asterisk/asterisk.git
9 years agoMerged revisions 311799 via svnmerge from
Tilghman Lesher [Tue, 29 Mar 2011 08:33:44 +0000 (08:33 +0000)]
Merged revisions 311799 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r311799 | tilghman | 2011-03-29 02:08:39 -0500 (Tue, 29 Mar 2011) | 7 lines

  Remove extraneous check from integer-type fields.

  (closes issue #19027)
   Reported by: mlehner

  Review: https://reviewboard.asterisk.org/r/1149/
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9 years agoMerged revisions 311751 via svnmerge from
Russell Bryant [Mon, 28 Mar 2011 22:00:46 +0000 (22:00 +0000)]
Merged revisions 311751 via svnmerge from
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  r311751 | russell | 2011-03-28 17:00:01 -0500 (Mon, 28 Mar 2011) | 2 lines

  Cross-reference VoiceMail() and VoiceMailMain() in the xml docs.
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9 years agoMerged revisions 311687 via svnmerge from
Alexandr Anikin [Sun, 27 Mar 2011 21:49:03 +0000 (21:49 +0000)]
Merged revisions 311687 via svnmerge from
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  r311687 | may | 2011-03-28 01:47:13 +0400 (Mon, 28 Mar 2011) | 2 lines

  correct return values in ooh323_indicate for AST_CONTROL_T38_PARAMETERS
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9 years agoMerged revisions 311615 via svnmerge from
Brett Bryant [Wed, 23 Mar 2011 21:55:54 +0000 (21:55 +0000)]
Merged revisions 311615 via svnmerge from
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  r311615 | bbryant | 2011-03-23 17:54:11 -0400 (Wed, 23 Mar 2011) | 8 lines

  This patch fixes a bug with MeetMe behavior where the 'P' option for always
  prompting for a pin is ignored for the first caller.

  (closes issue #18070)
  Reported by: mav3rick

  Review: https://reviewboard.asterisk.org/r/1132/
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9 years agoMerged revisions 311612 via svnmerge from
Brett Bryant [Wed, 23 Mar 2011 21:46:59 +0000 (21:46 +0000)]
Merged revisions 311612 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r311612 | bbryant | 2011-03-23 17:45:46 -0400 (Wed, 23 Mar 2011) | 9 lines

  Fix a possible crash in sip/reqresp_parser.c that is caused by a possible null
  value.

  (closes issue #18821)
  Reported by: cmaj
  Patches:
        patch-reqresp_parser_sip_uri_domain_cmp_c_locale-crash-1.8.3-rc2.diff.tx
        uploaded by cmaj (license 830)
........

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9 years agoMerged revisions 311558 via svnmerge from
Terry Wilson [Wed, 23 Mar 2011 02:51:09 +0000 (02:51 +0000)]
Merged revisions 311558 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r311558 | twilson | 2011-03-22 19:24:53 -0700 (Tue, 22 Mar 2011) | 5 lines

  Don't use static declared buf in parse_name_andor_addr

  This function isn't used anywhere yet, but we definitely don't want
  to keep the same value for buf between calls to the function.
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9 years agoMerged revisions 311497 via svnmerge from
David Vossel [Tue, 22 Mar 2011 15:26:51 +0000 (15:26 +0000)]
Merged revisions 311497 via svnmerge from
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  r311497 | dvossel | 2011-03-22 10:25:24 -0500 (Tue, 22 Mar 2011) | 9 lines

  Merged revisions 311496 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2

  ........
    r311496 | dvossel | 2011-03-22 10:24:45 -0500 (Tue, 22 Mar 2011) | 2 lines

    Fixes memory leak in MeetMe AMI action
  ........
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9 years agoAdds an option to FollowMe that isn't useful for the bug it was made to solve. Still...
Jonathan Rose [Fri, 18 Mar 2011 19:05:20 +0000 (19:05 +0000)]
Adds an option to FollowMe that isn't useful for the bug it was made to solve.  Still, due to the nature of FollowMe, it makes sense to have this option since it keeps apps bound to channels that would otherwise go away from being lost.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@311427 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoRemove libresample dependency from codec_resample.c
David Vossel [Fri, 18 Mar 2011 16:27:23 +0000 (16:27 +0000)]
Remove libresample dependency from codec_resample.c

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9 years agoMerged revisions 311352 via svnmerge from
Jonathan Rose [Fri, 18 Mar 2011 16:24:19 +0000 (16:24 +0000)]
Merged revisions 311352 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r311352 | jrose | 2011-03-18 11:19:05 -0500 (Fri, 18 Mar 2011) | 10 lines

  Changes some print statements/events to use a blank string in place of NULL if the string in question is NULL.

  This is supposed to improve Solaris compatibility since Solaris goes berserk when trying to output NULL strings.

  (closes issue #18759)
  Reported by: bklang
  Patches:
        null-strings.patch uploaded by bklang (license 919)
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9 years agoMerged revisions 311342 via svnmerge from
Matthew Nicholson [Fri, 18 Mar 2011 16:03:51 +0000 (16:03 +0000)]
Merged revisions 311342 via svnmerge from
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  r311342 | mnicholson | 2011-03-18 11:02:50 -0500 (Fri, 18 Mar 2011) | 2 lines

  Properly populate the LOCALSTATIONID channel variable.
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9 years agoMerged revisions 311297 via svnmerge from
Richard Mudgett [Fri, 18 Mar 2011 03:00:39 +0000 (03:00 +0000)]
Merged revisions 311297 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r311297 | rmudgett | 2011-03-17 21:59:05 -0500 (Thu, 17 Mar 2011) | 12 lines

  Race condition when ISDN CallRerouting/CallDeflection invoked.

  The queued AST_CONTROL_BUSY could sometimes be processed before the
  call_forward dial string is recognized.

  * Moved setting the call_forwarding dial string after sending a response
  to the initiator and just queue an empty frame to wake up the media thread
  instead of an AST_CONTROL_BUSY.

  * Added check for empty rerouting/deflection number and respond with an
  error.
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9 years agoMerged revisions 311295 via svnmerge from
Richard Mudgett [Fri, 18 Mar 2011 02:31:27 +0000 (02:31 +0000)]
Merged revisions 311295 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r311295 | rmudgett | 2011-03-17 21:22:07 -0500 (Thu, 17 Mar 2011) | 35 lines

  Merged revision 310986 from
  https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier

  ..........
    r310986 | rmudgett | 2011-03-16 13:56:28 -0500 (Wed, 16 Mar 2011) | 28 lines

    Dial() o option broke when connected line feature added.

    The patch restores the o option behavior and adds the ability to specify
    the CallerID.  The Dial o and f options are complementary to each other.
    The o option stores the CallerID on the outgoing channel as the channel's
    CallerID.  The f option forces the CallerID sent by the outgoing channel.

    o(x) - The argument 'x' is optional.  If not present, then specify that
    the CallerID that was present on the *calling* channel be stored as the
    CallerID on the *called* channel.  This was the behavior of Asterisk 1.0
    and earlier.  If present, then specify the CallerID stored on the *called*
    channel.  Note that o(${CALLERID(all)}) is similar to option o without
    parameters.

    f(x) - The argument 'x' is optional and its presence changes the behavior
    of this option.  If not present, then force the outgoing CallerID on a
    call-forward or deflection to the dialplan extension for this Dial() using
    a dialplan 'hint'.  For example, some PSTNs do not allow CallerID to be
    set to anything other than the numbers assigned to you.  If present, then
    force the outgoing CallerID to 'x'.

    Patches:
   jira_abe_2752_dial_fo_options.patch uploaded by rmudgett (license 664)
    Tested by: rmudgett

    JIRA ABE-2752
    JIRA SWP-3096
  ..........
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9 years agoMerged revisions 311197 via svnmerge from
Jonathan Rose [Thu, 17 Mar 2011 19:05:42 +0000 (19:05 +0000)]
Merged revisions 311197 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r311197 | jrose | 2011-03-17 14:03:34 -0500 (Thu, 17 Mar 2011) | 11 lines

  This fixes a nasty chanspy bug which was causing a channel leak every time a spied on channel made a call.

  In addition to the above, it makes certain channel destruction occurs so that applications don't get stuck waiting for datastore destruction while monitored by chanspy.

  (closes issue #18742)
  Reported by: jkister
  Tested by: jkister, jcovert, jrose

  Review: http://reviewboard.digium.internal/r/106/
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9 years agoMerged revisions 311141 via svnmerge from
Matthew Nicholson [Thu, 17 Mar 2011 15:02:12 +0000 (15:02 +0000)]
Merged revisions 311141 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r311141 | mnicholson | 2011-03-17 10:00:33 -0500 (Thu, 17 Mar 2011) | 11 lines

  Merged revisions 311140 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2

  ........
    r311140 | mnicholson | 2011-03-17 09:58:52 -0500 (Thu, 17 Mar 2011) | 4 lines

    Don't write items to the manager socket twice.

    AST-2011-003

    (closes issue 0018987)
    Reported by: ks-steven
  ........
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9 years agoMerged revisions 311050 via svnmerge from
Alec L Davis [Thu, 17 Mar 2011 10:51:57 +0000 (10:51 +0000)]
Merged revisions 311050 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r311050 | alecdavis | 2011-03-17 23:49:41 +1300 (Thu, 17 Mar 2011) | 24 lines

  Merged revisions 311049 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2

  ................
    r311049 | alecdavis | 2011-03-17 23:45:47 +1300 (Thu, 17 Mar 2011) | 17 lines

    Merged revisions 311048 via svnmerge from
    https://origsvn.digium.com/svn/asterisk/branches/1.4

    ........
      r311048 | alecdavis | 2011-03-17 23:43:35 +1300 (Thu, 17 Mar 2011) | 12 lines

      Remove extra quote in indications.conf

      Picking low hanging fruit.

      (closes issue #18971)
      Reported by: IgorG
      Patches:
            based on indications.conf.sample.diff uploaded by IgorG (license 20)
      Tested by: IgorG
    ........
  ................
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9 years agoMerged revisions 310999 via svnmerge from
Terry Wilson [Wed, 16 Mar 2011 19:51:55 +0000 (19:51 +0000)]
Merged revisions 310999 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r310999 | twilson | 2011-03-16 14:47:59 -0500 (Wed, 16 Mar 2011) | 18 lines

  Merged revisions 310998 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2

  ........
    r310998 | twilson | 2011-03-16 14:46:36 -0500 (Wed, 16 Mar 2011) | 11 lines

    Fix crash on fdopen failure

    See security advisory AST-2011-004

    (closes issue #18845)
    Reported by: cmaj
    Patches:
        patch-main-tcptls-1.8.3-rc2-open-session-crash-take2.diff.txt uploaded by cmaj (license 830)
        patch-main-tcptls-1.8.3-rc2-open-session-crash-take3.diff.txt uploaded by cmaj (license 830)
    Tested by: cmaj, twilson
  ........
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9 years agoMerged revisions 310993 via svnmerge from
Terry Wilson [Wed, 16 Mar 2011 19:51:04 +0000 (19:51 +0000)]
Merged revisions 310993 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r310993 | twilson | 2011-03-16 14:26:57 -0500 (Wed, 16 Mar 2011) | 11 lines

  Merged revisions 310992 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2

  ........
    r310992 | twilson | 2011-03-16 14:23:03 -0500 (Wed, 16 Mar 2011) | 4 lines

    Don't keep trying to write to a closed connection

    See security advisory AST-2011-003.
  ........
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9 years agoMerged revisions 310902 via svnmerge from
Terry Wilson [Wed, 16 Mar 2011 17:29:16 +0000 (17:29 +0000)]
Merged revisions 310902 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r310902 | twilson | 2011-03-16 12:19:57 -0500 (Wed, 16 Mar 2011) | 43 lines

  Merged revisions 310889 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2

  ................
    r310889 | twilson | 2011-03-16 12:03:27 -0500 (Wed, 16 Mar 2011) | 36 lines

    Merged revisions 310888 via svnmerge from
    https://origsvn.digium.com/svn/asterisk/branches/1.4

    ........
      r310888 | twilson | 2011-03-16 11:58:42 -0500 (Wed, 16 Mar 2011) | 29 lines

      Don't delay DTMF in core bridge while listening for DTMF features

      This patch is mostly the work of Olle Johansson. I did some cleanup and
      added the silence generating code if transmit_silence is set.

      When a channel listens for DTMF in the core bridge, the outbound DTMF is not
      sent until we have received DTMF_END. For a long DTMF, this is a disaster. We
      send 4 seconds of DTMF to Asterisk, which sends no audio for those 4 seconds.
      Some products see this delay and the time skew on RTP packets that results and
      start ignoring the audio that is sent afterward.

      With this change, the DTMF_BEGIN frame is inspected and checked. If it matches
      a feature code, we wait for DTMF_END and activate the feature as before. If
      transmit_silence=yes in asterisk.conf, silence is sent if we paritally match a
      multi-digit feature. If it doesn't match a feature, the frame is forwarded
      along with the DTMF_END without delay. By doing it this way, DTMF is not delayed.

      (closes issue #15642)
      Reported by: jasonshugart
      Patches:
            issue_15652_dtmf_ast-1.4.patch.txt uploaded by twilson (license 396)
      Tested by: globalnetinc, jde

      (closes issue #16625)
      Reported by: sharvanek

      Review: https://reviewboard.asterisk.org/r/1092/
      Review: https://reviewboard.asterisk.org/r/1125/
    ........
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9 years agoMerged revisions 310834 via svnmerge from
Tilghman Lesher [Tue, 15 Mar 2011 01:49:37 +0000 (01:49 +0000)]
Merged revisions 310834 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r310834 | tilghman | 2011-03-14 20:48:25 -0500 (Mon, 14 Mar 2011) | 2 lines

  Fix branch compile.
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9 years agoMerged revisions 310781 via svnmerge from
Alec L Davis [Tue, 15 Mar 2011 01:36:26 +0000 (01:36 +0000)]
Merged revisions 310781 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r310781 | alecdavis | 2011-03-15 14:00:55 +1300 (Tue, 15 Mar 2011) | 10 lines

  core show locks: display ThreadID in hexadecimal

  Allow easier cross referencing of thread ID's with GDB backtraces

  (closes issue #18968)
  Reported by: alecdavis
  Patches:
        bug18968.diff.txt uploaded by alecdavis (license 585)
........

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9 years agoMerged revisions 310734 via svnmerge from
Alexandr Anikin [Mon, 14 Mar 2011 21:51:35 +0000 (21:51 +0000)]
Merged revisions 310734 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
(closes issue #18693)

........
  r310734 | may | 2011-03-15 00:45:53 +0300 (Tue, 15 Mar 2011) | 12 lines

  Introduce t.38 parameters control functionality not full but enough for
  Send/RcvFax support

  Introduce t.38 controls between asterisk core and channel/proto layers.
  Not all parameters are transferred from proto layers but *Fax apps
  tested and work ok.

  (issue #18693)
  Reported by: benngard2
  Patches:
        issue-18693.patch uploaded by may213 (license 454)
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@310735 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 310636 via svnmerge from
Richard Mudgett [Mon, 14 Mar 2011 16:55:30 +0000 (16:55 +0000)]
Merged revisions 310636 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r310636 | rmudgett | 2011-03-14 11:50:59 -0500 (Mon, 14 Mar 2011) | 39 lines

  Merged revisions 310635 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2

  ................
    r310635 | rmudgett | 2011-03-14 11:47:54 -0500 (Mon, 14 Mar 2011) | 32 lines

    Merged revisions 310633 via svnmerge from
    https://origsvn.digium.com/svn/asterisk/branches/1.4

    ........
      r310633 | rmudgett | 2011-03-14 11:38:24 -0500 (Mon, 14 Mar 2011) | 25 lines

      "Caller*ID failed checksum" on Wildcard TDM2400P and TDM410

      The last character in the caller id message is getting a framing error.

      The checksum is the last character in the message.  A framing error in the
      checksum could be because:
      1) The sender did not send a full stop bit.
      2) The sender cut off the FSK carrier too soon.
      3) The sender opted to send zero of the specified zero to 10 trailing mark
      bits and round-off errors in the code resulted in the code not being where
      it thought it was in the demodulated bit stream.

      Bit 8 of 'b' is set when parity error.
      Bit 9 of 'b' is set when framing error.

      Made ignore the framing and parity error bits if the errored character is
      the checksum.  We can tolerate a framing/parity error there.  The checksum
      character validates the message.

      (closes issue #18474)
      Reported by: nivek
      Patches:
            callerid.c.1.patch uploaded by nivek (license 636) (with modifications)
      Tested by: nivek
    ........
  ................
................

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9 years agoMerged revisions 310587 via svnmerge from
Jonathan Rose [Mon, 14 Mar 2011 15:40:43 +0000 (15:40 +0000)]
Merged revisions 310587 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r310587 | jrose | 2011-03-14 10:27:57 -0500 (Mon, 14 Mar 2011) | 15 lines

  Merged revisions 310585 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2

  ........
    r310585 | jrose | 2011-03-14 08:56:22 -0500 (Mon, 14 Mar 2011) | 8 lines

    Adds 'p' as an option to func_volume.  When it is on, the old behavior with DTMF controlling volume adjustment will be enforced.
    When it is off, DTMF will not be processed by the function.

    Programmed by Jonathan Rose
    Reviewed by David Vossel, Leif Madsen, and Russell Bryant

    http://reviewboard.digium.internal/r/93/
  ........
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9 years agoFixes null reference bug introduced by audio hook changes that affects various OS...
Jonathan Rose [Mon, 14 Mar 2011 13:12:51 +0000 (13:12 +0000)]
Fixes null reference bug introduced by audio hook changes that affects various OS distributions.  Thanks David.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@310547 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 310462 via svnmerge from
Tilghman Lesher [Sat, 12 Mar 2011 20:42:33 +0000 (20:42 +0000)]
Merged revisions 310462 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r310462 | tilghman | 2011-03-12 14:27:54 -0600 (Sat, 12 Mar 2011) | 45 lines

  Merged revisions 310448 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2

  ................
    r310448 | tilghman | 2011-03-12 14:24:54 -0600 (Sat, 12 Mar 2011) | 38 lines

    Recorded merge of revisions 310435 via svnmerge from
    https://origsvn.digium.com/svn/asterisk/branches/1.4

    ........
      r310435 | tilghman | 2011-03-12 14:22:07 -0600 (Sat, 12 Mar 2011) | 31 lines

      Add AELSub, which provides a stable entry point into AEL subroutines.

      This commit needs some explanation, given that we're adding a new application
      into an existing release branch.  This is generally a violation of our release
      policy, except in very limited circumstances, and I believe this is one of
      those circumstances.

      The problem that this solves is one of the sanity of using multiple dialplan
      languages to define a dialplan.  In the case of the reporter, he or she is
      using AEL is define subroutines, while using Realtime extensions to invoke
      those subroutines.  While you can do this, it's based upon the reality of AEL
      using actual dialplan extensions; however, there is no guarantee that the
      details of _how_ AEL is compiled into extensions will remain stable.  In fact,
      at the time of this commit, it has already changed twice, once in a
      fundamental way.

      Now normally, a new application would only be added to trunk.  However, this
      application is explicitly to create a stable user-level API between versions,
      and adding it to trunk only will not solve the user's problem of switching
      between 1.6.2 and 1.8, nor will it help anybody switching from 1.8 to 1.10.
      Therefore, it needs to go into existing release branches.  For the sake of
      consistency, and also because one of the changes was between 1.4 and 1.6.x,
      I am also electing to commit this to 1.4.

      (closes issue #18910)
       Reported by: alexandrekeller
       Patches:
             20110304__issue18919__1.6.2.diff.txt uploaded by tilghman (license 14)
             20110304__issue18919__1.4.diff.txt uploaded by tilghman (license 14)
       Tested by: alexandrekeller
    ........
  ................
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9 years agoMerged revisions 310415 via svnmerge from
Tilghman Lesher [Sat, 12 Mar 2011 20:08:19 +0000 (20:08 +0000)]
Merged revisions 310415 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r310415 | tilghman | 2011-03-12 14:05:46 -0600 (Sat, 12 Mar 2011) | 14 lines

  Merged revisions 310414 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2

  ........
    r310414 | tilghman | 2011-03-12 13:51:23 -0600 (Sat, 12 Mar 2011) | 7 lines

    Transactional handles should be used for the insertbuf, if available.

    Also, fix a possible resource leak.

    (closes issue #18943)
     Reported by: irroot
  ........
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9 years agoMix Monitor: Now with r and t options.
Jonathan Rose [Fri, 11 Mar 2011 18:54:45 +0000 (18:54 +0000)]
Mix Monitor:  Now with r and t options.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@310373 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoUse "-march=native" when possible.
Kevin P. Fleming [Fri, 11 Mar 2011 15:09:23 +0000 (15:09 +0000)]
Use "-march=native" when possible.

Recent versions of GCC have a tuning option value of 'native', which causes
the compiler to optimize the build for the CPU the compile is performed on.
Since most people are building Asterisk on the machine they plan to run it on,
the configure script and build system will now use this value unless a different
value is specified by the user in CFLAGS when the configure script is executed.
In addition, this value will be used for building the GSM and LPC10 codecs as
well, in preference to the logic that has been in their Makefiles forever to
optimize for certain types of CPUs.

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9 years agoMerged revisions 310287 via svnmerge from
Alec L Davis [Fri, 11 Mar 2011 06:56:06 +0000 (06:56 +0000)]
Merged revisions 310287 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r310287 | alecdavis | 2011-03-11 19:47:44 +1300 (Fri, 11 Mar 2011) | 17 lines

  remote_bridge_loop: prevent segfault when after transfer of IAX2 of DAHDI call

  If the channel condition is one of the following after breaking out of the loop, don't try to update_peer
  (where x = 0/1)
   1). ZOMBIE
   2). cx->tech_pvt != pvtx
   3). gluex != ast_rtp_instance_get_glue(cx->tech->type))

  (closes issue #18781)
  Reported by: alecdavis
  Patches:
        bug18781.diff3.txt uploaded by alecdavis (license 585)
  Tested by: alecdavis, ZX81

  Review: https://reviewboard.asterisk.org/r/1128/
........

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9 years agoMerged revisions 310240 via svnmerge from
Terry Wilson [Thu, 10 Mar 2011 16:09:09 +0000 (16:09 +0000)]
Merged revisions 310240 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r310240 | twilson | 2011-03-10 10:05:45 -0600 (Thu, 10 Mar 2011) | 13 lines

  Add \r\n to remaining http headers passed to ast_http_send

  r309204 changed the behavior of ast_http_send. It now requires headers
  to be passed with trailing \r\n. This change updates the remaining
  instances in the code that did not pass the \r\n.

  (closes issue #18186)
  Reported by: nivaldomjunior
  Patches:
        res_phoneprov.c.diff uploaded by lathama (license 1028)
        manager.diff.txt uploaded by twilson (license 396)
  Tested by: lathama
........

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9 years agoMerged revisions 310231 via svnmerge from
Mark Michelson [Thu, 10 Mar 2011 15:28:55 +0000 (15:28 +0000)]
Merged revisions 310231 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r310231 | mmichelson | 2011-03-10 09:17:04 -0600 (Thu, 10 Mar 2011) | 9 lines

  Be more tolerant of what URI we accept for call completion PUBLISH requests.

  (closes issue #18946)
  Reported by: GeorgeKonopacki
  Patches:
        18946.patch uploaded by mmichelson (license 60)
  Tested by: GeorgeKonopacki
........

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9 years agoMerged revisions 310142 via svnmerge from
Tilghman Lesher [Thu, 10 Mar 2011 05:54:53 +0000 (05:54 +0000)]
Merged revisions 310142 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r310142 | tilghman | 2011-03-09 23:53:29 -0600 (Wed, 09 Mar 2011) | 19 lines

  Merged revisions 310141 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2

  ................
    r310141 | tilghman | 2011-03-09 23:51:37 -0600 (Wed, 09 Mar 2011) | 12 lines

    Merged revisions 310140 via svnmerge from
    https://origsvn.digium.com/svn/asterisk/branches/1.4

    ........
      r310140 | tilghman | 2011-03-09 23:38:44 -0600 (Wed, 09 Mar 2011) | 5 lines

      Initialize column size to 0 to deal with a potential UnixODBC bug on 64-bit systems.

      (closes issue #18295)
       Reported by: pruiz
    ........
  ................
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9 years agoMerged revisions 310088 via svnmerge from
Jonathan Rose [Tue, 8 Mar 2011 20:34:05 +0000 (20:34 +0000)]
Merged revisions 310088 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r310088 | jrose | 2011-03-08 14:19:32 -0600 (Tue, 08 Mar 2011) | 9 lines

  Returns with an error notice if CHANNEL function of SIP channel is read without arguments.

  (Closes issue #18653)
  Reported by: wuwu
  Patches:
        diff.patch uploaded by jrose (license 1225)
  Tested by: jrose
........

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9 years agoMerged revisions 310039 via svnmerge from
Terry Wilson [Tue, 8 Mar 2011 18:19:46 +0000 (18:19 +0000)]
Merged revisions 310039 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r310039 | twilson | 2011-03-08 10:10:50 -0800 (Tue, 08 Mar 2011) | 11 lines

  Spelling fix in "calendar show calendar"

  s/Cartegories/Catagories/

  (closes issue #18931)
  Reported by: pdugas
  Patches:
        res_calendar.c.patch uploaded by pdugas (license 1222)

  Review: [full review board URL with trailing slash]
........

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9 years agoMerged revisions 309994 via svnmerge from
Richard Mudgett [Tue, 8 Mar 2011 16:46:16 +0000 (16:46 +0000)]
Merged revisions 309994 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r309994 | rmudgett | 2011-03-08 10:37:02 -0600 (Tue, 08 Mar 2011) | 1 line

  Make pri parameter description consistent.
........

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9 years agoMerged revisions 309858 via svnmerge from
Jonathan Rose [Mon, 7 Mar 2011 22:16:33 +0000 (22:16 +0000)]
Merged revisions 309858 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r309858 | jrose | 2011-03-07 16:07:25 -0600 (Mon, 07 Mar 2011) | 22 lines

  Merged revisions 309857 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2

  ................
    r309857 | jrose | 2011-03-07 16:04:44 -0600 (Mon, 07 Mar 2011) | 15 lines

    Merged revisions 309856 via svnmerge from
    https://origsvn.digium.com/svn/asterisk/branches/1.4

    ........
      r309856 | jrose | 2011-03-07 16:02:12 -0600 (Mon, 07 Mar 2011) | 8 lines

      Bug fix for MixMonitor involving filenames with '.' not in the extension

      Closes issue #18391)
      Reported by: pabelanger
      Patches:
            bugfix.patch uploaded by jrose (license 1225)
      Tested by: jrose
    ........
  ................
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9 years agoMerged revisions 309808 via svnmerge from
Tilghman Lesher [Mon, 7 Mar 2011 01:01:08 +0000 (01:01 +0000)]
Merged revisions 309808 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r309808 | tilghman | 2011-03-06 18:54:42 -0600 (Sun, 06 Mar 2011) | 14 lines

  Merged revisions 309251 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2

  ........
    r309251 | tilghman | 2011-03-01 19:06:02 -0600 (Tue, 01 Mar 2011) | 7 lines

    Revert previous 2 commits, and instead conditionally redefine the same macro used in flex 2.5.35 that clashed with our workaround.

    Not surprisingly, the workaround was exactly the same code as was provided by
    the Flex maintainers, albeit in two different places, in different macros.

    This should fix the FreeBSD builds, which have an older version of Flex.
  ........
................

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9 years agoMerged revisions 309765 via svnmerge from
Mark Michelson [Mon, 7 Mar 2011 00:14:34 +0000 (00:14 +0000)]
Merged revisions 309765 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r309765 | mmichelson | 2011-03-06 18:13:36 -0600 (Sun, 06 Mar 2011) | 3 lines

  Indicate that Asterisk uses the Allow header to determine if MESSAGE requests should be sent.
........

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9 years agoMerged revisions 309720 via svnmerge from
Moises Silva [Sat, 5 Mar 2011 17:53:31 +0000 (17:53 +0000)]
Merged revisions 309720 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r309720 | moy | 2011-03-05 12:44:30 -0500 (Sat, 05 Mar 2011) | 6 lines

  Fix caller id passed to openr2_chan_make_call

  (closes issue #18894)
  Reported by: malufrj
  Tested by: moy
........

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9 years agoMerged revisions 309678 via svnmerge from
Tilghman Lesher [Sat, 5 Mar 2011 10:30:28 +0000 (10:30 +0000)]
Merged revisions 309678 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r309678 | tilghman | 2011-03-05 04:29:30 -0600 (Sat, 05 Mar 2011) | 14 lines

  Merged revisions 309677 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2

  ........
    r309677 | tilghman | 2011-03-05 04:28:24 -0600 (Sat, 05 Mar 2011) | 7 lines

    Missed part of the conversion when we started passing ppid to astcanary.

    (closes issue #18850)
     Reported by: viraptor
     Patches:
           canary_ppid.patch uploaded by viraptor (license 543)
  ........
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9 years agoAdd setvar option to calendaring
Terry Wilson [Fri, 4 Mar 2011 23:22:39 +0000 (23:22 +0000)]
Add setvar option to calendaring

Adding the setvar option with variable substitution on the value allows things
like setting the outbound caller id name to the summary of a calendar event,
etc. Values could be chained together as they are appended in order to do some
scripting if necessary.

Review: https://reviewboard.asterisk.org/r/1134/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309640 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 309585 via svnmerge from
Matthew Nicholson [Fri, 4 Mar 2011 19:38:59 +0000 (19:38 +0000)]
Merged revisions 309585 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r309585 | mnicholson | 2011-03-04 13:38:25 -0600 (Fri, 04 Mar 2011) | 9 lines

  Merged revisions 309584 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2

  ........
    r309584 | mnicholson | 2011-03-04 13:37:13 -0600 (Fri, 04 Mar 2011) | 2 lines

    Restore mysterious lua_pushvalue() call removed in r309494.  The mystery has been solved.
  ........
................

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309587 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 309542 via svnmerge from
Matthew Nicholson [Fri, 4 Mar 2011 19:02:31 +0000 (19:02 +0000)]
Merged revisions 309542 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r309542 | mnicholson | 2011-03-04 13:00:33 -0600 (Fri, 04 Mar 2011) | 11 lines

  Merged revisions 309541 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2

  ........
    r309541 | mnicholson | 2011-03-04 12:59:20 -0600 (Fri, 04 Mar 2011) | 4 lines

    Check for errors from fseek() when loading config file, properly abort on errors from fread(), and supply a traceback for errors generated when loading the config file.

    Also, prepend a newline to traceback output so that the main error message is on it's own line.
  ........
................

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309543 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 309495 via svnmerge from
Matthew Nicholson [Fri, 4 Mar 2011 18:11:43 +0000 (18:11 +0000)]
Merged revisions 309495 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r309495 | mnicholson | 2011-03-04 12:10:23 -0600 (Fri, 04 Mar 2011) | 9 lines

  Merged revisions 309494 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2

  ........
    r309494 | mnicholson | 2011-03-04 11:55:57 -0600 (Fri, 04 Mar 2011) | 2 lines

    remove mysterious lua_pushvalue() that is never used
  ........
................

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309496 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoAdd support for defining hints from pbx_lua
Matthew Nicholson [Fri, 4 Mar 2011 17:44:44 +0000 (17:44 +0000)]
Add support for defining hints from pbx_lua

(closes issue #16024)
Reported by: mnicholson

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309493 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoFix a buglet that prevented chan_nbs from loading (and subsequently stopped Asterisk).
Russell Bryant [Fri, 4 Mar 2011 17:40:02 +0000 (17:40 +0000)]
Fix a buglet that prevented chan_nbs from loading (and subsequently stopped Asterisk).

In passing, convert the return codes to be the proper AST_MODULE_LOAD_* constants.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309491 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 309448 via svnmerge from
Matthew Nicholson [Fri, 4 Mar 2011 16:00:05 +0000 (16:00 +0000)]
Merged revisions 309448 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r309448 | mnicholson | 2011-03-04 09:59:25 -0600 (Fri, 04 Mar 2011) | 8 lines

  Export global symbols from pbx_lua to allow modules to be loaded.  Fixes a regression introduced in r278132.

  (closes issue #18671)
  Reported by: Igels
  Patches:
        pbx_lua_global_symbols1.diff uploaded by mnicholson (license 96)
  Tested by: Igels
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309449 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 309445 via svnmerge from
Richard Mudgett [Fri, 4 Mar 2011 15:28:20 +0000 (15:28 +0000)]
Merged revisions 309445 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r309445 | rmudgett | 2011-03-04 09:22:04 -0600 (Fri, 04 Mar 2011) | 46 lines

  Get real channel of a DAHDI call.

  Starting with Asterisk v1.8, the DAHDI channel name format was changed for
  ISDN calls to: DAHDI/i<span>/<number>[:<subaddress>]-<sequence-number>

  There were several reasons that the channel name had to change.

  1) Call completion requires a device state for ISDN phones.  The generic
  device state uses the channel name.

  2) Calls do not necessarily have B channels.  Calls placed on hold by an
  ISDN phone do not have B channels.

  3) The B channel a call initially requests may not be the B channel the
  call ultimately uses.  Changes to the internal implementation of the
  Asterisk master channel list caused deadlock problems for chan_dahdi if it
  needed to change the channel name.  Chan_dahdi no longer changes the
  channel name.

  4) DTMF attended transfers now work with ISDN phones because the channel
  name is "dialable" like the chan_sip channel names.

  For various reasons, some people need to know which B channel a DAHDI call
  is using.

  * Added CHANNEL(dahdi_span), CHANNEL(dahdi_channel), and
  CHANNEL(dahdi_type) so the dialplan can determine the B channel currently
  in use by the channel.  Use CHANNEL(no_media_path) to determine if the
  channel even has a B channel.

  * Added AMI event DAHDIChannel to associate a DAHDI channel with an
  Asterisk channel so AMI applications can passively determine the B channel
  currently in use.  Calls with "no-media" as the DAHDIChannel do not have
  an associated B channel.  No-media calls are either on hold or
  call-waiting.

  (closes issue #17683)
  Reported by: mrwho
  Tested by: rmudgett

  (closes issue #18603)
  Reported by: arjankroon
  Patches:
        issue17683_18603_v1.8_v2.patch uploaded by rmudgett (license 664)
  Tested by: stever28, rmudgett
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309446 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 309403 via svnmerge from
David Ruggles [Fri, 4 Mar 2011 01:52:21 +0000 (01:52 +0000)]
Merged revisions 309403 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r309403 | diruggles | 2011-03-03 20:50:44 -0500 (Thu, 03 Mar 2011) | 23 lines

  Merged revisions 309356 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2

  ................
    r309356 | diruggles | 2011-03-03 19:42:28 -0500 (Thu, 03 Mar 2011) | 16 lines

    Merged revisions 309355 via svnmerge from
    https://origsvn.digium.com/svn/asterisk/branches/1.4

    ........
      r309355 | diruggles | 2011-03-03 19:34:13 -0500 (Thu, 03 Mar 2011) | 9 lines

      fix small memory leak

      fix small memory leak caused by a string allocation that wasn't freed

      (closes issue #18907)
      Reported by: andy11
      Patches:
            asterisk_trunk-app_externalivr-leak.patch uploaded by andy11 (license 1224)
    ........
  ................
................

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309404 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoAdd HangupRequest manager event, to specify when/where a channel gets hung up.
Jason Parker [Wed, 2 Mar 2011 21:08:39 +0000 (21:08 +0000)]
Add HangupRequest manager event, to specify when/where a channel gets hung up.

(closes issue #18226)
Reported by: clegall_proformatique
Patches:
      asterisk_1.8_293157_hanguprequests.svn.patch uploaded by clegall proformatique (license 1139)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309300 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 309256 via svnmerge from
Jason Parker [Wed, 2 Mar 2011 19:54:43 +0000 (19:54 +0000)]
Merged revisions 309256 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r309256 | qwell | 2011-03-02 13:54:20 -0600 (Wed, 02 Mar 2011) | 15 lines

  Merged revisions 309255 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2

  ........
    r309255 | qwell | 2011-03-02 13:53:47 -0600 (Wed, 02 Mar 2011) | 8 lines

    Fix usage of "hasvoicemail=yes" and "mailbox=" in users.conf for SIP.

    Since it's a duplicate, nothing is going to be done, so delme doesn't need to
    be set at all.  Strangely, when this was added, this was being set to 1 in 1.6,
    and 0 in trunk.

    (issue AST-439)
  ........
................

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309257 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 309204 via svnmerge from
Jason Parker [Tue, 1 Mar 2011 22:26:37 +0000 (22:26 +0000)]
Merged revisions 309204 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r309204 | qwell | 2011-03-01 16:25:44 -0600 (Tue, 01 Mar 2011) | 7 lines

  Fix consistency of CRLFs on HTTP headers that get sent out.

  (closes issue #18186)
  Reported by: nivaldomjunior
  Patches:
        18186-httpheadernewline.diff uploaded by qwell (license 4)
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309209 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 309170 via svnmerge from
Richard Mudgett [Tue, 1 Mar 2011 21:57:58 +0000 (21:57 +0000)]
Merged revisions 309170 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r309170 | rmudgett | 2011-03-01 15:57:26 -0600 (Tue, 01 Mar 2011) | 7 lines

  Document CHANNEL(keypad_digits) and CHANNEL(no_media_path).

  * Added XML documentation for CHANNEL(keypad_digits) and
  CHANNEL(no_media_path).

  * Tweaked XML documentation for CHANNEL(reversecharge).
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309171 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 309126 via svnmerge from
Richard Mudgett [Tue, 1 Mar 2011 18:50:07 +0000 (18:50 +0000)]
Merged revisions 309126 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r309126 | rmudgett | 2011-03-01 12:44:05 -0600 (Tue, 01 Mar 2011) | 16 lines

  Chan_dahdi does not retain CID when detecting DTMF CID without polarity reversal.

  Looks like an unintended change when sig_analog.c was extracted from
  chan_dahdi.c.

  Removed useless conditional around needed code and fixed resulting
  compiler warning.

  (closes issue #18667)
  Reported by: enegaard
  Patches:
        issue18667.patch uploaded by enegaard (license 1197)
  Tested by: enegaard

  JIRA SWP-2965
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309127 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 309084 via svnmerge from
David Vossel [Tue, 1 Mar 2011 16:22:27 +0000 (16:22 +0000)]
Merged revisions 309084 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r309084 | dvossel | 2011-03-01 10:09:11 -0600 (Tue, 01 Mar 2011) | 15 lines

  Merged revisions 309083 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2

  ........
    r309083 | dvossel | 2011-03-01 10:05:25 -0600 (Tue, 01 Mar 2011) | 9 lines

    Fixes thread blocking issue in the sip TCP/TLS implementation.

    (closes issue #18497)
    Reported by: vois
    Patches:
          issues_18497.diff uploaded by dvossel (license 671)
    Tested by: vois, rossbeer, kowalma, Freddi_Fonet
  ........
................

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309090 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 309035 via svnmerge from
Tilghman Lesher [Mon, 28 Feb 2011 11:16:06 +0000 (11:16 +0000)]
Merged revisions 309035 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r309035 | tilghman | 2011-02-28 05:10:28 -0600 (Mon, 28 Feb 2011) | 15 lines

  Merged revisions 309033-309034 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2

  ........
    r309033 | tilghman | 2011-02-28 04:43:12 -0600 (Mon, 28 Feb 2011) | 4 lines

    A later version of flex already includes the fwrite workaround code, which if used twice causes a compilation error.

    Detect whether Flex will compile without the workaround; if so, suppress our workaround code.
  ........
    r309034 | tilghman | 2011-02-28 05:07:52 -0600 (Mon, 28 Feb 2011) | 2 lines

    Clarify meaning, removing double negative (stupid!)
  ........
................

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309036 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 308991 via svnmerge from
Tilghman Lesher [Mon, 28 Feb 2011 09:34:16 +0000 (09:34 +0000)]
Merged revisions 308991 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r308991 | tilghman | 2011-02-28 03:33:22 -0600 (Mon, 28 Feb 2011) | 14 lines

  Merged revisions 308990 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2

  ........
    r308990 | tilghman | 2011-02-28 03:32:22 -0600 (Mon, 28 Feb 2011) | 7 lines

    Statements updating zero rows may return SQL_NO_DATA.  This is fine; it's handled.

    (closes issue #18815)
     Reported by: irroot
     Patches:
           func_odbc.insert_nodata.patch uploaded by irroot (license 52)
  ........
................

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308992 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 308945 via svnmerge from
Alec L Davis [Fri, 25 Feb 2011 18:58:10 +0000 (18:58 +0000)]
Merged revisions 308945 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r308945 | alecdavis | 2011-02-26 07:52:53 +1300 (Sat, 26 Feb 2011) | 21 lines

  Fix Deadlock with attended transfer of SIP call

  Call path
    sip_set_rtp_peer (locks chan then pvt)
     transmit_reinvite_with_sdp
      try_suggested_sip_codec
       pbx_builtin_getvar_helper (locks p->owner)

  But by the time p->owner lock was attempted, seems as though chan and p->owner were different.

  So in sip_set_rtp_peer, lock pvt first then lock p->owner using deadlocking methods.

  (closes issue #18837)
  Reported by: alecdavis
  Patches:
        bug18837-trunk.diff3.txt uploaded by alecdavis (license 585)
  Tested by: alecdavis, Irontec, ZX81, cmaj

  Review: [https://reviewboard.asterisk.org/r/1126/]
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308946 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 308903 via svnmerge from
Richard Mudgett [Thu, 24 Feb 2011 21:43:32 +0000 (21:43 +0000)]
Merged revisions 308903 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r308903 | rmudgett | 2011-02-24 15:38:41 -0600 (Thu, 24 Feb 2011) | 9 lines

  Invalid read in ast_channel_set_caller_event().

  Valgrind reported that ast_channel_set_caller_event() was reading data
  from a freed buffer when using the pre_set structure.

  Rearange things to pre-calculate the name and number pointer before
  updating the caller party structure to see if the name or number was
  changed.
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308904 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 308815 via svnmerge from
Terry Wilson [Thu, 24 Feb 2011 17:59:32 +0000 (17:59 +0000)]
Merged revisions 308815 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r308815 | twilson | 2011-02-24 11:57:18 -0600 (Thu, 24 Feb 2011) | 26 lines

  Merged revisions 308814 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2

  ................
    r308814 | twilson | 2011-02-24 11:54:49 -0600 (Thu, 24 Feb 2011) | 19 lines

    Merged revisions 308813 via svnmerge from
    https://origsvn.digium.com/svn/asterisk/branches/1.4

    ........
      r308813 | twilson | 2011-02-24 11:42:16 -0600 (Thu, 24 Feb 2011) | 12 lines

      Don't broadcast FullyBooted to every AMI connection

      The FullyBooted event should not be sent to every AMI connection every
      time someone connects via AMI. It should only be sent to the user who
      just connected.

      (closes issue #18168)
      Reported by: FeyFre
      Patches:
            bug0018168.patch uploaded by FeyFre (license 1142)
      Tested by: FeyFre, twilson
    ........
  ................
................

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308816 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 308723 via svnmerge from
Matthew Nicholson [Thu, 24 Feb 2011 15:10:58 +0000 (15:10 +0000)]
Merged revisions 308723 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r308723 | mnicholson | 2011-02-24 09:06:14 -0600 (Thu, 24 Feb 2011) | 16 lines

  Merged revisions 308722 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2

  ................
    r308722 | mnicholson | 2011-02-24 08:59:41 -0600 (Thu, 24 Feb 2011) | 9 lines

    Merged revisions 308721 via svnmerge from
    https://origsvn.digium.com/svn/asterisk/branches/1.4

    ........
      r308721 | mnicholson | 2011-02-24 08:54:56 -0600 (Thu, 24 Feb 2011) | 2 lines

      silence gcc 4.2 compiler warning
    ........
  ................
................

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308724 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 308679 via svnmerge from
Terry Wilson [Thu, 24 Feb 2011 03:49:07 +0000 (03:49 +0000)]
Merged revisions 308679 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r308679 | twilson | 2011-02-23 21:41:34 -0600 (Wed, 23 Feb 2011) | 15 lines

  Merged revisions 308678 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2

  ........
    r308678 | twilson | 2011-02-23 21:38:22 -0600 (Wed, 23 Feb 2011) | 8 lines

    Use remotesecret to authenticate with a remote party

    The remotesecret option was only being used for outbound registration
    and not for placing calls. This patch uses remotesecret on outbound
    calls if it is set, otherwise secret is still used.

    Review: https://reviewboard.asterisk.org/r/1107/
  ........
................

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308680 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoFix compiler warning.
Richard Mudgett [Wed, 23 Feb 2011 23:55:58 +0000 (23:55 +0000)]
Fix compiler warning.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308624 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 308622 via svnmerge from
Richard Mudgett [Wed, 23 Feb 2011 23:45:02 +0000 (23:45 +0000)]
Merged revisions 308622 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r308622 | rmudgett | 2011-02-23 17:38:04 -0600 (Wed, 23 Feb 2011) | 9 lines

  sig_pri_new_ast_channel() should return NULL when new_ast_channel() fails.

  (closes issue #18874)
  Reported by: cmaj
  Patches:
        patch-sig_pri-crash-possible-null-channel-pointer.diff.txt uploaded by cmaj (license 830)

  JIRA SWP-3172
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308623 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMedia Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio...
David Vossel [Tue, 22 Feb 2011 23:04:49 +0000 (23:04 +0000)]
Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff

-Functional changes
1. Dynamic global format list build by codecs defined in codecs.conf
2. SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf
3. Negotiation of SILK attributes in chan_sip.
4. SPEEX 32khz with translation
5. SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation
   using codec_resample.c
6. Various changes to RTP code required to properly handle the dynamic format list
   and formats with attributes.
7. ConfBridge now dynamically jumps to the best possible sample rate.  This allows
   for conferences to take advantage of HD audio (Which sounds awesome)
8. Audiohooks are no longer limited to 8khz audio, and most effects have been
   updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT.
9. codec_resample now uses its own code rather than depending on libresample.

-Organizational changes
Global format list is moved from frame.c to format.c
Various format specific functions moved from frame.c to format.c

Review: https://reviewboard.asterisk.org/r/1104/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308582 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoUse ast_debug for console logging
Andrew Latham [Tue, 22 Feb 2011 15:33:56 +0000 (15:33 +0000)]
Use ast_debug for console logging

Guessed the log levels based on info that level 3
is the soft roof.  Can we create a page / document
to define the levels?

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308527 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 308416 via svnmerge from
Matthew Nicholson [Mon, 21 Feb 2011 15:04:19 +0000 (15:04 +0000)]
Merged revisions 308416 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r308416 | mnicholson | 2011-02-21 09:02:20 -0600 (Mon, 21 Feb 2011) | 19 lines

  Merged revisions 308414 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2

  ................
    r308414 | mnicholson | 2011-02-21 09:00:22 -0600 (Mon, 21 Feb 2011) | 12 lines

    Merged revisions 308413 via svnmerge from
    https://origsvn.digium.com/svn/asterisk/branches/1.4

    ........
      r308413 | mnicholson | 2011-02-21 08:57:15 -0600 (Mon, 21 Feb 2011) | 5 lines

      Properly check the bounds of arrays when decoding UDPTL packets.  Also, remove broken support for receiving UDPTL packets larger than 16k.  That shouldn't ever happen anyway.

      AST-2011-002
      FAX-281
    ........
  ................
................

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308417 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoAdd HTTP URI Debug logging and update notice
Andrew Latham [Mon, 21 Feb 2011 14:14:41 +0000 (14:14 +0000)]
Add HTTP URI Debug logging and update notice

enable reporting of the request URI / URL in debugging
change funny debug note to a serious note.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308372 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agofix a memory leak in device state
Tzafrir Cohen [Mon, 21 Feb 2011 13:58:18 +0000 (13:58 +0000)]
fix a memory leak in device state

The callback handle_statechange (pbx.c) fails to release its data
pointer, leaking memory in the process.

Reported by: tzafrir
Patches:
      18735_pbx_free_callback.diff uploaded by tzafrir (license 46)

Review: https://reviewboard.asterisk.org/r/1110/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308371 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoAdd CSS MIME Type
Andrew Latham [Sat, 19 Feb 2011 14:07:38 +0000 (14:07 +0000)]
Add CSS MIME Type

Modern browsers are checking for the MIME Type of pages
and in some cases will not load a file if the type is
wrong.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308331 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 308288 via svnmerge from
Tilghman Lesher [Sat, 19 Feb 2011 11:03:44 +0000 (11:03 +0000)]
Merged revisions 308288 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r308288 | tilghman | 2011-02-19 05:02:49 -0600 (Sat, 19 Feb 2011) | 2 lines

  A few more (copies of) files to ignore in this directory.
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308289 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 308242 via svnmerge from
Alexandr Anikin [Fri, 18 Feb 2011 00:11:06 +0000 (00:11 +0000)]
Merged revisions 308242 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r308242 | may | 2011-02-18 03:07:20 +0300 (Fri, 18 Feb 2011) | 3 lines

  added g729onlyA option for announce only AnnexA g.729 codec in
  h.323 capabilities. Option can be global or per user/peer.
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308243 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoAdd more verbage to CLI command 'pri show channels' usage.
Richard Mudgett [Thu, 17 Feb 2011 20:21:56 +0000 (20:21 +0000)]
Add more verbage to CLI command 'pri show channels' usage.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308205 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 308150 via svnmerge from
Paul Belanger [Wed, 16 Feb 2011 22:02:41 +0000 (22:02 +0000)]
Merged revisions 308150 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r308150 | pabelanger | 2011-02-16 15:21:17 -0500 (Wed, 16 Feb 2011) | 2 lines

  Fix FreeBSD builds.
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308157 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 308098 via svnmerge from
Alexandr Anikin [Wed, 16 Feb 2011 08:06:01 +0000 (08:06 +0000)]
Merged revisions 308098 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r308098 | may | 2011-02-16 10:57:22 +0300 (Wed, 16 Feb 2011) | 2 lines

  ifdef __linux__ keepalive variables also
........

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9 years agoMerged revisions 308010 via svnmerge from
Jason Parker [Tue, 15 Feb 2011 23:34:27 +0000 (23:34 +0000)]
Merged revisions 308010 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r308010 | qwell | 2011-02-15 17:34:03 -0600 (Tue, 15 Feb 2011) | 24 lines

  Merged revisions 308007 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2

  ................
    r308007 | qwell | 2011-02-15 17:33:24 -0600 (Tue, 15 Feb 2011) | 17 lines

    Merged revisions 308002 via svnmerge from
    https://origsvn.digium.com/svn/asterisk/branches/1.4

    ........
      r308002 | qwell | 2011-02-15 17:32:20 -0600 (Tue, 15 Feb 2011) | 10 lines

      Fix regression that changed behavior of queues when ringing a queue member.

      This reverts r298596, which was to fix a highly bizarre and contrived issue
      with a queue member that called into his own queue being transferred back
      into his own queue.  I couldn't reproduce that issue in any way.  I think one
      of the other recent transfer fixes actually fixed this.

      (closes issue #18747)
      Reported by: vrban
    ........
  ................
................

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9 years agoinclude tcp keepalive socket calls only on linux, freebsd and others
Alexandr Anikin [Tue, 15 Feb 2011 23:07:47 +0000 (23:07 +0000)]
include tcp keepalive socket calls only on linux, freebsd and others
don't have these options on sockets.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307969 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoAdd CLI "pri show channels" command.
Richard Mudgett [Tue, 15 Feb 2011 21:42:55 +0000 (21:42 +0000)]
Add CLI "pri show channels" command.

List the current mapping of DAHDI B channels to Asterisk channel names and
which calls are on hold or call-waiting.  Calls on hold or call-waiting
are not associated with any B channel.

JIRA LIBPRI-27
JIRA SWP-2547

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307964 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 307962 via svnmerge from
Richard Mudgett [Tue, 15 Feb 2011 19:53:32 +0000 (19:53 +0000)]
Merged revisions 307962 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r307962 | rmudgett | 2011-02-15 13:52:45 -0600 (Tue, 15 Feb 2011) | 1 line

  Don't crash when forcing caller id.
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307963 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoFixes compile error in chan_phone for big endian
David Vossel [Tue, 15 Feb 2011 18:09:25 +0000 (18:09 +0000)]
Fixes compile error in chan_phone for big endian

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307927 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 307879 via svnmerge from
Richard Mudgett [Tue, 15 Feb 2011 16:18:43 +0000 (16:18 +0000)]
Merged revisions 307879 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r307879 | rmudgett | 2011-02-15 10:13:55 -0600 (Tue, 15 Feb 2011) | 37 lines

  No response sent for SIP CC subscribe/resubscribe request.

  Asterisk does not send a response if we try to subscribe for call
  completion after we have received a 180 Ringing.  You can only subscribe
  for call completion when the call has been cleared.

  When we receive the 180 Ringing, for this call, its call-completion state
  is 'CC_AVAILABLE'.  If we then send a subscribe message to Asterisk, it
  trys to change the call-completion state to 'CC_CALLER_REQUESTED'.
  Because this is an invalid state change, it just ignores the message.  The
  only state Asterisk will accept our subscribe message is in the
  'CC_CALLER_OFFERED' state.

  Asterisk will go into the 'CC_CALLER_OFFERED' when the SIP client clears
  the call by sending a CANCEL.

  Asterisk should always send a response.  Even if its a negative one.

  The fix is to allow for the CCSS core to notify a CC agent that a failure
  has occurred when CC is requested.  The "ack" callback is replaced with a
  "respond" callback.  The "respond" callback has a parameter indicating
  either a successful response or a specific type of failure that may need
  to be communicated to the requester.

  (closes issue #18336)
  Reported by: GeorgeKonopacki
  Tested by: mmichelson, rmudgett

  JIRA SWP-2633

  (closes issue #18337)
  Reported by: GeorgeKonopacki
  Tested by: mmichelson

  JIRA SWP-2634
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307883 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 307837 via svnmerge from
Tilghman Lesher [Tue, 15 Feb 2011 07:03:44 +0000 (07:03 +0000)]
Merged revisions 307837 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r307837 | tilghman | 2011-02-15 01:02:45 -0600 (Tue, 15 Feb 2011) | 15 lines

  Merged revisions 307836 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2

  ........
    r307836 | tilghman | 2011-02-15 01:01:37 -0600 (Tue, 15 Feb 2011) | 8 lines

    Need to retrieve the rows affected before using the associated variable.

    (closes issue #18795)
     Reported by: irroot
     Patches:
           20110211__issue18795.diff.txt uploaded by tilghman (license 14)
     Tested by: tilghman
  ........
................

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307838 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 307793 via svnmerge from
Tilghman Lesher [Mon, 14 Feb 2011 20:18:02 +0000 (20:18 +0000)]
Merged revisions 307793 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r307793 | tilghman | 2011-02-14 14:16:55 -0600 (Mon, 14 Feb 2011) | 15 lines

  Merged revisions 307792 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2

  ........
    r307792 | tilghman | 2011-02-14 14:10:28 -0600 (Mon, 14 Feb 2011) | 8 lines

    Increment usage count at first reference, to avoid a race condition with many threads creating connections all at once.

    (issue #18156)
     Reported by: asgaroth
     Patches:
           20110214__issue18156.diff.txt uploaded by tilghman (license 14)
     Tested by: tilghman
  ........
................

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307795 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMaking trunk compile again.
Tilghman Lesher [Mon, 14 Feb 2011 07:01:46 +0000 (07:01 +0000)]
Making trunk compile again.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307752 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 307750 via svnmerge from
Tilghman Lesher [Mon, 14 Feb 2011 06:54:08 +0000 (06:54 +0000)]
Merged revisions 307750 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r307750 | tilghman | 2011-02-14 00:50:23 -0600 (Mon, 14 Feb 2011) | 23 lines

  Calling a gosub routine defined in AEL from Dial/Queue ceased to work.

  A bug in AEL did not distinguish between the "s" extension generated by
  AEL and an "s" extension that was required to exist by the chan_dahdi
  (or another channel) that was not supplied with a starting extension.
  Therefore, AEL made incorrect assumptions about what commands were
  permissable in the context.  This was fixed by making AEL generate a
  different extension name.  However, Dial and Queue make additional
  assumptions about the name of the default gosub extension.  Therefore,
  they needed to be brought into line with a "macro" rendered by AEL (as
  a gosub), without breaking traditional dialplans written without the
  aid of AEL.

  Related to (issue #18480)
   Reported by: nivek

  (closes issue #18729)
   Reported by: kkm
   Patches:
         20110209__issue18729.diff.txt uploaded by tilghman (license 14)
         018729-dial-queue-gosub-try3.patch uploaded by kkm (license 888)
   Tested by: kkm
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307751 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agolc not found - it's warning, not error,
Alexandr Anikin [Sun, 13 Feb 2011 10:50:22 +0000 (10:50 +0000)]
lc not found - it's warning,  not error,
change malloc to ast_calloc again

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307713 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agochange malloc to ast_calloc calls to prevent crash of asterisk
Alexandr Anikin [Sat, 12 Feb 2011 23:25:58 +0000 (23:25 +0000)]
change malloc to ast_calloc calls to prevent crash of asterisk

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307677 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 307536 via svnmerge from
Jason Parker [Thu, 10 Feb 2011 22:43:51 +0000 (22:43 +0000)]
Merged revisions 307536 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r307536 | qwell | 2011-02-10 16:39:30 -0600 (Thu, 10 Feb 2011) | 22 lines

  Merged revisions 307535 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2

  ................
    r307535 | qwell | 2011-02-10 16:35:49 -0600 (Thu, 10 Feb 2011) | 15 lines

    Merged revisions 307534 via svnmerge from
    https://origsvn.digium.com/svn/asterisk/branches/1.4

    ........
      r307534 | qwell | 2011-02-10 16:33:09 -0600 (Thu, 10 Feb 2011) | 8 lines

      Remove color when executing commands via a remote console.

      Essentially this makes '-x' imply '-n' on rasterisk.  This was done in a
      different and incomplete way previously, which I'm reverting here.

      (issue #18776)
      Reported by: alecdavis
    ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307537 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 307467 via svnmerge from
Mark Michelson [Thu, 10 Feb 2011 17:45:24 +0000 (17:45 +0000)]
Merged revisions 307467 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r307467 | mmichelson | 2011-02-10 11:44:42 -0600 (Thu, 10 Feb 2011) | 5 lines

  Fix a gaffe in the CCSS sample configuration.

  Discovered by Philippe Lindheimer and pointed out on #asterisk-dev
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307468 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoFixes bug in chan_sip where nativeformats are not set correctly.
David Vossel [Thu, 10 Feb 2011 17:12:10 +0000 (17:12 +0000)]
Fixes bug in chan_sip where nativeformats are not set correctly.

The nativeformats field was being overwritten when it should have been
appended too.  This caused some format capabilities to be lost briefly and
some log warnings to be output.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307433 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoCorrections for properly work with H.323v2 (older) endpoints and other
Alexandr Anikin [Thu, 10 Feb 2011 13:29:19 +0000 (13:29 +0000)]
Corrections for properly work with H.323v2 (older) endpoints and other
small fixes.

Interpret remote side H.225 version.

Corrections for H.323v2 endpoints:
don't start TCS and MSD before connect,
don't start TCS and MSD by accepting H.245 connection,
start TCS and MSD by StartH245 facility message.

Other fixes:
fix non zeroended remoteDisplayName issue, small fixes in call clearing
by closing H.245 connection, tcp keepalive introduced on TCP
connections (now is hardcoded, will be configurable in the future),
don't force H.245tunneling if FastStart is active, don't send Alerting
singal more than once per call.

(closes issue #18542)
Reported by: vmikhelson
Patches:
      issue18542-final-3.patch uploaded by may213 (license 454)
Tested by: vmikhelson

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307396 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoAdd new manager action MeetmeListRooms.
Jeff Peeler [Wed, 9 Feb 2011 22:48:02 +0000 (22:48 +0000)]
Add new manager action MeetmeListRooms.

From the submitter:
I've added a new manager action to list only the active conferences on an
Asterisk system. It shows the same data displayed when you run a 'meetme list'
on the Asterisk CLI.

(closes issue #17905)
Reported by: rcasas
Patches:
      app_meetme.c.patch uploaded by rcasas (license 641)

Review: https://reviewboard.asterisk.org/r/874/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307359 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoDisable color during running test
Andrew Latham [Wed, 9 Feb 2011 21:46:24 +0000 (21:46 +0000)]
Disable color during running test

(closes issue #18776)
Reported by: alecdavis
Patches:
     ast_deb_init.diff uploaded by lathama (license 1028)
Tested by: andrel, lathama

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307315 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 307273 via svnmerge from
Jeff Peeler [Wed, 9 Feb 2011 21:08:22 +0000 (21:08 +0000)]
Merged revisions 307273 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r307273 | jpeeler | 2011-02-09 15:06:33 -0600 (Wed, 09 Feb 2011) | 8 lines

  Add missing debug info for ao2_link for use with REF_DEBUG in ao2 callback.

  (closes issue #18758)
  Reported by: rgagnon
  Patches:
        branch-1.8-r306540-astobj-fix.diff uploaded by rgagnon (license 1202)
        trunk-r306540-astobj-fix.diff uploaded by rgagnon (license 1202)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307274 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoAllow parkedmusicclass to be settable for non-default parking lots.
Jeff Peeler [Wed, 9 Feb 2011 20:11:11 +0000 (20:11 +0000)]
Allow parkedmusicclass to be settable for non-default parking lots.

(closes issue #17946)
Reported by: bluecrow76
Patches:
      asterisk-1.8.0-beta4-multipark-fixes-2010SEP02.diff

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307231 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 307228 via svnmerge from
Jeff Peeler [Wed, 9 Feb 2011 19:53:28 +0000 (19:53 +0000)]
Merged revisions 307228 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r307228 | jpeeler | 2011-02-09 13:52:51 -0600 (Wed, 09 Feb 2011) | 17 lines

  Merged revisions 307227 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2

  ........
    r307227 | jpeeler | 2011-02-09 13:52:12 -0600 (Wed, 09 Feb 2011) | 11 lines

    Make sure to set parking dial context for non-default parking lots.

    Since parking_con_dial isn't settable, set all parking lots to "park-dial".

    (closes issue #17946)
    Reported by: bluecrow76
    Patches:
          asterisk-1.8.0-beta4-multipark-fixes-2010SEP02.diff uploaded by bluecrow76 (license 270)
          modified by me
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307229 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoclarify warning when no loadable module support
Tzafrir Cohen [Wed, 9 Feb 2011 19:17:01 +0000 (19:17 +0000)]
clarify warning when no loadable module support

Clarify warning message when LOADABLE_MODULES is disabled but we still
try to load a module.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307192 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 307142 via svnmerge from
Tilghman Lesher [Wed, 9 Feb 2011 05:53:29 +0000 (05:53 +0000)]
Merged revisions 307142 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r307142 | tilghman | 2011-02-08 23:39:39 -0600 (Tue, 08 Feb 2011) | 3 lines

  Initialize tracking variable in structure properly.  Fixes a memory leak.
  (Reported by The_Boy_Wonder on IRC, fixed by me.)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307143 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 307092 via svnmerge from
Jason Parker [Tue, 8 Feb 2011 21:24:57 +0000 (21:24 +0000)]
Merged revisions 307092 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r307092 | qwell | 2011-02-08 15:24:01 -0600 (Tue, 08 Feb 2011) | 9 lines

  Fix issue with verbose messages not showing on remote console.

  This code was reworked recently, and since the logchannel list hadn't been
  created yet at this point, and it was a verbose message, it was being dropped
  on the floor.  Now it'll continue on to where it should be handled.

  (closes issue #18580)
  Reported by: pabelanger
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307097 65c4cc65-6c06-0410-ace0-fbb531ad65f3