Add full t.38 handshaking for OOH323 that are required for newest T.38
gateway codes.
Add fax detection (cng tone, t38) and dialplan redirection to fax ext on
fax event detected.
Add OOH323() function to set/get t38support and faxdetect parameters.

(closes issue ASTERISK-17754)
Reported by: irroot
Patches: 
      ooh323_faxdetect.patch uploaded by irroot (license 52)
      issue19183-final.patch uploaded by may213 (license 454)
Tested by: may213, irroot

Review: https://reviewboard.asterisk.org/r/1174/



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  r327211 | rmudgett | 2011-07-08 16:41:58 -0500 (Fri, 08 Jul 2011) | 9 lines
  
  INVITE 403 Forbidden response always retransmits the maximum times.
  
  Asterisk sends a 403 Forbidden response if authentication fails for an
  INVITE as required.  However, it ignores the ACK and keeps retransmitting
  the response.
  
  * Made not delete the to-tag in the dialog so the expected ACK can be
  matched with the dialog and stop the retransmissions.
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It was inconsistent to have the silk and celt format attribute
modules in the format file interpreter folder.


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  r327106 | mnicholson | 2011-07-08 14:52:51 -0500 (Fri, 08 Jul 2011) | 11 lines
  
  Reset our ast_str before passing it on to dialplan function backends.
  
  It is possible for a dialplan backend to not modify the given buffer or ast_str
  and still return success. This causes any previous value stored in the buffer
  to be used as if the new function call provided it. Some functions also append
  to the given buffer assuming it is empty.
  
  The test_substitution unit test has also been modified to detect this problem.
  
  (closes issue ASTERISK-17878)
........


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  r327046 | russell | 2011-07-08 11:00:05 -0500 (Fri, 08 Jul 2011) | 2 lines
  
  Fix an error and add more log message info to help see why this fails on FreeBSD.
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  r327044 | russell | 2011-07-08 10:28:44 -0500 (Fri, 08 Jul 2011) | 2 lines
  
  Resolve some set-but-unused-variable warnings.
........


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  r326985 | rmudgett | 2011-07-07 20:08:05 -0500 (Thu, 07 Jul 2011) | 12 lines
  
  Some code cleanup in pbx.c
  
  * Mostly comment and format changes.
  
  * ast_context_remove_extension_callerid() and ast_add_extension_nolock()
  will write lock the found specific context.
  
  * ast_context_find() will now tolerate a NULL name.
  
  * Eliminated some inlined versions of find_context() and
  find_context_locked().
........


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This patch adds pass-through support for CELT.  CELT
formats are defined in codecs.conf and can be configured
to any sample rate a CELT endpoint supports.  This patch also
addresses a crash in channel.c resulting from a frame list being
freed incorrectly.  This crash was discovered while testing a CELT
translator which had to split encoded audio into multiple frames.
The codec translator is not a part of this patch, but may be
contributed in the future.

Review: https://reviewboard.asterisk.org/r/1294/


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  r326830 | tilghman | 2011-07-07 14:17:19 -0500 (Thu, 07 Jul 2011) | 1 line
  
  libgen.h is also needed on Darwin for basename(3)
........


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  r326689 | jrose | 2011-07-07 11:04:51 -0500 (Thu, 07 Jul 2011) | 10 lines
  
  res_odbc patch by tilghman to fix integers with null values
  
  Addresses some improper sql statements in res_odbc that would cause an update to fail on
  realtime peers due to trying to set as "(NULL)" rather than an actual NULL.
  
  (closes issue #1922STERISK-17791)
  Reported by: marcelloceschia
  Patches: 
        20110505__issue19223.diff.txt uploaded by tilghman (license 14)
........


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  r326683 | mnicholson | 2011-07-07 10:28:25 -0500 (Thu, 07 Jul 2011) | 3 lines
  
  use sips: or sip: depending on the transport in use when building reply digest
  URIs
........


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  r326681 | mnicholson | 2011-07-07 10:25:49 -0500 (Thu, 07 Jul 2011) | 3 lines
  
  make the uri parameter used in reply digests more standards compliant in
  certain cases by prepending "sip:" or "sips:" to it
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326682 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Adds a new sub-command, "valgrind" to live_ast. It runs asterisk under
valgrind. The extra command-line parameters are passed to Asterisk as
usual, and parameters to valgrind are passed through LIVE_AST_VALGRIND_ARGS
in live.conf .

Review: https://reviewboard.asterisk.org/r/1109/


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There were some bugs in the very ancient version of Berkeley DB that Asterisk
used. Instead of spending the time tracking down the bugs in the Berkeley code
we move to the much better documented SQLite 3.

Conversion of the old astdb happens at runtime by running the included
astdb2sqlite3 utility. The ast_db API with SQLite 3 backend should behave
identically to the old Berkeley backend, but in the future we could offer a
much more robust interface.

We do not include the SQLite 3 library in the source tree, but instead rely
upon the distribution-provided libraries. SQLite is so ubiquitous that this
should not place undue burden on administrators.


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  r326484 | dvossel | 2011-07-06 10:26:49 -0500 (Wed, 06 Jul 2011) | 10 lines
  
  Reverts fix for timerfd locking issue.
  
  jrose discovered a performance issue with this
  fix that prevents his analog phones from working
  when using timerfd as a timing source.  Until
  it is understood what is causing this performance
  problem, this patch is being reverted.
........


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  r326411 | tilghman | 2011-07-05 17:08:29 -0500 (Tue, 05 Jul 2011) | 14 lines
  
  Add the attribute "type" to each "<use>" for menuselect.
  
  This matters only when autoconf fails to detect that weak linking is supported.
  External optional dependencies will become optional in both cases, as they are
  removed at compile time when not detected.  However, runtime-optional modules
  are made mandatory when weak linking is not found.  This change affects only
  the external optional dependencies; previously, they were incorrectly required
  when weak linking support was not detected.
  
  Patches:
  	20110702__issue18062__asterisk_trunk.diff.txt by tilghman (License #5003)
  
  Tested by: iasgoscouk
........


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Several variables in the script control which files are converted and the
source and destination formats.

Patch-by: Trey Blancher <support@digium.com>
(closes AST-560)


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  r326291 | rmudgett | 2011-07-05 12:22:59 -0500 (Tue, 05 Jul 2011) | 23 lines
  
  Used auth= parameter freed during "sip reload" causes crash.
  
  If you use the auth= parameter and do a "sip reload" while there is an
  ongoing call.  The peer->auth data points to free'd memory.
  
  The patch does several things:
  
  1) Puts the authentication list into an ao2 object for reference counting
  to fix the reported crash during a SIP reload.
  
  2) Converts the authentication list from open coding to AST list macros.
  
  3) Adds display of the global authentication list in "sip show settings".
  
  (closes issue ASTERISK-17939)
  Reported by: wdoekes
  Patches:
        jira_asterisk_17939_v1.8.patch (license #5621) patch uploaded by rmudgett
  
  Review: https://reviewboard.asterisk.org/r/1303/
  
  JIRA SWP-3526
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This adds a new action, FilterAdd to the manager interface that allows control over event filters for the current session

(closes issue ASTERISK-16795)
Reported by: kobaz
Tested by: kobaz,loloski



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  r326209 | mjordan | 2011-07-05 08:23:57 -0500 (Tue, 05 Jul 2011) | 7 lines
  
  Updated filestream destructor to block until move is complete when cache is used
  
  When a cache directory is used, the process is forked and a mv command is executed to move the temporary file to the permanent location.  This caused issues with voicemail, where a race condition occurred when the parent expected the file to be in the permanent location prior to the mv command completing.  The parent process is now blocked until the mv command completes.
  
  (closes issue ASTERISK-17724)
  Reported by: Adiren P.
  Tested by: mjordan
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  r326144 | rmudgett | 2011-07-01 16:07:22 -0500 (Fri, 01 Jul 2011) | 16 lines
  
  Better way to get chan and pvt lock for issue ASTERISK-17431.
  
  Redoes -r308945 for issue ASTERISK-17431 deadlock fix for
  sip_set_udptl_peer() and sip_set_rtp_peer().
  
  * Lock the channels in the defined order and avoid the need for a deadlock
  avoidance loop.
  
  * Lock the channel before getting the pointer to the private structure to
  be sure that the pointer will not change due to a masquerade or channel
  hangup.
  
  * To preserve sanity, check that chan and p->owner are the same.  (Pointer
  rearangements should not happen without the protection of locks because
  bad things tend to happen otherwise.)
........


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r296249 r318141 Application changes

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  r325935 | rmudgett | 2011-06-30 15:39:45 -0500 (Thu, 30 Jun 2011) | 11 lines
  
  Misc minor changes in chan_sip.
  
  * Add load failure exit if primary SIP container(s) could not get created
  in chan_sip.c:load_module().
  
  * Removed a redundant static prototype.
  
  * Some typos.
  
  * Some whitespace.
........


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Review: https://reviewboard.asterisk.org/r/1288/



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  r325877 | mjordan | 2011-06-30 15:09:48 -0500 (Thu, 30 Jun 2011) | 9 lines
  
  Patched voicemail user option for emailbody / emailsubject
  
  Incorporated changes per ASTERISK-16795; updated unit tests to check for vmu->emailbody / vmu->emailsubject
  
  (closes issue ASTERISK-16795)
  Reported by: mdeneen
  Tested by: mjordan
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  r325821 | jrose | 2011-06-30 14:17:32 -0500 (Thu, 30 Jun 2011) | 10 lines
  
  Fixes an issue with Music on Hold classes losing files in playlist when realtime is used.
  
  The bug occurs rather intermittently and I relied on the reporters to test the patch.
  After a sanity check and some testing, I'm giving it an OK.
  
  (closes issue ASTERISK-17875)
  Reported by: David Cunningham
  Patches: 
        res_musiconhold.c.mohrt17875_v1 uploaded by Igor Goncharovsky (license #5009)
........


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