asterisk/asterisk.git
8 years agoMerged revisions 338719 via svnmerge from
Jonathan Rose [Fri, 30 Sep 2011 18:59:01 +0000 (18:59 +0000)]
Merged revisions 338719 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

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  r338719 | jrose | 2011-09-30 13:55:27 -0500 (Fri, 30 Sep 2011) | 9 lines

  Merged revisions 338718 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.8

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    r338718 | jrose | 2011-09-30 13:54:30 -0500 (Fri, 30 Sep 2011) | 1 line

    Adds documentation for QueueMemberStatus event generation
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@338720 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoFix formatting of AMI header for SIP show peer.
Richard Mudgett [Fri, 30 Sep 2011 16:40:14 +0000 (16:40 +0000)]
Fix formatting of AMI header for SIP show peer.

ASTERISK-17486 exposed the problem for AMI parsers.

(closes issue ASTERISK-18649)
Reported by: Jacek Konieczny
Patches:
      asterisk-sipshowpeer_response_end.patch (license #6298) patch uploaded by Jacek Konieczny
........

Merged revisions 338663 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 338664 from http://svn.asterisk.org/svn/asterisk/branches/10

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@338665 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoPreserve DTMF length in main/features.c
Olle Johansson [Fri, 30 Sep 2011 13:21:17 +0000 (13:21 +0000)]
Preserve DTMF length in main/features.c

Review: https://reviewboard.asterisk.org/r/1463/

A small part of much larger work with DTMF duration in Asterisk,
funded  by IPvision AS in Denmark.

Thanks to irroot for the review!

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@338623 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoMerged revisions 338556 via svnmerge from
Paul Belanger [Thu, 29 Sep 2011 21:16:07 +0000 (21:16 +0000)]
Merged revisions 338556 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

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  r338556 | pabelanger | 2011-09-29 17:14:34 -0400 (Thu, 29 Sep 2011) | 9 lines

  Merged revisions 338555 via svnmerge from
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    r338555 | pabelanger | 2011-09-29 17:12:21 -0400 (Thu, 29 Sep 2011) | 2 lines

    Test modules should depend on the TEST_FRAMEWORK flag
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@338557 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoMerged revisions 338552 via svnmerge from
Jason Parker [Thu, 29 Sep 2011 20:55:15 +0000 (20:55 +0000)]
Merged revisions 338552 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

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  r338552 | qwell | 2011-09-29 15:54:55 -0500 (Thu, 29 Sep 2011) | 9 lines

  Merged revisions 338551 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.8

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    r338551 | qwell | 2011-09-29 15:54:13 -0500 (Thu, 29 Sep 2011) | 1 line

    Test modules have a support level of core.
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@338553 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoBlocked revisions 338493 via svnmerge
Leif Madsen [Thu, 29 Sep 2011 18:33:48 +0000 (18:33 +0000)]
Blocked revisions 338493 via svnmerge

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  r338493 | lmadsen | 2011-09-29 13:32:28 -0500 (Thu, 29 Sep 2011) | 14 lines

  Merged revisions 338492 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.8

  ........
    r338492 | lmadsen | 2011-09-29 13:31:33 -0500 (Thu, 29 Sep 2011) | 6 lines

    Update documentation for SIP_HEADER.

    The SIP_HEADER function only works on the the initial SIP INVITE. The documentation was updated
    in trunk, but not in 1.8 or 10, so I'm making them match.

    (Closes issue ASTERISK-18640)
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@338494 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoMerged revisions 338417 via svnmerge from
Gregory Nietsky [Thu, 29 Sep 2011 12:22:43 +0000 (12:22 +0000)]
Merged revisions 338417 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

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  r338417 | irroot | 2011-09-29 14:16:42 +0200 (Thu, 29 Sep 2011) | 19 lines

  Merged revisions 338416 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.8

  ........
    r338416 | irroot | 2011-09-29 14:13:05 +0200 (Thu, 29 Sep 2011) | 12 lines

    The rtptimeout setting is ignored on a per peer basis.

    Not only is the rtptimeout ignored in some cases but
    rtpkeepalive and rtpholdtimeout is affected.

    this commit also removes rtptimeout/rtpholdtimeout on
    text rtp.

    (closes issue ASTERISK-18559)

    Review: https://reviewboard.asterisk.org/r/1452
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@338435 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoAdd CLI command "cdr show pgsql status" based on "cdr mysql status"
Olle Johansson [Thu, 29 Sep 2011 12:03:23 +0000 (12:03 +0000)]
Add CLI command "cdr show pgsql status" based on "cdr mysql status"

Review: https://reviewboard.asterisk.org/r/923/

Thanks all for the code reviews and feedback.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@338415 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoJust formatting.
Olle Johansson [Thu, 29 Sep 2011 09:32:34 +0000 (09:32 +0000)]
Just formatting.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@338377 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoMerged revisions 338323 via svnmerge from
Richard Mudgett [Wed, 28 Sep 2011 22:38:00 +0000 (22:38 +0000)]
Merged revisions 338323 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

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  r338323 | rmudgett | 2011-09-28 17:36:57 -0500 (Wed, 28 Sep 2011) | 12 lines

  Merged revisions 338322 via svnmerge from
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  ........
    r338322 | rmudgett | 2011-09-28 17:35:52 -0500 (Wed, 28 Sep 2011) | 5 lines

    Make duplicate call ptr warning message more helpful.

    * Adds the value of the call ptr to the duplicate call ptr message to help
    trace why there is a duplicate call ptr.
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@338324 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoMerged revisions 338253 via svnmerge from
Richard Mudgett [Wed, 28 Sep 2011 21:30:14 +0000 (21:30 +0000)]
Merged revisions 338253 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

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  r338253 | rmudgett | 2011-09-28 16:22:05 -0500 (Wed, 28 Sep 2011) | 14 lines

  Merged revisions 338235 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.8

  ........
    r338235 | rmudgett | 2011-09-28 16:17:45 -0500 (Wed, 28 Sep 2011) | 7 lines

    Fix inconsistency in LOG_VERBOSE/AST_LOG_VERBOSE declaration.

    (closes issue ASTERISK-17973)
    Reported by: Luke H
    Patches:
          logger_h.patch (license #6278) patch uploaded by Luke H
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@338284 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoMerged revisions 338228 via svnmerge from
Jason Parker [Wed, 28 Sep 2011 20:55:42 +0000 (20:55 +0000)]
Merged revisions 338228 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

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  r338228 | qwell | 2011-09-28 15:54:35 -0500 (Wed, 28 Sep 2011) | 9 lines

  Merged revisions 338227 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.8

  ........
    r338227 | qwell | 2011-09-28 15:52:47 -0500 (Wed, 28 Sep 2011) | 1 line

    Add support levels to non-module sections of menuselect (cflags, utils, etc).
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@338229 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoMerged revisions 338225 via svnmerge from
Richard Mudgett [Wed, 28 Sep 2011 20:28:14 +0000 (20:28 +0000)]
Merged revisions 338225 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

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  r338225 | rmudgett | 2011-09-28 15:26:39 -0500 (Wed, 28 Sep 2011) | 12 lines

  Merged revisions 338224 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.8

  ........
    r338224 | rmudgett | 2011-09-28 15:24:41 -0500 (Wed, 28 Sep 2011) | 5 lines

    Fix chan_dahd compiling with gcc 4.6 when PRI and SS7 not present.

    (closes issue ASTERISK-18357)
    Reported by: Matthew Nicholson
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@338226 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoUpdate CHANGES to reflect autopausebusy not being in Asterisk 10
Terry Wilson [Wed, 28 Sep 2011 17:00:35 +0000 (17:00 +0000)]
Update CHANGES to reflect autopausebusy not being in Asterisk 10

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@338188 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoAdd autopausebusy and autopauseunavail queue options
Terry Wilson [Wed, 28 Sep 2011 16:59:11 +0000 (16:59 +0000)]
Add autopausebusy and autopauseunavail queue options

Make it possible to autopause on a busy or unavailable response from
a device.

(closes issue ASTERISK-16112)
Reported by: jlpedrosa
Patches:
autopausebusy.txt by twilson

Review: https://reviewboard.asterisk.org/r/1399/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@338187 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoUpdated for checking OSP Toolkit version 4.0.0.
TransNexus OSP Development [Wed, 28 Sep 2011 07:30:49 +0000 (07:30 +0000)]
Updated for checking OSP Toolkit version 4.0.0.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@338139 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoUpdated for OSP Toolkit 4.0.0.
TransNexus OSP Development [Wed, 28 Sep 2011 07:25:49 +0000 (07:25 +0000)]
Updated for OSP Toolkit 4.0.0.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@338136 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoMerged revisions 338085 via svnmerge from
Paul Belanger [Tue, 27 Sep 2011 20:15:30 +0000 (20:15 +0000)]
Merged revisions 338085 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

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  r338085 | pabelanger | 2011-09-27 16:13:14 -0400 (Tue, 27 Sep 2011) | 9 lines

  Merged revisions 338084 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.8

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    r338084 | pabelanger | 2011-09-27 16:10:13 -0400 (Tue, 27 Sep 2011) | 2 lines

    Upgrade app_macro to core
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@338086 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoWhitespace (red blobs) fixes
Olle Johansson [Tue, 27 Sep 2011 12:45:25 +0000 (12:45 +0000)]
Whitespace (red blobs) fixes

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@338042 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoMerged revisions 337974 via svnmerge from
Richard Mudgett [Mon, 26 Sep 2011 19:40:12 +0000 (19:40 +0000)]
Merged revisions 337974 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

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  r337974 | rmudgett | 2011-09-26 14:35:23 -0500 (Mon, 26 Sep 2011) | 37 lines

  Merged revisions 337973 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.8

  ........
    r337973 | rmudgett | 2011-09-26 14:30:39 -0500 (Mon, 26 Sep 2011) | 30 lines

    Fix deadlock when using dummy channels.

    Dummy channels created by ast_dummy_channel_alloc() should be destoyed by
    ast_channel_unref().  Using ast_channel_release() needlessly grabs the
    channel container lock and can cause a deadlock as a result.

    * Analyzed use of ast_dummy_channel_alloc() and made use
    ast_channel_unref() when done with the dummy channel.  (Primary reason for
    the reported deadlock.)

    * Made app_dial.c:dial_exec_full() not call ast_call() holding any channel
    locks.  Chan_local could not perform deadlock avoidance correctly.
    (Potential deadlock exposed by this issue.  Secondary reason for the
    reported deadlock since the held lock was part of the deadlock chain.)

    * Fixed some uses of ast_dummy_channel_alloc() not checking the returned
    channel pointer for failure.

    * Fixed some potential chan=NULL pointer usage in func_odbc.c.  Protected
    by testing the bogus_chan value.

    * Fixed needlessly clearing a 1024 char auto array when setting the first
    char to zero is enough in manager.c:action_getvar().

    (closes issue ASTERISK-18613)
    Reported by: Thomas Arimont
    Patches:
          jira_asterisk_18613_v1.8.patch (license #5621) patch uploaded by rmudgett
    Tested by: Thomas Arimont
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@337975 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoMerged revisions 337902 via svnmerge from
Gregory Nietsky [Fri, 23 Sep 2011 19:20:41 +0000 (19:20 +0000)]
Merged revisions 337902 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

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  r337902 | irroot | 2011-09-23 21:18:14 +0200 (Fri, 23 Sep 2011) | 10 lines

  Merged revisions 337898 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.8

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    r337898 | irroot | 2011-09-23 21:14:30 +0200 (Fri, 23 Sep 2011) | 4 lines

    Spelling fix
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@337910 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoMerged revisions 337840 via svnmerge from
Gregory Nietsky [Fri, 23 Sep 2011 09:35:32 +0000 (09:35 +0000)]
Merged revisions 337840 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

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  r337840 | irroot | 2011-09-23 10:39:22 +0200 (Fri, 23 Sep 2011) | 17 lines

  Merged revisions 337839 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.8

  ........
    r337839 | irroot | 2011-09-23 10:34:03 +0200 (Fri, 23 Sep 2011) | 11 lines

    Make sure a CDR is on the stack for call in the Queue.
    Only let update_cdr act on the last CDR in the stack.

    In some circumstances [Attended transfer to queue] a
    CDR record is not inserted for this call where it should.

    (closes issue ASTERISK-18567)

    Review: https://reviewboard.asterisk.org/r/1266
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@337855 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoMerged revisions 337775 via svnmerge from
Russell Bryant [Fri, 23 Sep 2011 00:47:18 +0000 (00:47 +0000)]
Merged revisions 337775 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

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  r337775 | russell | 2011-09-22 19:45:35 -0500 (Thu, 22 Sep 2011) | 18 lines

  Merged revisions 337774 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.8

  ........
    r337774 | russell | 2011-09-22 19:44:19 -0500 (Thu, 22 Sep 2011) | 11 lines

    Comment out entries in sample res_pktccops.conf.

    With these options enabled, they can cause Asterisk to freak out by
    SYN flooding a network and eating the CPU.  Obviously it would be good to
    fix the code so that this can't happen, but we can at least change the default
    configuration so it doesn't happen.

    This was reported downstream to the Fedora issue tracker:

        https://bugzilla.redhat.com/show_bug.cgi?id=658431
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@337776 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoMerged revisions 337721 via svnmerge from
Richard Mudgett [Thu, 22 Sep 2011 21:42:35 +0000 (21:42 +0000)]
Merged revisions 337721 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

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  r337721 | rmudgett | 2011-09-22 16:37:41 -0500 (Thu, 22 Sep 2011) | 25 lines

  Merged revisions 337720 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.8

  ........
    r337720 | rmudgett | 2011-09-22 16:29:46 -0500 (Thu, 22 Sep 2011) | 18 lines

    Made ISDN not add numbering plan prefix strings to empty numbers.

    When the Caller-ID is restricted, the expected behavior is for the
    Caller-ID to be blank.  In chan_dahdi, the national prefix is placed onto
    the Caller-ID number even if it is restricted (empty) causing the
    Caller-ID to be the national prefix rather than blank.

    This behavior was lost when sig_pri was extracted from chan_dahdi.

    * Made not add prefix strings to empty connected line, calling, and ANI
    number strings.

    (closes issue ASTERISK-18577)
    Reported by: Kris Shaw
    Patches:
          jira_asterisk_18577_v1.8.patch (license #5621) patch uploaded by rmudgett
    Tested by: Kris Shaw
  ........
................

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@337722 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoBlocked revisions 337433 via svnmerge
Gregory Nietsky [Thu, 22 Sep 2011 20:03:33 +0000 (20:03 +0000)]
Blocked revisions 337433 via svnmerge

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  r337433 | irroot | 2011-09-22 08:42:42 +0200 (Thu, 22 Sep 2011) | 12 lines

  Revert commit r337261

  This commit is for trunk not version 10

  -----
  Adds a timeout argument to app_originate

  the default is 30s this will be used if the timout supplied is invalid or
  no timeout is supplied.
  -----
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@337681 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoBlocked revisions 337640 via svnmerge
Paul Belanger [Thu, 22 Sep 2011 18:44:26 +0000 (18:44 +0000)]
Blocked revisions 337640 via svnmerge

........
  r337640 | pabelanger | 2011-09-22 14:43:35 -0400 (Thu, 22 Sep 2011) | 5 lines

  Revert previous commit

  New feature should be added into trunk, unfortunately it is too late for the
  Asterisk 10 branch.
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@337641 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoMerged revisions 337595,337597 via svnmerge from
Jonathan Rose [Thu, 22 Sep 2011 16:35:20 +0000 (16:35 +0000)]
Merged revisions 337595,337597 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

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  r337595 | jrose | 2011-09-22 10:35:50 -0500 (Thu, 22 Sep 2011) | 12 lines

  Generate Security events in chan_sip using new Security Events Framework

  Security Events Framework was added in 1.8 and support was added for AMI to generate
  events at that time. This patch adds support for chan_sip to generate security events.

  (closes issue ASTERISK-18264)
  Reported by: Michael L. Young
  Patches:
       security_events_chan_sip_v4.patch (license #5026) by Michael L. Young
  Review: https://reviewboard.asterisk.org/r/1362/
........
  r337597 | jrose | 2011-09-22 10:47:05 -0500 (Thu, 22 Sep 2011) | 10 lines

  Forgot to svn add new files to r337595

  Part of Generating security events for chan_sip

  (issue ASTERISK-18264)
  Reported by: Michael L. Young
  Patches:
      security_events_chan_sip_v4.patch (License #5026) by Michael L. Young
  Reviewboard: https://reviewboard.asterisk.org/r/1362/
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@337600 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoMerged revisions 337542 via svnmerge from
Gregory Nietsky [Thu, 22 Sep 2011 11:46:35 +0000 (11:46 +0000)]
Merged revisions 337542 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

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  r337542 | irroot | 2011-09-22 13:44:22 +0200 (Thu, 22 Sep 2011) | 14 lines

  Merged revisions 337541 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.8

  ........
    r337541 | irroot | 2011-09-22 13:39:49 +0200 (Thu, 22 Sep 2011) | 8 lines

    Add warned to ast_srtp to prevent errors on each frame from libsrtp

    The first 9 frames are not reported as some devices dont use srtp
    from first frame these are suppresed.

    the warning is then output only once every 100 frames.
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@337543 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoMerged revisions 337487 via svnmerge from
Gregory Nietsky [Thu, 22 Sep 2011 09:31:41 +0000 (09:31 +0000)]
Merged revisions 337487 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

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  r337487 | irroot | 2011-09-22 11:26:26 +0200 (Thu, 22 Sep 2011) | 16 lines

  Merged revisions 337486 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.8

  ........
    r337486 | irroot | 2011-09-22 11:22:26 +0200 (Thu, 22 Sep 2011) | 10 lines

    If IP address is used in chan_h323 host parameter of peer configuration.
    module tries to resolve IP address to IP address and fails.

    Simple fix to set family of socket this is a hangover from ipv6 changes.

    (closes issue ASTERISK-18237)
    (issue ASTERISK-17278)
    (issue ASTERISK-17500)
  ........
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8 years agoMerged revisions 337431 via svnmerge from
Gregory Nietsky [Thu, 22 Sep 2011 06:39:01 +0000 (06:39 +0000)]
Merged revisions 337431 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

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  r337431 | irroot | 2011-09-22 08:29:09 +0200 (Thu, 22 Sep 2011) | 25 lines

  Merged revisions 337430 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.8

  ........
    r337430 | irroot | 2011-09-22 08:18:33 +0200 (Thu, 22 Sep 2011) | 19 lines

    Its possible to loose audio on ast_write when the channel is not transcoded correctly.
    in the case of DAHDI the channel is hungup.

    This patch tries to "fix" the problem and make the channel compatiable and warn the user of
    this problem.

    Please note there is a underlying problem with codec negotion this does not fix the problem
    it does try to rectify it and prevent loss of service.

    Review: https://reviewboard.asterisk.org/r/1442/

    (closes issue ASTERISK-17541)
    (closes issue ASTERISK-18063)
    (issue ASTERISK-14384)
    (issue ASTERISK-17502)
    (issue ASTERISK-18325)
    (issue ASTERISK-18422)
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@337432 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoMore silly spacing changes
Tilghman Lesher [Wed, 21 Sep 2011 21:26:34 +0000 (21:26 +0000)]
More silly spacing changes

.....
Merged revisions 337353 from http://svn.asterisk.org/svn/asterisk/branches/1.8

.....
Merged revisions 337380 from http://svn.asterisk.org/svn/asterisk/branches/10

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8 years ago................
Tilghman Lesher [Wed, 21 Sep 2011 21:10:14 +0000 (21:10 +0000)]
................

........
Dumb little spacing fix.
........
Merged revisions 337344 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 337345 from http://svn.asterisk.org/svn/asterisk/branches/10

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8 years ago................
Tilghman Lesher [Wed, 21 Sep 2011 20:53:13 +0000 (20:53 +0000)]
................

........
Escape commas in keys and values, when keys and values are enumerated by commas.

Review: https://reviewboard.asterisk.org/r/1433
........
Merged revisions 337325 from https://origsvn.digium.com/svn/asterisk/branches/1.8
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Merged revisions 337342 from https://origsvn.digium.com/svn/asterisk/branches/10

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8 years agoMerged revisions 337263 via svnmerge from
Gregory Nietsky [Wed, 21 Sep 2011 11:21:49 +0000 (11:21 +0000)]
Merged revisions 337263 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

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  r337263 | irroot | 2011-09-21 13:15:48 +0200 (Wed, 21 Sep 2011) | 1 line

  Whitespace fixup from SRTP patch
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8 years agoMerged revisions 337261 via svnmerge from
Gregory Nietsky [Wed, 21 Sep 2011 10:46:09 +0000 (10:46 +0000)]
Merged revisions 337261 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

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  r337261 | irroot | 2011-09-21 12:42:06 +0200 (Wed, 21 Sep 2011) | 10 lines

  Adds a timeout argument to app_originate

  the default is 30s this will be used if the timout supplied is invalid or
  no timeout is supplied.

  Contributed by: jacco (thank you for the work)

  Review: https://reviewboard.asterisk.org/r/1310/
........

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8 years agoMerged revisions 337219 via svnmerge from
Olle Johansson [Wed, 21 Sep 2011 09:39:13 +0000 (09:39 +0000)]
Merged revisions 337219 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

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  r337219 | oej | 2011-09-21 11:32:50 +0200 (Ons, 21 Sep 2011) | 13 lines

  Make ast_pbx_run() not default to s@default if extension is not found

  Review: https://reviewboard.asterisk.org/r/1446/

  This is a bug - or architecture mistake - that has been in Asterisk for a
  very long time. It was exposed by the AMI originate action and possibly
  some other applications. Most channel drivers checks if an extension
  exists BEFORE starting a pbx on an inbound call, so most calls will
  not depend on this issue.

  Thanks everyone involved in the review and on IRC and the mailing list
  for a quick review and all the feedback.

  (closes issue ASTERISK-18578)
........

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8 years agoMerged revisions 337178 via svnmerge from
Olle Johansson [Wed, 21 Sep 2011 09:06:22 +0000 (09:06 +0000)]
Merged revisions 337178 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

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  r337178 | oej | 2011-09-21 10:51:41 +0200 (Ons, 21 Sep 2011) | 14 lines

  Change strictrtp option to default to yes in the RTP module

  Suggested by Kapejod on Facebook

  Review: https://reviewboard.asterisk.org/r/1448/
  (closes issue ASTERISK-18587)

  Thanks for quick feedback to kpfleming and Tilghman
  --Denna och nedanstående rader kommer inte med i loggmeddelandet--

  M    CHANGES
  M    configs/rtp.conf.sample
  M    res/res_rtp_asterisk.c
........

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8 years agoMerged revisions 337120 via svnmerge from
Matthew Jordan [Tue, 20 Sep 2011 23:02:25 +0000 (23:02 +0000)]
Merged revisions 337120 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

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  r337120 | mjordan | 2011-09-20 17:49:36 -0500 (Tue, 20 Sep 2011) | 28 lines

  Merged revisions 337118 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.8

  ........
    r337118 | mjordan | 2011-09-20 17:38:54 -0500 (Tue, 20 Sep 2011) | 21 lines

    Fix for incorrect voicemail duration in external notifications

    This patch fixes an issue where the voicemail duration was being reported
    with a duration significantly less than the actual sound file duration.
    Voicemails that contained mostly silence were reporting the duration of
    only the sound in the file, as opposed to the duration of the file with
    the silence.  This patch fixes this by having two durations reported in
    the __ast_play_and_record family of functions - the sound_duration and the
    actual duration of the file.  The sound_duration, which is optional, now
    reports the duration of the sound in the file, while the actual full duration
    of the file is reported in the duration parameter.  This allows the voicemail
    applications to use the sound_duration for minimum duration checking, while
    reporting the full duration to external parties if the voicemail is kept.

    (issue ASTERISK-2234)
    (closes issue ASTERISK-16981)
    Reported by: Mary Ciuciu, Byron Clark, Brad House, Karsten Wemheuer, KevinH
    Tested by: Matt Jordan

    Review: https://reviewboard.asterisk.org/r/1443
  ........
................

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8 years agoMerged revisions 337119 via svnmerge from
Richard Mudgett [Tue, 20 Sep 2011 22:54:21 +0000 (22:54 +0000)]
Merged revisions 337119 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

........
  r337119 | rmudgett | 2011-09-20 17:47:45 -0500 (Tue, 20 Sep 2011) | 16 lines

  Fix crash with STRREPLACE function.

  The ast_func_read() function calls the .read2 callback with the len
  parameter set to zero indicating no size restrictions on the supplied
  ast_str buffer.  The value was used to dimension a local starts[] array
  with the array subsequently used.

  * Reworked the strreplace() function to perform the string replacement in
  a straight forward manner.  Eliminated the need for the starts[] array.

  (closes issue ASTERISK-18545)
  Reported by: Federico Alves
  Patches:
        jira_asterisk_18545_v10.patch (license #5621) patch uploaded by rmudgett
  Tested by: rmudgett, Federico Alves
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@337123 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoUpdated 10 merge property.
Richard Mudgett [Tue, 20 Sep 2011 22:53:12 +0000 (22:53 +0000)]
Updated 10 merge property.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@337122 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoRestore branch-10 merge properties.
Richard Mudgett [Tue, 20 Sep 2011 22:51:41 +0000 (22:51 +0000)]
Restore branch-10 merge properties.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@337121 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoMerged revisions 337115 via svnmerge from
Leif Madsen [Tue, 20 Sep 2011 22:29:24 +0000 (22:29 +0000)]
Merged revisions 337115 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r337115 | lmadsen | 2011-09-20 17:18:25 -0500 (Tue, 20 Sep 2011) | 7 lines

  Update RedHat Init script to work with Heartbeat.

  The current RedHat init script was not LSB compatible. This change will make it LSB compatible so that
  it can work correctly with Heartbeat.

  (Closes issue ASTERISK-18253)
  Reported by: c0rnoTa
........

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8 years agoMerged revisions 337062 via svnmerge from
Kinsey Moore [Tue, 20 Sep 2011 21:05:42 +0000 (21:05 +0000)]
Merged revisions 337062 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

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  r337062 | kmoore | 2011-09-20 16:05:01 -0500 (Tue, 20 Sep 2011) | 18 lines

  Merged revisions 337061 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.8

  ........
    r337061 | kmoore | 2011-09-20 16:04:11 -0500 (Tue, 20 Sep 2011) | 11 lines

    Make CANMATCH with the new pattern match engine behave more like the old one

    When checking an extension for E_CANMATCH using the new extension matching
    algorithm, an exact match was not returned as a possible match resulting in the
    queue failing to allow a caller to exit on DTMF.  This removes the requirement
    that an extension be longer than acquired digits for an E_CANMATCH operation
    to succeed.

    (closes issue ASTERISK-18044)
    Review: https://reviewboard.asterisk.org/r/1367/
  ........
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8 years agoMerged revisions 337008 via svnmerge from
Richard Mudgett [Tue, 20 Sep 2011 19:13:36 +0000 (19:13 +0000)]
Merged revisions 337008 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

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  r337008 | rmudgett | 2011-09-20 14:12:24 -0500 (Tue, 20 Sep 2011) | 22 lines

  Merged revisions 337007 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.8

  ........
    r337007 | rmudgett | 2011-09-20 14:10:30 -0500 (Tue, 20 Sep 2011) | 15 lines

    Check if a channel was created before using the pointer in sig_ss7_new_ast_channel().

    Fixes the crash in ASTERISK-17955 gdb-11918.txt backtrace.

    * Added some missing libss7 access lock protection.

    * Prevent cancelling the ss7_linkset() thread at inoportune times just
    like the pri_dchannel() thread.

    (issue ASTERISK-17955)
    Reported by: Ian M Sherman
    Patches:
          jira_asterisk_17955_v1.8.patch (license #5621) patch uploaded by rmudgett
          (attached to related ASTERISK-17966)
  ........
................

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8 years agoMerged revisions 336978 via svnmerge from
Richard Mudgett [Tue, 20 Sep 2011 18:20:10 +0000 (18:20 +0000)]
Merged revisions 336978 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

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  r336978 | rmudgett | 2011-09-20 13:14:40 -0500 (Tue, 20 Sep 2011) | 28 lines

  Merged revisions 336977 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.8

  ........
    r336977 | rmudgett | 2011-09-20 13:12:17 -0500 (Tue, 20 Sep 2011) | 21 lines

    Fix deadlock from not releasing SS7 linkset lock.

    sig_ss7_hangup() failed to release the SS7 linkset lock if the call had
    the alreadyhungup flag set.

    * Made unlock the SS7 linkset lock in sig_ss7_hangup() if the
    alreadyhungup flag is set.

    * Made ss7_start_call() not hold any locks while creating the channel for
    an incoming call to prevent deadlock.

    * Made ss7_grab() a void function, since it could never fail, to simplify
    calling code.

    * Made obtain the channel lock to do softhangup in some places.

    Patches:
          jira_ast_668_v1.8.patch (license #5621) patch uploaded by rmudgett

    JIRA AST-668
  ........
................

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8 years agoMerged revisions 336936 via svnmerge from
Gregory Nietsky [Tue, 20 Sep 2011 16:56:11 +0000 (16:56 +0000)]
Merged revisions 336936 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

........
  r336936 | irroot | 2011-09-20 18:51:59 +0200 (Tue, 20 Sep 2011) | 14 lines

  Allow Setting Auth Tag Bit length Based on invite or config option

  Update the SIP SRTP API to allow use of 32 or 80 bit taglen.
  Curently only 80 bit is supported.

  The outgoing invite will use the taglen of the incoming invite preventing
  one-way audio.

  (Closes issue ASTERISK-17895)

  Review: https://reviewboard.asterisk.org/r/1173/
........

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8 years agoMerged revisions 336878 via svnmerge from
Russell Bryant [Tue, 20 Sep 2011 01:11:18 +0000 (01:11 +0000)]
Merged revisions 336878 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

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  r336878 | russell | 2011-09-19 20:03:55 -0500 (Mon, 19 Sep 2011) | 43 lines

  Merged revisions 336877 via svnmerge from
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  ........
    r336877 | russell | 2011-09-19 19:56:20 -0500 (Mon, 19 Sep 2011) | 36 lines

    Fix crashes in ast_rtcp_write().

    This patch addresses crashes related to RTCP handling.  The backtraces just
    show a crash in ast_rtcp_write() where it appears that the RTP instance is no
    longer valid.  There is a race condition with scheduled RTCP transmissions and
    the destruction of the RTP instance.  This patch utilizes the fact that
    ast_rtp_instance is a reference counted object and ensures that it will not get
    destroyed while a reference is still around due to scheduled RTCP
    transmissions.

    RTCP transmissions are scheduled and executed from the chan_sip scheduler
    context.  This scheduler context is processed in the SIP monitor thread.  The
    destruction of an RTP instance occurs when the associated sip_pvt gets
    destroyed (which happens when the sip_pvt reference count reaches 0).  However,
    the SIP monitor thread is not the only thread that can cause a sip_pvt to get
    destroyed.  The sip_hangup function, executed from a channel thread, also
    decrements the reference count on a sip_pvt and could cause it to get
    destroyed.

    While this is being changed anyway, the patch also removes calling
    ast_sched_del() from within the RTCP scheduler callback.  It's not helpful.
    Simply returning 0 prevents the callback from being rescheduled.

    (closes issue ASTERISK-18570)

    Related issues that look like they are the same problem:

    (issue ASTERISK-17560)
    (issue ASTERISK-15406)
    (issue ASTERISK-15257)
    (issue ASTERISK-13334)
    (issue ASTERISK-9977)
    (issue ASTERISK-9716)

    Review: https://reviewboard.asterisk.org/r/1444/
  ........
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8 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
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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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8 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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8 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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8 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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8 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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8 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.

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8 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

8 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.

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8 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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8 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
https://origsvn.digium.com/svn/asterisk/branches/10

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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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8 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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8 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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8 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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8 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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8 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
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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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8 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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8 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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8 years agoMerged revisions 336235 via svnmerge from
Sean Bright [Fri, 16 Sep 2011 19:11:22 +0000 (19:11 +0000)]
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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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8 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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8 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
https://origsvn.digium.com/svn/asterisk/branches/10

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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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8 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
https://origsvn.digium.com/svn/asterisk/branches/10

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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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8 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
https://origsvn.digium.com/svn/asterisk/branches/10

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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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8 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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8 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
  https://origsvn.digium.com/svn/asterisk/branches/1.8

  ........
    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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8 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
  https://origsvn.digium.com/svn/asterisk/branches/1.8

  ........
    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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8 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
  https://origsvn.digium.com/svn/asterisk/branches/1.8

  ........
    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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@335792 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 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
  https://origsvn.digium.com/svn/asterisk/branches/1.8

  ........
    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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8 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

8 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

8 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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8 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
  https://origsvn.digium.com/svn/asterisk/branches/1.8

  ........
    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

8 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

8 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

8 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

8 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

................
  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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8 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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@335435 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 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

8 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)
  ........
................

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@335354 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 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

8 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

8 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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8 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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8 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/
  ........
................

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

8 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

8 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/

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

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8 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/
  ........
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8 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

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  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/
........

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8 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

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  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.
  ........
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8 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

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8 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

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  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
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8 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

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  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
  ........
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8 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

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  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.
  ........
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