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/


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  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.
  ........
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  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.
  ........
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  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
  ........
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(closes issue #16024)
Reported by: mnicholson


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

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


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  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
........


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  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
........


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

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


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  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)
  ........
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  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)
........


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  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).
........


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  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
........


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  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
  ........
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  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!)
  ........
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  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)
  ........
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  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/]
........


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  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.
........


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  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
    ........
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  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
    ........
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  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/
  ........
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  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
........


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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

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



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  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
    ........
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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

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

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



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  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.
........


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  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.
........


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


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  r308098 | may | 2011-02-16 10:57:22 +0300 (Wed, 16 Feb 2011) | 2 lines
  
  ifdef __linux__ keepalive variables also
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  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
    
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      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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308013 65c4cc65-6c06-0410-ace0-fbb531ad65f3

don't have these options on sockets.


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

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

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

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  r307962 | rmudgett | 2011-02-15 13:52:45 -0600 (Tue, 15 Feb 2011) | 1 line
  
  Don't crash when forcing caller id.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307963 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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