon, 12 Sep 2011) | 16 lines

  Merged revisions 335320 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.8

  ........
    r335320 | kmoore | 2011-09-12 08:25:42 -0500 (Mon, 12 Sep 2011) | 9 lines

    Prevent IAX2 from getting IPv6 addresses via DNS

    IAX2 does not support IPv6 and getting such addresses from DNS can cause error
    messages on the remote end involving bad IPv4 address casts in the presence of
    IPv6/IPv4 tunnels.  This patch ensures that IAX2 will not encounter IPv6
    addresses via DNS queries.

    (closes issue ASTERISK-18090)
  ........
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9 years agoMerged revisions 335260 via svnmerge from
Stefan Schmidt [Mon, 12 Sep 2011 11:15:01 +0000 (11:15 +0000)]
Merged revisions 335260 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

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  r335260 | schmidts | 2011-09-12 11:11:45 +0000 (Mon, 12 Sep 2011) | 12 lines

  Merged revisions 335259 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.8

  ........
    r335259 | schmidts | 2011-09-12 11:09:19 +0000 (Mon, 12 Sep 2011) | 6 lines

    build_peer doesnt unlink a peer object from peers_by_ip container which leads to a wrong refcounter value.
    adding an ao2_unlink from the peers_by_ip container fix it.

    Review: https://reviewboard.asterisk.org/r/1428/
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@335261 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoBe more specific on which section has changed.
Paul Belanger [Mon, 12 Sep 2011 03:10:21 +0000 (03:10 +0000)]
Be more specific on which section has changed.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@335212 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoIterate though cdr.conf setting
Paul Belanger [Sun, 11 Sep 2011 18:21:39 +0000 (18:21 +0000)]
Iterate though cdr.conf setting

Review: https://reviewboard.asterisk.org/r/1426/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@335170 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoAdd SQLite 3 realtime support
Terry Wilson [Sun, 11 Sep 2011 17:09:36 +0000 (17:09 +0000)]
Add SQLite 3 realtime support

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@335129 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 335078 via svnmerge from
Matthew Jordan [Fri, 9 Sep 2011 16:28:23 +0000 (16:28 +0000)]
Merged revisions 335078 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

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  r335078 | mjordan | 2011-09-09 11:27:01 -0500 (Fri, 09 Sep 2011) | 29 lines

  Merged revisions 335064 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.8

  ........
    r335064 | mjordan | 2011-09-09 11:09:09 -0500 (Fri, 09 Sep 2011) | 23 lines

    Updated SIP 484 handling; added Incomplete control frame

    When a SIP phone uses the dial application and receives a 484 Address
    Incomplete response, if overlapped dialing is enabled for SIP, then
    the 484 Address Incomplete is forwarded back to the SIP phone and the
    HANGUPCAUSE channel variable is set to 28.  Previously, the Incomplete
    application dialplan logic was automatically triggered; now, explicit
    dialplan usage of the application is required.

    Additionally, this patch adds a new AST_CONTOL_FRAME type called
    AST_CONTROL_INCOMPLETE.  If a channel driver receives this control frame,
    it is an indication that the dialplan expects more digits back from the
    device.  If the device supports overlap dialing it should attempt to
    notify the device that the dialplan is waiting for more digits; otherwise,
    it can handle the frame in a manner appropriate to the channel driver.

    (closes issue ASTERISK-17288)
    Reported by: Mikael Carlsson
    Tested by: Matthew Jordan

    Review: https://reviewboard.asterisk.org/r/1416/
  ........
................

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@335079 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 335014 via svnmerge from
Gregory Nietsky [Fri, 9 Sep 2011 07:28:42 +0000 (07:28 +0000)]
Merged revisions 335014 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

........
  r335014 | irroot | 2011-09-09 09:23:53 +0200 (Fri, 09 Sep 2011) | 9 lines

  Move code for VALID_EXTEN from app_readexten to func_dialplan

  Mark VALID_EXTEN deprecated.

  Review: https://reviewboard.asterisk.org/r/1396/
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@335015 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 334954 via svnmerge from
Richard Mudgett [Thu, 8 Sep 2011 22:30:42 +0000 (22:30 +0000)]
Merged revisions 334954 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

................
  r334954 | rmudgett | 2011-09-08 17:28:56 -0500 (Thu, 08 Sep 2011) | 17 lines

  Merged revisions 334953 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.8

  ........
    r334953 | rmudgett | 2011-09-08 17:27:40 -0500 (Thu, 08 Sep 2011) | 10 lines

    Fix crash with res_fax when MALLOC_DEBUG and "core stop gracefully" are used.

    Asterisk crashes if MALLOC_DEBUG is enabled when res_fax tries to
    unregister its logger level.

    * Make ast_logger_unregister_level() use ast_free() instead of free().
    When MALLOC_DEBUG is enabled, ast_free() does not degenerate into a call
    to free().  Therefore, if you allocated memory with a form of ast_malloc
    you must free it with ast_free.
  ........
................

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@334955 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoRemoves colorful verb statements erroneously commited with r332760
Jonathan Rose [Thu, 8 Sep 2011 13:36:11 +0000 (13:36 +0000)]
Removes colorful verb statements erroneously commited with r332760

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@334907 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 334844 via svnmerge from
Paul Belanger [Wed, 7 Sep 2011 19:38:58 +0000 (19:38 +0000)]
Merged revisions 334844 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

................
  r334844 | pabelanger | 2011-09-07 15:37:24 -0400 (Wed, 07 Sep 2011) | 11 lines

  Merged revisions 334843 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.8

  ........
    r334843 | pabelanger | 2011-09-07 15:35:52 -0400 (Wed, 07 Sep 2011) | 4 lines

    Cleanup chan_iax2.c log messages

    Review: https://code.asterisk.org/code/cru/CR-AST-11
  ........
................

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@334845 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 334841 via svnmerge from
Richard Mudgett [Wed, 7 Sep 2011 19:35:18 +0000 (19:35 +0000)]
Merged revisions 334841 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

................
  r334841 | rmudgett | 2011-09-07 14:33:38 -0500 (Wed, 07 Sep 2011) | 17 lines

  Merged revisions 334840 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.8

  ........
    r334840 | rmudgett | 2011-09-07 14:31:44 -0500 (Wed, 07 Sep 2011) | 10 lines

    Fix AMI action Park crash.

    * Made AMI action Park not say anything to the parker channel (AMI header
    Channel2) since the AMI action is a third party parking the call.  (This
    is a change in behavior that cannot be preserved without a lot of effort.)

    * Made not play pbx-parkingfailed if the Park 's' option is used.

    JIRA AST-660
  ........
................

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@334842 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 334747 via svnmerge from
Stefan Schmidt [Wed, 7 Sep 2011 15:37:32 +0000 (15:37 +0000)]
Merged revisions 334747 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

................
  r334747 | schmidts | 2011-09-07 15:10:37 +0000 (Wed, 07 Sep 2011) | 9 lines

  Merged revisions 334682 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.8

  ........
    r334682 | schmidts | 2011-09-07 13:26:50 +0000 (Wed, 07 Sep 2011) | 3 lines

    Adding the Feature to sent a Reason Header in a SIP Cancel message by set the flag AST_FLAG_ANSWERED_ELSEWHERE before doing a masquerade in the pickup function.
  ........
................

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@334792 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoclean up wrong merged stuff
Stefan Schmidt [Wed, 7 Sep 2011 14:47:03 +0000 (14:47 +0000)]
clean up wrong merged stuff

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@334744 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 334682 via svnmerge from
Stefan Schmidt [Wed, 7 Sep 2011 14:23:38 +0000 (14:23 +0000)]
Merged revisions 334682 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

........
  r334682 | schmidts | 2011-09-07 13:26:50 +0000 (Wed, 07 Sep 2011) | 3 lines

  Adding the Feature to sent a Reason Header in a SIP Cancel message by set the flag AST_FLAG_ANSWERED_ELSEWHERE before doing a masquerade in the pickup function.
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@334712 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoAdding the Feature to sent a Reason Header in a SIP Cancel message by set the flag...
Stefan Schmidt [Wed, 7 Sep 2011 13:31:13 +0000 (13:31 +0000)]
Adding the Feature to sent a Reason Header in a SIP Cancel message by set the flag AST_FLAG_ANSWERED_ELSEWHERE before doing a masquerade in the pickup function.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@334683 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 334621 via svnmerge from
Alec L Davis [Wed, 7 Sep 2011 08:17:24 +0000 (08:17 +0000)]
Merged revisions 334621 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

................
  r334621 | alecdavis | 2011-09-07 20:14:50 +1200 (Wed, 07 Sep 2011) | 9 lines

  Merged revisions 334620 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.8

  ........
    r334620 | alecdavis | 2011-09-07 20:12:49 +1200 (Wed, 07 Sep 2011) | 2 lines

    peroid typo
  ........
................

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@334623 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agolog Asterisk Version number, Build etc into each log file
Alec L Davis [Wed, 7 Sep 2011 08:06:32 +0000 (08:06 +0000)]
log Asterisk Version number, Build etc into each log file

Allow tracking of previous versions through log file records to be tracked.
Each time log file is created or opened, log Asterisk Version, Buildinfo. etc.

alecdavis (license 585)
Tested by: alecdavis

Review: https://reviewboard.asterisk.org/r/1409/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@334619 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 334617 via svnmerge from
Alec L Davis [Wed, 7 Sep 2011 07:48:25 +0000 (07:48 +0000)]
Merged revisions 334617 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

................
  r334617 | alecdavis | 2011-09-07 19:45:00 +1200 (Wed, 07 Sep 2011) | 17 lines

  Merged revisions 334616 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.8

  ........
    r334616 | alecdavis | 2011-09-07 19:33:39 +1200 (Wed, 07 Sep 2011) | 10 lines

    Prevent segfault if call arrives before Asterisk is fully booted.

    Prevent ast_pbx_start and ast_run_start from starting a new thread unless asterisk
    is fully booted.

    alecdavis (license 585)
    Tested by: alecdavis

    Review: https://reviewboard.asterisk.org/r/1407/
  ........
................

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@334618 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoImplement the '!' negation element to negate codecs directly in the allow keyword.
Tilghman Lesher [Wed, 7 Sep 2011 00:54:36 +0000 (00:54 +0000)]
Implement the '!' negation element to negate codecs directly in the allow keyword.

This permits the list of codecs to be specified in one configuration line,
instead of two or more, generally with the aim of either allowing all codecs
with the exception of a few or disallowing most but permitting a few.

Review: https://reviewboard.asterisk.org/r/1411/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@334574 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 334455 via svnmerge from
Gregory Nietsky [Tue, 6 Sep 2011 16:15:50 +0000 (16:15 +0000)]
Merged revisions 334455 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

................
  r334455 | irroot | 2011-09-06 15:58:56 +0200 (Tue, 06 Sep 2011) | 18 lines

  Merged revisions 334453 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.8

  ........
    r334453 | irroot | 2011-09-06 15:48:03 +0200 (Tue, 06 Sep 2011) | 13 lines

    Make SQL query in app_voicemail.c portable LIMIT is not portable.

    Regression from r312212

    (closes issue ASTERISK-18255)
    Reported by: Leif Madsen
    Tested by: Leif Madsen

    Review: https://reviewboard.asterisk.org/r/1415/
  ........
................

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@334519 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 334514 via svnmerge from
Paul Belanger [Tue, 6 Sep 2011 16:08:10 +0000 (16:08 +0000)]
Merged revisions 334514 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

........
  r334514 | pabelanger | 2011-09-06 11:47:59 -0400 (Tue, 06 Sep 2011) | 6 lines

  authdebug is now disabled by default

  To enable this functionaility again set authdebug = yes in iax.conf

  Review: https://reviewboard.asterisk.org/r/1414/
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@334517 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoRevert r334472 due to properties going missing
Gregory Nietsky [Tue, 6 Sep 2011 16:04:02 +0000 (16:04 +0000)]
Revert r334472 due to properties going missing

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@334515 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 334455 via svnmerge from
Gregory Nietsky [Tue, 6 Sep 2011 14:24:07 +0000 (14:24 +0000)]
Merged revisions 334455 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

................
  r334455 | irroot | 2011-09-06 15:58:56 +0200 (Tue, 06 Sep 2011) | 18 lines

  Merged revisions 334453 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.8

  ........
    r334453 | irroot | 2011-09-06 15:48:03 +0200 (Tue, 06 Sep 2011) | 13 lines

    Make SQL query in app_voicemail.c portable LIMIT is not portable.

    Regression from r312212

    (closes issue ASTERISK-18255)
    Reported by: Leif Madsen
    Tested by: Leif Madsen

    Review: https://reviewboard.asterisk.org/r/1415/
  ........
................

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@334472 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 334357 via svnmerge from
Richard Mudgett [Fri, 2 Sep 2011 21:09:31 +0000 (21:09 +0000)]
Merged revisions 334357 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

................
  r334357 | rmudgett | 2011-09-02 16:08:16 -0500 (Fri, 02 Sep 2011) | 26 lines

  Merged revisions 334355 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.8

  ........
    r334355 | rmudgett | 2011-09-02 15:59:49 -0500 (Fri, 02 Sep 2011) | 19 lines

    MusicOnHold has extra unref which may lead to memory corruption and crash.

    The problem happens when a call is disconnected and you had started a MOH
    class that does not use the files mode.  If you define REF_DEBUG and
    recreate the problem, it will announce itself with the following warning:
    Attempt to unref mohclass 0xb70722e0 (default) when only 1 ref remained,
    and class is still in a container!

    * Fixed moh_alloc() and moh_release() functions not handling the
    state->class reference consistently.

    (closes issue ASTERISK-18346)
    Reported by: Mark Murawski
    Patches:
          jira_asterisk_18346_v1.8.patch (license #5621) patch uploaded by rmudgett
    Tested by: rmudgett, Mark Murawski

    Review: https://reviewboard.asterisk.org/r/1404/
  ........
................

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@334358 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 334297 via svnmerge from
Richard Mudgett [Fri, 2 Sep 2011 17:19:17 +0000 (17:19 +0000)]
Merged revisions 334297 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

................
  r334297 | rmudgett | 2011-09-02 12:15:08 -0500 (Fri, 02 Sep 2011) | 46 lines

  Merged revisions 334296 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.8

  ........
    r334296 | rmudgett | 2011-09-02 12:10:58 -0500 (Fri, 02 Sep 2011) | 39 lines

    Fix potential memory allocation failure crashes in config.c.

    * Added required checks to the returned memory allocation pointers to
    prevent crashes.

    * Made ast_include_rename() create a replacement ast_variable list node if
    the new filename is longer than the available space.  Fixes potential
    crash and memory leak.

    * Factored out ast_variable_move() from ast_variable_update() so
    ast_include_rename() can also use it when creating a replacement
    ast_variable list node.

    * Made the filename stuffed at the end of the struct a minimum allocated
    size in ast_variable_new() in case ast_include_rename() changes the stored
    filename.

    * Constify struct char pointers pointing to strings stuffed at the end of
    the struct for: ast_variable, cache_file_mtime, and ast_config_map.

    * Factored out cfmtime_new() to remove inlined code and allow some struct
    pointers to become const.

    * Removed the list lock from struct cache_file_mtime that was never used.

    * Added doxygen comments to several structure elements and better
    documented what strings are stuffed at the struct end char array.

    * Reworked ast_config_text_file_save() and set_fn() to handle allocation
    failure of the include file scratch pad object tracking blank lines.

    * Made ast_config_text_file_save() fn[] declared with PATH_MAX to ensure
    it is long enough for any filename with path.  Also reduced the number of
    container fileset buckets from a rediculus 180,000 to 1023.

    JIRA AST-618

    Review: https://reviewboard.asterisk.org/r/1378/
  ........
................

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@334304 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 334235 via svnmerge from
Tilghman Lesher [Thu, 1 Sep 2011 17:41:09 +0000 (17:41 +0000)]
Merged revisions 334235 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

................
  r334235 | tilghman | 2011-09-01 12:39:32 -0500 (Thu, 01 Sep 2011) | 9 lines

  Merged revisions 334234 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.8

  ........
    r334234 | tilghman | 2011-09-01 12:38:33 -0500 (Thu, 01 Sep 2011) | 2 lines

    Remove 1.6 compatibility documentation from 1.8, as it no longer applies.
  ........
................

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@334236 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 334230 via svnmerge from
Tilghman Lesher [Thu, 1 Sep 2011 17:31:34 +0000 (17:31 +0000)]
Merged revisions 334230 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

................
  r334230 | tilghman | 2011-09-01 12:30:19 -0500 (Thu, 01 Sep 2011) | 25 lines

  Merged revisions 334229 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.8

  ........
    r334229 | tilghman | 2011-09-01 12:28:09 -0500 (Thu, 01 Sep 2011) | 18 lines

    Create a local alias for ast_odbc_clear_cache.

    As a function pointer, the reference has to be resolved at load time
    irrespective of the RTLD_LAZY flag.  Creating a local alias solves
    this problem, because the structure is initialized with that local
    function pointer, while the actual function can remain lazily linked
    until runtime.

    The reason why this is important is because we lazily load function
    references during the module loading process, in order to obtain
    priority values for each module, ensuring that modules are loaded in
    the correct order.  Previous to this change, when this module was
    initially loaded, the module loader would emit a symbol resolution
    error, because of the above requirement.

    Closes ASTERISK-18399 (reported by Mikael Carlsson, fix suggested by
    Walter Doekes, patch by me)
  ........
................

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@334231 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 334157 via svnmerge from
Matthew Nicholson [Wed, 31 Aug 2011 18:54:33 +0000 (18:54 +0000)]
Merged revisions 334157 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

................
  r334157 | mnicholson | 2011-08-31 13:53:40 -0500 (Wed, 31 Aug 2011) | 11 lines

  Merged revisions 334156 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.8

  ........
    r334156 | mnicholson | 2011-08-31 13:50:33 -0500 (Wed, 31 Aug 2011) | 4 lines

    Disable T.38 when we get a invite with image media port set to 0

    ASTERISK-17678
  ........
................

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@334158 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoOptimize chan_sip.c check_rtp_timeout() function.
Richard Mudgett [Wed, 31 Aug 2011 18:11:23 +0000 (18:11 +0000)]
Optimize chan_sip.c check_rtp_timeout() function.

* Make check_rtp_timeout() remember the values returned by
ast_rtp_instance_get_timeout(), ast_rtp_instance_get_hold_timeout(), and
ast_rtp_instance_get_keepalive() instead of repeatedly calling them.

(closes issue ASTERISK-18319)
Reported by: Rob Gagnon
Patches:
      issue-18319-trunk-r333066.diff (License #6159) patch uploaded by Rob Gagnon

Review: https://reviewboard.asterisk.org/r/1377/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@334115 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 334064 via svnmerge from
Matthew Nicholson [Wed, 31 Aug 2011 16:31:30 +0000 (16:31 +0000)]
Merged revisions 334064 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

........
  r334064 | mnicholson | 2011-08-31 11:31:00 -0500 (Wed, 31 Aug 2011) | 4 lines

  only alter the gateway_timeout when attching the gateway to a channel

  ASTERISK-18219
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@334067 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 334013 via svnmerge from
Richard Mudgett [Wed, 31 Aug 2011 16:02:11 +0000 (16:02 +0000)]
Merged revisions 334013 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

................
  r334013 | rmudgett | 2011-08-31 11:00:49 -0500 (Wed, 31 Aug 2011) | 30 lines

  Merged revisions 334012 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.8

  ........
    r334012 | rmudgett | 2011-08-31 10:57:12 -0500 (Wed, 31 Aug 2011) | 23 lines

    No DAHDI channel available for conference, user introduction disabled.

    The following error will consistently occur when trying to dial into a
    MeetMe conference when the server does not have DAHDI hardware installed:

    app_meetme.c: No DAHDI channel available for conference, user introduction
    disabled (is chan_dahdi loaded?)

    While chan_dahdi is loaded correctly during compilation and install of
    Asterisk/Dahdi, including associated modules, etc., a chan_dahdi.conf
    configuration file in /etc/asterisk is not created by FreePBX if hardware
    does not exist, causing MeetMe to be unable to open a DAHDI pseudo
    channel.

    * Allow chan_dahdi to create a pseudo channel when there is no
    chan_dahdi.conf file to load.

    (closes issue ASTERISK-17398)
    Reported by: Preston Edwards
    Patches:
          jira_asterisk_17398_v1.8.patch (license #5621) patch uploaded by rmudgett
    Tested by: rmudgett
  ........
................

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@334014 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 334010 via svnmerge from
Richard Mudgett [Wed, 31 Aug 2011 15:25:35 +0000 (15:25 +0000)]
Merged revisions 334010 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

................
  r334010 | rmudgett | 2011-08-31 10:23:11 -0500 (Wed, 31 Aug 2011) | 50 lines

  Merged revisions 334009 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.8

  ........
    r334009 | rmudgett | 2011-08-31 10:20:31 -0500 (Wed, 31 Aug 2011) | 43 lines

    Call pickup race leaves orphaned channels or crashes.

    Multiple users attempting to pickup a call that has been forked to
    multiple extensions either crashes or fails a masquerade with a "bad
    things may happen" message.

    This is the scenario that is causing all the grief:
    1) Pickup target is selected
    2) target is marked as being picked up in ast_do_pickup()
    3) target is unlocked by ast_do_pickup()
    4) app dial or queue gets a chance to hang up losing calls and calls
    ast_hangup() on target
    5) SINCE A MASQUERADE HAS NOT BEEN SETUP YET BY ast_do_pickup() with
    ast_channel_masquerade(), ast_hangup() completes successfully and the
    channel is no longer in the channels container.
    6) ast_do_pickup() then calls ast_channel_masquerade() to schedule the
    masquerade on the dead channel.
    7) ast_do_pickup() then calls ast_do_masquerade() on the dead channel
    8) bad things happen while doing the masquerade and in the process
    ast_do_masquerade() puts the dead channel back into the channels container
    9) The "orphaned" channel is visible in the channels list if a crash does
    not happen.

    This patch does the following:

    * Made ast_hangup() set AST_FLAG_ZOMBIE on a successfully hung-up channel
    and not release the channel lock until that has happened.

    * Made __ast_channel_masquerade() not setup a masquerade if either channel
    has AST_FLAG_ZOMBIE set.

    * Fix chan_agent misuse of AST_FLAG_ZOMBIE since it would no longer work.

    (closes issue ASTERISK-18222)
    Reported by: Alec Davis
    Tested by: rmudgett, Alec Davis, irroot, Karsten Wemheuer

    (closes issue ASTERISK-18273)
    Reported by: Karsten Wemheuer
    Tested by: rmudgett, Alec Davis, irroot, Karsten Wemheuer

    Review: https://reviewboard.asterisk.org/r/1400/
  ........
................

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@334011 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 334007 via svnmerge from
Kinsey Moore [Wed, 31 Aug 2011 15:20:21 +0000 (15:20 +0000)]
Merged revisions 334007 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

................
  r334007 | kmoore | 2011-08-31 10:19:30 -0500 (Wed, 31 Aug 2011) | 14 lines

  Merged revisions 334006 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.8

  ........
    r334006 | kmoore | 2011-08-31 10:18:37 -0500 (Wed, 31 Aug 2011) | 7 lines

    Correct an AMI protocol violation with SIPshowpeer

    The response of SIPshowpeer ends with "\r\n\r\n". Since other commands are
    ended by using \r\n this confuses any interfacing script.

    (closes issue ASTERISK-17486)
  ........
................

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@334008 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 333961-333962 via svnmerge from
Alexandr Anikin [Tue, 30 Aug 2011 22:16:13 +0000 (22:16 +0000)]
Merged revisions 333961-333962 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

................
  r333961 | may | 2011-08-31 01:21:53 +0400 (Wed, 31 Aug 2011) | 11 lines

  Merged revisions 333947 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.8

  ........
    r333947 | may | 2011-08-31 01:16:30 +0400 (Wed, 31 Aug 2011) | 5 lines

    cleanups in ACF/ARJ GK replies processing
    fixed long (24 sec) pause if acf/arj proccessed
    before ast_cond_wait called to wait this
  ........
................
  r333962 | may | 2011-08-31 01:53:42 +0400 (Wed, 31 Aug 2011) | 3 lines

  security fix. really drop call if signalling addr is not same as socket
  addr
................

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@333963 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 333895 via svnmerge from
Matthew Nicholson [Tue, 30 Aug 2011 14:03:02 +0000 (14:03 +0000)]
Merged revisions 333895 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

........
  r333895 | mnicholson | 2011-08-30 09:01:31 -0500 (Tue, 30 Aug 2011) | 6 lines

  Replaced FAXOPT(gwtimeout) with a second parameter to FAXOPT(gateway).

  Patch by: irroot
  Review: https://reviewboard.asterisk.org/r/1385/
  ASTERISK-18219
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@333896 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 333837 via svnmerge from
Terry Wilson [Mon, 29 Aug 2011 21:43:33 +0000 (21:43 +0000)]
Merged revisions 333837 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

................
  r333837 | twilson | 2011-08-29 16:41:13 -0500 (Mon, 29 Aug 2011) | 22 lines

  Merged revisions 333836 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.8

  ........
    r333836 | twilson | 2011-08-29 16:38:31 -0500 (Mon, 29 Aug 2011) | 15 lines

    Refresh peer address if DNS unavailable at peer creation

    If Asterisk starts and no DNS is available, outbound registrations will fail
    indefinitely. This patch copies the address from the sip_registry struct, which
    will be updated, to the peer->addr when necessary.

    If dnsmgr is enabled, the registration fails without the patch because even
    though the address on the registry is updated via dnsmgr, the address is just
    copied on the first try. Since we use ast_sockaddr_copy, dnsmgr can't update
    the address that is copied to the sip_pvt or peers.

    Closes issue ASTERISK-18000

    Review: https://reviewboard.asterisk.org/r/1335/
  ........
................

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@333838 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 333786 via svnmerge from
Richard Mudgett [Mon, 29 Aug 2011 21:17:51 +0000 (21:17 +0000)]
Merged revisions 333786 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

................
  r333786 | rmudgett | 2011-08-29 16:12:29 -0500 (Mon, 29 Aug 2011) | 13 lines

  Merged revisions 333784-333785 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.8

  ........
    r333784 | rmudgett | 2011-08-29 16:05:43 -0500 (Mon, 29 Aug 2011) | 2 lines

    Fix deadlock potential of chan_mobile.c:mbl_ast_hangup().
  ........
    r333785 | rmudgett | 2011-08-29 16:06:16 -0500 (Mon, 29 Aug 2011) | 1 line

    Add some do not hold locks notes to channel.h
  ........
................

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@333789 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 333716 via svnmerge from
Matthew Nicholson [Mon, 29 Aug 2011 18:28:02 +0000 (18:28 +0000)]
Merged revisions 333716 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

........
  r333716 | mnicholson | 2011-08-29 13:22:58 -0500 (Mon, 29 Aug 2011) | 5 lines

  It is possible for the gateway to be attached when the channel is still
  negotiating T.38. This change handles that case.

  ASTERISK-18329
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@333736 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 333681 via svnmerge from
Terry Wilson [Mon, 29 Aug 2011 17:31:40 +0000 (17:31 +0000)]
Merged revisions 333681 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

........
  r333681 | twilson | 2011-08-29 12:28:59 -0500 (Mon, 29 Aug 2011) | 7 lines

  Use realtime text when it is negotiated

  This patch make use of wirte_text() realtime text instead of
  send_text() if T.140 is in native formats. ASTERISK-17937

  Review: https://reviewboard.asterisk.org/r/1356/
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@333689 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 333631 via svnmerge from
Matthew Jordan [Mon, 29 Aug 2011 17:14:26 +0000 (17:14 +0000)]
Merged revisions 333631 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

................
  r333631 | mjordan | 2011-08-29 12:12:55 -0500 (Mon, 29 Aug 2011) | 9 lines

  Merged revisions 333630 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.8

  ........
    r333630 | mjordan | 2011-08-29 12:11:15 -0500 (Mon, 29 Aug 2011) | 1 line

    Fixed improperly formatted TestEvent AMI message in app_voicemail
  ........
................

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@333632 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 333570 via svnmerge from
Jonathan Rose [Mon, 29 Aug 2011 15:58:24 +0000 (15:58 +0000)]
Merged revisions 333570 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

................
  r333570 | jrose | 2011-08-29 10:56:56 -0500 (Mon, 29 Aug 2011) | 11 lines

  Merged revisions 333569 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.8

  ........
    r333569 | jrose | 2011-08-29 10:55:34 -0500 (Mon, 29 Aug 2011) | 4 lines

    Accidental use of variable client->status instead of client->state in from ASTERISK-18078

    (issue ASTERISK-18078)
  ........
................

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@333571 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agochan_vpb: remove unused variables (gcc4.6)
Tzafrir Cohen [Sun, 28 Aug 2011 09:57:47 +0000 (09:57 +0000)]
chan_vpb: remove unused variables (gcc4.6)

GCC 4.6 detects variables that get assined to, but never used later.
Also removes some remmed-out lines that become invalid.

(closes issue ASTERISK-18336)
Signed-off-by: Tzafrir Cohen (License #5035) <tzafrir.cohen@xorcom.com>,

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@333509 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 333410 via svnmerge from
Jonathan Rose [Fri, 26 Aug 2011 16:38:37 +0000 (16:38 +0000)]
Merged revisions 333410 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

................
  r333410 | jrose | 2011-08-26 11:28:03 -0500 (Fri, 26 Aug 2011) | 19 lines

  Merged revisions 333378 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.8

  ........
    r333378 | jrose | 2011-08-26 11:19:07 -0500 (Fri, 26 Aug 2011) | 13 lines

    [patch] Buddies are always auto-registered when processing the roster

    Reporter said autoregister flag was ignored for registering 'buddies' which
    had a subscription to us. Verified that this was the case and observed how
    the patch addressed this and made sure it didn't break anything.

    (closes issue ASTERISK-14233)
    Reported by: Simon Arlott
    Patches:
          asterisk-0015229.patch (license #5756) patch uploaded by Simon Arlott
    Tested by: Jonathan Rose
  ........
................

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@333428 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 333370 via svnmerge from
Matthew Jordan [Fri, 26 Aug 2011 16:12:13 +0000 (16:12 +0000)]
Merged revisions 333370 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

................
  r333370 | mjordan | 2011-08-26 10:58:37 -0500 (Fri, 26 Aug 2011) | 26 lines

  Merged revisions 333339 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.8

  ........
    r333339 | mjordan | 2011-08-26 08:36:36 -0500 (Fri, 26 Aug 2011) | 20 lines

    Bug fixes for voicemail user emailsubject / emailbody.

    This code change fixes a few issues with the voicemail user override of
    emailbody and emailsubject, including escaping the strings, potential memory
    leaks, and not overriding the voicemail defaults.  Revision 325877 fixed this
    for ASTERISK-16795, but did not fix it for ASTERISK-16781.  A subsequent
    check-in prevented 325877 from being applied to 10.  This check-in resolves
    both issues, and applies the changes to 1.8, 10, and trunk.

    (closes issue ASTERISK-16781)
    Reported by: Sebastien Couture
    Tested by: mjordan

    (closes issue ASTERISK-16795)
    Reported by: mdeneen
    Tested by: mjordan

    Review: https://reviewboard.asterisk.org/r/1374
  ........
................

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@333371 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 333266 via svnmerge from
Jonathan Rose [Thu, 25 Aug 2011 19:13:23 +0000 (19:13 +0000)]
Merged revisions 333266 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

................
  r333266 | jrose | 2011-08-25 14:00:05 -0500 (Thu, 25 Aug 2011) | 20 lines

  Merged revisions 333265 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.8

  ........
    r333265 | jrose | 2011-08-25 13:47:42 -0500 (Thu, 25 Aug 2011) | 14 lines

    Segfault when publishing device states via XMPP and not connected

    When using publishing device state with res_jabber, Asterisk will attempt
    to send a device state using the unconnected client using iks_send_raw
    and crash. This patch checks the validity of the connection before
    attempting to send the device state.

    (closes issue ASTERISK-18078)
    Reported by: Michael L. Young
    Patches:
          res_jabber-segfault-pubsub-not-connected2.patch (license #5026) patch uploaded by Michael L. Young
    Tested by: Jonathan Rose
  ........
................

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@333276 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 333268 via svnmerge from
Jason Parker [Thu, 25 Aug 2011 19:01:39 +0000 (19:01 +0000)]
Merged revisions 333268 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

................
  r333268 | qwell | 2011-08-25 14:01:18 -0500 (Thu, 25 Aug 2011) | 9 lines

  Merged revisions 333267 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.8

  ........
    r333267 | qwell | 2011-08-25 14:00:55 -0500 (Thu, 25 Aug 2011) | 2 lines

    Fix for DESTDIR spaces patch.
  ........
................

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9 years agoMerged revisions 333203 via svnmerge from
Jason Parker [Thu, 25 Aug 2011 15:30:46 +0000 (15:30 +0000)]
Merged revisions 333203 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

................
  r333203 | qwell | 2011-08-25 10:29:56 -0500 (Thu, 25 Aug 2011) | 15 lines

  Merged revisions 333201 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.8

  ........
    r333201 | qwell | 2011-08-25 10:27:06 -0500 (Thu, 25 Aug 2011) | 8 lines

    Fix installation into directories containing spaces.

    This also vastly simplifies the logic in sounds/Makefile

    (Closes issue ASTERISK-18290)
    Reported by: Paul Belanger
    Review: https://reviewboard.asterisk.org/r/1379/
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@333204 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoFix typo from r333070
Jason Parker [Wed, 24 Aug 2011 19:27:15 +0000 (19:27 +0000)]
Fix typo from r333070

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@333159 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 333115 via svnmerge from
Matthew Nicholson [Wed, 24 Aug 2011 16:52:56 +0000 (16:52 +0000)]
Merged revisions 333115 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

........
  r333115 | mnicholson | 2011-08-24 11:51:42 -0500 (Wed, 24 Aug 2011) | 4 lines

  Changed the "timeout" option to "gwtimeout".

  ASTERISK-18219
........

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9 years agoFormatting changes - Removing some red white space and adding some curly brackets.
Olle Johansson [Wed, 24 Aug 2011 09:17:48 +0000 (09:17 +0000)]
Formatting changes - Removing some red white space and adding some curly brackets.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@333075 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoAdd documentation for new manager event in chan_local
Olle Johansson [Wed, 24 Aug 2011 09:12:23 +0000 (09:12 +0000)]
Add documentation for new manager event in chan_local

AST-17623

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@333071 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoAdd manager event for local channel semi-bridge
Olle Johansson [Wed, 24 Aug 2011 09:09:53 +0000 (09:09 +0000)]
Add manager event for local channel semi-bridge

(issue AST-17623)

Review: https://reviewboard.asterisk.org/r/1154

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@333070 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 333011 via svnmerge from
Richard Mudgett [Tue, 23 Aug 2011 18:17:52 +0000 (18:17 +0000)]
Merged revisions 333011 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

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  r333011 | rmudgett | 2011-08-23 13:15:49 -0500 (Tue, 23 Aug 2011) | 19 lines

  Merged revisions 333010 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.8

  ........
    r333010 | rmudgett | 2011-08-23 13:14:01 -0500 (Tue, 23 Aug 2011) | 12 lines

    Memory Leak in app_queue

    The patch that was committed in the 1.6.x versions of Asterisk for
    ASTERISK-15862 actually fixed two issues.  One was not applicable to 1.8
    but the other is.  queue_leak.patch fixes the portion applicable to 1.8.

    (closes issue ASTERISK-18265)
    Reported by: Fred Schroeder
    Patches:
          queue_leak.patch (license #5049) patch uploaded by mmichelson
    Tested by: Thomas Arimont
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@333014 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 332940 via svnmerge from
Richard Mudgett [Mon, 22 Aug 2011 21:25:11 +0000 (21:25 +0000)]
Merged revisions 332940 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

................
  r332940 | rmudgett | 2011-08-22 16:23:40 -0500 (Mon, 22 Aug 2011) | 14 lines

  Merged revisions 332939 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.8

  ........
    r332939 | rmudgett | 2011-08-22 16:22:24 -0500 (Mon, 22 Aug 2011) | 7 lines

    Minor code optimizations.

    * Simplify ast_category_browse() logic for easier understanding.

    * Remove dead code in ast_variable_delete() and simplify some of its
    logic.
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@332941 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 332875,332878 via svnmerge from
Richard Mudgett [Mon, 22 Aug 2011 20:01:30 +0000 (20:01 +0000)]
Merged revisions 332875,332878 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

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  r332875 | rmudgett | 2011-08-22 14:41:03 -0500 (Mon, 22 Aug 2011) | 1 line

  Fix merge property.
................
  r332878 | rmudgett | 2011-08-22 14:46:25 -0500 (Mon, 22 Aug 2011) | 25 lines

  Merged revisions 332874 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.8

  ........
    r332874 | rmudgett | 2011-08-22 14:32:19 -0500 (Mon, 22 Aug 2011) | 18 lines

    Reference leaks in app_queue.

    * Fixed load_realtime_queue() leaking a queue reference when it overwrites
    q when processing a realtime queue.
    (issue ASTERISK-18265)

    * Make join_queue() unreference the queue returned by
    load_realtime_queue() when it is done with the pointer.  The
    load_realtime_queue() returns a reference to the just loaded realtime
    queue.

    * Fixed queues container reference leak in queues_data_provider_get().

    * queue_unref() should not return q that was just unreferenced.

    * Made logic in __queues_show() and queues_data_provider_get() when
    calling load_realtime_queue() easier to understand.
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@332881 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 332877 via svnmerge from
Paul Belanger [Mon, 22 Aug 2011 19:56:00 +0000 (19:56 +0000)]
Merged revisions 332877 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

................
  r332877 | pabelanger | 2011-08-22 15:43:33 -0400 (Mon, 22 Aug 2011) | 13 lines

  Merged revisions 332876 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.8

  ........
    r332876 | pabelanger | 2011-08-22 15:41:24 -0400 (Mon, 22 Aug 2011) | 6 lines

    Revert previous commit

    It seems google is still making changes to the protocol.

    (issue ASTERISK-18301)
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@332880 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoFix merge 10 branch merge properties.
Richard Mudgett [Mon, 22 Aug 2011 19:52:30 +0000 (19:52 +0000)]
Fix merge 10 branch merge properties.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@332879 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 332817 via svnmerge from
Matthew Jordan [Mon, 22 Aug 2011 19:19:44 +0000 (19:19 +0000)]
Merged revisions 332817 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r332817 | mjordan | 2011-08-22 13:15:51 -0500 (Mon, 22 Aug 2011) | 4 lines

  Review: https://reviewboard.asterisk.org/r/1364/

  This update adds a new AMI event, TestEvent, which is enabled when the TEST_FRAMEWORK compiler flag is defined.  It also adds initial usage of this event to app_voicemail.  The TestEvent AMI event is used extensively by the voicemail tests in the Asterisk Test Suite.
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@332844 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 332830 via svnmerge from
Richard Mudgett [Mon, 22 Aug 2011 18:33:27 +0000 (18:33 +0000)]
Merged revisions 332830 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

................
  r332830 | rmudgett | 2011-08-22 13:32:09 -0500 (Mon, 22 Aug 2011) | 15 lines

  Merged revisions 332816 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.8

  ........
    r332816 | rmudgett | 2011-08-22 13:14:59 -0500 (Mon, 22 Aug 2011) | 8 lines

    Memory leaks in realtime_multi_xxx() when database access returns error.

    * Fix realtime_multi_pgsql() configuration memory leak when the database
    access returns an error.

    * Fix realtime_multi_odbc() configuration category use after free when the
    database access returns an error.
  ........
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